GIT b348e6e752bc8287448226980a61d412c64f2ce8 git+ssh://master.kernel.org/pub/scm/linux/kernel/git/tiwai/sound-2.6.git b348e6e752bc8287448226980a61d412c64f2ce8 commit b348e6e752bc8287448226980a61d412c64f2ce8 Author: Roel Kluin <12o3l@tiscali.nl> Date: Thu Apr 17 18:58:34 2008 +0200 [ALSA] SOC: fix tests in cs4270_hw_params() cs4270_hw_params does several times: ret = snd_soc_write() if (ret < 0) ... This only works when ret is signed. Signed-off-by: Roel Kluin <12o3l@tiscali.nl> Signed-off-by: Takashi Iwai commit 8c447fb022cc81c5a06c6e9c28be8e7ac32aa8b8 Author: Takashi Iwai Date: Thu Apr 17 12:53:26 2008 +0200 [ALSA] usb-audio - Fix race in reconnection Fix the race at reconnection of the device. The disconnected usb_chip[] must be cleared before the next probe call properly. Signed-off-by: Takashi Iwai commit da69d3045ca8d57fffdb6fc252aa6ae43f9fb8e4 Author: Takashi Iwai Date: Thu Apr 17 12:52:02 2008 +0200 [ALSA] Clean up snd_card_free*() A little clean up of snd_card_free*(). Removed snd_card_free_prepare() since it's actually almost identical with snd_card_disconnect(). Signed-off-by: Takashi Iwai commit 3870ac8d753482558a10718562448d689498ce7d Author: Takashi Iwai Date: Thu Apr 17 12:50:47 2008 +0200 [ALSA] Fix the race of card instance unregistration Move the call of device_unregister() for the card instance in snd_card_disconnect() to avoid the race of sysfs card entry, which can be typically found on usb-audio reconnection. Signed-off-by: Takashi Iwai commit 0eebf2d006a524cb5973842b149d9651fde27186 Author: Risto Suominen Date: Wed Apr 16 19:45:51 2008 +0200 [ALSA] snd-powermac: style burgundy.c Coding style corrections for burgundy.c. Signed-off-by: Risto Suominen Signed-off-by: Takashi Iwai commit e6315178573f06210c89c9494f06f373a6d47c65 Author: Risto Suominen Date: Wed Apr 16 19:45:31 2008 +0200 [ALSA] snd-powermac: Burgundy mixers for B&W and iMac Add mixer controls and correct headphone detection bits for PowerMac G3 B&W and iMac G3 Tray-loading, both having Burgundy chipset. Signed-off-by: Risto Suominen Signed-off-by: Takashi Iwai commit 0493adcf452af93b4f954b53629e4f28a319d2fa Author: Risto Suominen Date: Wed Apr 16 19:39:27 2008 +0200 [ALSA] snd-powermac: style awacs.s and awacs.h Coding style corrections for awacs.c and awacs.h. Signed-off-by: Risto Suominen Signed-off-by: Takashi Iwai commit c28b86e872bad0b0943d12f81e977db948019ec1 Author: Risto Suominen Date: Thu Apr 17 17:55:30 2008 +0200 [ALSA] snd-powermac: AWACS and Screamer mixers for PM7500, Beige, and iMac SL Add mixer controls and correct headphone detection bits for PowerMacs 7300/7500 (AWACS) and G3 Beige (Screamer), and iMac G3 Slot-loading (Screamer). Signed-off-by: Risto Suominen Signed-off-by: Takashi Iwai commit 39f754549010151892cfb96699cb133d6bfaf365 Author: Risto Suominen Date: Wed Apr 16 13:16:05 2008 +0200 [ALSA] snd-powermac: style pmac.c Coding style corrections for pmac.c. Signed-off-by: Risto Suominen Signed-off-by: Takashi Iwai commit 1a85e6fcccf543ac4fd736053fdded936ef9687a Author: Risto Suominen Date: Wed Apr 16 13:15:38 2008 +0200 [ALSA] snd-powermac: enable headphone detection Enable port change interrupt while initialising AWACS, Screamer, and Burgundy chipsets. Signed-off-by: Risto Suominen Signed-off-by: Takashi Iwai commit d9de07425110896b211ea80b3fe85526e1953967 Author: Roel Kluin <12o3l@tiscali.nl> Date: Wed Apr 16 19:30:30 2008 +0200 [ALSA] sound/drivers/dummy.c: fix negative snd_pcm_format_width() check bps is unsigned, a negative snd_pcm_format_width() return value is not noticed Signed-off-by: Roel Kluin <12o3l@tiscali.nl> Signed-off-by: Takashi Iwai commit f992272ea4df850258a96c7dbd1aa0f632045792 Author: Takashi Iwai Date: Wed Apr 16 17:29:09 2008 +0200 [ALSA] hda - Avoid unexpected breakage with ALC889A hack The last ALC889A hack may break on some devices with certain model presets since patch_alc*() have different model tables. So, now it's handled in the original patch_alc882() but fly to patch_alc883() in model=auto appropriately. Signed-off-by: Takashi Iwai commit 16779c0845b26487e6edd5c5f89b7a30fc36c7c8 Author: Takashi Iwai Date: Wed Apr 16 14:13:29 2008 +0200 [ALSA] hda - Fix ALC889A codec support ALC889A is recognized ALC885/ALC882 but it's actually closer to ALC888/ALC883. Cc: Kasper Sandberg Signed-off-by: Takashi Iwai commit 09c18514a3bcffa23a203f1fb6797b968771ce7e Author: Matthew Ranostay Date: Wed Apr 16 13:13:59 2008 +0200 [ALSA] hda: Add 5.1 support for second headphone jack Several 92hd7xxx and STAC9228 laptops have multiple headphone jacks, the second headphone jack should be used for the 5.1 surround sound. Add support for 'Headphone as Line Out' switch, which allows it be used in 5.1 surround sound. Signed-off-by: Matthew Ranostay Signed-off-by: Takashi Iwai commit 3ad08f1bdf6748e9af4034efd64dd5d7170e38f8 Author: Mark Brown Date: Wed Apr 16 12:59:55 2008 +0200 [ALSA] soc - wm9712: Remove unneeded AC97_EXTENDED_MID updates Signed-off-by: Mark Brown Signed-off-by: Takashi Iwai commit 79654aec55145367fd53bb2435f136ab003dae67 Author: Clemens Ladisch Date: Wed Apr 16 09:15:45 2008 +0200 [ALSA] oxygen: generalize DAC volume TLV handling Add a pointer for DAC volume TLV data to the model structure so that the model driver do not need to manually assign it in their control filter. Signed-off-by: Clemens Ladisch Signed-off-by: Takashi Iwai commit 31a77798e97b5cc491ee8ef6ebe4c062bd4380f2 Author: Clemens Ladisch Date: Wed Apr 16 09:14:30 2008 +0200 [ALSA] oxygen: mute by default Initialize the playback volume controls as being muted and having minimal volume. Signed-off-by: Clemens Ladisch Signed-off-by: Takashi Iwai commit 69b34061e902e2c5d0f29c97acc13821a5da93ed Author: Clemens Ladisch Date: Wed Apr 16 09:13:36 2008 +0200 [ALSA] oxygen: generalize handling of DAC volume limits Add fields for the DAC volume limits to the module structure so that model drivers do not need to install their own control info handlers. Signed-off-by: Clemens Ladisch Signed-off-by: Takashi Iwai commit 0ca04dde7aa3175fbb45c07b79fee03921acd148 Author: Clemens Ladisch Date: Wed Apr 16 09:12:27 2008 +0200 [ALSA] hifier: remove empty hifier_mixer_init() The empty hifier_mixer_init() function is useless; remove it. Signed-off-by: Clemens Ladisch Signed-off-by: Takashi Iwai commit 0c0ef231e1bfe9ea530b76723b3c092795ffaa04 Author: Takashi Iwai Date: Tue Apr 15 18:46:42 2008 +0200 [ALSA] hda - Add support of AD1989A/AD1989B Added the support of AD1989A and AD1989B codecs. These codecs can have multiple SPDIF devices, but currently we handle only one SPDIF. If any real devices with two SPDIF interfaces (likely one for SPDIF and one for HDMI), we'll fix this rightly. Otherwise, these codecs are pretty similar with AD1988. Signed-off-by: Takashi Iwai commit a528b35308ab11032c51b586525b5bc750373b93 Author: Pavel Machek Date: Mon Apr 14 18:31:35 2008 +0200 [ALSA] sound/core.h: evil #ifdefs snd_minor_info_oss_* is an function returning int _or_ comment, depending on config parameters. That is truly evil, fix it. Signed-off-by: Pavel Machek Signed-off-by: Takashi Iwai commit 1ab9e43fa571858966071d89fb243945bad804da Author: Clemens Ladisch Date: Tue Apr 15 08:57:31 2008 +0200 [ALSA] virtuoso: fix DX front panel I/O Fix the GPIO 1 mixer control to enable I/O through the front panel connector of the Xonar DX. Signed-off-by: Clemens Ladisch Signed-off-by: Takashi Iwai commit 203d72ed3e2b000479685f98750063fdf49cfe79 Author: Daniel Mack Date: Mon Apr 14 15:40:31 2008 +0200 [ALSA] snd_usb_caiaq: make high sample rates work with A8DJ This patch for snd_usb_caiaq makes sample rates higher dann 48KHz work with devices which have more than 2 stereo input/output pairs. Signed-off-by: Daniel Mack Signed-off-by: Takashi Iwai commit d14ff4a5c7bd96b2c0dcf9e060909f675b0d2af6 Author: Daniel Mack Date: Mon Apr 14 15:39:47 2008 +0200 [ALSA] snd_usb_caiaq: correct input channel order This patch corrects the input channel order of hardware supported by snd_usb_caiaq. Signed-off-by: Daniel Mack Signed-off-by: Takashi Iwai commit 577f7b3be4989a080618da9a6d4e531527fa8d4b Author: Daniel Mack Date: Mon Apr 14 15:39:14 2008 +0200 [ALSA] snd_usb_caiaq: fix potential lockups locking This patch fixes potential lockups in snd_usb_caiaq by refining the locking mechanims and by using usb_kill_urb() in favor to usb_unlink_urb(). Signed-off-by: Daniel Mack Signed-off-by: Takashi Iwai commit 79edd1b09890336f8813fd0391b5ce6222612e5a Author: Jarkko Nikula Date: Mon Apr 14 15:28:19 2008 +0200 [ALSA] ASoC: Add support for 19.2 MHz MCLK in TLV320AIC3X Signed-off-by: Jarkko Nikula Signed-off-by: Mark Brown Signed-off-by: Takashi Iwai commit 62ea290c4206e2e1ac6e73142ff0a9b70228d7b6 Author: Mark Brown Date: Mon Apr 14 15:27:30 2008 +0200 [ALSA] wm9713: Don't control touch screen power on suspend Leave the power bit for the touch screen alone when suspending the WM9713 so that the touch screen driver can handle it. This allows the touch screen to be used as a wakeup source when the system is suspended. Signed-off-by: Mark Brown Signed-off-by: Takashi Iwai commit 69fc83c420b3e5560fc06cdff9f5f69ce283f355 Author: Nick Andrew Date: Mon Apr 14 15:22:11 2008 +0200 [ALSA] sound: this amplifier only goes up to 7 sound: kernel log levels are 0-7 Kernel log levels are 0-7, not 0-9. Signed-off-by: Nick Andrew Signed-off-by: Takashi Iwai commit 1f8a514b63f2b990dea835251a00854b53350ade Author: Herton Ronaldo Krzesinski Date: Mon Apr 14 13:46:28 2008 +0200 [ALSA] hda-intel: Add Quanta IL1 ALC267 model This adds support for Quanta IL1 mini-notebook to alsa, defining a new model for it. It comes with an ALC267 codec chip. Some notes about this model: * In headphone automute, I use AC_VERB_SET_PIN_WIDGET_CONTROL instead of common amp mute, to avoid conflict with mixer switch (mixer and automute use the same nid). * The only connected capture sources in the hardware are the internal mic and external mic jack. So instead of using an input source selector like on other ALC268 models, the mic automute automatically switch between captures. Signed-off-by: Herton Ronaldo Krzesinski Signed-off-by: Takashi Iwai commit b66ea221517a204e11d0ddabc09b1eb4ba2bd8e7 Author: Kay Sievers Date: Mon Apr 14 13:33:36 2008 +0200 [ALSA] sound: fix platform driver hotplug/coldplug Since 43cc71eed1250755986da4c0f9898f9a635cb3bf, the platform modalias is prefixed with "platform:". Add MODULE_ALIAS() to the hotpluggable sound platform drivers, to re-enable auto loading. [dbrownell@users.sourceforge.net: more drivers, registration fixes] Signed-off-by: Kay Sievers Signed-off-by: David Brownell Signed-off-by: Andrew Morton Signed-off-by: Takashi Iwai commit 967c765db9af6bfd035e10292eff5e2d46197d16 Author: Matthew Ranostay Date: Mon Apr 14 13:32:54 2008 +0200 [ALSA] hda: EAPD power management Power management support for EAPD enabled laptops, when headphones are sensed it pulls the EAPD GPIO line low to power it down. Signed-off-by: Matthew Ranostay Signed-off-by: Takashi Iwai commit f3ed484393d1175493c4453971772c8ab1591505 Author: Matthew Ranostay Date: Mon Apr 14 13:32:27 2008 +0200 [ALSA] hda: Correct SPDIF out default config Several laptops have have the SPDIF out defined as 'Digital other out' when it should be 'SPDIF out' in the default config. Signed-off-by: Matthew Ranostay Signed-off-by: Takashi Iwai commit d85fe659bf8ed59be6284a315bc33f6b7e38f226 Author: Tony Vroon Date: Mon Apr 14 13:31:45 2008 +0200 [ALSA] hda - Fujitsu Lifebook PC speaker signal The legacy PC speaker signal was not routed to outputs. The codec is not prevented from powering down in this patch, although I suppose one could argue that perhaps it should be. Let me know if anyone feels strongly one way or the other. Signed-off-by: Tony Vroon Signed-off-by: Takashi Iwai commit 914c83bcdc4c572aef0f3fb226d4129db155ab01 Author: Jiang zhe Date: Mon Apr 14 13:26:53 2008 +0200 [ALSA] hda - PCI quirk for laptop LG which use CMI9880 Please refer to [0003874] on the alsa mantis. This patch added the pci quirk. Signed-off-by: Jiang zhe Signed-off-by: Takashi Iwai commit f0babf9921af5427bc876af38b0297963614f299 Author: Jiang zhe Date: Mon Apr 14 13:26:21 2008 +0200 [ALSA] hda - Should use HDA_OUTPUT instead of HDA_INPUT to mute pin 15 of ALC880 To mute the output of Pin widget 15 in ALC880, we should use the HDA_OUTPUT. However, current code looks like : snd_hda_codec_amp_stereo(codec, 0x15, HDA_INPUT, 0, HDA_AMP_MUTE, bits); It may be a misspelling. Signed-off-by: Jiang zhe Signed-off-by: Takashi Iwai commit fc5c523376d964b5cb429470af343f06509392b9 Author: Pavel Machek Date: Mon Apr 14 13:15:56 2008 +0200 [ALSA] sound/usb/usbaudio.c: coding style Putting space between ! and variable is a strange coding style, fix that, also make it fit into 80 columns where that is easy. Signed-off-by: Pavel Machek Signed-off-by: Takashi Iwai commit f49f0f4ecf8a35f941c490c9c8c2974707d20ed3 Author: Pavel Machek Date: Mon Apr 14 13:14:22 2008 +0200 [ALSA] usb audio: make quirk handling more readable, and fix commented-out code usb audio contains useful debugging code, protected by #if 0. Unfortunately, it will not compile because variable names changed; fix it. Dallas workaround is formatted in a way where it is not quite obvious what is normal code and what is quirk. Reformat it to make it obvious. Signed-off-by: Pavel Machek Signed-off-by: Andrew Morton Signed-off-by: Takashi Iwai commit 5a1ca9292c3391e2197ab2be855997b86a065fc2 Author: Pavel Machek Date: Mon Apr 14 13:12:47 2008 +0200 [ALSA] usb audio: Fix another Dallas quirk Dallas USB speakers are buggy in more than one way. One of configs they offer does not work at all. Signed-off-by: Pavel Machek Signed-off-by: Andrew Morton Signed-off-by: Takashi Iwai commit 798899249e8ca19ba3b708cf8eac076a376a57ce Author: Frederik Deweerdt Date: Mon Apr 14 13:11:44 2008 +0200 [ALSA] hda-codec - Fix unbalanced mutex On Wed, Apr 02, 2008 at 08:19:29AM -0400, Miles Lane wrote: > [ 48.765906] [ BUG: bad unlock balance detected! ] > [ 48.765912] ------------------------------------- > [ 48.765918] pulseaudio/4277 is trying to release lock > (&codec->spdif_mutex) at: > [ 48.765930] [] mutex_unlock+0x8/0xa > [ 48.765945] but there are no more locks to release! > [ 48.765950] > [ 48.765952] other info that might help us debug this: > [ 48.765959] 2 locks held by pulseaudio/4277: > [ 48.765965] #0: (&pcm->open_mutex){--..}, at: [] > snd_pcm_open+0xc1/0x1ba [snd_pcm] > [ 48.766003] #1: (&chip->open_mutex){--..}, at: [] > azx_pcm_open+0x36/0x184 [snd_hda_intel] > [ 48.766057] > [ 48.766059] stack backtrace: > [ 48.766066] Pid: 4277, comm: pulseaudio Not tainted 2.6.25-rc8-mm1 #12 > [ 48.766086] [] print_unlock_inbalance_bug+0xce/0xd8 > [ 48.766107] [] ? save_stack_trace+0x1d/0x3b > [ 48.766130] [] ? __kernel_text_address+0x1b/0x27 > [ 48.766146] [] ? dump_trace+0xcd/0xd9 > [ 48.766160] [] ? save_stack_address+0x0/0x2c > [ 48.766176] [] ? find_usage_backwards+0xa4/0xc3 > [ 48.766193] [] lock_release_non_nested+0x84/0x120 > [ 48.766209] [] ? mutex_unlock+0x8/0xa > [ 48.766222] [] lock_release+0x16a/0x199 > [ 48.766238] [] __mutex_unlock_slowpath+0xa9/0x121 > [ 48.766252] [] mutex_unlock+0x8/0xa > [ 48.766263] [] snd_hda_multi_out_analog_open+0xd3/0xef > [snd_hda_intel] The following patch should fix it. Cc: "Miles Lane" Signed-off-by: Andrew Morton Signed-off-by: Takashi Iwai commit 447c5fe8170806b47c2378e2683cd8f15091e170 Author: Andrew Morton Date: Mon Apr 14 13:09:33 2008 +0200 [ALSA] es1968 - fix coding style in the last patch WARNING: braces {} are not necessary for single statement blocks #40: FILE: sound/pci/es1968.c:1831: + if (diff > 1) { + __maestro_write(chip, IDR0_DATA_PORT, cp1); + } total: 0 errors, 1 warnings, 35 lines checked ./patches/es1968-fix-jitter-on-some-maestro-cards.patch has style problems, please review. If any of these errors are false positives report them to the maintainer, see CHECKPATCH in MAINTAINERS. Please run checkpatch prior to sending patches Cc: Andreas Mueller Tested-by: Rene Herman Signed-off-by: Andrew Morton Signed-off-by: Takashi Iwai commit 613a6227d9d55d1ff90940f1f8e63507cda93f49 Author: Andreas Mueller Date: Mon Apr 14 13:08:05 2008 +0200 [ALSA] es1968: fix jitter on some maestro cards This patch suppresses jitter on several Maestro cards in stereo mode (ALSA of course). The patch is also incorporated in the *BSD drivers where I "ported" it from. Without this patch most of the stereo audio gets out of sync and really distorted (oss-emulation with mplayer at 48000khz worked somehow). Signed-off-by: Andrew Morton Signed-off-by: Takashi Iwai commit c394b0eaaf3b26bb2a208aefbc12ac27cfd7fe8e Author: Denys Vlasenko Date: Mon Apr 14 13:04:18 2008 +0200 [ALSA] sound/pci/rme9652/hdspm.c: stop inlining largish static functions sound/pci/rme9652/hdspm.c has unusually large number of static inline functions - 22. I looked through them and some of them seem to be too big to warrant inlining. This patch removes "inline" from these static functions (regardless of number of callsites - gcc nowadays auto-inlines statics with one callsite). Size difference on 32bit x86: text data bss dec hex filename 20437 2160 516 23113 5a49 linux-2.6-ALLYES/sound/pci/rme9652/hdspm.o 18036 2160 516 20712 50e8 linux-2.6.inline-ALLYES/sound/pci/rme9652/hdspm.o [coding fix by Takashi Iwai ] Signed-off-by: Denys Vlasenko Signed-off-by: Takashi Iwai commit d9a878901beff8bfa929681b36174aac2748645f Author: Mark Brown Date: Mon Apr 14 12:59:27 2008 +0200 [ALSA] soc - Include register in DAPM debug output When logging register changes in DAPM debug output include the register number. Signed-off-by: Mark Brown Signed-off-by: Takashi Iwai commit f73b2b63389fbdcf3d58c808471ae1325570d29c Author: Jiang zhe Date: Mon Apr 14 12:58:57 2008 +0200 [ALSA] hda-codec - PCI quirk for MSI laptop Please refer to [0003848] on the alsa mantis. This patch adds the pci quirk and Mic-Int controller. Signed-off-by: Jiang zhe Signed-off-by: Takashi Iwai commit cfd9161f73803ef78cb84d79f95d81a0d8801fd4 Author: Clemens Ladisch Date: Fri Apr 11 10:25:40 2008 +0200 [ALSA] virtuoso: initialize two-wire control register On the Xonar DX, initialize all bits of the two-wire control register. Signed-off-by: Clemens Ladisch Signed-off-by: Takashi Iwai commit bcb82dfab2bd64b9963b9cce0e6a4e69466fbdec Author: Clemens Ladisch Date: Fri Apr 11 10:24:48 2008 +0200 [ALSA] virtuoso: add GPIO 1 mixer control Add a mixer control for switching whatever it is that is connected to GPIO pin 1 on the Xonar DX. Signed-off-by: Clemens Ladisch Signed-off-by: Takashi Iwai commit c513bc20440bb31f67d3432be9032ec380030ed8 Author: Clemens Ladisch Date: Wed Apr 9 09:16:33 2008 +0200 [ALSA] oxygen: use SPDIF input only if present If the card model does not have a digital input or an AC97 codec, disable the respective interrupt and mixer controls. Signed-off-by: Clemens Ladisch Signed-off-by: Takashi Iwai commit c8fe7327f62096b5261fe6af56de710689f4553c Author: Clemens Ladisch Date: Wed Apr 9 09:16:14 2008 +0200 [ALSA] virtuoso: correctly switch input jack on Xonar DX When selecting the capture source on the Xonar DX, the input jack must be routed to either the line input or the microphone input by setting a GPIO pin. This requires an additional callback so that the model driver can hook into the toggling of AC97 switches. Signed-off-by: Clemens Ladisch Signed-off-by: Takashi Iwai commit d70f457105bbbf104c5d3a2e1aa0177301a8ddb1 Author: Clemens Ladisch Date: Mon Apr 7 10:29:44 2008 +0200 [ALSA] virtuoso: add Xonar DX support Add support for the Asus Xonar DX. Signed-off-by: Clemens Ladisch Signed-off-by: Takashi Iwai commit 094a9de9c21703d07d4991be8e584279c02acb6a Author: Clemens Ladisch Date: Mon Apr 7 10:27:01 2008 +0200 [ALSA] virtuoso: fix typo Fix a (fortunately harmless) typo. Signed-off-by: Clemens Ladisch Signed-off-by: Takashi Iwai commit b7751efa4c9568932e89272061b287305a9c777a Author: Clemens Ladisch Date: Mon Apr 7 10:26:45 2008 +0200 [ALSA] virtuoso: change card short name Change the card short name to show to show the card name instead of the chip name. Signed-off-by: Clemens Ladisch Signed-off-by: Takashi Iwai commit 53a5fc137b6b5aab22f5ba09d574b1f7916e7f92 Author: Clemens Ladisch Date: Mon Apr 7 10:26:26 2008 +0200 [ALSA] virtuoso: set PCM1796 oversampling rate When playing data at 96 kHz or higher, reduce the DAC oversampling rate to 32. Signed-off-by: Clemens Ladisch Signed-off-by: Takashi Iwai commit 29dbe0517791ddd090e2026603e4f4b64004fe99 Author: Clemens Ladisch Date: Mon Apr 7 10:26:03 2008 +0200 [ALSA] virtuoso: move some code to xonar_common_init() Move the code that is common to all Xonar models to a separate function, and make it more generic in preparation for another model. Signed-off-by: Clemens Ladisch Signed-off-by: Takashi Iwai commit 6fd0ae0364fbdb95d87c289e8c1932baea6e2a22 Author: Clemens Ladisch Date: Mon Apr 7 10:25:30 2008 +0200 [ALSA] virtuoso: allow both CS5381 and CS5361 Rename all CS5381 symbols to CS53x1 because they can also be used for Xonar models with a CS5361. Signed-off-by: Clemens Ladisch Signed-off-by: Takashi Iwai commit 68fcb4d9fcacb832cafa30fffd0bc23a3907c225 Author: Clemens Ladisch Date: Mon Apr 7 10:24:22 2008 +0200 [ALSA] virtuoso: separate D2/D2X init functions Use separate model structures for the D2 and D2X so that the init function does not have to check for the model again. Signed-off-by: Clemens Ladisch Signed-off-by: Takashi Iwai commit 35ff25c5e7b757bf09f2d7760d95e02c3b899a3b Author: Clemens Ladisch Date: Mon Apr 7 10:23:37 2008 +0200 [ALSA] oxygen: add I2C support Add a function to write I2C registers. Signed-off-by: Clemens Ladisch Signed-off-by: Takashi Iwai commit 842e4ae7dcb2826f0365884aa493c8424b334301 Author: Clemens Ladisch Date: Wed Apr 2 10:56:30 2008 +0200 [ALSA] aw2: remove duplicate MODULE_LICENSE "GPL 2" does not mean that there have to be two MODULE_LICENSE("GPL") entries. ;-) Signed-off-by: Clemens Ladisch Signed-off-by: Takashi Iwai commit 218dc9bfa1f356e1596f711a0b44d17dd447ed9d Author: Pavel Machek Date: Tue Apr 1 15:33:22 2008 +0200 [ALSA] fix comments in sound/core.h Two sentences seem to be spliced into one in comment, fix that and fix english. Also fix codingstyle. Signed-off-by: Pavel Machek Signed-off-by: Jaroslav Kysela Signed-off-by: Takashi Iwai commit 17a25342313252b69c211504a5277f04cc0f808c Author: Clemens Ladisch Date: Tue Apr 1 10:02:18 2008 +0200 [ALSA] oxygen: fix line-in recording selection (now for real) On C-Media cards, the GPIO pin 0 of the CM9780 must be handled exactly like on Xonar cards, so move the Xonar code to the common mixer code. Signed-off-by: Clemens Ladisch Signed-off-by: Takashi Iwai commit b95f8ea27247522b7ef394ac757ef4d13bc34175 Author: Herton Ronaldo Krzesinski Date: Sat Mar 22 10:26:05 2008 +0100 [ALSA] hda-codec - Support mic automute for Clevo M720R/SR Add support for mic automute in clevo-m720r ALC883 model, and rename it to more generic clevo-m720. Also change model entry in ALSA-Configuration.txt accordingly. Signed-off-by: Herton Ronaldo Krzesinski Signed-off-by: Takashi Iwai commit 19ef02c7214b28f91c156c611c19e38c019d9dd4 Author: Herton Ronaldo Krzesinski Date: Sat Mar 22 10:25:30 2008 +0100 [ALSA] hda-codec - Map clevo-m720r ALC883 model for Clevo M720SR Map clevo-m720r ALC883 model for Clevo M720SR. Signed-off-by: Herton Ronaldo Krzesinski Signed-off-by: Takashi Iwai commit 91b20a3e7fcb0a6d56e866b54980af15289c2355 Author: Stas Sergeev Date: Sat Mar 22 10:12:37 2008 +0100 [ALSA] pcsp: remove downsampling pcsp: remove S16->U8 downsampling as dmix now supports U8 natively. Signed-off-by: Stas Sergeev Signed-off-by: Takashi Iwai commit 9388bab2858f5f50b8a893b495e6a53b70909f96 Author: Takashi Iwai Date: Sat Mar 22 10:11:08 2008 +0100 [ALSA] ymfpci - Fix race at removal free_irq() must be called first to avoid races at removal. Signed-off-by: Takashi Iwai commit 530cdd787a8f3d958d1ee7914dd3cb9ec7103ae3 Author: Takashi Iwai Date: Thu Mar 20 12:30:36 2008 +0100 [ALSA] hda-codec - Add missing models in ALSA-Configuration.txt Signed-off-by: Takashi Iwai commit b213edc6029801169e6db202402cfc8940598a88 Author: Herton Ronaldo Krzesinski Date: Thu Mar 20 12:14:59 2008 +0100 [ALSA] hda-codec - Use common 3stack-6ch mixer for 3stack-hp model Forgot one more: 3stack-hp model also have now the same mixer as 3stack-6ch (after DAC assignment fix in ALC883), so use it avoiding duplicating the same mixer definition. Signed-off-by: Herton Ronaldo Krzesinski Signed-off-by: Takashi Iwai commit 9a49a72d78e2eab199bfa1840a7146961aa0743a Author: Herton Ronaldo Krzesinski Date: Thu Mar 20 12:14:28 2008 +0100 [ALSA] hda-codec - Use base ALC883 mixer for 6stack-dell model After DAC assignment fix in ALC883, alc888_6st_dell_mixer is now the same as alc883_base_mixer. Avoid duplicated code and use alc883_base_mixer in 6stack-dell model, removing alc888_6st_dell_mixer definition. Signed-off-by: Herton Ronaldo Krzesinski Signed-off-by: Takashi Iwai commit b0341b749e798e1e3519167d8a1c9559dfc47e84 Author: Herton Ronaldo Krzesinski Date: Thu Mar 20 12:13:46 2008 +0100 [ALSA] hda-codec - Remove now uneeded 6stack-hp model from ALC883 After DAC assignment fix in ALC883, the 6stack-hp model is now the same as 6stack-dig. So just remove 6stack-hp model and replace its use with 6stack-dig. Signed-off-by: Herton Ronaldo Krzesinski Signed-off-by: Takashi Iwai commit 8465260b6fe2fd9ab6592eea2ffda02b1600fb43 Author: Jiang zhe Date: Thu Mar 20 12:12:39 2008 +0100 [ALSA] hda-codec - model for alc262 to support Lenovo 3000 This model is to support the Lenovo 3000 y410. ALSA bug#3856: https://bugtrack.alsa-project.org/alsa-bug/view.php?id=3856 Signed-off-by: Jiang zhe Signed-off-by: Takashi Iwai commit 79709dab2d506dc2702784edf09ce8dd3144b3c3 Author: Matthew Ranostay Date: Thu Mar 20 12:10:57 2008 +0100 [ALSA] hda: 92hd71bxxx DMIC nid Added missing DMIC verb to dell_4_1_pin_configs[]. Signed-off-by: Matthew Ranostay Signed-off-by: Takashi Iwai commit de92a6285a8eb425bcec101886f53662e9099175 Author: Pavel Hofman Date: Thu Mar 20 12:10:27 2008 +0100 [ALSA] ice1724 - Improved the Juli rate setting * moving most of clock-specific code to card-specific routines * support for ESI Juli * to-be-researched - monitoring of analog/digital inputs Signed-off-by: Pavel Hofman Signed-off-by: Takashi Iwai commit 478d1ef677028e05ee6ec126a6790f9be0ebf4ff Author: Andrew Morton Date: Thu Mar 20 12:07:31 2008 +0100 [ALSA] sound/pci/pcxhr/pcxhr.c: fix warnings sparc64: sound/pci/pcxhr/pcxhr.c: In function `pcxhr_update_r_buffer': sound/pci/pcxhr/pcxhr.c:459: warning: cast to pointer from integer of different size sound/pci/pcxhr/pcxhr.c: In function `pcxhr_trigger_tasklet': sound/pci/pcxhr/pcxhr.c:628: warning: long int format, different type arg (arg 4) Signed-off-by: Andrew Morton Signed-off-by: Takashi Iwai commit 461254996308c01626f210595c4030119c5d5929 Author: Andrew Morton Date: Thu Mar 20 12:05:33 2008 +0100 [ALSA] sound/pci/pcxhr/pcxhr_core.c: fix printk warning sound/pci/pcxhr/pcxhr_core.c: In function `pcxhr_set_pipe_state': sound/pci/pcxhr/pcxhr_core.c:899: warning: long int format, different type arg (arg 4) suseconds_t is int on sparc64. Signed-off-by: Andrew Morton Signed-off-by: Takashi Iwai commit 99ed3f144f7218a37970cc639ccbdf205d88bad3 Author: Andrew Morton Date: Thu Mar 20 12:04:46 2008 +0100 [ALSA] sound/pci/aw2/aw2-alsa.c needs dma-mapping.h sparc32: sound/pci/aw2/aw2-alsa.c: In function 'snd_aw2_create': sound/pci/aw2/aw2-alsa.c:282: error: 'DMA_32BIT_MASK' undeclared (first use in this function) sound/pci/aw2/aw2-alsa.c:282: error: (Each undeclared identifier is reported only once sound/pci/aw2/aw2-alsa.c:282: error: for each function it appears in.) Signed-off-by: Andrew Morton Signed-off-by: Takashi Iwai commit 216b5129801e29ffb064e58d574b9f63cfa2196b Author: Clemens Ladisch Date: Wed Mar 19 08:21:32 2008 +0100 [ALSA] oxygen: disable clock of unused I2S inputs Disable the master clock outputs of any unused I2S inputs. Signed-off-by: Clemens Ladisch Signed-off-by: Takashi Iwai commit ba74da35281ee610d4b0e254124fd563f15d94b6 Author: Clemens Ladisch Date: Wed Mar 19 08:20:59 2008 +0100 [ALSA] oxygen: move MIDI flag to model struct Put the flag that enables the MIDI port into the model structure instead of passing it as a separate parameter to oxygen_pci_probe(). Signed-off-by: Clemens Ladisch Signed-off-by: Takashi Iwai commit c0e1c0e360d9b80f4e5a91d03777b6896a54c85d Author: Clemens Ladisch Date: Wed Mar 19 08:20:13 2008 +0100 [ALSA] oxygen: make SPI/2-wire configuration model-specific Allow the model drivers to specify if the codec communication goes over SPI or a 2-wire bus. Signed-off-by: Clemens Ladisch Signed-off-by: Takashi Iwai commit c44a30fd0214a96bf7b93a7420bef7a295e7e49f Author: Clemens Ladisch Date: Wed Mar 19 08:19:41 2008 +0100 [ALSA] oxygen: change model-specific PCM device configuration When specifying which PCM devices to use, model drivers now use flags that also specify the routing between PCM devices and DMA channels instead of just DMA channel bits. This simplifies some code that checks for these flags. Signed-off-by: Clemens Ladisch Signed-off-by: Takashi Iwai commit 14dcb68d95230fad91f84afdb8e03eed16c9b2c9 Author: Clemens Ladisch Date: Wed Mar 19 08:17:33 2008 +0100 [ALSA] oxygen: add monitor controls Add controls to enable monitoring of the analog and digital inputs. To allow monitoring after loading the driver when nothing has been played back or recorded yet, the I2S input and outputs are initialized to a valid configuration. Signed-off-by: Clemens Ladisch Signed-off-by: Takashi Iwai commit b7b7fa163da375246a71e98a44aeaac3ac42a50b Author: Clemens Ladisch Date: Wed Mar 19 08:16:40 2008 +0100 [ALSA] virtuoso: move PCM1796 symbols to a header file Move the PCM1796 register symbol definitions to their own header file. Signed-off-by: Clemens Ladisch Signed-off-by: Takashi Iwai commit 9893e14255a3b579f5564ff8c2a15591f1df02ea Author: Clemens Ladisch Date: Wed Mar 19 08:14:01 2008 +0100 [ALSA] oxygen: move WM8785 symbols to a header file Move the WM8786 register symbol definitions to their own header file. Signed-off-by: Clemens Ladisch Signed-off-by: Takashi Iwai commit 01470fbe2665f1c2ce1eb7984d105c31b459786e Author: Timur Tabi Date: Tue Mar 18 17:18:18 2008 +0100 [ALSA] Removed deprecated sound/driver.h from Freescale MPC8610 drivers With commit 9004acc70e8c49c50c4c7b652f906f1e0ed5709d, include/sound/driver.h is deprecated. This patch removes the #include from fsl_ssi.c and fsl_dma.c. Signed-off-by: Timur Tabi Signed-off-by: Takashi Iwai commit e6198111cfaae18257da58024d02c1b6974155c3 Author: Takashi Iwai Date: Tue Mar 18 17:11:05 2008 +0100 [ALSA] hda-intel - Add sync support Addded the support of sync streams to hda-intel driver. Signed-off-by: Takashi Iwai commit e9e881864a199952d864b165bb2ed47570494191 Author: Takashi Iwai Date: Tue Mar 18 12:13:03 2008 +0100 [ALSA] hda-codec - Support of Lenovo Thinkpad X300 Added the model thinkpad for Lenovo Thinkpad X300 with AD1984A codec. Signed-off-by: Takashi Iwai commit 06b62648fe46f1111c7aa1af2aa55517730a10be Author: Robert Jarzmik Date: Tue Mar 18 12:08:35 2008 +0100 [ALSA] soc - Add missing audio path between Mono Mixer and Mic PGAs Signed-off-by: Robert Jarzmik Signed-off-by: Mark Brown Signed-off-by: Takashi Iwai commit b760095d88add7f68ee244feb8bb26e9666034a8 Author: Takashi Iwai Date: Tue Mar 18 09:57:50 2008 +0100 [ALSA] hda-codec - keep the format verb at closing PCM streams Keep the format verb at closing PCM streams. Introduced snd_hda_codec_cleanup_stream() for the parcicular purpose. Signed-off-by: Takashi Iwai commit 3f1e4408325604349430f8aaca4652763d193385 Author: Takashi Iwai Date: Tue Mar 18 09:53:23 2008 +0100 [ALSA] hda-codec - Fix spekaer output of Panasonic CF-74 Add a new model "panasonic" for Panasonic CF-74 with STAC9200 codec to fix the speaker output. Signed-off-by: Takashi Iwai commit afae1a2e0f06ada3e5753242b19ecaa4d2333ede Author: Takashi Iwai Date: Tue Mar 18 09:47:06 2008 +0100 [ALSA] hda-intel - Add barrier Add proper barriers in the RIRB communication code. Signed-off-by: Takashi Iwai commit c952684711435d35e64b2a175f575c84b1ea11fb Author: Herton Ronaldo Krzesinski Date: Tue Mar 18 09:27:59 2008 +0100 [ALSA] hda-codec - Map 3stack-6ch-dig ALC883 model for MSI 945GCM5 V2 (MSI-7267) Map 3stack-6ch-dig ALC883 model for MSI 945GCM5 V2 (MSI-7267). Signed-off-by: Herton Ronaldo Krzesinski Signed-off-by: Takashi Iwai commit cf3daca26cfab82c8def853b73391c7f42bb00cd Author: Herton Ronaldo Krzesinski Date: Tue Mar 18 09:27:08 2008 +0100 [ALSA] hda-codec - Fix DAC assignment order in ALC883 Actually clfe and surround DACs are inverted in alc883_dac_nids array (see ALC883 datasheet). I discovered this while testing multi-channel setup (using 3stack-6ch-dig model) on MSI 945GCM5 V2 motherboard that has an ALC883 codec. Simply Rear Left/Right and Center/LFE were swapped in 6 channel mode (also in 4 channel mode you didn't get rear left/right output). Other models also were affected by this bug, as can be seen by the mixer layouts that "workaround" this (the real bug was not noticed, and some other models simply played with mixer and initial verbs). Thus along with fixing the order of dac nids, also change the models that relied on previous dac ordering properly. Signed-off-by: Herton Ronaldo Krzesinski Signed-off-by: Takashi Iwai commit 368044635b7c43e0e0b26f650c180f1805f14ad1 Author: Mariusz Kozlowski Date: Tue Mar 18 09:03:03 2008 +0100 [ALSA] sound/drivers/pcsp/pcsp.c build fix sound/drivers/pcsp/pcsp.c: In function 'snd_pcsp_create': sound/drivers/pcsp/pcsp.c:54: error: 'loops_per_jiffy' undeclared (first use in\ this function) sound/drivers/pcsp/pcsp.c:54: error: (Each undeclared identifier is reported on\ ly once sound/drivers/pcsp/pcsp.c:54: error: for each function it appears in.) Signed-off-by: Mariusz Kozlowski Signed-off-by: Andrew Morton Signed-off-by: Takashi Iwai commit db6c3680cfe3d32f18d89ffd6294f357d1ca6f10 Author: Atsushi Nemoto Date: Mon Mar 17 14:36:24 2008 +0100 [ALSA] at73c213: Add constraints for periods value The interrupt handler always provide runtime->period_size data, so it works correctly only if buffer_size was a multiple of period_size. This patch fixes periodic click noise. Signed-off-by: Atsushi Nemoto Signed-off-by: Takashi Iwai commit b57ef63a3b4b3fde02e4edb6e52168ffe2bb47ca Author: Julia Lawall Date: Mon Mar 17 10:23:35 2008 +0100 [ALSA] sound/pci: remove unused variable The variable is_capture is initialized but never used otherwise. The semantic patch that makes this change is as follows: (http://www.emn.fr/x-info/coccinelle/) // @@ type T; identifier i; constant C; @@ ( extern T i; | - T i; <+... when != i - i = C; ...+> ) // Signed-off-by: Julia Lawall Signed-off-by: Takashi Iwai commit db6131446efc0e748902f13196b45f3cfa017546 Author: Takashi Iwai Date: Mon Mar 17 10:16:37 2008 +0100 [ALSA] ice1724 - Fix return codes in some pointis callbacks Fixed the return codes (1 for changed values) in put callbacks of pontis. Signed-off-by: Takashi Iwai commit 9c4a35ebfe5a2efd67ee7ff2780933ec4b1dae8a Author: Takashi Iwai Date: Mon Mar 17 09:59:32 2008 +0100 [ALSA] usb-audio - Add a proper error check The error in check_hw_params_convention() has to be checked and handled properly. Signed-off-by: Takashi Iwai commit 1471b94ed36c9dc48359cdf0bf0d621b2e3ca169 Author: Pavel Hofman Date: Mon Mar 17 08:45:33 2008 +0100 [ALSA] some fixes and cleanup for ICE1724 cards * removing the hack with NON_AKM ak4xxx type * support for card-specific flags in ak4114_stats * definition of the flags for corresponding cards Signed-off-by: Pavel Hofman Signed-off-by: Takashi Iwai commit 8e368255384042654c599d18077fb20041a975ce Author: Joachim Foerster Date: Mon Mar 17 08:40:12 2008 +0100 [ALSA] [ML403-AC97CR] Remove duplicate snd_card_set_dev() We want to have snd_card_set_dev() in _probe(), but not a second one in snd_ml403_ac97cr_create(). Signed-off-by: Joachim Foerster Signed-off-by: Takashi Iwai commit cafcfce5a9a1a10b255b0e6bb492b72d01d67609 Author: Takashi Iwai Date: Fri Mar 14 17:17:09 2008 +0100 [ALSA] ice1724 - Fix the SPDIF input sample-rate on Juli@ AK4114 on Juli@ has the SPDIF input sample rate detection and causes errors when an incompatible sample rate is chosen. The patch adds the open hook to check the current rate and limit the hw constraints. Signed-off-by: Takashi Iwai commit af34a6d91865bf8e7bdbf4f79bfc429e714ad2a9 Author: Tony Vroon Date: Fri Mar 14 17:09:18 2008 +0100 [ALSA] hda-codec - Fujitsu Lifebook port replicator/dock headphone jack sense The docking station headphone output had no audio and jack sense was not considered. Jack information from the laptop itself and the dock are combined, as the dock does not obscure the connector. Signed-off-by: Tony Vroon Signed-off-by: Takashi Iwai commit f9d1bbb94d759b3153549bbadb9ed073b8402983 Author: Takashi Iwai Date: Fri Mar 14 15:52:20 2008 +0100 [ALSA] hda-intel - Fix power-off hang on ASUS P5AD2 The hda-intel driver has a problem at power-off on ASUS P5AD2. It's caused when the position-buffer is enabled -- most likely a hardware-specific problem. This patch adds a quirk to avoid the unnecessary enablement of position-buffer. Signed-off-by: Takashi Iwai commit 624a0a79231faab6a144d5a069d989f3ea8dcb05 Author: Herton Ronaldo Krzesinski Date: Fri Mar 14 12:52:59 2008 +0100 [ALSA] hda-codec - Map 3stack-6ch-dig ALC662 model for Asus P5GC-MX Map 3stack-6ch-dig ALC662 model for Asus P5GC-MX. Signed-off-by: Herton Ronaldo Krzesinski Signed-off-by: Takashi Iwai commit 101af73accf5ede05cc06c0b8550a9396ad3160b Author: Herton Ronaldo Krzesinski Date: Fri Mar 14 12:52:20 2008 +0100 [ALSA] hda-codec - Fix ALC662 DAC mixer mutes Currently ALC662 doesn't suport amp mute for AmpOut in nids 0x02, 0x03, 0x04 (see block diagram in ALC662 datasheet page 3, does M correspond to mute?). The result is that currently mute for "Front Playback Switch", "Surround Playback Switch", "Center Playback Switch" and "LFE Playback Switch" mixer items doesn't work (tested on Asus P5GC-MX motherboard with 3stack-6ch model). The solution I found for this is to mute the proper inputs in 0x0c, 0x0d, 0x0e audio mixers. Signed-off-by: Herton Ronaldo Krzesinski Signed-off-by: Takashi Iwai commit a616b2ebd14317a8761f0c37bd8663e5c21f4cff Author: Takashi Iwai Date: Fri Mar 14 09:18:32 2008 +0100 [ALSA] hda-codec - Fix orphan Headphone controls in STAC codecs Currently, the headphone controls are created as Master wrongly in some cases, and this prevents the virtual master controls. The patch fixes the problem by simply using "Headphone" always for headphone controls. Signed-off-by: Takashi Iwai commit edb3660fe66ef6f1b5fbf85dc04f3e816375fab6 Author: Matthew Ranostay Date: Fri Mar 14 08:46:51 2008 +0100 [ALSA] hda: 92HD73xxx distortion fix Fixed issue on some laptops that if the Master mixer and DAC mixers are turned all the way up that will cause distortion. This is fixed by limiting the max volume with the volume knob nid. Signed-off-by: Matthew Ranostay Signed-off-by: Takashi Iwai commit ddaa9d7c8768cc09415fffed413b158ab2a66ba9 Author: Stas Sergeev Date: Wed Mar 12 13:12:15 2008 +0100 [ALSA] pcsp: locking fix pcsp: locking fix. Signed-off-by: Stas Sergeev Signed-off-by: Takashi Iwai commit 0acafb78e7cf917a4e4fc22da7de709be99fc7b7 Author: Takashi Iwai Date: Wed Mar 12 12:51:09 2008 +0100 [ALSA] hda-codec - Improve ALC262 ultra model Improved ALC262 ultra model for Samsung Q1 Ultra series. - clean up mixers - support of input from HP jack as a mic - add quirk for Q1 EL Signed-off-by: Takashi Iwai commit f1cb858c82129da2c37e654754aa6bc9afca1e85 Author: Atsushi Nemoto Date: Tue Mar 11 08:15:30 2008 +0100 [ALSA] at73c213: remove redundant private_free routine snd_pcm_lib_preallocate_free_for_all() is called from snd_pcm_free() just after calling the private_free routine. So there should be no need to call it in driver's private_free routine. Signed-off-by: Atsushi Nemoto Signed-off-by: Andrew Morton Signed-off-by: Takashi Iwai commit 5dda4838452f69e762bee1afda26f88a6c523cca Author: Takashi Iwai Date: Mon Mar 10 12:19:12 2008 +0100 [ALSA] aw2 - Rename aw2-tsl.h to aw2-tsl.c aw2-tsl.h should be rather a C file to be included since it's referred only in aw2-saa6146.c and includes a table data. Signed-off-by: Takashi Iwai commit 633486dec6dafb445d477d361d86e4411a3d3a4b Author: Michael Gruber Date: Mon Mar 10 11:30:59 2008 +0100 [ALSA] hda-intel - Fix microphone capture with ALC880 F1734 model The default capture source should be the mic which is 0x01 on this model. In addition to that the change to VREF50 allows for higher capture volume. Signed-off-by: Michael Gruber Signed-off-by: Takashi Iwai commit e13c400bf0ce425f1be74e649257787272d999f2 Author: Matthew Ranostay Date: Mon Mar 10 11:30:04 2008 +0100 [ALSA] hda: Reorganized DAC outputs Changed so that internal speakers point to the Front mixer instead of Surround. Signed-off-by: Matthew Ranostay Signed-off-by: Takashi Iwai commit d77c33c691c9412a51f27a0cd41564d5299d5050 Author: Takashi Iwai Date: Mon Mar 10 11:21:30 2008 +0100 [ALSA] release 1.0.16 Signed-off-by: Takashi Iwai commit 94bea593fdb606e4264fd0f6addd30e132415cf9 Author: Atsushi Nemoto Date: Sat Mar 8 11:08:32 2008 +0100 [ALSA] at73c213: monaural support Add support for monaural playback to at73c213 driver. The sound will be apear on L-channel. Tested on AT91SAM9260-EK. Signed-off-by: Atsushi Nemoto Signed-off-by: Andrew Morton Signed-off-by: Takashi Iwai commit c8c85876439e96afca5e25ad81945f3aacf3de54 Author: Atsushi Nemoto Date: Sat Mar 8 11:07:26 2008 +0100 [ALSA] at73c213: fix error checking for clk API The clk_round_rate() and clk_set_rate() will return int, so not store thier return value to unsigned long variable. This bug hides real error on these API. Signed-off-by: Atsushi Nemoto Signed-off-by: Andrew Morton Signed-off-by: Takashi Iwai commit 9771eb9eb00790dec5896894873ccec0904b3298 Author: Tobin Davis Date: Fri Mar 7 11:57:51 2008 +0100 [ALSA] HDA Codecs: add support for Toshiba Equium L30 This patch adds support for the Toshiba Equium L30 laptop and renames the mixer controls to match Laptop usages. Signed-off-by: Tobin Davis Signed-off-by: Takashi Iwai commit 2b5ed369333721cb3e83436c615258701f0d3530 Author: Julia Lawall Date: Tue Mar 4 15:07:24 2008 -0800 sound: Use BUG_ON if (...) BUG(); should be replaced with BUG_ON(...) when the test has no side-effects to allow a definition of BUG_ON that drops the code completely. The semantic patch that makes this change is as follows: (http://www.emn.fr/x-info/coccinelle/) // @ disable unlikely @ expression E,f; @@ ( if (<... f(...) ...>) { BUG(); } | - if (unlikely(E)) { BUG(); } + BUG_ON(E); ) @@ expression E,f; @@ ( if (<... f(...) ...>) { BUG(); } | - if (E) { BUG(); } + BUG_ON(E); ) // Signed-off-by: Julia Lawall Signed-off-by: Andrew Morton commit 03e96fa17a58847b0542128c12f1b3e393a82f0d Author: Takashi Iwai Date: Thu Mar 6 16:58:35 2008 +0100 [ALSA] hda-codec - Add internal mic item for ALC268 acer model Added the internal mic as a capture source item for ALC268 acer model. Signed-off-by: Takashi Iwai Signed-off-by: Takashi Iwai commit d7e38d4892230eb4281752720fb89a3c5ceb1061 Author: Takashi Iwai Date: Thu Mar 6 16:58:17 2008 +0100 [ALSA] hda-codec - Fix dmics on ALC268 in auto configuration Fixed the handling of dmics on ALC268 in the auto-configuration mode. Signed-off-by: Takashi Iwai commit 108b41d210a8b1c853f22b044e81ff0d719203de Author: Peer Chen Date: Thu Mar 6 15:15:11 2008 +0100 [ALSA] hda_intel: Add the DIDs of nvidia MCP79 HD audio controller to hda_intel.c Add the Device IDs of nvidia MCP79 HD audio controller to hda_intel.c Signed-off-by: Peer Chen Signed-off-by: Takashi Iwai commit 14eb2db61f1a20d1e5915a27f3c9ec3b9d7f4ac0 Author: Jiang zhe Date: Thu Mar 6 11:09:09 2008 +0100 [ALSA] hda-codec - model for cx20549 to support laptop HP530 Currently the model laptop-hpsense use the 0x12 as ExtMic, and use 0x14 as Internal IntMic. But the hp530 only have one ExtMic, the Pin widget is 0x14. In this patch, I changed the mixer item for them. I still reserved the IntMic item, it will be helpful if other machine may use this model. ALSA bug#3821. Signed-off-by: Jiang zhe Signed-off-by: Takashi Iwai commit 258658a4cc121f04fa800717a81e241263785fb5 Author: Jiang zhe Date: Thu Mar 6 11:07:11 2008 +0100 [ALSA] hda-codec - model for alc883 to support FUJITSU Pi2515 There is no suitable model for Pi2515. This model is to support it. ALSA bug#3800. Signed-off-by: Jiang zhe Signed-off-by: Takashi Iwai commit cf7a373a7ba78406b797063e531dda5885f90bc1 Author: Stas Sergeev Date: Thu Mar 6 11:01:44 2008 +0100 [ALSA] pcsp: add description update ALSA-Configuration.txt Signed-off-by: Stas Sergeev Signed-off-by: Takashi Iwai commit a29f7e22bfa5f0279dcf832f7a1d5d4113efeef5 Author: Stas Sergeev Date: Thu Mar 6 11:01:16 2008 +0100 [ALSA] pcsp: improve "enable" option handling Simplify init code. Signed-off-by: Stas Sergeev Signed-off-by: Takashi Iwai commit 1381b0b3fb184a67919409ce67ffff6a0f614006 Author: Pascal Terjan Date: Tue Mar 4 11:33:28 2008 +0100 [ALSA] ALC288 - Add NEC S970 to the quirk table NEC S970 has no sound in the internal speakers when autodetection is used. With targa-dig model, there is sound in the speakers and it gets correctly muted when pluging headphones. Signed-off-by: Takashi Iwai commit 3053f60f6ebe8575ed171aaa44ed2ec36a6aaf2e Author: Stas Sergeev Date: Tue Mar 4 11:28:43 2008 +0100 [ALSA] pcsp - clean ups - make pcsp_start_timer_tasklet static - remove redundant includes. is not available on all platforms. Signed-off-by: Stas Sergeev Signed-off-by: Takashi Iwai commit ca805254455d623cbbd65af44e9b76bab2ccec41 Author: Jiang zhe Date: Tue Mar 4 11:20:33 2008 +0100 [ALSA] hda-codec - model for alc883 to support M720R There is no suitable model for M720R (ALSA bug#3781). This patch is to support HP jack-sensing and mixer. Signed-off-by: Jiang zhe Signed-off-by: Takashi Iwai commit 2bbad55122677812401a20557f42d7011f37509f Author: Takashi Iwai Date: Tue Mar 4 11:06:26 2008 +0100 [ALSA] aw2 - Remove endian dependency Removed unnecessary dependency on the little-endianess. Signed-off-by: Takashi Iwai commit 2c4a2ae692f59d26e5bb6bc5e166ddda30996d7c Author: Harvey Harrison Date: Mon Mar 3 15:32:18 2008 -0800 [ALSA] sound: replace remaining __FUNCTION__ occurences __FUNCTION__ is gcc-specific, use __func__ Signed-off-by: Harvey Harrison Signed-off-by: Takashi Iwai commit 36c052b7ac4734889eafda05909c619feb1c20a8 Author: Andrew Morton Date: Tue Mar 4 10:08:58 2008 +0100 [ALSA] hda_intel needs dma-mapping.h sparc32: sound/pci/hda/hda_intel.c: In function 'azx_create': sound/pci/hda/hda_intel.c:1838: error: 'DMA_64BIT_MASK' undeclared (first use in this function) sound/pci/hda/hda_intel.c:1838: error: (Each undeclared identifier is reported only once sound/pci/hda/hda_intel.c:1838: error: for each function it appears in.) Signed-off-by: Andrew Morton Signed-off-by: Takashi Iwai commit 5ed9ab2d4c289cd76bbb8a50b19e42385b476949 Author: Graeme Gregory Date: Mon Mar 3 17:19:45 2008 +0100 [ALSA] soc - Add Invert Switch for ROUT2 GTA02 device has a speaker between LOUT2 & ROUT2 and in this mode ROUT2 needs to be inverted. This patch adds a mixer control for this. Signed-off-by: Graeme Gregory Signed-off-by: Mark Brown Signed-off-by: Takashi Iwai commit fcb0c4b0e493588206f945b927bd3bda5bf1b875 Author: Stas Sergeev Date: Mon Mar 3 10:53:54 2008 +0100 [ALSA] Add PC-speaker sound driver Added PC-speaker sound driver (snd-pcsp). Signed-off-by: Stas Sergeev Signed-off-by: Takashi Iwai commit a0d875dc58e58f51c4130ad590d15b484792f124 Author: Takashi Iwai Date: Fri Feb 29 14:16:17 2008 +0100 [ALSA] hda-codec - Fix the array over-range access with stac92hd71bxx codec Add the check of the array range for dac_nids to prevent the over-range access. Signed-off-by: Takashi Iwai commit 21af229fd72e68d16383e6fea5b1f5e3d3b27772 Author: Pawel MOLL Date: Fri Feb 29 12:41:31 2008 +0100 [ALSA] IEC958 definitions for consumer status channel, byte 4 Added definition for byte 4 of SPDIF channel status, according to second edition of IEC 60958-3 (consumer) spec. Signed-off-by: Pawel MOLL Signed-off-by: Takashi Iwai commit 8d6cbdf85ff348b77496d70ba33b0c91e9b44e8d Author: Matthew Ranostay Date: Fri Feb 29 12:08:20 2008 +0100 [ALSA] hda: add verbs for 92hd73xxx laptops Added core_init[] for several 92hd73xxx laptops. Signed-off-by: Matthew Ranostay Signed-off-by: Takashi Iwai commit 9353e8434df3d9704b38164376a9ce01d4d36bcc Author: Matthew Ranostay Date: Fri Feb 29 12:07:43 2008 +0100 [ALSA] hda: disable power management on fixed ports Power management can't be enabled on fixed ports, since the presence will always return false and prevent output. Signed-off-by: Matthew Ranostay Signed-off-by: Takashi Iwai commit 0d776b1574f01bce3af6d0a9ea48bfba54411bdb Author: Harvey Harrison Date: Fri Feb 29 11:59:26 2008 +0100 [ALSA] sound: hda: missing includes of hda_patch.h Move the array declaration to hda_codec.c where it is used and add includes where the individual presets are declared. Fixes the following sparse warnings: sound/pci/hda/patch_realtek.c:13744:25: warning: symbol 'snd_hda_preset_realtek' was not declared. Should it be static? sound/pci/hda/patch_cmedia.c:729:25: warning: symbol 'snd_hda_preset_cmedia' was not declared. Should it be static? sound/pci/hda/patch_analog.c:3656:25: warning: symbol 'snd_hda_preset_analog' was not declared. Should it be static? sound/pci/hda/patch_sigmatel.c:3995:25: warning: symbol 'snd_hda_preset_sigmatel' was not declared. Should it be static? sound/pci/hda/patch_si3054.c:286:25: warning: symbol 'snd_hda_preset_si3054' was not declared. Should it be static? sound/pci/hda/patch_atihdmi.c:156:25: warning: symbol 'snd_hda_preset_atihdmi' was not declared. Should it be static? sound/pci/hda/patch_conexant.c:1721:25: warning: symbol 'snd_hda_preset_conexant' was not declared. Should it be static? sound/pci/hda/patch_via.c:1962:25: warning: symbol 'snd_hda_preset_via' was not declared. Should it be static? Signed-off-by: Harvey Harrison Signed-off-by: Takashi Iwai commit dd588147ca7d175c5918d3433fdc1ade39895752 Author: Takashi Iwai Date: Fri Feb 29 11:57:53 2008 +0100 [ALSA] hda-codec - Use int instead of long in patch_sigmatel.c The HD-audio parameters are at most 32bit int. Signed-off-by: Takashi Iwai commit 3d08598188bc536960b9416fa1a2702d8deeef7b Author: Harvey Harrison Date: Fri Feb 29 11:56:48 2008 +0100 [ALSA] sound: patch_sigmatel.c fix shadowed variable warning Temp variable in the loop shadows the second argument (which is otherwise unused in this function). Change this to defcfg as it is used to hold the default config. sound/pci/hda/patch_sigmatel.c:2759:18: warning: symbol 'cfg' shadows an earlier one sound/pci/hda/patch_sigmatel.c:2734:26: originally declared here Signed-off-by: Harvey Harrison Signed-off-by: Takashi Iwai commit 809b7c7cdede6c551982b4531191c91c4f6dc01e Author: Harvey Harrison Date: Fri Feb 29 11:54:49 2008 +0100 [ALSA] sound: hdspm.c fix returning void expression warnings Just drop the returns. sound/pci/rme9652/hdspm.c:1031:3: warning: returning void-valued expression sound/pci/rme9652/hdspm.c:1033:3: warning: returning void-valued expression Signed-off-by: Harvey Harrison Signed-off-by: Takashi Iwai commit 2b8a9a79eee8192b1137f878ce65fc98ca73310e Author: Harvey Harrison Date: Fri Feb 29 11:54:26 2008 +0100 [ALSA] sound: riptide.c fix shadowed variable warnings In both cases we are passing around the substream number, use sub_num for this. sound/pci/riptide/riptide.c:1633:6: warning: symbol 'index' shadows an earlier one sound/pci/riptide/riptide.c:121:12: originally declared here sound/pci/riptide/riptide.c:1673:6: warning: symbol 'index' shadows an earlier one sound/pci/riptide/riptide.c:121:12: originally declared here Signed-off-by: Harvey Harrison Signed-off-by: Takashi Iwai commit 4b1a4d2b03f49b8049491d057717f3dc0096c095 Author: Harvey Harrison Date: Fri Feb 29 11:53:59 2008 +0100 [ALSA] sound: pcxhr_core.c fix shadowed variable warning Inner loop redeclares err with u32 rather than int, stupid fix here is to change the inner err to err2. sound/pci/pcxhr/pcxhr_core.c:1008:8: warning: symbol 'err' shadows an earlier one sound/pci/pcxhr/pcxhr_core.c:983:6: originally declared here Signed-off-by: Harvey Harrison Signed-off-by: Takashi Iwai commit f69656c224b8449f9205a98edcdfb4b481eff43b Author: Harvey Harrison Date: Fri Feb 29 11:46:57 2008 +0100 [ALSA] sound: virtuoso.c fix shadowed variable warning Use priv_idx as an identifier. sound/pci/oxygen/virtuoso.c:277:15: warning: symbol 'index' shadows an earlier one sound/pci/oxygen/virtuoso.c:56:12: originally declared here Signed-off-by: Harvey Harrison Signed-off-by: Takashi Iwai commit 1a1480134a0ca799135e72e4d53b5e111624aa17 Author: Harvey Harrison Date: Fri Feb 29 11:52:50 2008 +0100 [ALSA] sound: ice1712.c fix shadowed variable warnings In all four case, adding a private value to the iooff index, call it priv_idx. sound/pci/ice1712/ice1712.c:1300:6: warning: symbol 'index' shadows an earlier one sound/pci/ice1712/ice1712.c:85:12: originally declared here sound/pci/ice1712/ice1712.c:1312:6: warning: symbol 'index' shadows an earlier one sound/pci/ice1712/ice1712.c:85:12: originally declared here sound/pci/ice1712/ice1712.c:1338:6: warning: symbol 'index' shadows an earlier one sound/pci/ice1712/ice1712.c:85:12: originally declared here sound/pci/ice1712/ice1712.c:1350:6: warning: symbol 'index' shadows an earlier one sound/pci/ice1712/ice1712.c:85:12: originally declared here [tiwai - fixed coding issues as well] Signed-off-by: Harvey Harrison Signed-off-by: Takashi Iwai commit 610f5764e825b0056051f749bf3523a0fda3b9e0 Author: Harvey Harrison Date: Fri Feb 29 11:44:57 2008 +0100 [ALSA] sound: emu10k1x.c fix shadowed variable warnings enable in these contexts refers specifically to intr enable, as per the two functions it is found in. Use intr_enable instead. sound/pci/emu10k1/emu10k1x.c:330:15: warning: symbol 'enable' shadows an earlier one sound/pci/emu10k1/emu10k1x.c:53:12: originally declared here sound/pci/emu10k1/emu10k1x.c:341:15: warning: symbol 'enable' shadows an earlier one sound/pci/emu10k1/emu10k1x.c:53:12: originally declared here instead of shadowing, use cap_voice as we test for the capture voice in this statement. sound/pci/emu10k1/emu10k1x.c:798:25: warning: symbol 'pvoice' shadows an earlier one sound/pci/emu10k1/emu10k1x.c:787:24: originally declared here Signed-off-by: Harvey Harrison Signed-off-by: Takashi Iwai commit e1e360cba68960290a3ca8dcb2b7fec39c976776 Author: Harvey Harrison Date: Fri Feb 29 11:44:26 2008 +0100 [ALSA] sound: emuproc.c fix signedness warning Reading regs from the fpga into an int instead of a u32, trivial fix. sound/pci/emu10k1/emuproc.c:422:34: warning: incorrect type in argument 3 (different signedness) sound/pci/emu10k1/emuproc.c:422:34: expected unsigned int [usertype] *value sound/pci/emu10k1/emuproc.c:422:34: got int * Signed-off-by: Harvey Harrison Signed-off-by: Takashi Iwai commit ce9e8627067727ef458adfb427bd5f12102e5b25 Author: Harvey Harrison Date: Fri Feb 29 11:41:56 2008 +0100 [ALSA] sound: au88x0_pcm.c fix integer as NULL pointer warning sound/pci/au88x0/au88x0_pcm.c:508:15: warning: Using plain integer as NULL pointer Also some small codingstyle fixes. Signed-off-by: Harvey Harrison Signed-off-by: Takashi Iwai commit 8ee5fb36fb93a6711ce84e6bd42caa8daa63e5c5 Author: Ahmet İnan Date: Thu Feb 28 12:46:32 2008 +0100 [ALSA] snd-dummy - better realtime app support when the time interval for a period is smaller than kernel HZ, then snd-aloop and snd-dummy cannot call snd_pcm_period_elapsed as fast enough annymore. this happens for example with games. but the app still needs to see, that the buffer actually did go further, which is provided by these patches. Signed-off-by: Ahmet İnan mathematik.uni-freiburg.de> Signed-off-by: Takashi Iwai commit ffdbd5feb8522b3937312ad2d805785b92f02925 Author: Harvey Harrison Date: Thu Feb 28 12:02:56 2008 +0100 [ALSA] sound: ca0106_mixer.c fix shadowed variable warnings Change the variable err to _err within the ADD_CTLS macro to avoid shadowing the local variable. sound/pci/ca0106/ca0106_mixer.c:710:2: warning: symbol 'err' shadows an earlier one sound/pci/ca0106/ca0106_mixer.c:663:6: originally declared here sound/pci/ca0106/ca0106_mixer.c:712:3: warning: symbol 'err' shadows an earlier one sound/pci/ca0106/ca0106_mixer.c:663:6: originally declared here sound/pci/ca0106/ca0106_mixer.c:721:3: warning: symbol 'err' shadows an earlier one sound/pci/ca0106/ca0106_mixer.c:663:6: originally declared here Signed-off-by: Harvey Harrison Signed-off-by: Takashi Iwai commit 7f71708cfee3bc41b756071990fcd4bf6a91559a Author: Harvey Harrison Date: Thu Feb 28 12:02:22 2008 +0100 [ALSA] sound: ca0106_main.c fix shadowed variable warnings change to intr_enable as per the two functions it is defined in. sound/pci/ca0106/ca0106_main.c:438:15: warning: symbol 'enable' shadows an earlier one sound/pci/ca0106/ca0106_main.c:159:12: originally declared here sound/pci/ca0106/ca0106_main.c:449:15: warning: symbol 'enable' shadows an earlier one sound/pci/ca0106/ca0106_main.c:159:12: originally declared here Signed-off-by: Harvey Harrison Signed-off-by: Takashi Iwai commit 40de0039a52bf821ee7296640246fb4d6c330d5e Author: Harvey Harrison Date: Thu Feb 28 12:00:48 2008 +0100 [ALSA] sound: ali5451.c fix shadowed variable warnings enable is used to test for whether or not spdif should be enabled, change to spdif_enable. sound/pci/ali5451/ali5451.c:1812:15: warning: symbol 'enable' shadows an earlier one sound/pci/ali5451/ali5451.c:63:12: originally declared here sound/pci/ali5451/ali5451.c:1840:27: warning: symbol 'enable' shadows an earlier one sound/pci/ali5451/ali5451.c:63:12: originally declared here Signed-off-by: Harvey Harrison Signed-off-by: Takashi Iwai commit 6eb12bb7c07713baaaf8175e743cf2a07623c7b4 Author: Harvey Harrison Date: Thu Feb 28 11:58:18 2008 +0100 [ALSA] sound: ac97_pcm.c fix shadowed variable warning err is always assigned before it is used, no need to declare another inside the if statement. sound/pci/ac97/ac97_pcm.c:577:7: warning: symbol 'err' shadows an earlier one sound/pci/ac97/ac97_pcm.c:572:6: originally declared here Signed-off-by: Harvey Harrison Signed-off-by: Takashi Iwai commit a6f2ccca5c5df00c604fa5c7f9eeb717a0a92983 Author: Harvey Harrison Date: Thu Feb 28 11:57:47 2008 +0100 [ALSA] sound: rme96.c fix integer as NULL pointer warning kernel style does assignment outside of if() block sound/pci/rme96.c:1562:71: warning: Using plain integer as NULL pointer Signed-off-by: Harvey Harrison Signed-off-by: Takashi Iwai commit 676ac0436363e858424fbf86e0b440208640841f Author: Harvey Harrison Date: Thu Feb 28 11:57:23 2008 +0100 [ALSA] sound: rme32.c fix integer as NULL pointer warning kernel style does assignment outside of if() statements. sound/pci/rme32.c:1353:71: warning: Using plain integer as NULL pointer Signed-off-by: Harvey Harrison Signed-off-by: Takashi Iwai commit 9aa9352c80c737c358cd077141f1f9b595f62a57 Author: Harvey Harrison Date: Thu Feb 28 11:56:37 2008 +0100 [ALSA] sound: maestro3.c fix shadowed variable warnings change id to elem_id as it is used to initialize each mixer element sound/pci/maestro3.c:2071:25: warning: symbol 'id' shadows an earlier one sound/pci/maestro3.c:67:13: originally declared here index is used in each of these places to count over the dsp's memory, change to the name dsp_index sound/pci/maestro3.c:2572:9: warning: symbol 'index' shadows an earlier one sound/pci/maestro3.c:66:12: originally declared here sound/pci/maestro3.c:2604:9: warning: symbol 'index' shadows an earlier one sound/pci/maestro3.c:66:12: originally declared here [tiwai - fixed coding style issues as well] Signed-off-by: Harvey Harrison Signed-off-by: Takashi Iwai commit 1b941b08bb3bd3045488de5156fcd1854fa8f600 Author: Harvey Harrison Date: Thu Feb 28 11:55:07 2008 +0100 [ALSA] sound: fm801.c fix shadowed variable warning id was only used as a counter in a for loop, move the declaration to where it is used and change it to i. sound/pci/fm801.c:1288:6: warning: symbol 'id' shadows an earlier one sound/pci/fm801.c:51:13: originally declared here [tiwai - fixed a coding style issue as well] Signed-off-by: Harvey Harrison Signed-off-by: Takashi Iwai commit a07b2fd2466f26d30a7e51a1c220e346ddc3ee7a Author: Harvey Harrison Date: Thu Feb 28 11:53:41 2008 +0100 [ALSA] sound: es1968.c fox shadowed variable warning id is used when initializing the mixer elements, use elem_id here instead. sound/pci/es1968.c:1963:25: warning: symbol 'id' shadows an earlier one sound/pci/es1968.c:129:13: originally declared here Signed-off-by: Harvey Harrison Signed-off-by: Takashi Iwai commit 345ff3f915c602239cf99e0fdd3f8482a7b32ed9 Author: Harvey Harrison Date: Thu Feb 28 11:53:07 2008 +0100 [ALSA] sound: ens1370.c fix shadowed variable warning index is incremented only when AC97_EI_SPDIF and then assigned to the index field. Change the temporary name to is_spdif. sound/pci/ens1370.c:1638:10: warning: symbol 'index' shadows an earlier one sound/pci/ens1370.c:84:12: originally declared here Signed-off-by: Harvey Harrison Signed-off-by: Takashi Iwai commit 7e5e7b88b559e0cc2f09a9926d2caed2ce1ad4cd Author: Harvey Harrison Date: Thu Feb 28 11:52:17 2008 +0100 [ALSA] sound: cmipci.c fix shadowed variable warning A temporary variable for each mixer element is used in an initialization loop, use the name elem_id. sound/pci/cmipci.c:2747:26: warning: symbol 'id' shadows an earlier one sound/pci/cmipci.c:56:13: originally declared here [tiwai - fixed a coding style issue as well] Signed-off-by: Harvey Harrison Signed-off-by: Takashi Iwai commit 3f65c65f86bea26a57c240f74a07954f4ba33fb0 Author: Mark Brown Date: Tue Feb 26 13:16:08 2008 +0100 [ALSA] soc - Report errors from snd_soc_dapm_set_endpoint() Signed-off-by: Mark Brown Signed-off-by: Takashi Iwai commit 03a72ac44920794f38095d5ecc56d88806a83239 Author: Takashi Iwai Date: Tue Feb 26 11:56:35 2008 +0100 [ALSA] hda-codec - Add docking-station mic input for Thinkpad X61 Added the docking-stationc mic input to the capture source list for Thinkpad X61. Signed-off-by: Takashi Iwai commit 7b7f1339e3135ec0ba31cb82e8aa29dc3f4ac42e Author: Takashi Iwai Date: Mon Feb 25 18:26:41 2008 +0100 [ALSA] hda-codec - Fix initial DAC numbers of 92HD71bxx codecs Fix the initial num_dacs of 92HD71bxx codecs. Signed-off-by: Takashi Iwai commit 99695dfab5c6e38f5b65045ccdffd4e58af572e3 Author: Clemens Ladisch Date: Mon Feb 25 11:04:19 2008 +0100 [ALSA] usb-audio: sort quirks list Move some entries to their proper position so that the list is again sorted by vendor/product ID. Signed-off-by: Clemens Ladisch commit aa349e9cc8ca136db6e234c969fcb86c3cd8480a Author: Clemens Ladisch Date: Mon Feb 25 10:59:52 2008 +0100 [ALSA] mpu401: reduce tx loop timeout Reduce the number of times to check for a non-empty Tx FIFO from 100 to 2 because there is no MPU-401 implementation that needs more than one or two reads to determine the actual FIFO status. Signed-off-by: Clemens Ladisch commit 9c0d110605cd47ad8d7595dcd50d1aeff11f99de Author: Remy Bruno Date: Fri Feb 22 17:57:02 2008 +0100 [ALSA] hdsp - RME 9632 fix at 192kHz The bits indicating SPDIF frequency in the status register are not the same for the 9632 than for the other cards, because it also supports 192kHz. A specific bitmask has thus been added (used in hdsp_spdif_sample_rate()). The 9632 does not seem to report external sample rates greater than 96kHz. In this case, the best seems to report spdif rate when autosync reference is spdif. This also required to move function hdsp_spdif_sample_rate(). Signed-off-by: Remy Bruno Signed-off-by: Takashi Iwai commit 628c96acef04dca25f651817e2cb6877243cf9d5 Author: Matthew Ranostay Date: Fri Feb 22 17:55:05 2008 +0100 [ALSA] hda: Mic as output fix Added logic to check if AUTO_PIN_FRONT_MIC is available for output switch, if AUTO_PIN_MIC isn't. Signed-off-by: Matthew Ranostay Signed-off-by: Takashi Iwai commit 44488ea29f8a94adacdca399595f0c3b98fc3b62 Author: Takashi Iwai Date: Thu Feb 21 14:11:09 2008 +0100 [ALSA] hda-codec - Add missing descriptions for STAC codec models Added the missing descriptions for STAC codec models. Signed-off-by: Takashi Iwai commit b3c44dc847096663e9f148053b2871a4d7be3d90 Author: Takashi Iwai Date: Thu Feb 21 12:40:00 2008 +0100 [ALSA] seq-oss - Remove invalid BUG() Removed invalid BUG() - the driver should handle the error case properly rather than issuing BUG(). Signed-off-by: Takashi Iwai commit fd931e07a5648be7a96d555215be2bbfd3cdb104 Author: Takashi Iwai Date: Thu Feb 21 08:13:11 2008 +0100 [ALSA] hda-intel - Use PCI_DEVICE() macro Clean up the pci id table using PCI_DEVICE() macro. Signed-off-by: Takashi Iwai commit 31dd9eb4ecbf5f4e0aed69cbd1435540a7ddbf43 Author: Ahmet İnan Date: Thu Feb 21 07:55:30 2008 +0100 [ALSA] snd-dummy - improved timing, silence on prepare Signed-off-by: Ahmet İnan mathematik.uni-freiburg.de> Signed-off-by: Takashi Iwai commit c55617d51a75651a2525e4768c73e2f8b1fd982e Author: Matthew Ranostay Date: Thu Feb 21 07:51:46 2008 +0100 [ALSA] hda: STAC927x analog mic Some laptops have a internal analog microphone that is not setup by the BIOS. Signed-off-by: Matthew Ranostay Signed-off-by: Takashi Iwai commit 196cf28dae1b38814d76612aed18dcbb15b41f18 Author: Matthew Ranostay Date: Thu Feb 21 07:51:14 2008 +0100 [ALSA] hda: 92HDxxxx PCI Quirks Added PCI_QUIRKS for laptop that have the 92HDxxx family of codecs. Signed-off-by: Matthew Ranostay Signed-off-by: Takashi Iwai commit d429fa9e82769e6708f02790504593a1bc493174 Author: Matthew Ranostay Date: Thu Feb 21 07:50:12 2008 +0100 [ALSA] hda: STAC927x invalid association value STAC_DELL_BIOS quirks were setting the association value wrong for port 0x0f, which prevented it from being included in hp_outs[]. Signed-off-by: Matthew Ranostay Signed-off-by: Takashi Iwai commit a8ad3b707b1934bef444360a346368e938b4ff6b Author: Matthew Ranostay Date: Thu Feb 21 07:49:31 2008 +0100 [ALSA] hda: fix STAC927x power management Fix issue on STAC927x codecs that first DAC was getting powered down even if was being used. Signed-off-by: Matthew Ranostay Signed-off-by: Takashi Iwai commit dd5b1dc5368fbceb350586a95a332815131e4f81 Author: Jarkko Nikula Date: Wed Feb 20 17:13:44 2008 +0100 [ALSA] ASoC: Add support for 12 MHz MCLK in TLV320AIC3X Signed-off-by: Jarkko Nikula Signed-off-by: Takashi Iwai commit 6df8f412f838d800b872cb03229acbe3675951ba Author: Takashi Iwai Date: Wed Feb 20 12:13:29 2008 +0100 [ALSA] Add description of aw2 driver Added a brief description of aw2 driver to ALSA-Configuration.txt. Signed-off-by: Takashi Iwai commit 21d04d101f9a5b32d1136808058b305bf5d9cd2b Author: Takashi Iwai Date: Wed Feb 20 12:12:58 2008 +0100 [ALSA] aw2 - Add missing module parameters Added the missing declarations for module parameters. Signed-off-by: Takashi Iwai commit 5652a3d959a493e0035f95ba21392650c95ef1b1 Author: Cedric Bregardis Date: Wed Feb 20 12:05:13 2008 +0100 [ALSA] Emagic Audiowerk 2 ALSA driver. Signed-off-by: Cedric Bregardis Signed-off-by: Jean-Christian Hassler Signed-off-by: Takashi Iwai commit 997c5d40a7e7214ff988ddaaefc3eea762a42922 Author: Takashi Iwai Date: Tue Feb 19 15:03:57 2008 +0100 [ALSA] hda-codec - Don't create multiple capture streams for single inputs When the device has only one input source, it makes no sense to have multiple capture streams. Signed-off-by: Takashi Iwai commit 31788fbb234252bd352f8431056dc0179cb054aa Author: Takashi Iwai Date: Tue Feb 19 15:00:15 2008 +0100 [ALSA] hda-codec - Fix ALC268 capture source Initialize the capture source properly for auto model. It's especially important for cases that only mic is detected. Signed-off-by: Takashi Iwai commit d6c9493448e98e4779c1bef8cdec348adaf7eef3 Author: Takashi Iwai Date: Tue Feb 19 13:16:41 2008 +0100 [ALSA] hda-codec - Add beep volume control to ALC268 Added the beep volume control to ALC268 codec support code. Since the codec doesn't return the correct AMP caps, we need to override the value. Signed-off-by: Takashi Iwai commit fe9ad9ac8685cdb720a6ee27767d2edd4fb47457 Author: Kailang Yang Date: Tue Feb 19 11:38:05 2008 +0100 [ALSA] hda-codec - Fix ALC662 recording Fixed ALC662 recording issue. Signed-off-by: Kailang Yang Signed-off-by: Takashi Iwai commit c24d1ff09aa3021adce12f06452de2e0bcef5026 Author: Takashi Iwai Date: Tue Feb 19 11:36:35 2008 +0100 [ALSA] hda-intel - Clean up stream definitions Clean up the code to define playback/capture streams. Signed-off-by: Takashi Iwai commit f43db174747644c5e2479b8e3dad6280e0293d28 Author: Takashi Iwai Date: Mon Feb 18 13:06:49 2008 +0100 [ALSA] ca0106 - Add master volume controls Added master volume and switch controls for ca0106 using vmaster. Signed-off-by: Takashi Iwai commit 7ffb2d74331d77e76beec7f11a3f1bf8bb90d7b9 Author: Takashi Iwai Date: Mon Feb 18 13:05:50 2008 +0100 [ALSA] Keep private TLV entry in vmaster itself Use a private array for TLV entries of virtual master controls instead of (supposed) static array. This cleans up the existing codes. Also, now vmaster assumes the simple dB-range TLV that is the only type it can handle. Signed-off-by: Takashi Iwai commit f7b70fc4818d2ef78b9942b4bb823e83b68ed0f2 Author: Takashi Iwai Date: Mon Feb 18 13:03:13 2008 +0100 [ALSA] Move vmaster code to sound core Move the codes for virtual master controls to sound core part so that not only hda-intel drivers can use it. Signed-off-by: Takashi Iwai commit 4a31c2027849d44c5226f259528cdb6bb3b02ee5 Author: Takashi Iwai Date: Mon Feb 18 12:23:13 2008 +0100 [ALSA] intel8x0 - Add support of 8 channel sound Added the support of 8 channel sound for codecs that are known to work. So far, only ALC850 is marked as a 8ch-support codec. This fix is a modified version of the patch on ALSA BTS#2097 by Martin Ellis: https://bugtrack.alsa-project.org/alsa-bug/view.php?id=2097 Signed-off-by: Takashi Iwai commit 32fec3331383b5b773d4dc43ff1d3d81d3124ca7 Author: Hans-Christian Egtvedt Date: Mon Feb 18 11:44:56 2008 +0100 [ALSA] Add __devinit macro to at73c213 sound driver probe functions This patch adds __devinit to the functions used when probing. Will also reduce the memory footprint a bit if CONFIG_HOTPLUG is not enabled. Signed-off-by: Hans-Christian Egtvedt Signed-off-by: Takashi Iwai commit deed0ea37aa1f8db2550c449d4aa8c465230ca49 Author: Vladimir Barinov Date: Mon Feb 18 11:40:22 2008 +0100 [ALSA] Davinci ASoC support Add ASoC support for the TI Davinci SoC and the Davicni-EVM reference board. It includes: - ASoC Davinci DMA driver - ASoC Davinci I2S (Davinci McBSP module based) driver - ASoC Davinci-EVM reference board Signed-off-by: Vladimir Barinov Signed-off-by: Takashi Iwai commit 36b9940bb30c6ff8debcc886074447cb221e41d6 Author: Takashi Iwai Date: Sat Feb 16 09:44:56 2008 +0100 [ALSA] hda-codec - Add model=mobile for AD1884A & co Added the new model mobile for AD1884A and compatible codecs. It's a reduced version of model=laptop. Signed-off-by: Takashi Iwai commit 74b583415a1cfa91a69b0d1aabbcbf5c9a17e7bc Author: Takashi Iwai Date: Sat Feb 16 09:43:56 2008 +0100 [ALSA] hda-codec - Add support of AD1883/1884A/1984A/1984B Added the support of new AD codecs: AD1883, AD1884A, AD1984A and AD1984B. These are almost compatible except for additional digital pins, etc. Signed-off-by: Takashi Iwai commit 32524af90e49cd2d45cd72fc1cd2b831b6835b93 Author: Liam Girdwood Date: Fri Feb 15 16:43:11 2008 +0100 [ALSA] ASoC: WM9713 driver This patch adds an ASoC driver for the WM9713 AC97 codec. Signed-off-by: Liam Girdwood Signed-off-by: Mark Brown Signed-off-by: Takashi Iwai commit b60ac36fbcb80fbfecdef85c19833c2c7d15820a Author: Takashi Iwai Date: Thu Feb 14 17:27:17 2008 +0100 [ALSA] hda-codec - Fix missing capsrc_nids for ALC262 ALC262 must have capsrc_nids defined as well as in ALC882. Also, add a NULL check in alc882_mux_enum_put to avoid Oops. Signed-off-by: Takashi Iwai commit db93a7c2fe28a537cccb034f6f2e82e79a9df967 Author: Libin Yang Date: Thu Feb 14 12:55:13 2008 +0100 [ALSA] HDA-Intel - Patch to support RV7xx HDMI Audio This patch is to add R7xx HDMI audio support. Signed-off-by: Libin Yang Signed-off-by: Takashi Iwai commit 7629d603e72d33ba5e5ca428f3c767794938af86 Author: Takashi Iwai Date: Wed Feb 13 17:19:35 2008 +0100 [ALSA] hda-codec - Fix breakage of resume in auto-config of realtek codecs The last patch for fixing the auto-config pin setting breaks the resume due to a wrong use of snd_hda_codec_amp_stereo(). The code in the init hook shouldn't touch the amp cache. Signed-off-by: Takashi Iwai commit fb5c6b770633d76b19463648727dade8eb7d8b58 Author: Takashi Iwai Date: Wed Feb 13 16:59:29 2008 +0100 [ALSA] hda-codec - Add more names to vendor list Added more known names to the vendor id list. Signed-off-by: Takashi Iwai commit 589a1fece35bbd27040fb5ebab328862d426e641 Author: Takashi Iwai Date: Tue Feb 12 18:37:26 2008 +0100 [ALSA] hda-codec - Add "IEC958 Default PCM" switch Added a new mixer switch to enable/disable the sharing of the default PCM stream with analog and SPDIF outputs. When "IEC958 Default PCM" switch is on, the PCM stream is routed both to analog and SPDIF outputs. This is the behavior in the earlier version. Turning this switch off has a merit for some codecs, though. Some codec chips don't support 24bit formats for SPDIF but only for analog outputs. In this case, you can use 24bit format by disabling this switch. Signed-off-by: Takashi Iwai commit 74c260357234cff9f5938ae741e4f3614249dea8 Author: Takashi Iwai Date: Tue Feb 12 18:32:23 2008 +0100 [ALSA] hda-codec - Fix auto-configuration of Realtek codecs This patch fixes some bugs in the auto-configurator of Realtek codecs: - add missing pin set-up for speaker pins - fix the speaker auto-mute function not to conflict with the existing "Speaker" mixer switch Signed-off-by: Takashi Iwai commit fa82b9d3cc6411f611cd3349ffb1f8e3b2b5a5d6 Author: Takashi Iwai Date: Tue Feb 12 18:30:12 2008 +0100 [ALSA] hda-codec - More fix-up for auto-configuration In some cases, the BIOS sets up only the HP pins with different assoc and sequence numbers, e.g. on FSC Esprimo with ALC262. This patch adds a fix-up for such a case. When multiple HPs are defined and no line-outs is found, the configurator tries to re-assign some pins from HP list to line-out, judging from the sequence number. Signed-off-by: Takashi Iwai commit 238f14b56a7b63e6f7e4fb1c4da83e213c17608c Author: Takashi Iwai Date: Tue Feb 12 12:11:36 2008 +0100 [ALSA] hda-codec - Implement auto-mic jack sensing on Samsung laptops Implemented the auto-mic jack sensing for Samsung laptops with AD1986A codec chip (model=laptop-eapd). The hardware uses pin 0x1d and 0x1f for the internal and external mics, respectively. Signed-off-by: Takashi Iwai commit 68072555cb9b9da7e32a370bffe94986b81ca9df Author: Takashi Iwai Date: Mon Feb 11 18:32:32 2008 +0100 [ALSA] hda-codec - Clean up capture source selection of Realtek codecs Clean up the codes of the capture source selection for Realtek codecs. Now using common helper functions with the new capsrc_nids field. Signed-off-by: Takashi Iwai commit 520a638e640e169369996f431734e9580cf5347f Author: Takashi Iwai Date: Mon Feb 11 15:54:34 2008 +0100 [ALSA] hda-codec - Fix automute of AD1981HD hp model Reprogram the speaker-pin setting at each HP pin plug to make sure the spekaer auto-muting on AD1981HD hp model. Signed-off-by: Takashi Iwai commit 474c24feb97b43e3a766a5cd6ef4de8aae1265aa Author: Takashi Iwai Date: Mon Feb 11 14:52:36 2008 +0100 [ALSA] hda-codec - Fix ALC880 F1734 model Fixed some issues with ALC880 F1734 model - fix capture via mic - enable volume-wheel control Signed-off-by: Takashi Iwai commit 9f5215a9dc25b65e2910a768184ae35f3e532e72 Author: Pavel Hofman Date: Mon Feb 11 14:48:06 2008 +0100 [ALSA] AK4114 - listing regs in proc A simple patch for listing AK4114 regs in proc. Signed-off-by: Pavel Hofman Signed-off-by: Takashi Iwai commit 6ad3512ada2538d61c9ea2c110037aa6542cee38 Author: Jonathan Woithe Date: Fri Feb 8 12:44:17 2008 +0100 [ALSA] hda-codec - remove duplicate controls in alc268 test mixer I've just noticed that there are a handful of duplicate controls in the ALC268 test model mixer. This patch (against alsa-driver 1.0.16) removes them. Signed-off-by: Jonathan Woithe Signed-off-by: Takashi Iwai commit 8629d2e62f880f6c78b64be874b4eaec00e1443c Author: Takashi Iwai Date: Thu Feb 7 17:12:01 2008 +0100 [ALSA] hda-codec - Correct HDMI transmitter names Give better names to the new HDMI transmitter chips. Signed-off-by: Takashi Iwai commit b5a01fcba4c1057ce5e1de153845322b8a004c63 Author: Takashi Iwai Date: Thu Feb 7 12:06:32 2008 +0100 [ALSA] hda-intel - Fix a compile error with CONFIG_SND_DEBUG_DETECT=y Forgot to get rid of the obsolete fragsize field from a debug print. Signed-off-by: Takashi Iwai commit ad0f94df3d149eebe155aeab5c836ad93b22eaa8 Author: Jaroslav Kysela Date: Wed Feb 6 20:04:49 2008 +0100 [ALSA] ice1712 - added support for M-Audio Delta 66E See ALSA bug#3327 for more details. Experimental. Also fix support for M-Audio Delta 1010E - subdevice check. Signed-off-by: Jaroslav Kysela commit 618165aee87478cc4a066b4bb4af135686401abb Author: Jaroslav Kysela Date: Wed Feb 6 15:48:06 2008 +0100 [ALSA] Added support for Delta1010E (newer revisions of Delta1010) For more details, see ALSA bug#3327 . Signed-off-by: Jaroslav Kysela commit 24e74dc52d7ed7331f23dfb7576e01767cbd33d2 Author: Takashi Iwai Date: Wed Feb 6 15:05:57 2008 +0100 [ALSA] hda-intel - Support 64bit buffer allocation The HD-audio hardware usually supports 64bit address for DMA and other buffers. The patch enables the feature if supported. Signed-off-by: Takashi Iwai commit b68e6a26bc7e439bc3ab096d59c6c54f61da5241 Author: Takashi Iwai Date: Wed Feb 6 14:50:19 2008 +0100 [ALSA] hda-intel - Use SG buffer Use SG buffers for the HD-audio instead of linear buffers. Signed-off-by: Takashi Iwai commit c410d58c70b89ddfd2e2c0c70e834a095c137046 Author: Matthew Ranostay Date: Wed Feb 6 14:49:44 2008 +0100 [ALSA] hda: STAC927x power down inactive DACs On several laptops that have STAC9228 codecs have unused DACs, this powers them down to a D3 state. Signed-off-by: Matthew Ranostay Signed-off-by: Takashi Iwai commit 8b27d908183e5956d25737426ba503ed8e86a232 Author: Alan Horstmann Date: Wed Feb 6 14:43:54 2008 +0100 [ALSA] ice1712 - Fix hoontech MIDI input Fixes the problems with Midi In on Hoontech/STA dsp24 cards, for example with DSP2000 box, without restricting the box configurations available. Also adds mpu_401 name strings. Signed-off-by: Alan Horstmann Signed-off-by: Takashi Iwai commit 3001a5fbc3b4db875a3958bd41282505ace3c4ca Author: Takashi Iwai Date: Wed Feb 6 14:41:59 2008 +0100 [ALSA] hda-codec - Add ID for an unknown HDMI codec chip Added the ID for an unknown HDMI codec chip on Jetway J9F2. Signed-off-by: Takashi Iwai commit 620b66126f30f4fddcc7bd306dd36f74fe326c3c Author: Takashi Iwai Date: Wed Feb 6 14:03:20 2008 +0100 [ALSA] hda-intel - Fix PCM device number assignment In the current scheme, PCM device numbers are assigned incrementally in the order of codecs. This causes problems when the codec number is irregular, e.g. codec #0 for HDMI and codec #1 for analog. Then the HDMI becomes the first PCM, which is picked up as the default output device. Unfortuantely this doesn't work well with normal setups. This patch introduced the fixed device numbers for the PCM types, namely, analog, SPDIF, HDMI and modem. The PCM devices are assigned according to the corresponding PCM type. After this patch, HDMI will be always assigned to PCM #3, SPDIF to PCM #1, and the first analog to PCM #0, etc. Signed-off-by: Takashi Iwai commit afadb87651b6223794c8fd1cad5dac1fd28bf24a Author: Takashi Iwai Date: Mon Feb 4 12:44:11 2008 +0100 [ALSA] Add more fallbacks to OSS PHONEOUT mixer map Added more fallbacks to OSS PHONEOUT mixer mapping. This corresponds to the speaker output in general, so now "Mono" and "Speaker" are assigned. Signed-off-by: Takashi Iwai commit e159c21106c2da7cee7c674e3a2116f2e2878fd2 Author: Takashi Iwai Date: Mon Feb 4 12:36:32 2008 +0100 [ALSA] ice1724 - Add ADC setup in set_rate callback for Audiophile192 Added the missing GPIO setup for the AK5385A ADC codec on Audiophile192. Signed-off-by: Takashi Iwai commit 6a672a1bbfed2481375b5bb860a6f43644433ce3 Author: Takashi Iwai Date: Mon Feb 4 12:34:59 2008 +0100 [ALSA] ice1724 - Enable AK4114 support for Audiophile192 Fixed and enabled the support of AK4114 chip on Audiophile192. Signed-off-by: Takashi Iwai commit c0c4bf5d6c216dc3707c263156526366ae24ae7f Author: Mirco Tischler Date: Mon Feb 4 12:33:59 2008 +0100 [ALSA] hda-codec - Add support of Zepto laptops Adds support for zepto laptops with alc268 intel_hda codec. Signed-off-by: Mirco Tischler Signed-off-by: Takashi Iwai commit ee0c4958ac93af0b8f2fc5bce7635f6139dcb9fc Author: Takashi Iwai Date: Mon Feb 4 12:32:20 2008 +0100 [ALSA] hda-codec - Add SI HDMI codec support Added the support of SI HDMI codec, found in ASUS machines. ALSA bug#3654 https://bugtrack.alsa-project.org/alsa-bug/view.php?id=3654 Signed-off-by: Takashi Iwai commit ee30f87178444d796f3ddf7a750d7b2d70b5245e Author: Takashi Iwai Date: Mon Feb 4 12:31:13 2008 +0100 [ALSA] hda-codec - Allow multiple SPDIF devices The current code doesn't allow multiple SPDIF devices, and causes errors when multiple SPDIF devices are found (e.g. SPDIF out and HDMI). This patch allows multiple SPDIF devices by incrementing the index automatically. Signed-off-by: Takashi Iwai commit e2c96dd1b85d4cef5db9f9969440a87bdc96f27f Author: Tobin Davis Date: Sun Feb 3 20:31:47 2008 +0100 [ALSA] HDA - Add support for the OQO Model 2 This patch adds support for the OQO Model 2 Ultra Mobile PC. Signed-off-by: Tobin Davis Signed-off-by: Takashi Iwai commit 8557b1c1f7f32068fc57886bd365415fb959cf5e Author: Kristoffer Ericson Date: Fri Feb 1 13:16:10 2008 +0100 [ALSA] Add SUPERH depends to sound/soc/sh/Kconfig Currently you will see an empty "SoC Audio support for SuperH" menu when building for other archs (example pxa). This patch adds "depends on SUPERH" to remove that empty menu. Signed-off-by: Kristoffer Ericson Signed-off-by: Takashi Iwai commit b014f5ecdce9e82eafdf35762ec0ff6a0c00cafd Author: Mike Montour Date: Fri Feb 1 13:12:12 2008 +0100 [ALSA] soc - Mono voice playback volume for WM8753 Voice playback volume is in register bits 0:2, not 4:6. From: Mike Montour Signed-off-by: Mark Brown Cc: Werner Almesberger Signed-off-by: Takashi Iwai Documentation/sound/alsa/ALSA-Configuration.txt | 47 +- include/sound/ac97_codec.h | 1 + include/sound/ak4114.h | 1 + include/sound/ak4xxx-adda.h | 2 +- include/sound/asoundef.h | 8 + include/sound/control.h | 7 + include/sound/core.h | 10 +- include/sound/version.h | 4 +- sound/arm/pxa2xx-ac97.c | 10 +- sound/core/Kconfig | 4 + sound/core/Makefile | 1 + sound/core/init.c | 38 +- sound/core/misc.c | 4 +- sound/core/oss/mixer_oss.c | 2 + sound/core/seq/oss/seq_oss_synth.c | 9 +- sound/core/vmaster.c | 371 +++++++ sound/drivers/Kconfig | 17 + sound/drivers/Makefile | 2 +- sound/drivers/dummy.c | 37 +- sound/drivers/ml403-ac97cr.c | 6 +- sound/drivers/mpu401/mpu401_uart.c | 13 +- sound/drivers/pcsp/Makefile | 2 + sound/drivers/pcsp/pcsp.c | 239 +++++ sound/drivers/pcsp/pcsp.h | 82 ++ sound/drivers/pcsp/pcsp_input.c | 116 ++ sound/drivers/pcsp/pcsp_input.h | 14 + sound/drivers/pcsp/pcsp_lib.c | 338 ++++++ sound/drivers/pcsp/pcsp_mixer.c | 143 +++ sound/i2c/other/ak4114.c | 24 +- sound/i2c/other/ak4xxx-adda.c | 16 +- sound/isa/sb/sb16_csp.c | 28 +- sound/isa/sb/sb_common.c | 6 +- sound/oss/trident.c | 12 +- sound/oss/trident.h | 2 +- sound/oss/vwsnd.c | 6 +- sound/pci/Kconfig | 22 +- sound/pci/Makefile | 1 + sound/pci/ac97/ac97_patch.c | 46 +- sound/pci/ac97/ac97_pcm.c | 1 - sound/pci/ad1889.c | 4 +- sound/pci/ali5451/ali5451.c | 28 +- sound/pci/als300.c | 4 +- sound/pci/au88x0/au88x0_pcm.c | 10 +- sound/pci/aw2/Makefile | 3 + sound/pci/aw2/aw2-alsa.c | 794 ++++++++++++++ sound/pci/aw2/aw2-saa7146.c | 465 ++++++++ sound/pci/aw2/aw2-saa7146.h | 105 ++ sound/pci/aw2/aw2-tsl.c | 110 ++ sound/pci/aw2/saa7146.h | 168 +++ sound/pci/azt3328.c | 4 +- sound/pci/ca0106/ca0106_main.c | 16 +- sound/pci/ca0106/ca0106_mixer.c | 59 +- sound/pci/cmipci.c | 11 +- sound/pci/emu10k1/emu10k1x.c | 22 +- sound/pci/emu10k1/emuproc.c | 2 +- sound/pci/ens1370.c | 6 +- sound/pci/es1968.c | 39 +- sound/pci/fm801.c | 8 +- sound/pci/hda/Makefile | 2 +- sound/pci/hda/hda_codec.c | 201 ++++- sound/pci/hda/hda_codec.h | 13 +- sound/pci/hda/hda_generic.c | 4 +- sound/pci/hda/hda_intel.c | 454 +++++--- sound/pci/hda/hda_local.h | 20 +- sound/pci/hda/hda_patch.h | 28 - sound/pci/hda/patch_analog.c | 578 ++++++++++- sound/pci/hda/patch_atihdmi.c | 8 + sound/pci/hda/patch_cmedia.c | 13 +- sound/pci/hda/patch_conexant.c | 68 ++- sound/pci/hda/patch_realtek.c | 1247 ++++++++++++++--------- sound/pci/hda/patch_si3054.c | 4 +- sound/pci/hda/patch_sigmatel.c | 391 ++++++-- sound/pci/hda/patch_via.c | 14 +- sound/pci/hda/vmaster.c | 364 ------- sound/pci/ice1712/delta.c | 22 +- sound/pci/ice1712/delta.h | 2 + sound/pci/ice1712/hoontech.c | 21 +- sound/pci/ice1712/ice1712.c | 40 +- sound/pci/ice1712/ice1712.h | 9 + sound/pci/ice1712/ice1724.c | 321 +++--- sound/pci/ice1712/juli.c | 486 ++++++++- sound/pci/ice1712/pontis.c | 4 +- sound/pci/ice1712/prodigy192.c | 33 +- sound/pci/ice1712/revo.c | 55 +- sound/pci/intel8x0.c | 30 +- sound/pci/maestro3.c | 34 +- sound/pci/oxygen/cs4362a.h | 69 ++ sound/pci/oxygen/cs4398.h | 69 ++ sound/pci/oxygen/hifier.c | 36 +- sound/pci/oxygen/oxygen.c | 129 +-- sound/pci/oxygen/oxygen.h | 23 +- sound/pci/oxygen/oxygen_io.c | 23 +- sound/pci/oxygen/oxygen_lib.c | 109 ++- sound/pci/oxygen/oxygen_mixer.c | 217 ++++- sound/pci/oxygen/oxygen_pcm.c | 78 +- sound/pci/oxygen/pcm1796.h | 58 + sound/pci/oxygen/virtuoso.c | 594 +++++++---- sound/pci/oxygen/wm8785.h | 45 + sound/pci/pcxhr/pcxhr.c | 7 +- sound/pci/pcxhr/pcxhr_core.c | 33 +- sound/pci/riptide/riptide.c | 14 +- sound/pci/rme32.c | 3 +- sound/pci/rme96.c | 3 +- sound/pci/rme9652/hdsp.c | 54 +- sound/pci/rme9652/hdspm.c | 19 +- sound/pci/ymfpci/ymfpci_main.c | 4 +- sound/ppc/awacs.c | 265 ++++-- sound/ppc/awacs.h | 21 +- sound/ppc/burgundy.c | 465 +++++++-- sound/ppc/burgundy.h | 31 +- sound/ppc/pmac.c | 10 +- sound/soc/Kconfig | 1 + sound/soc/Makefile | 2 +- sound/soc/codecs/Kconfig | 4 + sound/soc/codecs/Makefile | 2 + sound/soc/codecs/cs4270.c | 2 +- sound/soc/codecs/tlv320aic3x.c | 22 + sound/soc/codecs/wm8753.c | 5 +- sound/soc/codecs/wm9712.c | 8 - sound/soc/codecs/wm9713.c | 1300 +++++++++++++++++++++++ sound/soc/codecs/wm9713.h | 53 + sound/soc/davinci/Kconfig | 19 + sound/soc/davinci/Makefile | 11 + sound/soc/davinci/davinci-evm.c | 208 ++++ sound/soc/davinci/davinci-i2s.c | 407 +++++++ sound/soc/davinci/davinci-i2s.h | 17 + sound/soc/davinci/davinci-pcm.c | 389 +++++++ sound/soc/davinci/davinci-pcm.h | 29 + sound/soc/fsl/fsl_dma.c | 1 - sound/soc/fsl/fsl_ssi.c | 1 - sound/soc/pxa/pxa2xx-ac97.c | 8 +- sound/soc/s3c24xx/s3c24xx-i2s.c | 20 +- sound/soc/s3c24xx/s3c24xx-pcm.c | 28 +- sound/soc/sh/Kconfig | 1 + sound/soc/soc-core.c | 2 + sound/soc/soc-dapm.c | 7 +- sound/spi/at73c213.c | 44 +- sound/usb/caiaq/caiaq-audio.c | 81 +- sound/usb/caiaq/caiaq-device.c | 4 +- sound/usb/usbaudio.c | 98 +- sound/usb/usbquirks.h | 75 +- 141 files changed, 10767 insertions(+), 2477 deletions(-) diff --git a/Documentation/sound/alsa/ALSA-Configuration.txt b/Documentation/sound/alsa/ALSA-Configuration.txt index e985cf5..fd4c32a 100644 --- a/Documentation/sound/alsa/ALSA-Configuration.txt +++ b/Documentation/sound/alsa/ALSA-Configuration.txt @@ -284,6 +284,13 @@ Prior to version 0.9.0rc4 options had a 'snd_' prefix. This was removed. control correctly. If you have problems regarding this, try another ALSA compliant mixer (alsamixer works). + Module snd-aw2 + -------------- + + Module for Audiowerk2 sound card + + This module supports multiple cards. + Module snd-azt2320 ------------------ @@ -818,19 +825,25 @@ Prior to version 0.9.0rc4 options had a 'snd_' prefix. This was removed. hippo_1 Hippo (Benq) with jack detection sony-assamd Sony ASSAMD ultra Samsung Q1 Ultra Vista model + lenovo-3000 Lenovo 3000 y410 basic fixed pin assignment w/o SPDIF auto auto-config reading BIOS (default) - ALC268 + ALC267/268 + quanta-il1 Quanta IL1 mini-notebook 3stack 3-stack model toshiba Toshiba A205 acer Acer laptops dell Dell OEM laptops (Vostro 1200) + zepto Zepto laptops test for testing/debugging purpose, almost all controls can adjusted. Appearing only when compiled with $CONFIG_SND_DEBUG=y auto auto-config reading BIOS (default) + ALC269 + basic Basic preset + ALC662 3stack-dig 3-stack (2-channel) with SPDIF 3stack-6ch 3-stack (6-channel) @@ -871,10 +884,11 @@ Prior to version 0.9.0rc4 options had a 'snd_' prefix. This was removed. lenovo-nb0763 Lenovo NB0763 lenovo-ms7195-dig Lenovo MS7195 haier-w66 Haier W66 - 6stack-hp HP machines with 6stack (Nettle boards) 3stack-hp HP machines with 3stack (Lucknow, Samba boards) 6stack-dell Dell machines with 6stack (Inspiron 530) mitac Mitac 8252D + clevo-m720 Clevo M720 laptop series + fujitsu-pi2515 Fujitsu AMILO Pi2515 auto auto-config reading BIOS (default) ALC861/660 @@ -911,6 +925,12 @@ Prior to version 0.9.0rc4 options had a 'snd_' prefix. This was removed. 3stack 3-stack mode (default) 6stack 6-stack mode + AD1884A / AD1883 / AD1984A / AD1984B + desktop 3-stack desktop (default) + laptop laptop with HP jack sensing + mobile mobile devices with HP jack sensing + thinkpad Lenovo Thinkpad X300 + AD1884 N/A @@ -936,7 +956,7 @@ Prior to version 0.9.0rc4 options had a 'snd_' prefix. This was removed. laptop-automute 2-channel with EAPD and HP-automute (Lenovo N100) ultra 2-channel with EAPD (Samsung Ultra tablet PC) - AD1988 + AD1988/AD1988B/AD1989A/AD1989B 6stack 6-jack 6stack-dig ditto with SPDIF 3stack 3-jack @@ -979,6 +999,7 @@ Prior to version 0.9.0rc4 options had a 'snd_' prefix. This was removed. dell-m26 Dell Inspiron 1501 dell-m27 Dell Inspiron E1705/9400 gateway Gateway laptops with EAPD control + panasonic Panasonic CF-74 STAC9205/9254 ref Reference board @@ -1017,6 +1038,16 @@ Prior to version 0.9.0rc4 options had a 'snd_' prefix. This was removed. 3stack D965 3stack 5stack D965 5stack + SPDIF dell-3stack Dell Dimension E520 + dell-bios Fixes with Dell BIOS setup + + STAC92HD71B* + ref Reference board + dell-m4-1 Dell desktops + dell-m4-2 Dell desktops + + STAC92HD73* + ref Reference board + dell-m6 Dell desktops STAC9872 vaio Setup for VAIO FE550G/SZ110 @@ -1590,6 +1621,16 @@ Prior to version 0.9.0rc4 options had a 'snd_' prefix. This was removed. Power management is _not_ supported. + Module snd-pcsp + ----------------- + + Module for internal PC-Speaker. + + nforce_wa - enable NForce chipset workaround. Expect bad sound. + + This module supports system beeps, some kind of PCM playback and + even a few mixer controls. + Module snd-pcxhr ---------------- diff --git a/include/sound/ac97_codec.h b/include/sound/ac97_codec.h index 0148058..049edc5 100644 --- a/include/sound/ac97_codec.h +++ b/include/sound/ac97_codec.h @@ -397,6 +397,7 @@ #define AC97_HAS_NO_TONE (1<<16) /* no Tone volume */ #define AC97_HAS_NO_STD_PCM (1<<17) /* no standard AC97 PCM volume and mute */ #define AC97_HAS_NO_AUX (1<<18) /* no standard AC97 AUX volume and mute */ +#define AC97_HAS_8CH (1<<19) /* supports 8-channel output */ /* rates indexes */ #define AC97_RATES_FRONT_DAC 0 diff --git a/include/sound/ak4114.h b/include/sound/ak4114.h index 4e80d3f..d293d36 100644 --- a/include/sound/ak4114.h +++ b/include/sound/ak4114.h @@ -182,6 +182,7 @@ struct ak4114 { unsigned char rcs0; unsigned char rcs1; struct delayed_work work; + unsigned int check_flags; void *change_callback_private; void (*change_callback)(struct ak4114 *ak4114, unsigned char c0, unsigned char c1); }; diff --git a/include/sound/ak4xxx-adda.h b/include/sound/ak4xxx-adda.h index 6153b91..891cf1a 100644 --- a/include/sound/ak4xxx-adda.h +++ b/include/sound/ak4xxx-adda.h @@ -68,7 +68,7 @@ struct snd_akm4xxx { enum { SND_AK4524, SND_AK4528, SND_AK4529, SND_AK4355, SND_AK4358, SND_AK4381, - SND_AK5365, NON_AKM + SND_AK5365 } type; /* (array) information of combined codecs */ diff --git a/include/sound/asoundef.h b/include/sound/asoundef.h index 024ce62..a6e0fac 100644 --- a/include/sound/asoundef.h +++ b/include/sound/asoundef.h @@ -112,6 +112,14 @@ #define IEC958_AES3_CON_CLOCK_1000PPM (0<<4) /* 1000 ppm */ #define IEC958_AES3_CON_CLOCK_50PPM (1<<4) /* 50 ppm */ #define IEC958_AES3_CON_CLOCK_VARIABLE (2<<4) /* variable pitch */ +#define IEC958_AES4_CON_MAX_WORDLEN_24 (1<<0) /* 0 = 20-bit, 1 = 24-bit */ +#define IEC958_AES4_CON_WORDLEN (7<<1) /* mask - sample word length */ +#define IEC958_AES4_CON_WORDLEN_NOTID (0<<1) /* not indicated */ +#define IEC958_AES4_CON_WORDLEN_20_16 (1<<1) /* 20-bit or 16-bit */ +#define IEC958_AES4_CON_WORDLEN_22_18 (2<<1) /* 22-bit or 18-bit */ +#define IEC958_AES4_CON_WORDLEN_23_19 (4<<1) /* 23-bit or 19-bit */ +#define IEC958_AES4_CON_WORDLEN_24_20 (5<<1) /* 24-bit or 20-bit */ +#define IEC958_AES4_CON_WORDLEN_21_17 (6<<1) /* 21-bit or 17-bit */ /***************************************************************************** * * diff --git a/include/sound/control.h b/include/sound/control.h index e79baa6..3dc1291 100644 --- a/include/sound/control.h +++ b/include/sound/control.h @@ -169,4 +169,11 @@ int snd_ctl_boolean_mono_info(struct snd_kcontrol *kcontrol, int snd_ctl_boolean_stereo_info(struct snd_kcontrol *kcontrol, struct snd_ctl_elem_info *uinfo); +/* + * virtual master control + */ +struct snd_kcontrol *snd_ctl_make_virtual_master(char *name, + const unsigned int *tlv); +int snd_ctl_add_slave(struct snd_kcontrol *master, struct snd_kcontrol *slave); + #endif /* __SOUND_CONTROL_H */ diff --git a/include/sound/core.h b/include/sound/core.h index 4fc0235..695ee53 100644 --- a/include/sound/core.h +++ b/include/sound/core.h @@ -277,8 +277,8 @@ int snd_minor_info_done(void); int snd_minor_info_oss_init(void); int snd_minor_info_oss_done(void); #else -#define snd_minor_info_oss_init() /*NOP*/ -#define snd_minor_info_oss_done() /*NOP*/ +static inline int snd_minor_info_oss_init(void) { return 0; } +static inline int snd_minor_info_oss_done(void) { return 0; } #endif /* memory.c */ @@ -310,7 +310,7 @@ int snd_card_file_add(struct snd_card *card, struct file *file); int snd_card_file_remove(struct snd_card *card, struct file *file); #ifndef snd_card_set_dev -#define snd_card_set_dev(card,devptr) ((card)->dev = (devptr)) +#define snd_card_set_dev(card, devptr) ((card)->dev = (devptr)) #endif /* device.c */ @@ -373,7 +373,7 @@ void snd_verbose_printd(const char *file, int line, const char *format, ...) * snd_printd - debug printk * @fmt: format string * - * Compiled only when Works like snd_printk() for debugging purpose. + * Works like snd_printk() for debugging purposes. * Ignored when CONFIG_SND_DEBUG is not set. */ #define snd_printd(fmt, args...) \ @@ -417,7 +417,7 @@ void snd_verbose_printd(const char *file, int line, const char *format, ...) * snd_printdd - debug printk * @format: format string * - * Compiled only when Works like snd_printk() for debugging purpose. + * Works like snd_printk() for debugging purposes. * Ignored when CONFIG_SND_DEBUG_DETECT is not set. */ #define snd_printdd(format, args...) snd_printk(format, ##args) diff --git a/include/sound/version.h b/include/sound/version.h index fac66c4..ed6fb2e 100644 --- a/include/sound/version.h +++ b/include/sound/version.h @@ -1,3 +1,3 @@ /* include/version.h. Generated by alsa/ksync script. */ -#define CONFIG_SND_VERSION "1.0.16rc2" -#define CONFIG_SND_DATE " (Thu Jan 31 16:40:16 2008 UTC)" +#define CONFIG_SND_VERSION "1.0.16" +#define CONFIG_SND_DATE "" diff --git a/sound/arm/pxa2xx-ac97.c b/sound/arm/pxa2xx-ac97.c index 5d86e68..dc870f3 100644 --- a/sound/arm/pxa2xx-ac97.c +++ b/sound/arm/pxa2xx-ac97.c @@ -66,7 +66,7 @@ static unsigned short pxa2xx_ac97_read(struct snd_ac97 *ac97, unsigned short reg if (wait_event_timeout(gsr_wq, (GSR | gsr_bits) & GSR_SDONE, 1) <= 0 && !((GSR | gsr_bits) & GSR_SDONE)) { printk(KERN_ERR "%s: read error (ac97_reg=%d GSR=%#lx)\n", - __FUNCTION__, reg, GSR | gsr_bits); + __func__, reg, GSR | gsr_bits); val = -1; goto out; } @@ -98,7 +98,7 @@ static void pxa2xx_ac97_write(struct snd_ac97 *ac97, unsigned short reg, unsigne if (wait_event_timeout(gsr_wq, (GSR | gsr_bits) & GSR_CDONE, 1) <= 0 && !((GSR | gsr_bits) & GSR_CDONE)) printk(KERN_ERR "%s: write error (ac97_reg=%d GSR=%#lx)\n", - __FUNCTION__, reg, GSR | gsr_bits); + __func__, reg, GSR | gsr_bits); mutex_unlock(&car_mutex); } @@ -125,7 +125,7 @@ static void pxa2xx_ac97_reset(struct snd_ac97 *ac97) if (!((GSR | gsr_bits) & (GSR_PCR | GSR_SCR))) { printk(KERN_INFO "%s: cold reset timeout (GSR=%#lx)\n", - __FUNCTION__, gsr_bits); + __func__, gsr_bits); /* let's try warm reset */ gsr_bits = 0; @@ -144,7 +144,7 @@ static void pxa2xx_ac97_reset(struct snd_ac97 *ac97) if (!((GSR | gsr_bits) & (GSR_PCR | GSR_SCR))) printk(KERN_INFO "%s: warm reset timeout (GSR=%#lx)\n", - __FUNCTION__, gsr_bits); + __func__, gsr_bits); } GCR &= ~(GCR_PRIRDY_IEN|GCR_SECRDY_IEN); @@ -391,6 +391,7 @@ static struct platform_driver pxa2xx_ac97_driver = { .resume = pxa2xx_ac97_resume, .driver = { .name = "pxa2xx-ac97", + .owner = THIS_MODULE, }, }; @@ -410,3 +411,4 @@ module_exit(pxa2xx_ac97_exit); MODULE_AUTHOR("Nicolas Pitre"); MODULE_DESCRIPTION("AC97 driver for the Intel PXA2xx chip"); MODULE_LICENSE("GPL"); +MODULE_ALIAS("platform:pxa2xx-ac97"); diff --git a/sound/core/Kconfig b/sound/core/Kconfig index 829ca38..a8d71c6 100644 --- a/sound/core/Kconfig +++ b/sound/core/Kconfig @@ -181,3 +181,7 @@ config SND_PCM_XRUN_DEBUG It is usually not required, but if you have trouble with sound clicking when system is loaded, it may help to determine the process or driver which causes the scheduling gaps. + +config SND_VMASTER + bool + depends on SND diff --git a/sound/core/Makefile b/sound/core/Makefile index 267039a..da8e685 100644 --- a/sound/core/Makefile +++ b/sound/core/Makefile @@ -6,6 +6,7 @@ snd-y := sound.o init.o memory.o info.o control.o misc.o device.o snd-$(CONFIG_ISA_DMA_API) += isadma.o snd-$(CONFIG_SND_OSSEMUL) += sound_oss.o info_oss.o +snd-$(CONFIG_SND_VMASTER) += vmaster.o snd-pcm-objs := pcm.o pcm_native.o pcm_lib.o pcm_timer.o pcm_misc.o \ pcm_memory.o diff --git a/sound/core/init.c b/sound/core/init.c index e3338d6..ac05734 100644 --- a/sound/core/init.c +++ b/sound/core/init.c @@ -254,7 +254,7 @@ static int snd_disconnect_release(struct inode *inode, struct file *file) if (likely(df)) return df->disconnected_f_op->release(inode, file); - panic("%s(%p, %p) failed!", __FUNCTION__, inode, file); + panic("%s(%p, %p) failed!", __func__, inode, file); } static unsigned int snd_disconnect_poll(struct file * file, poll_table * wait) @@ -311,6 +311,9 @@ int snd_card_disconnect(struct snd_card *card) struct file *file; int err; + if (!card) + return -EINVAL; + spin_lock(&card->files_lock); if (card->shutdown) { spin_unlock(&card->files_lock); @@ -322,6 +325,7 @@ int snd_card_disconnect(struct snd_card *card) /* phase 1: disable fops (user space) operations for ALSA API */ mutex_lock(&snd_card_mutex); snd_cards[card->number] = NULL; + snd_cards_lock &= ~(1 << card->number); mutex_unlock(&snd_card_mutex); /* phase 2: replace file->f_op with special dummy operations */ @@ -360,6 +364,15 @@ int snd_card_disconnect(struct snd_card *card) snd_printk(KERN_ERR "not all devices for card %i can be disconnected\n", card->number); snd_info_card_disconnect(card); +#ifndef CONFIG_SYSFS_DEPRECATED + if (card->card_dev) { + device_unregister(card->card_dev); + card->card_dev = NULL; + } +#endif +#ifdef CONFIG_PM + wake_up(&card->power_sleep); +#endif return 0; } @@ -401,33 +414,14 @@ static int snd_card_do_free(struct snd_card *card) snd_printk(KERN_WARNING "unable to free card info\n"); /* Not fatal error */ } -#ifndef CONFIG_SYSFS_DEPRECATED - if (card->card_dev) - device_unregister(card->card_dev); -#endif kfree(card); return 0; } -static int snd_card_free_prepare(struct snd_card *card) -{ - if (card == NULL) - return -EINVAL; - (void) snd_card_disconnect(card); - mutex_lock(&snd_card_mutex); - snd_cards[card->number] = NULL; - snd_cards_lock &= ~(1 << card->number); - mutex_unlock(&snd_card_mutex); -#ifdef CONFIG_PM - wake_up(&card->power_sleep); -#endif - return 0; -} - int snd_card_free_when_closed(struct snd_card *card) { int free_now = 0; - int ret = snd_card_free_prepare(card); + int ret = snd_card_disconnect(card); if (ret) return ret; @@ -447,7 +441,7 @@ EXPORT_SYMBOL(snd_card_free_when_closed); int snd_card_free(struct snd_card *card) { - int ret = snd_card_free_prepare(card); + int ret = snd_card_disconnect(card); if (ret) return ret; diff --git a/sound/core/misc.c b/sound/core/misc.c index 102d1c3..38524f6 100644 --- a/sound/core/misc.c +++ b/sound/core/misc.c @@ -39,7 +39,7 @@ void snd_verbose_printk(const char *file, int line, const char *format, ...) { va_list args; - if (format[0] == '<' && format[1] >= '0' && format[1] <= '9' && format[2] == '>') { + if (format[0] == '<' && format[1] >= '0' && format[1] <= '7' && format[2] == '>') { char tmp[] = "<0>"; tmp[1] = format[1]; printk("%sALSA %s:%d: ", tmp, file, line); @@ -60,7 +60,7 @@ void snd_verbose_printd(const char *file, int line, const char *format, ...) { va_list args; - if (format[0] == '<' && format[1] >= '0' && format[1] <= '9' && format[2] == '>') { + if (format[0] == '<' && format[1] >= '0' && format[1] <= '7' && format[2] == '>') { char tmp[] = "<0>"; tmp[1] = format[1]; printk("%sALSA %s:%d: ", tmp, file, line); diff --git a/sound/core/oss/mixer_oss.c b/sound/core/oss/mixer_oss.c index 75daed2..581aa2c 100644 --- a/sound/core/oss/mixer_oss.c +++ b/sound/core/oss/mixer_oss.c @@ -1257,6 +1257,8 @@ static void snd_mixer_oss_build(struct snd_mixer_oss *mixer) { SOUND_MIXER_DIGITAL3, "Digital", 2 }, { SOUND_MIXER_PHONEIN, "Phone", 0 }, { SOUND_MIXER_PHONEOUT, "Master Mono", 0 }, + { SOUND_MIXER_PHONEOUT, "Speaker", 0 }, /*fallback*/ + { SOUND_MIXER_PHONEOUT, "Mono", 0 }, /*fallback*/ { SOUND_MIXER_PHONEOUT, "Phone", 0 }, /* fallback */ { SOUND_MIXER_VIDEO, "Video", 0 }, { SOUND_MIXER_RADIO, "Radio", 0 }, diff --git a/sound/core/seq/oss/seq_oss_synth.c b/sound/core/seq/oss/seq_oss_synth.c index ab570a0..558dadb 100644 --- a/sound/core/seq/oss/seq_oss_synth.c +++ b/sound/core/seq/oss/seq_oss_synth.c @@ -245,8 +245,13 @@ snd_seq_oss_synth_setup(struct seq_oss_devinfo *dp) info->nr_voices = rec->nr_voices; if (info->nr_voices > 0) { info->ch = kcalloc(info->nr_voices, sizeof(struct seq_oss_chinfo), GFP_KERNEL); - if (!info->ch) - BUG(); + if (!info->ch) { + snd_printk(KERN_ERR "Cannot malloc\n"); + rec->oper.close(&info->arg); + module_put(rec->oper.owner); + snd_use_lock_free(&rec->use_lock); + continue; + } reset_channels(info); } debug_printk(("synth %d assigned\n", i)); diff --git a/sound/core/vmaster.c b/sound/core/vmaster.c new file mode 100644 index 0000000..4cc57f9 --- /dev/null +++ b/sound/core/vmaster.c @@ -0,0 +1,371 @@ +/* + * Virtual master and slave controls + * + * Copyright (c) 2008 by Takashi Iwai + * + * This program is free software; you can redistribute it and/or + * modify it under the terms of the GNU General Public License as + * published by the Free Software Foundation, version 2. + * + */ + +#include +#include +#include +#include + +/* + * a subset of information returned via ctl info callback + */ +struct link_ctl_info { + int type; /* value type */ + int count; /* item count */ + int min_val, max_val; /* min, max values */ +}; + +/* + * link master - this contains a list of slave controls that are + * identical types, i.e. info returns the same value type and value + * ranges, but may have different number of counts. + * + * The master control is so far only mono volume/switch for simplicity. + * The same value will be applied to all slaves. + */ +struct link_master { + struct list_head slaves; + struct link_ctl_info info; + int val; /* the master value */ + unsigned int tlv[4]; +}; + +/* + * link slave - this contains a slave control element + * + * It fakes the control callbacsk with additional attenuation by the + * master control. A slave may have either one or two channels. + */ + +struct link_slave { + struct list_head list; + struct link_master *master; + struct link_ctl_info info; + int vals[2]; /* current values */ + struct snd_kcontrol slave; /* the copy of original control entry */ +}; + +/* get the slave ctl info and save the initial values */ +static int slave_init(struct link_slave *slave) +{ + struct snd_ctl_elem_info *uinfo; + struct snd_ctl_elem_value *uctl; + int err, ch; + + if (slave->info.count) + return 0; /* already initialized */ + + uinfo = kmalloc(sizeof(*uinfo), GFP_KERNEL); + if (!uinfo) + return -ENOMEM; + uinfo->id = slave->slave.id; + err = slave->slave.info(&slave->slave, uinfo); + if (err < 0) { + kfree(uinfo); + return err; + } + slave->info.type = uinfo->type; + slave->info.count = uinfo->count; + if (slave->info.count > 2 || + (slave->info.type != SNDRV_CTL_ELEM_TYPE_INTEGER && + slave->info.type != SNDRV_CTL_ELEM_TYPE_BOOLEAN)) { + snd_printk(KERN_ERR "invalid slave element\n"); + kfree(uinfo); + return -EINVAL; + } + slave->info.min_val = uinfo->value.integer.min; + slave->info.max_val = uinfo->value.integer.max; + kfree(uinfo); + + uctl = kmalloc(sizeof(*uctl), GFP_KERNEL); + if (!uctl) + return -ENOMEM; + uctl->id = slave->slave.id; + err = slave->slave.get(&slave->slave, uctl); + for (ch = 0; ch < slave->info.count; ch++) + slave->vals[ch] = uctl->value.integer.value[ch]; + kfree(uctl); + return 0; +} + +/* initialize master volume */ +static int master_init(struct link_master *master) +{ + struct link_slave *slave; + + if (master->info.count) + return 0; /* already initialized */ + + list_for_each_entry(slave, &master->slaves, list) { + int err = slave_init(slave); + if (err < 0) + return err; + master->info = slave->info; + master->info.count = 1; /* always mono */ + /* set full volume as default (= no attenuation) */ + master->val = master->info.max_val; + return 0; + } + return -ENOENT; +} + +static int slave_get_val(struct link_slave *slave, + struct snd_ctl_elem_value *ucontrol) +{ + int err, ch; + + err = slave_init(slave); + if (err < 0) + return err; + for (ch = 0; ch < slave->info.count; ch++) + ucontrol->value.integer.value[ch] = slave->vals[ch]; + return 0; +} + +static int slave_put_val(struct link_slave *slave, + struct snd_ctl_elem_value *ucontrol) +{ + int err, ch, vol; + + err = master_init(slave->master); + if (err < 0) + return err; + + switch (slave->info.type) { + case SNDRV_CTL_ELEM_TYPE_BOOLEAN: + for (ch = 0; ch < slave->info.count; ch++) + ucontrol->value.integer.value[ch] &= + !!slave->master->val; + break; + case SNDRV_CTL_ELEM_TYPE_INTEGER: + for (ch = 0; ch < slave->info.count; ch++) { + /* max master volume is supposed to be 0 dB */ + vol = ucontrol->value.integer.value[ch]; + vol += slave->master->val - slave->master->info.max_val; + if (vol < slave->info.min_val) + vol = slave->info.min_val; + else if (vol > slave->info.max_val) + vol = slave->info.max_val; + ucontrol->value.integer.value[ch] = vol; + } + break; + } + return slave->slave.put(&slave->slave, ucontrol); +} + +/* + * ctl callbacks for slaves + */ +static int slave_info(struct snd_kcontrol *kcontrol, + struct snd_ctl_elem_info *uinfo) +{ + struct link_slave *slave = snd_kcontrol_chip(kcontrol); + return slave->slave.info(&slave->slave, uinfo); +} + +static int slave_get(struct snd_kcontrol *kcontrol, + struct snd_ctl_elem_value *ucontrol) +{ + struct link_slave *slave = snd_kcontrol_chip(kcontrol); + return slave_get_val(slave, ucontrol); +} + +static int slave_put(struct snd_kcontrol *kcontrol, + struct snd_ctl_elem_value *ucontrol) +{ + struct link_slave *slave = snd_kcontrol_chip(kcontrol); + int err, ch, changed = 0; + + err = slave_init(slave); + if (err < 0) + return err; + for (ch = 0; ch < slave->info.count; ch++) { + if (slave->vals[ch] != ucontrol->value.integer.value[ch]) { + changed = 1; + slave->vals[ch] = ucontrol->value.integer.value[ch]; + } + } + if (!changed) + return 0; + return slave_put_val(slave, ucontrol); +} + +static int slave_tlv_cmd(struct snd_kcontrol *kcontrol, + int op_flag, unsigned int size, + unsigned int __user *tlv) +{ + struct link_slave *slave = snd_kcontrol_chip(kcontrol); + /* FIXME: this assumes that the max volume is 0 dB */ + return slave->slave.tlv.c(&slave->slave, op_flag, size, tlv); +} + +static void slave_free(struct snd_kcontrol *kcontrol) +{ + struct link_slave *slave = snd_kcontrol_chip(kcontrol); + if (slave->slave.private_free) + slave->slave.private_free(&slave->slave); + if (slave->master) + list_del(&slave->list); + kfree(slave); +} + +/* + * Add a slave control to the group with the given master control + * + * All slaves must be the same type (returning the same information + * via info callback). The fucntion doesn't check it, so it's your + * responsibility. + * + * Also, some additional limitations: + * - at most two channels + * - logarithmic volume control (dB level), no linear volume + * - master can only attenuate the volume, no gain + */ +int snd_ctl_add_slave(struct snd_kcontrol *master, struct snd_kcontrol *slave) +{ + struct link_master *master_link = snd_kcontrol_chip(master); + struct link_slave *srec; + + srec = kzalloc(sizeof(*srec) + + slave->count * sizeof(*slave->vd), GFP_KERNEL); + if (!srec) + return -ENOMEM; + srec->slave = *slave; + memcpy(srec->slave.vd, slave->vd, slave->count * sizeof(*slave->vd)); + srec->master = master_link; + + /* override callbacks */ + slave->info = slave_info; + slave->get = slave_get; + slave->put = slave_put; + if (slave->vd[0].access & SNDRV_CTL_ELEM_ACCESS_TLV_CALLBACK) + slave->tlv.c = slave_tlv_cmd; + slave->private_data = srec; + slave->private_free = slave_free; + + list_add_tail(&srec->list, &master_link->slaves); + return 0; +} + +EXPORT_SYMBOL(snd_ctl_add_slave); + +/* + * ctl callbacks for master controls + */ +static int master_info(struct snd_kcontrol *kcontrol, + struct snd_ctl_elem_info *uinfo) +{ + struct link_master *master = snd_kcontrol_chip(kcontrol); + int ret; + + ret = master_init(master); + if (ret < 0) + return ret; + uinfo->type = master->info.type; + uinfo->count = master->info.count; + uinfo->value.integer.min = master->info.min_val; + uinfo->value.integer.max = master->info.max_val; + return 0; +} + +static int master_get(struct snd_kcontrol *kcontrol, + struct snd_ctl_elem_value *ucontrol) +{ + struct link_master *master = snd_kcontrol_chip(kcontrol); + int err = master_init(master); + if (err < 0) + return err; + ucontrol->value.integer.value[0] = master->val; + return 0; +} + +static int master_put(struct snd_kcontrol *kcontrol, + struct snd_ctl_elem_value *ucontrol) +{ + struct link_master *master = snd_kcontrol_chip(kcontrol); + struct link_slave *slave; + struct snd_ctl_elem_value *uval; + int err, old_val; + + err = master_init(master); + if (err < 0) + return err; + old_val = master->val; + if (ucontrol->value.integer.value[0] == old_val) + return 0; + + uval = kmalloc(sizeof(*uval), GFP_KERNEL); + if (!uval) + return -ENOMEM; + list_for_each_entry(slave, &master->slaves, list) { + master->val = old_val; + uval->id = slave->slave.id; + slave_get_val(slave, uval); + master->val = ucontrol->value.integer.value[0]; + slave_put_val(slave, uval); + } + kfree(uval); + return 1; +} + +static void master_free(struct snd_kcontrol *kcontrol) +{ + struct link_master *master = snd_kcontrol_chip(kcontrol); + struct link_slave *slave; + + list_for_each_entry(slave, &master->slaves, list) + slave->master = NULL; + kfree(master); +} + + +/* + * Create a virtual master control with the given name + */ +struct snd_kcontrol *snd_ctl_make_virtual_master(char *name, + const unsigned int *tlv) +{ + struct link_master *master; + struct snd_kcontrol *kctl; + struct snd_kcontrol_new knew; + + memset(&knew, 0, sizeof(knew)); + knew.iface = SNDRV_CTL_ELEM_IFACE_MIXER; + knew.name = name; + knew.info = master_info; + + master = kzalloc(sizeof(*master), GFP_KERNEL); + if (!master) + return NULL; + INIT_LIST_HEAD(&master->slaves); + + kctl = snd_ctl_new1(&knew, master); + if (!kctl) { + kfree(master); + return NULL; + } + /* override some callbacks */ + kctl->info = master_info; + kctl->get = master_get; + kctl->put = master_put; + kctl->private_free = master_free; + + /* additional (constant) TLV read */ + if (tlv && tlv[0] == SNDRV_CTL_TLVT_DB_SCALE) { + kctl->vd[0].access |= SNDRV_CTL_ELEM_ACCESS_TLV_READ; + memcpy(master->tlv, tlv, sizeof(master->tlv)); + kctl->tlv.p = master->tlv; + } + + return kctl; +} + +EXPORT_SYMBOL(snd_ctl_make_virtual_master); diff --git a/sound/drivers/Kconfig b/sound/drivers/Kconfig index 75d4fe0..78648c4 100644 --- a/sound/drivers/Kconfig +++ b/sound/drivers/Kconfig @@ -4,6 +4,23 @@ menu "Generic devices" depends on SND!=n +config SND_PCSP + tristate "Internal PC speaker support" + depends on X86_PC && HIGH_RES_TIMERS + help + If you don't have a sound card in your computer, you can include a + driver for the PC speaker which allows it to act like a primitive + sound card. + This driver also replaces the pcspkr driver for beeps. + + You can compile this as a module which will be called snd-pcsp. + + You don't need this driver if you only want your pc-speaker to beep. + You don't need this driver if you have a tablet piezo beeper + in your PC instead of the real speaker. + + It should not hurt to say Y or M here in all other cases. + config SND_MPU401_UART tristate select SND_RAWMIDI diff --git a/sound/drivers/Makefile b/sound/drivers/Makefile index 8e55300..d4a07f9 100644 --- a/sound/drivers/Makefile +++ b/sound/drivers/Makefile @@ -20,4 +20,4 @@ obj-$(CONFIG_SND_MTS64) += snd-mts64.o obj-$(CONFIG_SND_PORTMAN2X4) += snd-portman2x4.o obj-$(CONFIG_SND_ML403_AC97CR) += snd-ml403-ac97cr.o -obj-$(CONFIG_SND) += opl3/ opl4/ mpu401/ vx/ +obj-$(CONFIG_SND) += opl3/ opl4/ mpu401/ vx/ pcsp/ diff --git a/sound/drivers/dummy.c b/sound/drivers/dummy.c index a240eae..4e4c69e 100644 --- a/sound/drivers/dummy.c +++ b/sound/drivers/dummy.c @@ -181,10 +181,10 @@ struct snd_dummy_pcm { struct snd_dummy *dummy; spinlock_t lock; struct timer_list timer; - unsigned int pcm_size; - unsigned int pcm_count; + unsigned int pcm_buffer_size; + unsigned int pcm_period_size; unsigned int pcm_bps; /* bytes per second */ - unsigned int pcm_jiffie; /* bytes per one jiffie */ + unsigned int pcm_hz; /* HZ */ unsigned int pcm_irq_pos; /* IRQ position */ unsigned int pcm_buf_pos; /* position in buffer */ struct snd_pcm_substream *substream; @@ -230,19 +230,24 @@ static int snd_card_dummy_pcm_prepare(struct snd_pcm_substream *substream) { struct snd_pcm_runtime *runtime = substream->runtime; struct snd_dummy_pcm *dpcm = runtime->private_data; - unsigned int bps; + int bps; + + bps = snd_pcm_format_width(runtime->format) * runtime->rate * + runtime->channels / 8; - bps = runtime->rate * runtime->channels; - bps *= snd_pcm_format_width(runtime->format); - bps /= 8; if (bps <= 0) return -EINVAL; + dpcm->pcm_bps = bps; - dpcm->pcm_jiffie = bps / HZ; - dpcm->pcm_size = snd_pcm_lib_buffer_bytes(substream); - dpcm->pcm_count = snd_pcm_lib_period_bytes(substream); + dpcm->pcm_hz = HZ; + dpcm->pcm_buffer_size = snd_pcm_lib_buffer_bytes(substream); + dpcm->pcm_period_size = snd_pcm_lib_period_bytes(substream); dpcm->pcm_irq_pos = 0; dpcm->pcm_buf_pos = 0; + + snd_pcm_format_set_silence(runtime->format, runtime->dma_area, + bytes_to_samples(runtime, runtime->dma_bytes)); + return 0; } @@ -254,11 +259,11 @@ static void snd_card_dummy_pcm_timer_function(unsigned long data) spin_lock_irqsave(&dpcm->lock, flags); dpcm->timer.expires = 1 + jiffies; add_timer(&dpcm->timer); - dpcm->pcm_irq_pos += dpcm->pcm_jiffie; - dpcm->pcm_buf_pos += dpcm->pcm_jiffie; - dpcm->pcm_buf_pos %= dpcm->pcm_size; - if (dpcm->pcm_irq_pos >= dpcm->pcm_count) { - dpcm->pcm_irq_pos %= dpcm->pcm_count; + dpcm->pcm_irq_pos += dpcm->pcm_bps; + dpcm->pcm_buf_pos += dpcm->pcm_bps; + dpcm->pcm_buf_pos %= dpcm->pcm_buffer_size * dpcm->pcm_hz; + if (dpcm->pcm_irq_pos >= dpcm->pcm_period_size * dpcm->pcm_hz) { + dpcm->pcm_irq_pos %= dpcm->pcm_period_size * dpcm->pcm_hz; spin_unlock_irqrestore(&dpcm->lock, flags); snd_pcm_period_elapsed(dpcm->substream); } else @@ -270,7 +275,7 @@ static snd_pcm_uframes_t snd_card_dummy_pcm_pointer(struct snd_pcm_substream *su struct snd_pcm_runtime *runtime = substream->runtime; struct snd_dummy_pcm *dpcm = runtime->private_data; - return bytes_to_frames(runtime, dpcm->pcm_buf_pos); + return bytes_to_frames(runtime, dpcm->pcm_buf_pos / dpcm->pcm_hz); } static struct snd_pcm_hardware snd_card_dummy_playback = diff --git a/sound/drivers/ml403-ac97cr.c b/sound/drivers/ml403-ac97cr.c index 05a871a..ecdbeb6 100644 --- a/sound/drivers/ml403-ac97cr.c +++ b/sound/drivers/ml403-ac97cr.c @@ -1191,8 +1191,6 @@ snd_ml403_ac97cr_create(struct snd_card *card, struct platform_device *pfdev, return err; } - snd_card_set_dev(card, &pfdev->dev); - *rml403_ac97cr = ml403_ac97cr; return 0; } @@ -1330,11 +1328,15 @@ static int snd_ml403_ac97cr_remove(struct platform_device *pfdev) return 0; } +/* work with hotplug and coldplug */ +MODULE_ALIAS("platform:" SND_ML403_AC97CR_DRIVER); + static struct platform_driver snd_ml403_ac97cr_driver = { .probe = snd_ml403_ac97cr_probe, .remove = snd_ml403_ac97cr_remove, .driver = { .name = SND_ML403_AC97CR_DRIVER, + .owner = THIS_MODULE, }, }; diff --git a/sound/drivers/mpu401/mpu401_uart.c b/sound/drivers/mpu401/mpu401_uart.c index 5993864..dd6ec42 100644 --- a/sound/drivers/mpu401/mpu401_uart.c +++ b/sound/drivers/mpu401/mpu401_uart.c @@ -425,16 +425,17 @@ static void snd_mpu401_uart_input_read(struct snd_mpu401 * mpu) static void snd_mpu401_uart_output_write(struct snd_mpu401 * mpu) { unsigned char byte; - int max = 256, timeout; + int max = 256; do { if (snd_rawmidi_transmit_peek(mpu->substream_output, &byte, 1) == 1) { - for (timeout = 100; timeout > 0; timeout--) { - if (snd_mpu401_output_ready(mpu)) - break; - } - if (timeout == 0) + /* + * Try twice because there is hardware that insists on + * setting the output busy bit after each write. + */ + if (!snd_mpu401_output_ready(mpu) && + !snd_mpu401_output_ready(mpu)) break; /* Tx FIFO full - try again later */ mpu->write(mpu, byte, MPU401D(mpu)); snd_rawmidi_transmit_ack(mpu->substream_output, 1); diff --git a/sound/drivers/pcsp/Makefile b/sound/drivers/pcsp/Makefile new file mode 100644 index 0000000..b19555b --- /dev/null +++ b/sound/drivers/pcsp/Makefile @@ -0,0 +1,2 @@ +snd-pcsp-objs := pcsp.o pcsp_lib.o pcsp_mixer.o pcsp_input.o +obj-$(CONFIG_SND_PCSP) += snd-pcsp.o diff --git a/sound/drivers/pcsp/pcsp.c b/sound/drivers/pcsp/pcsp.c new file mode 100644 index 0000000..d8f9621 --- /dev/null +++ b/sound/drivers/pcsp/pcsp.c @@ -0,0 +1,239 @@ +/* + * PC-Speaker driver for Linux + * + * Copyright (C) 1997-2001 David Woodhouse + * Copyright (C) 2001-2008 Stas Sergeev + */ + +#include +#include +#include +#include +#include +#include +#include +#include +#include +#include "pcsp_input.h" +#include "pcsp.h" + +MODULE_AUTHOR("Stas Sergeev "); +MODULE_DESCRIPTION("PC-Speaker driver"); +MODULE_LICENSE("GPL"); +MODULE_SUPPORTED_DEVICE("{{PC-Speaker, pcsp}}"); +MODULE_ALIAS("platform:pcspkr"); + +static int index = SNDRV_DEFAULT_IDX1; /* Index 0-MAX */ +static char *id = SNDRV_DEFAULT_STR1; /* ID for this card */ +static int enable = SNDRV_DEFAULT_ENABLE1; /* Enable this card */ + +module_param(index, int, 0444); +MODULE_PARM_DESC(index, "Index value for pcsp soundcard."); +module_param(id, charp, 0444); +MODULE_PARM_DESC(id, "ID string for pcsp soundcard."); +module_param(enable, bool, 0444); +MODULE_PARM_DESC(enable, "Enable PC-Speaker sound."); + +struct snd_pcsp pcsp_chip; + +static int __devinit snd_pcsp_create(struct snd_card *card) +{ + static struct snd_device_ops ops = { }; + struct timespec tp; + int err; + int div, min_div, order; + + hrtimer_get_res(CLOCK_MONOTONIC, &tp); + if (tp.tv_sec || tp.tv_nsec > PCSP_MAX_PERIOD_NS) { + printk(KERN_ERR "PCSP: Timer resolution is not sufficient " + "(%linS)\n", tp.tv_nsec); + printk(KERN_ERR "PCSP: Make sure you have HPET and ACPI " + "enabled.\n"); + return -EIO; + } + + if (loops_per_jiffy >= PCSP_MIN_LPJ && tp.tv_nsec <= PCSP_MIN_PERIOD_NS) + min_div = MIN_DIV; + else + min_div = MAX_DIV; +#if PCSP_DEBUG + printk("PCSP: lpj=%li, min_div=%i, res=%li\n", + loops_per_jiffy, min_div, tp.tv_nsec); +#endif + + div = MAX_DIV / min_div; + order = fls(div) - 1; + + pcsp_chip.max_treble = min(order, PCSP_MAX_TREBLE); + pcsp_chip.treble = min(pcsp_chip.max_treble, PCSP_DEFAULT_TREBLE); + pcsp_chip.playback_ptr = 0; + pcsp_chip.period_ptr = 0; + atomic_set(&pcsp_chip.timer_active, 0); + pcsp_chip.enable = 1; + pcsp_chip.pcspkr = 1; + + spin_lock_init(&pcsp_chip.substream_lock); + + pcsp_chip.card = card; + pcsp_chip.port = 0x61; + pcsp_chip.irq = -1; + pcsp_chip.dma = -1; + + /* Register device */ + err = snd_device_new(card, SNDRV_DEV_LOWLEVEL, &pcsp_chip, &ops); + if (err < 0) + return err; + + return 0; +} + +static int __devinit snd_card_pcsp_probe(int devnum, struct device *dev) +{ + struct snd_card *card; + int err; + + if (devnum != 0) + return -EINVAL; + + hrtimer_init(&pcsp_chip.timer, CLOCK_MONOTONIC, HRTIMER_MODE_REL); + pcsp_chip.timer.cb_mode = HRTIMER_CB_IRQSAFE; + pcsp_chip.timer.function = pcsp_do_timer; + + card = snd_card_new(index, id, THIS_MODULE, 0); + if (!card) + return -ENOMEM; + + err = snd_pcsp_create(card); + if (err < 0) { + snd_card_free(card); + return err; + } + err = snd_pcsp_new_pcm(&pcsp_chip); + if (err < 0) { + snd_card_free(card); + return err; + } + err = snd_pcsp_new_mixer(&pcsp_chip); + if (err < 0) { + snd_card_free(card); + return err; + } + + snd_card_set_dev(pcsp_chip.card, dev); + + strcpy(card->driver, "PC-Speaker"); + strcpy(card->shortname, "pcsp"); + sprintf(card->longname, "Internal PC-Speaker at port 0x%x", + pcsp_chip.port); + + err = snd_card_register(card); + if (err < 0) { + snd_card_free(card); + return err; + } + + return 0; +} + +static int __devinit alsa_card_pcsp_init(struct device *dev) +{ + int err; + + err = snd_card_pcsp_probe(0, dev); + if (err) { + printk(KERN_ERR "PC-Speaker initialization failed.\n"); + return err; + } + +#ifdef CONFIG_DEBUG_PAGEALLOC + /* Well, CONFIG_DEBUG_PAGEALLOC makes the sound horrible. Lets alert */ + printk(KERN_WARNING + "PCSP: Warning, CONFIG_DEBUG_PAGEALLOC is enabled!\n" + "You have to disable it if you want to use the PC-Speaker " + "driver.\n" + "Unless it is disabled, enjoy the horrible, distorted " + "and crackling noise.\n"); +#endif + + return 0; +} + +static void __devexit alsa_card_pcsp_exit(struct snd_pcsp *chip) +{ + snd_card_free(chip->card); +} + +static int __devinit pcsp_probe(struct platform_device *dev) +{ + int err; + + err = pcspkr_input_init(&pcsp_chip.input_dev, &dev->dev); + if (err < 0) + return err; + + err = alsa_card_pcsp_init(&dev->dev); + if (err < 0) { + pcspkr_input_remove(pcsp_chip.input_dev); + return err; + } + + platform_set_drvdata(dev, &pcsp_chip); + return 0; +} + +static int __devexit pcsp_remove(struct platform_device *dev) +{ + struct snd_pcsp *chip = platform_get_drvdata(dev); + alsa_card_pcsp_exit(chip); + pcspkr_input_remove(chip->input_dev); + platform_set_drvdata(dev, NULL); + return 0; +} + +static void pcsp_stop_beep(struct snd_pcsp *chip) +{ + spin_lock_irq(&chip->substream_lock); + if (!chip->playback_substream) + pcspkr_stop_sound(); + spin_unlock_irq(&chip->substream_lock); +} + +static int pcsp_suspend(struct platform_device *dev, pm_message_t state) +{ + struct snd_pcsp *chip = platform_get_drvdata(dev); + pcsp_stop_beep(chip); + snd_pcm_suspend_all(chip->pcm); + return 0; +} + +static void pcsp_shutdown(struct platform_device *dev) +{ + struct snd_pcsp *chip = platform_get_drvdata(dev); + pcsp_stop_beep(chip); +} + +static struct platform_driver pcsp_platform_driver = { + .driver = { + .name = "pcspkr", + .owner = THIS_MODULE, + }, + .probe = pcsp_probe, + .remove = __devexit_p(pcsp_remove), + .suspend = pcsp_suspend, + .shutdown = pcsp_shutdown, +}; + +static int __init pcsp_init(void) +{ + if (!enable) + return -ENODEV; + return platform_driver_register(&pcsp_platform_driver); +} + +static void __exit pcsp_exit(void) +{ + platform_driver_unregister(&pcsp_platform_driver); +} + +module_init(pcsp_init); +module_exit(pcsp_exit); diff --git a/sound/drivers/pcsp/pcsp.h b/sound/drivers/pcsp/pcsp.h new file mode 100644 index 0000000..f07cc1e --- /dev/null +++ b/sound/drivers/pcsp/pcsp.h @@ -0,0 +1,82 @@ +/* + * PC-Speaker driver for Linux + * + * Copyright (C) 1993-1997 Michael Beck + * Copyright (C) 1997-2001 David Woodhouse + * Copyright (C) 2001-2008 Stas Sergeev + */ + +#ifndef __PCSP_H__ +#define __PCSP_H__ + +#include +#if defined(CONFIG_MIPS) || defined(CONFIG_X86) +/* Use the global PIT lock ! */ +#include +#else +#include +static DEFINE_SPINLOCK(i8253_lock); +#endif + +#define PCSP_SOUND_VERSION 0x400 /* read 4.00 */ +#define PCSP_DEBUG 0 + +/* default timer freq for PC-Speaker: 18643 Hz */ +#define DIV_18KHZ 64 +#define MAX_DIV DIV_18KHZ +#define CUR_DIV() (MAX_DIV >> chip->treble) +#define PCSP_MAX_TREBLE 1 + +/* unfortunately, with hrtimers 37KHz does not work very well :( */ +#define PCSP_DEFAULT_TREBLE 0 +#define MIN_DIV (MAX_DIV >> PCSP_MAX_TREBLE) + +/* wild guess */ +#define PCSP_MIN_LPJ 1000000 +#define PCSP_DEFAULT_SDIV (DIV_18KHZ >> 1) +#define PCSP_DEFAULT_SRATE (PIT_TICK_RATE / PCSP_DEFAULT_SDIV) +#define PCSP_INDEX_INC() (1 << (PCSP_MAX_TREBLE - chip->treble)) +#define PCSP_RATE() (PIT_TICK_RATE / CUR_DIV()) +#define PCSP_MIN_RATE__1 MAX_DIV/PIT_TICK_RATE +#define PCSP_MAX_RATE__1 MIN_DIV/PIT_TICK_RATE +#define PCSP_MAX_PERIOD_NS (1000000000ULL * PCSP_MIN_RATE__1) +#define PCSP_MIN_PERIOD_NS (1000000000ULL * PCSP_MAX_RATE__1) +#define PCSP_CALC_NS(div) ({ \ + u64 __val = 1000000000ULL * (div); \ + do_div(__val, PIT_TICK_RATE); \ + __val; \ +}) +#define PCSP_PERIOD_NS() PCSP_CALC_NS(CUR_DIV()) + +#define PCSP_MAX_PERIOD_SIZE (64*1024) +#define PCSP_MAX_PERIODS 512 +#define PCSP_BUFFER_SIZE (128*1024) + +struct snd_pcsp { + struct snd_card *card; + struct snd_pcm *pcm; + struct input_dev *input_dev; + struct hrtimer timer; + unsigned short port, irq, dma; + spinlock_t substream_lock; + struct snd_pcm_substream *playback_substream; + size_t playback_ptr; + size_t period_ptr; + atomic_t timer_active; + int thalf; + u64 ns_rem; + unsigned char val61; + int enable; + int max_treble; + int treble; + int pcspkr; +}; + +extern struct snd_pcsp pcsp_chip; + +extern enum hrtimer_restart pcsp_do_timer(struct hrtimer *handle); + +extern int snd_pcsp_new_pcm(struct snd_pcsp *chip); +extern int snd_pcsp_new_mixer(struct snd_pcsp *chip); + +#endif diff --git a/sound/drivers/pcsp/pcsp_input.c b/sound/drivers/pcsp/pcsp_input.c new file mode 100644 index 0000000..cd9b83e --- /dev/null +++ b/sound/drivers/pcsp/pcsp_input.c @@ -0,0 +1,116 @@ +/* + * PC Speaker beeper driver for Linux + * + * Copyright (c) 2002 Vojtech Pavlik + * Copyright (c) 1992 Orest Zborowski + * + */ + +/* + * This program is free software; you can redistribute it and/or modify it + * under the terms of the GNU General Public License version 2 as published by + * the Free Software Foundation + */ + +#include +#include +#include +#include "pcsp.h" + +static void pcspkr_do_sound(unsigned int count) +{ + unsigned long flags; + + spin_lock_irqsave(&i8253_lock, flags); + + if (count) { + /* enable counter 2 */ + outb_p(inb_p(0x61) | 3, 0x61); + /* set command for counter 2, 2 byte write */ + outb_p(0xB6, 0x43); + /* select desired HZ */ + outb_p(count & 0xff, 0x42); + outb((count >> 8) & 0xff, 0x42); + } else { + /* disable counter 2 */ + outb(inb_p(0x61) & 0xFC, 0x61); + } + + spin_unlock_irqrestore(&i8253_lock, flags); +} + +void pcspkr_stop_sound(void) +{ + pcspkr_do_sound(0); +} + +static int pcspkr_input_event(struct input_dev *dev, unsigned int type, + unsigned int code, int value) +{ + unsigned int count = 0; + + if (atomic_read(&pcsp_chip.timer_active) || !pcsp_chip.pcspkr) + return 0; + + switch (type) { + case EV_SND: + switch (code) { + case SND_BELL: + if (value) + value = 1000; + case SND_TONE: + break; + default: + return -1; + } + break; + + default: + return -1; + } + + if (value > 20 && value < 32767) + count = PIT_TICK_RATE / value; + + pcspkr_do_sound(count); + + return 0; +} + +int __devinit pcspkr_input_init(struct input_dev **rdev, struct device *dev) +{ + int err; + + struct input_dev *input_dev = input_allocate_device(); + if (!input_dev) + return -ENOMEM; + + input_dev->name = "PC Speaker"; + input_dev->phys = "isa0061/input0"; + input_dev->id.bustype = BUS_ISA; + input_dev->id.vendor = 0x001f; + input_dev->id.product = 0x0001; + input_dev->id.version = 0x0100; + input_dev->dev.parent = dev; + + input_dev->evbit[0] = BIT(EV_SND); + input_dev->sndbit[0] = BIT(SND_BELL) | BIT(SND_TONE); + input_dev->event = pcspkr_input_event; + + err = input_register_device(input_dev); + if (err) { + input_free_device(input_dev); + return err; + } + + *rdev = input_dev; + return 0; +} + +int pcspkr_input_remove(struct input_dev *dev) +{ + pcspkr_stop_sound(); + input_unregister_device(dev); /* this also does kfree() */ + + return 0; +} diff --git a/sound/drivers/pcsp/pcsp_input.h b/sound/drivers/pcsp/pcsp_input.h new file mode 100644 index 0000000..e66738c --- /dev/null +++ b/sound/drivers/pcsp/pcsp_input.h @@ -0,0 +1,14 @@ +/* + * PC-Speaker driver for Linux + * + * Copyright (C) 2001-2008 Stas Sergeev + */ + +#ifndef __PCSP_INPUT_H__ +#define __PCSP_INPUT_H__ + +int __devinit pcspkr_input_init(struct input_dev **rdev, struct device *dev); +int pcspkr_input_remove(struct input_dev *dev); +void pcspkr_stop_sound(void); + +#endif diff --git a/sound/drivers/pcsp/pcsp_lib.c b/sound/drivers/pcsp/pcsp_lib.c new file mode 100644 index 0000000..ac6238e --- /dev/null +++ b/sound/drivers/pcsp/pcsp_lib.c @@ -0,0 +1,338 @@ +/* + * PC-Speaker driver for Linux + * + * Copyright (C) 1993-1997 Michael Beck + * Copyright (C) 1997-2001 David Woodhouse + * Copyright (C) 2001-2008 Stas Sergeev + */ + +#include +#include +#include +#include +#include +#include "pcsp.h" + +static int nforce_wa; +module_param(nforce_wa, bool, 0444); +MODULE_PARM_DESC(nforce_wa, "Apply NForce chipset workaround " + "(expect bad sound)"); + +static void pcsp_start_timer(unsigned long dummy) +{ + hrtimer_start(&pcsp_chip.timer, ktime_set(0, 0), HRTIMER_MODE_REL); +} + +/* + * We need the hrtimer_start as a tasklet to avoid + * the nasty locking problem. :( + * The problem: + * - The timer handler is called with the cpu_base->lock + * already held by hrtimer code. + * - snd_pcm_period_elapsed() takes the + * substream->self_group.lock. + * So far so good. + * But the snd_pcsp_trigger() is called with the + * substream->self_group.lock held, and it calls + * hrtimer_start(), which takes the cpu_base->lock. + * You see the problem. We have the code pathes + * which take two locks in a reverse order. This + * can deadlock and the lock validator complains. + * The only solution I could find was to move the + * hrtimer_start() into a tasklet. -stsp + */ +static DECLARE_TASKLET(pcsp_start_timer_tasklet, pcsp_start_timer, 0); + +enum hrtimer_restart pcsp_do_timer(struct hrtimer *handle) +{ + unsigned long flags; + unsigned char timer_cnt, val; + int periods_elapsed; + u64 ns; + size_t period_bytes, buffer_bytes; + struct snd_pcm_substream *substream; + struct snd_pcm_runtime *runtime; + struct snd_pcsp *chip = container_of(handle, struct snd_pcsp, timer); + + if (chip->thalf) { + outb(chip->val61, 0x61); + chip->thalf = 0; + if (!atomic_read(&chip->timer_active)) + return HRTIMER_NORESTART; + hrtimer_forward(&chip->timer, chip->timer.expires, + ktime_set(0, chip->ns_rem)); + return HRTIMER_RESTART; + } + + /* hrtimer calls us from both hardirq and softirq contexts, + * so irqsave :( */ + spin_lock_irqsave(&chip->substream_lock, flags); + /* Takashi Iwai says regarding this extra lock: + + If the irq handler handles some data on the DMA buffer, it should + do snd_pcm_stream_lock(). + That protects basically against all races among PCM callbacks, yes. + However, there are two remaining issues: + 1. The substream pointer you try to lock isn't protected _before_ + this lock yet. + 2. snd_pcm_period_elapsed() itself acquires the lock. + The requirement of another lock is because of 1. When you get + chip->playback_substream, it's not protected. + Keeping this lock while snd_pcm_period_elapsed() assures the substream + is still protected (at least, not released). And the other status is + handled properly inside snd_pcm_stream_lock() in + snd_pcm_period_elapsed(). + + */ + if (!chip->playback_substream) + goto exit_nr_unlock1; + substream = chip->playback_substream; + snd_pcm_stream_lock(substream); + if (!atomic_read(&chip->timer_active)) + goto exit_nr_unlock2; + + runtime = substream->runtime; + /* assume it is u8 mono */ + val = runtime->dma_area[chip->playback_ptr]; + timer_cnt = val * CUR_DIV() / 256; + + if (timer_cnt && chip->enable) { + spin_lock(&i8253_lock); + if (!nforce_wa) { + outb_p(chip->val61, 0x61); + outb_p(timer_cnt, 0x42); + outb(chip->val61 ^ 1, 0x61); + } else { + outb(chip->val61 ^ 2, 0x61); + chip->thalf = 1; + } + spin_unlock(&i8253_lock); + } + + period_bytes = snd_pcm_lib_period_bytes(substream); + buffer_bytes = snd_pcm_lib_buffer_bytes(substream); + chip->playback_ptr += PCSP_INDEX_INC(); + periods_elapsed = chip->playback_ptr - chip->period_ptr; + if (periods_elapsed < 0) { + printk(KERN_WARNING "PCSP: playback_ptr inconsistent " + "(%zi %zi %zi)\n", + chip->playback_ptr, period_bytes, buffer_bytes); + periods_elapsed += buffer_bytes; + } + periods_elapsed /= period_bytes; + /* wrap the pointer _before_ calling snd_pcm_period_elapsed(), + * or ALSA will BUG on us. */ + chip->playback_ptr %= buffer_bytes; + + snd_pcm_stream_unlock(substream); + + if (periods_elapsed) { + snd_pcm_period_elapsed(substream); + chip->period_ptr += periods_elapsed * period_bytes; + chip->period_ptr %= buffer_bytes; + } + + spin_unlock_irqrestore(&chip->substream_lock, flags); + + if (!atomic_read(&chip->timer_active)) + return HRTIMER_NORESTART; + + chip->ns_rem = PCSP_PERIOD_NS(); + ns = (chip->thalf ? PCSP_CALC_NS(timer_cnt) : chip->ns_rem); + chip->ns_rem -= ns; + hrtimer_forward(&chip->timer, chip->timer.expires, ktime_set(0, ns)); + return HRTIMER_RESTART; + +exit_nr_unlock2: + snd_pcm_stream_unlock(substream); +exit_nr_unlock1: + spin_unlock_irqrestore(&chip->substream_lock, flags); + return HRTIMER_NORESTART; +} + +static void pcsp_start_playing(struct snd_pcsp *chip) +{ +#if PCSP_DEBUG + printk(KERN_INFO "PCSP: start_playing called\n"); +#endif + if (atomic_read(&chip->timer_active)) { + printk(KERN_ERR "PCSP: Timer already active\n"); + return; + } + + spin_lock(&i8253_lock); + chip->val61 = inb(0x61) | 0x03; + outb_p(0x92, 0x43); /* binary, mode 1, LSB only, ch 2 */ + spin_unlock(&i8253_lock); + atomic_set(&chip->timer_active, 1); + chip->thalf = 0; + + tasklet_schedule(&pcsp_start_timer_tasklet); +} + +static void pcsp_stop_playing(struct snd_pcsp *chip) +{ +#if PCSP_DEBUG + printk(KERN_INFO "PCSP: stop_playing called\n"); +#endif + if (!atomic_read(&chip->timer_active)) + return; + + atomic_set(&chip->timer_active, 0); + spin_lock(&i8253_lock); + /* restore the timer */ + outb_p(0xb6, 0x43); /* binary, mode 3, LSB/MSB, ch 2 */ + outb(chip->val61 & 0xFC, 0x61); + spin_unlock(&i8253_lock); +} + +static int snd_pcsp_playback_close(struct snd_pcm_substream *substream) +{ + struct snd_pcsp *chip = snd_pcm_substream_chip(substream); +#if PCSP_DEBUG + printk(KERN_INFO "PCSP: close called\n"); +#endif + if (atomic_read(&chip->timer_active)) { + printk(KERN_ERR "PCSP: timer still active\n"); + pcsp_stop_playing(chip); + } + spin_lock_irq(&chip->substream_lock); + chip->playback_substream = NULL; + spin_unlock_irq(&chip->substream_lock); + return 0; +} + +static int snd_pcsp_playback_hw_params(struct snd_pcm_substream *substream, + struct snd_pcm_hw_params *hw_params) +{ + int err; + err = snd_pcm_lib_malloc_pages(substream, + params_buffer_bytes(hw_params)); + if (err < 0) + return err; + return 0; +} + +static int snd_pcsp_playback_hw_free(struct snd_pcm_substream *substream) +{ +#if PCSP_DEBUG + printk(KERN_INFO "PCSP: hw_free called\n"); +#endif + return snd_pcm_lib_free_pages(substream); +} + +static int snd_pcsp_playback_prepare(struct snd_pcm_substream *substream) +{ + struct snd_pcsp *chip = snd_pcm_substream_chip(substream); +#if PCSP_DEBUG + printk(KERN_INFO "PCSP: prepare called, " + "size=%zi psize=%zi f=%zi f1=%i\n", + snd_pcm_lib_buffer_bytes(substream), + snd_pcm_lib_period_bytes(substream), + snd_pcm_lib_buffer_bytes(substream) / + snd_pcm_lib_period_bytes(substream), + substream->runtime->periods); +#endif + chip->playback_ptr = 0; + chip->period_ptr = 0; + return 0; +} + +static int snd_pcsp_trigger(struct snd_pcm_substream *substream, int cmd) +{ + struct snd_pcsp *chip = snd_pcm_substream_chip(substream); +#if PCSP_DEBUG + printk(KERN_INFO "PCSP: trigger called\n"); +#endif + switch (cmd) { + case SNDRV_PCM_TRIGGER_START: + case SNDRV_PCM_TRIGGER_RESUME: + pcsp_start_playing(chip); + break; + case SNDRV_PCM_TRIGGER_STOP: + case SNDRV_PCM_TRIGGER_SUSPEND: + pcsp_stop_playing(chip); + break; + default: + return -EINVAL; + } + return 0; +} + +static snd_pcm_uframes_t snd_pcsp_playback_pointer(struct snd_pcm_substream + *substream) +{ + struct snd_pcsp *chip = snd_pcm_substream_chip(substream); + return bytes_to_frames(substream->runtime, chip->playback_ptr); +} + +static struct snd_pcm_hardware snd_pcsp_playback = { + .info = (SNDRV_PCM_INFO_INTERLEAVED | + SNDRV_PCM_INFO_HALF_DUPLEX | + SNDRV_PCM_INFO_MMAP | SNDRV_PCM_INFO_MMAP_VALID), + .formats = SNDRV_PCM_FMTBIT_U8, + .rates = SNDRV_PCM_RATE_KNOT, + .rate_min = PCSP_DEFAULT_SRATE, + .rate_max = PCSP_DEFAULT_SRATE, + .channels_min = 1, + .channels_max = 1, + .buffer_bytes_max = PCSP_BUFFER_SIZE, + .period_bytes_min = 64, + .period_bytes_max = PCSP_MAX_PERIOD_SIZE, + .periods_min = 2, + .periods_max = PCSP_MAX_PERIODS, + .fifo_size = 0, +}; + +static int snd_pcsp_playback_open(struct snd_pcm_substream *substream) +{ + struct snd_pcsp *chip = snd_pcm_substream_chip(substream); + struct snd_pcm_runtime *runtime = substream->runtime; +#if PCSP_DEBUG + printk(KERN_INFO "PCSP: open called\n"); +#endif + if (atomic_read(&chip->timer_active)) { + printk(KERN_ERR "PCSP: still active!!\n"); + return -EBUSY; + } + runtime->hw = snd_pcsp_playback; + spin_lock_irq(&chip->substream_lock); + chip->playback_substream = substream; + spin_unlock_irq(&chip->substream_lock); + return 0; +} + +static struct snd_pcm_ops snd_pcsp_playback_ops = { + .open = snd_pcsp_playback_open, + .close = snd_pcsp_playback_close, + .ioctl = snd_pcm_lib_ioctl, + .hw_params = snd_pcsp_playback_hw_params, + .hw_free = snd_pcsp_playback_hw_free, + .prepare = snd_pcsp_playback_prepare, + .trigger = snd_pcsp_trigger, + .pointer = snd_pcsp_playback_pointer, +}; + +int __devinit snd_pcsp_new_pcm(struct snd_pcsp *chip) +{ + int err; + + err = snd_pcm_new(chip->card, "pcspeaker", 0, 1, 0, &chip->pcm); + if (err < 0) + return err; + + snd_pcm_set_ops(chip->pcm, SNDRV_PCM_STREAM_PLAYBACK, + &snd_pcsp_playback_ops); + + chip->pcm->private_data = chip; + chip->pcm->info_flags = SNDRV_PCM_INFO_HALF_DUPLEX; + strcpy(chip->pcm->name, "pcsp"); + + snd_pcm_lib_preallocate_pages_for_all(chip->pcm, + SNDRV_DMA_TYPE_CONTINUOUS, + snd_dma_continuous_data + (GFP_KERNEL), PCSP_BUFFER_SIZE, + PCSP_BUFFER_SIZE); + + return 0; +} diff --git a/sound/drivers/pcsp/pcsp_mixer.c b/sound/drivers/pcsp/pcsp_mixer.c new file mode 100644 index 0000000..64a695f --- /dev/null +++ b/sound/drivers/pcsp/pcsp_mixer.c @@ -0,0 +1,143 @@ +/* + * PC-Speaker driver for Linux + * + * Mixer implementation. + * Copyright (C) 2001-2008 Stas Sergeev + */ + +#include +#include +#include "pcsp.h" + + +static int pcsp_enable_info(struct snd_kcontrol *kcontrol, + struct snd_ctl_elem_info *uinfo) +{ + uinfo->type = SNDRV_CTL_ELEM_TYPE_BOOLEAN; + uinfo->count = 1; + uinfo->value.integer.min = 0; + uinfo->value.integer.max = 1; + return 0; +} + +static int pcsp_enable_get(struct snd_kcontrol *kcontrol, + struct snd_ctl_elem_value *ucontrol) +{ + struct snd_pcsp *chip = snd_kcontrol_chip(kcontrol); + ucontrol->value.integer.value[0] = chip->enable; + return 0; +} + +static int pcsp_enable_put(struct snd_kcontrol *kcontrol, + struct snd_ctl_elem_value *ucontrol) +{ + struct snd_pcsp *chip = snd_kcontrol_chip(kcontrol); + int changed = 0; + int enab = ucontrol->value.integer.value[0]; + if (enab != chip->enable) { + chip->enable = enab; + changed = 1; + } + return changed; +} + +static int pcsp_treble_info(struct snd_kcontrol *kcontrol, + struct snd_ctl_elem_info *uinfo) +{ + struct snd_pcsp *chip = snd_kcontrol_chip(kcontrol); + uinfo->type = SNDRV_CTL_ELEM_TYPE_ENUMERATED; + uinfo->count = 1; + uinfo->value.enumerated.items = chip->max_treble + 1; + if (uinfo->value.enumerated.item > chip->max_treble) + uinfo->value.enumerated.item = chip->max_treble; + sprintf(uinfo->value.enumerated.name, "%d", PCSP_RATE()); + return 0; +} + +static int pcsp_treble_get(struct snd_kcontrol *kcontrol, + struct snd_ctl_elem_value *ucontrol) +{ + struct snd_pcsp *chip = snd_kcontrol_chip(kcontrol); + ucontrol->value.enumerated.item[0] = chip->treble; + return 0; +} + +static int pcsp_treble_put(struct snd_kcontrol *kcontrol, + struct snd_ctl_elem_value *ucontrol) +{ + struct snd_pcsp *chip = snd_kcontrol_chip(kcontrol); + int changed = 0; + int treble = ucontrol->value.enumerated.item[0]; + if (treble != chip->treble) { + chip->treble = treble; +#if PCSP_DEBUG + printk(KERN_INFO "PCSP: rate set to %i\n", PCSP_RATE()); +#endif + changed = 1; + } + return changed; +} + +static int pcsp_pcspkr_info(struct snd_kcontrol *kcontrol, + struct snd_ctl_elem_info *uinfo) +{ + uinfo->type = SNDRV_CTL_ELEM_TYPE_BOOLEAN; + uinfo->count = 1; + uinfo->value.integer.min = 0; + uinfo->value.integer.max = 1; + return 0; +} + +static int pcsp_pcspkr_get(struct snd_kcontrol *kcontrol, + struct snd_ctl_elem_value *ucontrol) +{ + struct snd_pcsp *chip = snd_kcontrol_chip(kcontrol); + ucontrol->value.integer.value[0] = chip->pcspkr; + return 0; +} + +static int pcsp_pcspkr_put(struct snd_kcontrol *kcontrol, + struct snd_ctl_elem_value *ucontrol) +{ + struct snd_pcsp *chip = snd_kcontrol_chip(kcontrol); + int changed = 0; + int spkr = ucontrol->value.integer.value[0]; + if (spkr != chip->pcspkr) { + chip->pcspkr = spkr; + changed = 1; + } + return changed; +} + +#define PCSP_MIXER_CONTROL(ctl_type, ctl_name) \ +{ \ + .iface = SNDRV_CTL_ELEM_IFACE_MIXER, \ + .name = ctl_name, \ + .info = pcsp_##ctl_type##_info, \ + .get = pcsp_##ctl_type##_get, \ + .put = pcsp_##ctl_type##_put, \ +} + +static struct snd_kcontrol_new __devinitdata snd_pcsp_controls[] = { + PCSP_MIXER_CONTROL(enable, "Master Playback Switch"), + PCSP_MIXER_CONTROL(treble, "BaseFRQ Playback Volume"), + PCSP_MIXER_CONTROL(pcspkr, "PC Speaker Playback Switch"), +}; + +int __devinit snd_pcsp_new_mixer(struct snd_pcsp *chip) +{ + struct snd_card *card = chip->card; + int i, err; + + for (i = 0; i < ARRAY_SIZE(snd_pcsp_controls); i++) { + err = snd_ctl_add(card, + snd_ctl_new1(snd_pcsp_controls + i, + chip)); + if (err < 0) + return err; + } + + strcpy(card->mixername, "PC-Speaker"); + + return 0; +} diff --git a/sound/i2c/other/ak4114.c b/sound/i2c/other/ak4114.c index 15061bd..d20d893 100644 --- a/sound/i2c/other/ak4114.c +++ b/sound/i2c/other/ak4114.c @@ -27,6 +27,7 @@ #include #include #include +#include MODULE_AUTHOR("Jaroslav Kysela "); MODULE_DESCRIPTION("AK4114 IEC958 (S/PDIF) receiver by Asahi Kasei"); @@ -446,6 +447,26 @@ static struct snd_kcontrol_new snd_ak4114_iec958_controls[] = { } }; + +static void snd_ak4114_proc_regs_read(struct snd_info_entry *entry, + struct snd_info_buffer *buffer) +{ + struct ak4114 *ak4114 = entry->private_data; + int reg, val; + /* all ak4114 registers 0x00 - 0x1f */ + for (reg = 0; reg < 0x20; reg++) { + val = reg_read(ak4114, reg); + snd_iprintf(buffer, "0x%02x = 0x%02x\n", reg, val); + } +} + +static void snd_ak4114_proc_init(struct ak4114 *ak4114) +{ + struct snd_info_entry *entry; + if (!snd_card_proc_new(ak4114->card, "ak4114", &entry)) + snd_info_set_text_ops(entry, ak4114, snd_ak4114_proc_regs_read); +} + int snd_ak4114_build(struct ak4114 *ak4114, struct snd_pcm_substream *ply_substream, struct snd_pcm_substream *cap_substream) @@ -478,6 +499,7 @@ int snd_ak4114_build(struct ak4114 *ak4114, return err; ak4114->kctls[idx] = kctl; } + snd_ak4114_proc_init(ak4114); /* trigger workq */ schedule_delayed_work(&ak4114->work, HZ / 10); return 0; @@ -590,7 +612,7 @@ static void ak4114_stats(struct work_struct *work) struct ak4114 *chip = container_of(work, struct ak4114, work.work); if (!chip->init) - snd_ak4114_check_rate_and_errors(chip, 0); + snd_ak4114_check_rate_and_errors(chip, chip->check_flags); schedule_delayed_work(&chip->work, HZ / 10); } diff --git a/sound/i2c/other/ak4xxx-adda.c b/sound/i2c/other/ak4xxx-adda.c index 35fbbf2..288926d 100644 --- a/sound/i2c/other/ak4xxx-adda.c +++ b/sound/i2c/other/ak4xxx-adda.c @@ -70,7 +70,8 @@ static void ak4524_reset(struct snd_akm4xxx *ak, int state) } /* reset procedure for AK4355 and AK4358 */ -static void ak4355_reset(struct snd_akm4xxx *ak, int state) +static void ak435X_reset(struct snd_akm4xxx *ak, int state, + unsigned char total_regs) { unsigned char reg; @@ -78,7 +79,7 @@ static void ak4355_reset(struct snd_akm4xxx *ak, int state) snd_akm4xxx_write(ak, 0, 0x01, 0x02); /* reset and soft-mute */ return; } - for (reg = 0x00; reg < 0x0b; reg++) + for (reg = 0x00; reg < total_regs; reg++) if (reg != 0x01) snd_akm4xxx_write(ak, 0, reg, snd_akm4xxx_get(ak, 0, reg)); @@ -118,8 +119,10 @@ void snd_akm4xxx_reset(struct snd_akm4xxx *ak, int state) /* FIXME: needed for ak4529? */ break; case SND_AK4355: + ak435X_reset(ak, state, 0x0b); + break; case SND_AK4358: - ak4355_reset(ak, state); + ak435X_reset(ak, state, 0x10); break; case SND_AK4381: ak4381_reset(ak, state); @@ -292,11 +295,6 @@ void snd_akm4xxx_init(struct snd_akm4xxx *ak) case SND_AK5365: /* FIXME: any init sequence? */ return; - case NON_AKM: - /* fake value for non-akm codecs using akm infrastructure - * (e.g. of ice1724) - certainly FIXME - */ - return; default: snd_BUG(); return; @@ -374,6 +372,8 @@ static int put_ak_reg(struct snd_kcontrol *kcontrol, int addr, nval = mask - nval; if (AK_GET_NEEDSMSB(kcontrol->private_value)) nval |= 0x80; + /* printk(KERN_DEBUG "DEBUG - AK writing reg: chip %x addr %x, + nval %x\n", chip, addr, nval); */ snd_akm4xxx_write(ak, chip, addr, nval); return 1; } diff --git a/sound/isa/sb/sb16_csp.c b/sound/isa/sb/sb16_csp.c index bed29ca..f3fd7b4 100644 --- a/sound/isa/sb/sb16_csp.c +++ b/sound/isa/sb/sb16_csp.c @@ -331,7 +331,7 @@ static int snd_sb_csp_riff_load(struct snd_sb_csp * p, return -EFAULT; if ((file_h.name != RIFF_HEADER) || (le32_to_cpu(file_h.len) >= SNDRV_SB_CSP_MAX_MICROCODE_FILE_SIZE - sizeof(file_h))) { - snd_printd("%s: Invalid RIFF header\n", __FUNCTION__); + snd_printd("%s: Invalid RIFF header\n", __func__); return -EINVAL; } data_ptr += sizeof(file_h); @@ -340,7 +340,7 @@ static int snd_sb_csp_riff_load(struct snd_sb_csp * p, if (copy_from_user(&item_type, data_ptr, sizeof(item_type))) return -EFAULT; if (item_type != CSP__HEADER) { - snd_printd("%s: Invalid RIFF file type\n", __FUNCTION__); + snd_printd("%s: Invalid RIFF file type\n", __func__); return -EINVAL; } data_ptr += sizeof (item_type); @@ -395,7 +395,7 @@ static int snd_sb_csp_riff_load(struct snd_sb_csp * p, return -EFAULT; if (code_h.name != MAIN_HEADER) { - snd_printd("%s: Missing 'main' microcode\n", __FUNCTION__); + snd_printd("%s: Missing 'main' microcode\n", __func__); return -EINVAL; } data_ptr += sizeof(code_h); @@ -439,7 +439,7 @@ static int snd_sb_csp_riff_load(struct snd_sb_csp * p, p->acc_format = p->acc_width = p->acc_rates = 0; p->mode = 0; snd_printd("%s: Unsupported CSP codec type: 0x%04x\n", - __FUNCTION__, + __func__, le16_to_cpu(funcdesc_h.VOC_type)); return -EINVAL; } @@ -458,7 +458,7 @@ static int snd_sb_csp_riff_load(struct snd_sb_csp * p, return 0; } } - snd_printd("%s: Function #%d not found\n", __FUNCTION__, info.func_req); + snd_printd("%s: Function #%d not found\n", __func__, info.func_req); return -EINVAL; } @@ -612,7 +612,7 @@ static int get_version(struct snd_sb *chip) static int snd_sb_csp_check_version(struct snd_sb_csp * p) { if (p->version < 0x10 || p->version > 0x1f) { - snd_printd("%s: Invalid CSP version: 0x%x\n", __FUNCTION__, p->version); + snd_printd("%s: Invalid CSP version: 0x%x\n", __func__, p->version); return 1; } return 0; @@ -631,7 +631,7 @@ static int snd_sb_csp_load(struct snd_sb_csp * p, const unsigned char *buf, int spin_lock_irqsave(&p->chip->reg_lock, flags); snd_sbdsp_command(p->chip, 0x01); /* CSP download command */ if (snd_sbdsp_get_byte(p->chip)) { - snd_printd("%s: Download command failed\n", __FUNCTION__); + snd_printd("%s: Download command failed\n", __func__); goto __fail; } /* Send CSP low byte (size - 1) */ @@ -658,7 +658,7 @@ static int snd_sb_csp_load(struct snd_sb_csp * p, const unsigned char *buf, int udelay (10); } if (status != 0x55) { - snd_printd("%s: Microcode initialization failed\n", __FUNCTION__); + snd_printd("%s: Microcode initialization failed\n", __func__); goto __fail; } } else { @@ -824,19 +824,19 @@ static int snd_sb_csp_start(struct snd_sb_csp * p, int sample_width, int channel unsigned long flags; if (!(p->running & (SNDRV_SB_CSP_ST_LOADED | SNDRV_SB_CSP_ST_AUTO))) { - snd_printd("%s: Microcode not loaded\n", __FUNCTION__); + snd_printd("%s: Microcode not loaded\n", __func__); return -ENXIO; } if (p->running & SNDRV_SB_CSP_ST_RUNNING) { - snd_printd("%s: CSP already running\n", __FUNCTION__); + snd_printd("%s: CSP already running\n", __func__); return -EBUSY; } if (!(sample_width & p->acc_width)) { - snd_printd("%s: Unsupported PCM sample width\n", __FUNCTION__); + snd_printd("%s: Unsupported PCM sample width\n", __func__); return -EINVAL; } if (!(channels & p->acc_channels)) { - snd_printd("%s: Invalid number of channels\n", __FUNCTION__); + snd_printd("%s: Invalid number of channels\n", __func__); return -EINVAL; } @@ -858,11 +858,11 @@ static int snd_sb_csp_start(struct snd_sb_csp * p, int sample_width, int channel s_type |= 0x22; /* 00dX 00dX (d = 1 if 8 bit samples) */ if (set_codec_parameter(p->chip, 0x81, s_type)) { - snd_printd("%s: Set sample type command failed\n", __FUNCTION__); + snd_printd("%s: Set sample type command failed\n", __func__); goto __fail; } if (set_codec_parameter(p->chip, 0x80, 0x00)) { - snd_printd("%s: Codec start command failed\n", __FUNCTION__); + snd_printd("%s: Codec start command failed\n", __func__); goto __fail; } p->run_width = sample_width; diff --git a/sound/isa/sb/sb_common.c b/sound/isa/sb/sb_common.c index d63c1af..b432d9a 100644 --- a/sound/isa/sb/sb_common.c +++ b/sound/isa/sb/sb_common.c @@ -51,7 +51,7 @@ int snd_sbdsp_command(struct snd_sb *chip, unsigned char val) outb(val, SBP(chip, COMMAND)); return 1; } - snd_printd("%s [0x%lx]: timeout (0x%x)\n", __FUNCTION__, chip->port, val); + snd_printd("%s [0x%lx]: timeout (0x%x)\n", __func__, chip->port, val); return 0; } @@ -68,7 +68,7 @@ int snd_sbdsp_get_byte(struct snd_sb *chip) return val; } } - snd_printd("%s [0x%lx]: timeout\n", __FUNCTION__, chip->port); + snd_printd("%s [0x%lx]: timeout\n", __func__, chip->port); return -ENODEV; } @@ -87,7 +87,7 @@ int snd_sbdsp_reset(struct snd_sb *chip) else break; } - snd_printdd("%s [0x%lx] failed...\n", __FUNCTION__, chip->port); + snd_printdd("%s [0x%lx] failed...\n", __func__, chip->port); return -ENODEV; } diff --git a/sound/oss/trident.c b/sound/oss/trident.c index d6af906..f43f91e 100644 --- a/sound/oss/trident.c +++ b/sound/oss/trident.c @@ -3076,8 +3076,7 @@ ali_ac97_get(struct trident_card *card, int secondary, u8 reg) u16 wcontrol; unsigned long flags; - if (!card) - BUG(); + BUG_ON(!card); address = ALI_AC97_READ; if (card->revision == ALI_5451_V02) { @@ -3148,8 +3147,7 @@ ali_ac97_set(struct trident_card *card, int secondary, u8 reg, u16 val) data = ((u32) val) << 16; - if (!card) - BUG(); + BUG_ON(!card); address = ALI_AC97_WRITE; mask = ALI_AC97_WRITE_ACTION | ALI_AC97_AUDIO_BUSY; @@ -3213,8 +3211,7 @@ ali_ac97_read(struct ac97_codec *codec, u8 reg) struct trident_card *card = NULL; /* Added by Matt Wu */ - if (!codec) - BUG(); + BUG_ON(!codec); card = (struct trident_card *) codec->private_data; @@ -3240,8 +3237,7 @@ ali_ac97_write(struct ac97_codec *codec, u8 reg, u16 val) struct trident_card *card; /* Added by Matt Wu */ - if (!codec) - BUG(); + BUG_ON(!codec); card = (struct trident_card *) codec->private_data; diff --git a/sound/oss/trident.h b/sound/oss/trident.h index 4713b49..ff30a1d 100644 --- a/sound/oss/trident.h +++ b/sound/oss/trident.h @@ -322,7 +322,7 @@ enum miscint_bits { #define VALIDATE_MAGIC(FOO,MAG) \ ({ \ if (!(FOO) || (FOO)->magic != MAG) { \ - printk(invalid_magic,__FUNCTION__); \ + printk(invalid_magic,__func__); \ return -ENXIO; \ } \ }) diff --git a/sound/oss/vwsnd.c b/sound/oss/vwsnd.c index d25249a..2c5aaa5 100644 --- a/sound/oss/vwsnd.c +++ b/sound/oss/vwsnd.c @@ -194,11 +194,11 @@ static void dbgassert(const char *fcn, int line, const char *expr) * DBGRV - debug print function return when verbose */ -#define ASSERT(e) ((e) ? (void) 0 : dbgassert(__FUNCTION__, __LINE__, #e)) +#define ASSERT(e) ((e) ? (void) 0 : dbgassert(__func__, __LINE__, #e)) #define DBGDO(x) x #define DBGX(fmt, args...) (in_interrupt() ? 0 : printk(KERN_ERR fmt, ##args)) -#define DBGP(fmt, args...) (DBGX("%s: " fmt, __FUNCTION__ , ##args)) -#define DBGE(fmt, args...) (DBGX("%s" fmt, __FUNCTION__ , ##args)) +#define DBGP(fmt, args...) (DBGX("%s: " fmt, __func__ , ##args)) +#define DBGE(fmt, args...) (DBGX("%s" fmt, __func__ , ##args)) #define DBGC(rtn) (DBGP("calling %s\n", rtn)) #define DBGR() (DBGP("returning\n")) #define DBGXV(fmt, args...) (shut_up ? 0 : DBGX(fmt, ##args)) diff --git a/sound/pci/Kconfig b/sound/pci/Kconfig index 812085d..581debf 100644 --- a/sound/pci/Kconfig +++ b/sound/pci/Kconfig @@ -122,6 +122,21 @@ config SND_AU8830 To compile this driver as a module, choose M here: the module will be called snd-au8830. +config SND_AW2 + tristate "Emagic Audiowerk 2" + depends on SND + help + Say Y here to include support for Emagic Audiowerk 2 soundcards. + + Supported features: Analog and SPDIF output. Analog or SPDIF input. + Note: Switch between analog and digital input does not always work. + It can produce continuous noise. The workaround is to switch again + (and again) between digital and analog input until it works. + + To compile this driver as a module, choose M here: the module + will be called snd-aw2. + + config SND_AZT3328 tristate "Aztech AZF3328 / PCI168 (EXPERIMENTAL)" depends on SND && EXPERIMENTAL @@ -162,6 +177,7 @@ config SND_CA0106 depends on SND select SND_AC97_CODEC select SND_RAWMIDI + select SND_VMASTER help Say Y here to include support for the Sound Blaster Audigy LS and Live 24bit. @@ -517,6 +533,7 @@ config SND_HDA_INTEL tristate "Intel HD Audio" depends on SND select SND_PCM + select SND_VMASTER help Say Y here to include support for Intel "High Definition Audio" (Azalia) motherboard devices. @@ -680,6 +697,7 @@ config SND_ICE1724 depends on SND select SND_MPU401_UART select SND_AC97_CODEC + select SND_VMASTER help Say Y here to include support for soundcards based on ICE/VT1724/1720 (Envy24HT/PT) chips. @@ -896,12 +914,12 @@ config SND_VIA82XX_MODEM will be called snd-via82xx-modem. config SND_VIRTUOSO - tristate "Asus Virtuoso 200 (Xonar)" + tristate "Asus Virtuoso 100/200 (Xonar)" depends on SND select SND_OXYGEN_LIB help Say Y here to include support for sound cards based on the - Asus AV200 chip, i.e., Xonar D2 and Xonar D2X. + Asus AV100/AV200 chips, i.e., Xonar D2, DX and D2X. To compile this driver as a module, choose M here: the module will be called snd-virtuoso. diff --git a/sound/pci/Makefile b/sound/pci/Makefile index 2d42fd2..85ef14b 100644 --- a/sound/pci/Makefile +++ b/sound/pci/Makefile @@ -58,6 +58,7 @@ obj-$(CONFIG_SND) += \ ac97/ \ ali5451/ \ au88x0/ \ + aw2/ \ ca0106/ \ cs46xx/ \ cs5535audio/ \ diff --git a/sound/pci/ac97/ac97_patch.c b/sound/pci/ac97/ac97_patch.c index 50c637e..39198e5 100644 --- a/sound/pci/ac97/ac97_patch.c +++ b/sound/pci/ac97/ac97_patch.c @@ -114,10 +114,9 @@ static int ac97_surround_jack_mode_put(struct snd_kcontrol *kcontrol, struct snd static int ac97_channel_mode_info(struct snd_kcontrol *kcontrol, struct snd_ctl_elem_info *uinfo) { - static const char *texts[] = { "2ch", "4ch", "6ch" }; - if (kcontrol->private_value) - return ac97_enum_text_info(kcontrol, uinfo, texts, 2); /* 4ch only */ - return ac97_enum_text_info(kcontrol, uinfo, texts, 3); + static const char *texts[] = { "2ch", "4ch", "6ch", "8ch" }; + return ac97_enum_text_info(kcontrol, uinfo, texts, + kcontrol->private_value); } static int ac97_channel_mode_get(struct snd_kcontrol *kcontrol, struct snd_ctl_elem_value *ucontrol) @@ -133,13 +132,8 @@ static int ac97_channel_mode_put(struct snd_kcontrol *kcontrol, struct snd_ctl_e struct snd_ac97 *ac97 = snd_kcontrol_chip(kcontrol); unsigned char mode = ucontrol->value.enumerated.item[0]; - if (kcontrol->private_value) { - if (mode >= 2) - return -EINVAL; - } else { - if (mode >= 3) - return -EINVAL; - } + if (mode >= kcontrol->private_value) + return -EINVAL; if (mode != ac97->channel_mode) { ac97->channel_mode = mode; @@ -158,6 +152,7 @@ static int ac97_channel_mode_put(struct snd_kcontrol *kcontrol, struct snd_ctl_e .get = ac97_surround_jack_mode_get, \ .put = ac97_surround_jack_mode_put, \ } +/* 6ch */ #define AC97_CHANNEL_MODE_CTL \ { \ .iface = SNDRV_CTL_ELEM_IFACE_MIXER, \ @@ -165,7 +160,9 @@ static int ac97_channel_mode_put(struct snd_kcontrol *kcontrol, struct snd_ctl_e .info = ac97_channel_mode_info, \ .get = ac97_channel_mode_get, \ .put = ac97_channel_mode_put, \ + .private_value = 3, \ } +/* 4ch */ #define AC97_CHANNEL_MODE_4CH_CTL \ { \ .iface = SNDRV_CTL_ELEM_IFACE_MIXER, \ @@ -173,7 +170,17 @@ static int ac97_channel_mode_put(struct snd_kcontrol *kcontrol, struct snd_ctl_e .info = ac97_channel_mode_info, \ .get = ac97_channel_mode_get, \ .put = ac97_channel_mode_put, \ - .private_value = 1, \ + .private_value = 2, \ + } +/* 8ch */ +#define AC97_CHANNEL_MODE_8CH_CTL \ + { \ + .iface = SNDRV_CTL_ELEM_IFACE_MIXER, \ + .name = "Channel Mode", \ + .info = ac97_channel_mode_info, \ + .get = ac97_channel_mode_get, \ + .put = ac97_channel_mode_put, \ + .private_value = 4, \ } static inline int is_surround_on(struct snd_ac97 *ac97) @@ -210,6 +217,10 @@ static inline int is_shared_micin(struct snd_ac97 *ac97) return !ac97->indep_surround && !is_clfe_on(ac97); } +static inline int alc850_is_aux_back_surround(struct snd_ac97 *ac97) +{ + return is_surround_on(ac97); +} /* The following snd_ac97_ymf753_... items added by David Shust (dshust@shustring.com) */ /* Modified for YMF743 by Keita Maehara */ @@ -2816,10 +2827,12 @@ static int patch_alc655(struct snd_ac97 * ac97) #define AC97_ALC850_JACK_SELECT 0x76 #define AC97_ALC850_MISC1 0x7a +#define AC97_ALC850_MULTICH 0x6a static void alc850_update_jacks(struct snd_ac97 *ac97) { int shared; + int aux_is_back_surround; /* shared Line-In / Surround Out */ shared = is_shared_surrout(ac97); @@ -2837,13 +2850,18 @@ static void alc850_update_jacks(struct snd_ac97 *ac97) /* MIC-IN = 1, CENTER-LFE = 5 */ snd_ac97_update_bits(ac97, AC97_ALC850_JACK_SELECT, 7 << 4, shared ? (5<<4) : (1<<4)); + + aux_is_back_surround = alc850_is_aux_back_surround(ac97); + /* Aux is Back Surround */ + snd_ac97_update_bits(ac97, AC97_ALC850_MULTICH, 1 << 10, + aux_is_back_surround ? (1<<10) : (0<<10)); } static const struct snd_kcontrol_new snd_ac97_controls_alc850[] = { AC97_PAGE_SINGLE("Duplicate Front", AC97_ALC650_MULTICH, 0, 1, 0, 0), AC97_SINGLE("Mic Front Input Switch", AC97_ALC850_JACK_SELECT, 15, 1, 1), AC97_SURROUND_JACK_MODE_CTL, - AC97_CHANNEL_MODE_CTL, + AC97_CHANNEL_MODE_8CH_CTL, }; static int patch_alc850_specific(struct snd_ac97 *ac97) @@ -2869,6 +2887,7 @@ static int patch_alc850(struct snd_ac97 *ac97) ac97->build_ops = &patch_alc850_ops; ac97->spec.dev_flags = 0; /* for IEC958 playback route - ALC655 compatible */ + ac97->flags |= AC97_HAS_8CH; /* assume only page 0 for writing cache */ snd_ac97_update_bits(ac97, AC97_INT_PAGING, AC97_PAGE_MASK, AC97_PAGE_VENDOR); @@ -2878,6 +2897,7 @@ static int patch_alc850(struct snd_ac97 *ac97) spdif-in monitor off, spdif-in PCM off center on mic off, surround on line-in off duplicate front off + NB default bit 10=0 = Aux is Capture, not Back Surround */ snd_ac97_write_cache(ac97, AC97_ALC650_MULTICH, 1<<15); /* SURR_OUT: on, Surr 1kOhm: on, Surr Amp: off, Front 1kOhm: off diff --git a/sound/pci/ac97/ac97_pcm.c b/sound/pci/ac97/ac97_pcm.c index 3674f35..48cbda9 100644 --- a/sound/pci/ac97/ac97_pcm.c +++ b/sound/pci/ac97/ac97_pcm.c @@ -574,7 +574,6 @@ int snd_ac97_pcm_open(struct ac97_pcm *pcm, unsigned int rate, r = rate > 48000; bus = pcm->bus; if (cfg == AC97_PCM_CFG_SPDIF) { - int err; for (cidx = 0; cidx < 4; cidx++) if (bus->codec[cidx] && (bus->codec[cidx]->ext_id & AC97_EI_SPDIF)) { err = set_spdif_rate(bus->codec[cidx], rate); diff --git a/sound/pci/ad1889.c b/sound/pci/ad1889.c index a66d515..1edb644 100644 --- a/sound/pci/ad1889.c +++ b/sound/pci/ad1889.c @@ -264,10 +264,10 @@ snd_ad1889_ac97_ready(struct snd_ad1889 *chip) mdelay(1); if (!retry) { snd_printk(KERN_ERR PFX "[%s] Link is not ready.\n", - __FUNCTION__); + __func__); return -EIO; } - ad1889_debug("[%s] ready after %d ms\n", __FUNCTION__, 400 - retry); + ad1889_debug("[%s] ready after %d ms\n", __func__, 400 - retry); return 0; } diff --git a/sound/pci/ali5451/ali5451.c b/sound/pci/ali5451/ali5451.c index 6a905ed..fc04d3d 100644 --- a/sound/pci/ali5451/ali5451.c +++ b/sound/pci/ali5451/ali5451.c @@ -1809,26 +1809,26 @@ static int snd_ali5451_spdif_get(struct snd_kcontrol *kcontrol, struct snd_ctl_elem_value *ucontrol) { struct snd_ali *codec = kcontrol->private_data; - unsigned int enable; + unsigned int spdif_enable; - enable = ucontrol->value.integer.value[0] ? 1 : 0; + spdif_enable = ucontrol->value.integer.value[0] ? 1 : 0; spin_lock_irq(&codec->reg_lock); switch (kcontrol->private_value) { case 0: - enable = (codec->spdif_mask & 0x02) ? 1 : 0; + spdif_enable = (codec->spdif_mask & 0x02) ? 1 : 0; break; case 1: - enable = ((codec->spdif_mask & 0x02) && + spdif_enable = ((codec->spdif_mask & 0x02) && (codec->spdif_mask & 0x04)) ? 1 : 0; break; case 2: - enable = (codec->spdif_mask & 0x01) ? 1 : 0; + spdif_enable = (codec->spdif_mask & 0x01) ? 1 : 0; break; default: break; } - ucontrol->value.integer.value[0] = enable; + ucontrol->value.integer.value[0] = spdif_enable; spin_unlock_irq(&codec->reg_lock); return 0; } @@ -1837,17 +1837,17 @@ static int snd_ali5451_spdif_put(struct snd_kcontrol *kcontrol, struct snd_ctl_elem_value *ucontrol) { struct snd_ali *codec = kcontrol->private_data; - unsigned int change = 0, enable = 0; + unsigned int change = 0, spdif_enable = 0; - enable = ucontrol->value.integer.value[0] ? 1 : 0; + spdif_enable = ucontrol->value.integer.value[0] ? 1 : 0; spin_lock_irq(&codec->reg_lock); switch (kcontrol->private_value) { case 0: change = (codec->spdif_mask & 0x02) ? 1 : 0; - change = change ^ enable; + change = change ^ spdif_enable; if (change) { - if (enable) { + if (spdif_enable) { codec->spdif_mask |= 0x02; snd_ali_enable_spdif_out(codec); } else { @@ -1859,9 +1859,9 @@ static int snd_ali5451_spdif_put(struct snd_kcontrol *kcontrol, break; case 1: change = (codec->spdif_mask & 0x04) ? 1 : 0; - change = change ^ enable; + change = change ^ spdif_enable; if (change && (codec->spdif_mask & 0x02)) { - if (enable) { + if (spdif_enable) { codec->spdif_mask |= 0x04; snd_ali_enable_spdif_chnout(codec); } else { @@ -1872,9 +1872,9 @@ static int snd_ali5451_spdif_put(struct snd_kcontrol *kcontrol, break; case 2: change = (codec->spdif_mask & 0x01) ? 1 : 0; - change = change ^ enable; + change = change ^ spdif_enable; if (change) { - if (enable) { + if (spdif_enable) { codec->spdif_mask |= 0x01; snd_ali_enable_spdif_in(codec); } else { diff --git a/sound/pci/als300.c b/sound/pci/als300.c index 0e990a7..8df6824 100644 --- a/sound/pci/als300.c +++ b/sound/pci/als300.c @@ -92,8 +92,8 @@ #if DEBUG_CALLS #define snd_als300_dbgcalls(format, args...) printk(format, ##args) -#define snd_als300_dbgcallenter() printk(KERN_ERR "--> %s\n", __FUNCTION__) -#define snd_als300_dbgcallleave() printk(KERN_ERR "<-- %s\n", __FUNCTION__) +#define snd_als300_dbgcallenter() printk(KERN_ERR "--> %s\n", __func__) +#define snd_als300_dbgcallleave() printk(KERN_ERR "<-- %s\n", __func__) #else #define snd_als300_dbgcalls(format, args...) #define snd_als300_dbgcallenter() diff --git a/sound/pci/au88x0/au88x0_pcm.c b/sound/pci/au88x0/au88x0_pcm.c index 526c6c5..f9a58b4 100644 --- a/sound/pci/au88x0/au88x0_pcm.c +++ b/sound/pci/au88x0/au88x0_pcm.c @@ -498,14 +498,14 @@ static struct snd_kcontrol_new snd_vortex_mixer_spdif[] __devinitdata = { }; /* create a pcm device */ -static int __devinit snd_vortex_new_pcm(vortex_t * chip, int idx, int nr) +static int __devinit snd_vortex_new_pcm(vortex_t *chip, int idx, int nr) { struct snd_pcm *pcm; struct snd_kcontrol *kctl; int i; int err, nr_capt; - if ((chip == 0) || (idx < 0) || (idx >= VORTEX_PCM_LAST)) + if (!chip || idx < 0 || idx >= VORTEX_PCM_LAST) return -ENODEV; /* idx indicates which kind of PCM device. ADB, SPDIF, I2S and A3D share the @@ -514,9 +514,9 @@ static int __devinit snd_vortex_new_pcm(vortex_t * chip, int idx, int nr) nr_capt = nr; else nr_capt = 0; - if ((err = - snd_pcm_new(chip->card, vortex_pcm_prettyname[idx], idx, nr, - nr_capt, &pcm)) < 0) + err = snd_pcm_new(chip->card, vortex_pcm_prettyname[idx], idx, nr, + nr_capt, &pcm); + if (err < 0) return err; strcpy(pcm->name, vortex_pcm_name[idx]); chip->pcm[idx] = pcm; diff --git a/sound/pci/aw2/Makefile b/sound/pci/aw2/Makefile new file mode 100644 index 0000000..842335d --- /dev/null +++ b/sound/pci/aw2/Makefile @@ -0,0 +1,3 @@ +snd-aw2-objs := aw2-alsa.o aw2-saa7146.o + +obj-$(CONFIG_SND_AW2) += snd-aw2.o diff --git a/sound/pci/aw2/aw2-alsa.c b/sound/pci/aw2/aw2-alsa.c new file mode 100644 index 0000000..56f87cd --- /dev/null +++ b/sound/pci/aw2/aw2-alsa.c @@ -0,0 +1,794 @@ +/***************************************************************************** + * + * Copyright (C) 2008 Cedric Bregardis and + * Jean-Christian Hassler + * + * This file is part of the Audiowerk2 ALSA driver + * + * The Audiowerk2 ALSA driver is free software; you can redistribute it and/or + * modify it under the terms of the GNU General Public License as published by + * the Free Software Foundation; version 2. + * + * The Audiowerk2 ALSA driver is distributed in the hope that it will be useful, + * but WITHOUT ANY WARRANTY; without even the implied warranty of + * MERCHANTABILITY or FITNESS FOR A PARTICULAR PURPOSE. See the + * GNU General Public License for more details. + * + * You should have received a copy of the GNU General Public License + * along with the Audiowerk2 ALSA driver; if not, write to the Free Software + * Foundation, Inc., 51 Franklin Street, Fifth Floor, Boston, MA 02110-1301, + * USA. + * + *****************************************************************************/ +#include +#include +#include +#include +#include +#include +#include +#include +#include +#include +#include +#include + +#include "saa7146.h" +#include "aw2-saa7146.h" + +MODULE_AUTHOR("Cedric Bregardis , " + "Jean-Christian Hassler "); +MODULE_DESCRIPTION("Emagic Audiowerk 2 sound driver"); +MODULE_LICENSE("GPL"); + +/********************************* + * DEFINES + ********************************/ +#define PCI_VENDOR_ID_SAA7146 0x1131 +#define PCI_DEVICE_ID_SAA7146 0x7146 + +#define CTL_ROUTE_ANALOG 0 +#define CTL_ROUTE_DIGITAL 1 + +/********************************* + * TYPEDEFS + ********************************/ + /* hardware definition */ +static struct snd_pcm_hardware snd_aw2_playback_hw = { + .info = (SNDRV_PCM_INFO_MMAP | + SNDRV_PCM_INFO_INTERLEAVED | + SNDRV_PCM_INFO_BLOCK_TRANSFER | SNDRV_PCM_INFO_MMAP_VALID), + .formats = SNDRV_PCM_FMTBIT_S16_LE, + .rates = SNDRV_PCM_RATE_44100, + .rate_min = 44100, + .rate_max = 44100, + .channels_min = 2, + .channels_max = 4, + .buffer_bytes_max = 32768, + .period_bytes_min = 4096, + .period_bytes_max = 32768, + .periods_min = 1, + .periods_max = 1024, +}; + +static struct snd_pcm_hardware snd_aw2_capture_hw = { + .info = (SNDRV_PCM_INFO_MMAP | + SNDRV_PCM_INFO_INTERLEAVED | + SNDRV_PCM_INFO_BLOCK_TRANSFER | SNDRV_PCM_INFO_MMAP_VALID), + .formats = SNDRV_PCM_FMTBIT_S16_LE, + .rates = SNDRV_PCM_RATE_44100, + .rate_min = 44100, + .rate_max = 44100, + .channels_min = 2, + .channels_max = 2, + .buffer_bytes_max = 32768, + .period_bytes_min = 4096, + .period_bytes_max = 32768, + .periods_min = 1, + .periods_max = 1024, +}; + +struct aw2_pcm_device { + struct snd_pcm *pcm; + unsigned int stream_number; + struct aw2 *chip; +}; + +struct aw2 { + struct snd_aw2_saa7146 saa7146; + + struct pci_dev *pci; + int irq; + spinlock_t reg_lock; + struct mutex mtx; + + unsigned long iobase_phys; + void __iomem *iobase_virt; + + struct snd_card *card; + + struct aw2_pcm_device device_playback[NB_STREAM_PLAYBACK]; + struct aw2_pcm_device device_capture[NB_STREAM_CAPTURE]; +}; + +/********************************* + * FUNCTION DECLARATIONS + ********************************/ +static int __init alsa_card_aw2_init(void); +static void __exit alsa_card_aw2_exit(void); +static int snd_aw2_dev_free(struct snd_device *device); +static int __devinit snd_aw2_create(struct snd_card *card, + struct pci_dev *pci, struct aw2 **rchip); +static int __devinit snd_aw2_probe(struct pci_dev *pci, + const struct pci_device_id *pci_id); +static void __devexit snd_aw2_remove(struct pci_dev *pci); +static int snd_aw2_pcm_playback_open(struct snd_pcm_substream *substream); +static int snd_aw2_pcm_playback_close(struct snd_pcm_substream *substream); +static int snd_aw2_pcm_capture_open(struct snd_pcm_substream *substream); +static int snd_aw2_pcm_capture_close(struct snd_pcm_substream *substream); +static int snd_aw2_pcm_hw_params(struct snd_pcm_substream *substream, + struct snd_pcm_hw_params *hw_params); +static int snd_aw2_pcm_hw_free(struct snd_pcm_substream *substream); +static int snd_aw2_pcm_prepare_playback(struct snd_pcm_substream *substream); +static int snd_aw2_pcm_prepare_capture(struct snd_pcm_substream *substream); +static int snd_aw2_pcm_trigger_playback(struct snd_pcm_substream *substream, + int cmd); +static int snd_aw2_pcm_trigger_capture(struct snd_pcm_substream *substream, + int cmd); +static snd_pcm_uframes_t snd_aw2_pcm_pointer_playback(struct snd_pcm_substream + *substream); +static snd_pcm_uframes_t snd_aw2_pcm_pointer_capture(struct snd_pcm_substream + *substream); +static int __devinit snd_aw2_new_pcm(struct aw2 *chip); + +static int snd_aw2_control_switch_capture_info(struct snd_kcontrol *kcontrol, + struct snd_ctl_elem_info *uinfo); +static int snd_aw2_control_switch_capture_get(struct snd_kcontrol *kcontrol, + struct snd_ctl_elem_value + *ucontrol); +static int snd_aw2_control_switch_capture_put(struct snd_kcontrol *kcontrol, + struct snd_ctl_elem_value + *ucontrol); + +/********************************* + * VARIABLES + ********************************/ +static int index[SNDRV_CARDS] = SNDRV_DEFAULT_IDX; +static char *id[SNDRV_CARDS] = SNDRV_DEFAULT_STR; +static int enable[SNDRV_CARDS] = SNDRV_DEFAULT_ENABLE_PNP; + +module_param_array(index, int, NULL, 0444); +MODULE_PARM_DESC(index, "Index value for Audiowerk2 soundcard."); +module_param_array(id, charp, NULL, 0444); +MODULE_PARM_DESC(id, "ID string for the Audiowerk2 soundcard."); +module_param_array(enable, bool, NULL, 0444); +MODULE_PARM_DESC(enable, "Enable Audiowerk2 soundcard."); + +static struct pci_device_id snd_aw2_ids[] = { + {PCI_VENDOR_ID_SAA7146, PCI_DEVICE_ID_SAA7146, PCI_ANY_ID, PCI_ANY_ID, + 0, 0, 0}, + {0} +}; + +MODULE_DEVICE_TABLE(pci, snd_aw2_ids); + +/* pci_driver definition */ +static struct pci_driver driver = { + .name = "Emagic Audiowerk 2", + .id_table = snd_aw2_ids, + .probe = snd_aw2_probe, + .remove = __devexit_p(snd_aw2_remove), +}; + +/* operators for playback PCM alsa interface */ +static struct snd_pcm_ops snd_aw2_playback_ops = { + .open = snd_aw2_pcm_playback_open, + .close = snd_aw2_pcm_playback_close, + .ioctl = snd_pcm_lib_ioctl, + .hw_params = snd_aw2_pcm_hw_params, + .hw_free = snd_aw2_pcm_hw_free, + .prepare = snd_aw2_pcm_prepare_playback, + .trigger = snd_aw2_pcm_trigger_playback, + .pointer = snd_aw2_pcm_pointer_playback, +}; + +/* operators for capture PCM alsa interface */ +static struct snd_pcm_ops snd_aw2_capture_ops = { + .open = snd_aw2_pcm_capture_open, + .close = snd_aw2_pcm_capture_close, + .ioctl = snd_pcm_lib_ioctl, + .hw_params = snd_aw2_pcm_hw_params, + .hw_free = snd_aw2_pcm_hw_free, + .prepare = snd_aw2_pcm_prepare_capture, + .trigger = snd_aw2_pcm_trigger_capture, + .pointer = snd_aw2_pcm_pointer_capture, +}; + +static struct snd_kcontrol_new aw2_control __devinitdata = { + .iface = SNDRV_CTL_ELEM_IFACE_MIXER, + .name = "PCM Capture Route", + .index = 0, + .access = SNDRV_CTL_ELEM_ACCESS_READWRITE, + .private_value = 0xffff, + .info = snd_aw2_control_switch_capture_info, + .get = snd_aw2_control_switch_capture_get, + .put = snd_aw2_control_switch_capture_put +}; + +/********************************* + * FUNCTION IMPLEMENTATIONS + ********************************/ + +/* initialization of the module */ +static int __init alsa_card_aw2_init(void) +{ + snd_printdd(KERN_DEBUG "aw2: Load aw2 module\n"); + return pci_register_driver(&driver); +} + +/* clean up the module */ +static void __exit alsa_card_aw2_exit(void) +{ + snd_printdd(KERN_DEBUG "aw2: Unload aw2 module\n"); + pci_unregister_driver(&driver); +} + +module_init(alsa_card_aw2_init); +module_exit(alsa_card_aw2_exit); + +/* component-destructor */ +static int snd_aw2_dev_free(struct snd_device *device) +{ + struct aw2 *chip = device->device_data; + + /* Free hardware */ + snd_aw2_saa7146_free(&chip->saa7146); + + /* release the irq */ + if (chip->irq >= 0) + free_irq(chip->irq, (void *)chip); + /* release the i/o ports & memory */ + if (chip->iobase_virt) + iounmap(chip->iobase_virt); + + pci_release_regions(chip->pci); + /* disable the PCI entry */ + pci_disable_device(chip->pci); + /* release the data */ + kfree(chip); + + return 0; +} + +/* chip-specific constructor */ +static int __devinit snd_aw2_create(struct snd_card *card, + struct pci_dev *pci, struct aw2 **rchip) +{ + struct aw2 *chip; + int err; + static struct snd_device_ops ops = { + .dev_free = snd_aw2_dev_free, + }; + + *rchip = NULL; + + /* initialize the PCI entry */ + err = pci_enable_device(pci); + if (err < 0) + return err; + pci_set_master(pci); + + /* check PCI availability (32bit DMA) */ + if ((pci_set_dma_mask(pci, DMA_32BIT_MASK) < 0) || + (pci_set_consistent_dma_mask(pci, DMA_32BIT_MASK) < 0)) { + printk(KERN_ERR "aw2: Impossible to set 32bit mask DMA\n"); + pci_disable_device(pci); + return -ENXIO; + } + chip = kzalloc(sizeof(*chip), GFP_KERNEL); + if (chip == NULL) { + pci_disable_device(pci); + return -ENOMEM; + } + + /* initialize the stuff */ + chip->card = card; + chip->pci = pci; + chip->irq = -1; + + /* (1) PCI resource allocation */ + err = pci_request_regions(pci, "Audiowerk2"); + if (err < 0) { + pci_disable_device(pci); + kfree(chip); + return err; + } + chip->iobase_phys = pci_resource_start(pci, 0); + chip->iobase_virt = + ioremap_nocache(chip->iobase_phys, + pci_resource_len(pci, 0)); + + if (chip->iobase_virt == NULL) { + printk(KERN_ERR "aw2: unable to remap memory region"); + pci_release_regions(pci); + pci_disable_device(pci); + kfree(chip); + return -ENOMEM; + } + + + if (request_irq(pci->irq, snd_aw2_saa7146_interrupt, + IRQF_SHARED, "Audiowerk2", chip)) { + printk(KERN_ERR "aw2: Cannot grab irq %d\n", pci->irq); + + iounmap(chip->iobase_virt); + pci_release_regions(chip->pci); + pci_disable_device(chip->pci); + kfree(chip); + return -EBUSY; + } + chip->irq = pci->irq; + + /* (2) initialization of the chip hardware */ + snd_aw2_saa7146_setup(&chip->saa7146, chip->iobase_virt); + err = snd_device_new(card, SNDRV_DEV_LOWLEVEL, chip, &ops); + if (err < 0) { + free_irq(chip->irq, (void *)chip); + iounmap(chip->iobase_virt); + pci_release_regions(chip->pci); + pci_disable_device(chip->pci); + kfree(chip); + return err; + } + + snd_card_set_dev(card, &pci->dev); + *rchip = chip; + + printk(KERN_INFO + "Audiowerk 2 sound card (saa7146 chipset) detected and " + "managed\n"); + return 0; +} + +/* constructor */ +static int __devinit snd_aw2_probe(struct pci_dev *pci, + const struct pci_device_id *pci_id) +{ + static int dev; + struct snd_card *card; + struct aw2 *chip; + int err; + + /* (1) Continue if device is not enabled, else inc dev */ + if (dev >= SNDRV_CARDS) + return -ENODEV; + if (!enable[dev]) { + dev++; + return -ENOENT; + } + + /* (2) Create card instance */ + card = snd_card_new(index[dev], id[dev], THIS_MODULE, 0); + if (card == NULL) + return -ENOMEM; + + /* (3) Create main component */ + err = snd_aw2_create(card, pci, &chip); + if (err < 0) { + snd_card_free(card); + return err; + } + + /* initialize mutex */ + mutex_init(&chip->mtx); + /* init spinlock */ + spin_lock_init(&chip->reg_lock); + /* (4) Define driver ID and name string */ + strcpy(card->driver, "aw2"); + strcpy(card->shortname, "Audiowerk2"); + + sprintf(card->longname, "%s with SAA7146 irq %i", + card->shortname, chip->irq); + + /* (5) Create other components */ + snd_aw2_new_pcm(chip); + + /* (6) Register card instance */ + err = snd_card_register(card); + if (err < 0) { + snd_card_free(card); + return err; + } + + /* (7) Set PCI driver data */ + pci_set_drvdata(pci, card); + + dev++; + return 0; +} + +/* destructor */ +static void __devexit snd_aw2_remove(struct pci_dev *pci) +{ + snd_card_free(pci_get_drvdata(pci)); + pci_set_drvdata(pci, NULL); +} + +/* open callback */ +static int snd_aw2_pcm_playback_open(struct snd_pcm_substream *substream) +{ + struct snd_pcm_runtime *runtime = substream->runtime; + + snd_printdd(KERN_DEBUG "aw2: Playback_open \n"); + runtime->hw = snd_aw2_playback_hw; + return 0; +} + +/* close callback */ +static int snd_aw2_pcm_playback_close(struct snd_pcm_substream *substream) +{ + return 0; + +} + +static int snd_aw2_pcm_capture_open(struct snd_pcm_substream *substream) +{ + struct snd_pcm_runtime *runtime = substream->runtime; + + snd_printdd(KERN_DEBUG "aw2: Capture_open \n"); + runtime->hw = snd_aw2_capture_hw; + return 0; +} + +/* close callback */ +static int snd_aw2_pcm_capture_close(struct snd_pcm_substream *substream) +{ + /* TODO: something to do ? */ + return 0; +} + + /* hw_params callback */ +static int snd_aw2_pcm_hw_params(struct snd_pcm_substream *substream, + struct snd_pcm_hw_params *hw_params) +{ + return snd_pcm_lib_malloc_pages(substream, + params_buffer_bytes(hw_params)); +} + +/* hw_free callback */ +static int snd_aw2_pcm_hw_free(struct snd_pcm_substream *substream) +{ + return snd_pcm_lib_free_pages(substream); +} + +/* prepare callback for playback */ +static int snd_aw2_pcm_prepare_playback(struct snd_pcm_substream *substream) +{ + struct aw2_pcm_device *pcm_device = snd_pcm_substream_chip(substream); + struct aw2 *chip = pcm_device->chip; + struct snd_pcm_runtime *runtime = substream->runtime; + unsigned long period_size, buffer_size; + + mutex_lock(&chip->mtx); + + period_size = snd_pcm_lib_period_bytes(substream); + buffer_size = snd_pcm_lib_buffer_bytes(substream); + + snd_aw2_saa7146_pcm_init_playback(&chip->saa7146, + pcm_device->stream_number, + runtime->dma_addr, period_size, + buffer_size); + + /* Define Interrupt callback */ + snd_aw2_saa7146_define_it_playback_callback(pcm_device->stream_number, + (snd_aw2_saa7146_it_cb) + snd_pcm_period_elapsed, + (void *)substream); + + mutex_unlock(&chip->mtx); + + return 0; +} + +/* prepare callback for capture */ +static int snd_aw2_pcm_prepare_capture(struct snd_pcm_substream *substream) +{ + struct aw2_pcm_device *pcm_device = snd_pcm_substream_chip(substream); + struct aw2 *chip = pcm_device->chip; + struct snd_pcm_runtime *runtime = substream->runtime; + unsigned long period_size, buffer_size; + + mutex_lock(&chip->mtx); + + period_size = snd_pcm_lib_period_bytes(substream); + buffer_size = snd_pcm_lib_buffer_bytes(substream); + + snd_aw2_saa7146_pcm_init_capture(&chip->saa7146, + pcm_device->stream_number, + runtime->dma_addr, period_size, + buffer_size); + + /* Define Interrupt callback */ + snd_aw2_saa7146_define_it_capture_callback(pcm_device->stream_number, + (snd_aw2_saa7146_it_cb) + snd_pcm_period_elapsed, + (void *)substream); + + mutex_unlock(&chip->mtx); + + return 0; +} + +/* playback trigger callback */ +static int snd_aw2_pcm_trigger_playback(struct snd_pcm_substream *substream, + int cmd) +{ + int status = 0; + struct aw2_pcm_device *pcm_device = snd_pcm_substream_chip(substream); + struct aw2 *chip = pcm_device->chip; + spin_lock(&chip->reg_lock); + switch (cmd) { + case SNDRV_PCM_TRIGGER_START: + snd_aw2_saa7146_pcm_trigger_start_playback(&chip->saa7146, + pcm_device-> + stream_number); + break; + case SNDRV_PCM_TRIGGER_STOP: + snd_aw2_saa7146_pcm_trigger_stop_playback(&chip->saa7146, + pcm_device-> + stream_number); + break; + default: + status = -EINVAL; + } + spin_unlock(&chip->reg_lock); + return status; +} + +/* capture trigger callback */ +static int snd_aw2_pcm_trigger_capture(struct snd_pcm_substream *substream, + int cmd) +{ + int status = 0; + struct aw2_pcm_device *pcm_device = snd_pcm_substream_chip(substream); + struct aw2 *chip = pcm_device->chip; + spin_lock(&chip->reg_lock); + switch (cmd) { + case SNDRV_PCM_TRIGGER_START: + snd_aw2_saa7146_pcm_trigger_start_capture(&chip->saa7146, + pcm_device-> + stream_number); + break; + case SNDRV_PCM_TRIGGER_STOP: + snd_aw2_saa7146_pcm_trigger_stop_capture(&chip->saa7146, + pcm_device-> + stream_number); + break; + default: + status = -EINVAL; + } + spin_unlock(&chip->reg_lock); + return status; +} + +/* playback pointer callback */ +static snd_pcm_uframes_t snd_aw2_pcm_pointer_playback(struct snd_pcm_substream + *substream) +{ + struct aw2_pcm_device *pcm_device = snd_pcm_substream_chip(substream); + struct aw2 *chip = pcm_device->chip; + unsigned int current_ptr; + + /* get the current hardware pointer */ + struct snd_pcm_runtime *runtime = substream->runtime; + current_ptr = + snd_aw2_saa7146_get_hw_ptr_playback(&chip->saa7146, + pcm_device->stream_number, + runtime->dma_area, + runtime->buffer_size); + + return bytes_to_frames(substream->runtime, current_ptr); +} + +/* capture pointer callback */ +static snd_pcm_uframes_t snd_aw2_pcm_pointer_capture(struct snd_pcm_substream + *substream) +{ + struct aw2_pcm_device *pcm_device = snd_pcm_substream_chip(substream); + struct aw2 *chip = pcm_device->chip; + unsigned int current_ptr; + + /* get the current hardware pointer */ + struct snd_pcm_runtime *runtime = substream->runtime; + current_ptr = + snd_aw2_saa7146_get_hw_ptr_capture(&chip->saa7146, + pcm_device->stream_number, + runtime->dma_area, + runtime->buffer_size); + + return bytes_to_frames(substream->runtime, current_ptr); +} + +/* create a pcm device */ +static int __devinit snd_aw2_new_pcm(struct aw2 *chip) +{ + struct snd_pcm *pcm_playback_ana; + struct snd_pcm *pcm_playback_num; + struct snd_pcm *pcm_capture; + struct aw2_pcm_device *pcm_device; + int err = 0; + + /* Create new Alsa PCM device */ + + err = snd_pcm_new(chip->card, "Audiowerk2 analog playback", 0, 1, 0, + &pcm_playback_ana); + if (err < 0) { + printk(KERN_ERR "aw2: snd_pcm_new error (0x%X)\n", err); + return err; + } + + /* Creation ok */ + pcm_device = &chip->device_playback[NUM_STREAM_PLAYBACK_ANA]; + + /* Set PCM device name */ + strcpy(pcm_playback_ana->name, "Analog playback"); + /* Associate private data to PCM device */ + pcm_playback_ana->private_data = pcm_device; + /* set operators of PCM device */ + snd_pcm_set_ops(pcm_playback_ana, SNDRV_PCM_STREAM_PLAYBACK, + &snd_aw2_playback_ops); + /* store PCM device */ + pcm_device->pcm = pcm_playback_ana; + /* give base chip pointer to our internal pcm device + structure */ + pcm_device->chip = chip; + /* Give stream number to PCM device */ + pcm_device->stream_number = NUM_STREAM_PLAYBACK_ANA; + + /* pre-allocation of buffers */ + /* Preallocate continuous pages. */ + err = snd_pcm_lib_preallocate_pages_for_all(pcm_playback_ana, + SNDRV_DMA_TYPE_DEV, + snd_dma_pci_data + (chip->pci), + 64 * 1024, 64 * 1024); + if (err) + printk(KERN_ERR "aw2: snd_pcm_lib_preallocate_pages_for_all " + "error (0x%X)\n", err); + + err = snd_pcm_new(chip->card, "Audiowerk2 digital playback", 1, 1, 0, + &pcm_playback_num); + + if (err < 0) { + printk(KERN_ERR "aw2: snd_pcm_new error (0x%X)\n", err); + return err; + } + /* Creation ok */ + pcm_device = &chip->device_playback[NUM_STREAM_PLAYBACK_DIG]; + + /* Set PCM device name */ + strcpy(pcm_playback_num->name, "Digital playback"); + /* Associate private data to PCM device */ + pcm_playback_num->private_data = pcm_device; + /* set operators of PCM device */ + snd_pcm_set_ops(pcm_playback_num, SNDRV_PCM_STREAM_PLAYBACK, + &snd_aw2_playback_ops); + /* store PCM device */ + pcm_device->pcm = pcm_playback_num; + /* give base chip pointer to our internal pcm device + structure */ + pcm_device->chip = chip; + /* Give stream number to PCM device */ + pcm_device->stream_number = NUM_STREAM_PLAYBACK_DIG; + + /* pre-allocation of buffers */ + /* Preallocate continuous pages. */ + err = snd_pcm_lib_preallocate_pages_for_all(pcm_playback_num, + SNDRV_DMA_TYPE_DEV, + snd_dma_pci_data + (chip->pci), + 64 * 1024, 64 * 1024); + if (err) + printk(KERN_ERR + "aw2: snd_pcm_lib_preallocate_pages_for_all error " + "(0x%X)\n", err); + + + + err = snd_pcm_new(chip->card, "Audiowerk2 capture", 2, 0, 1, + &pcm_capture); + + if (err < 0) { + printk(KERN_ERR "aw2: snd_pcm_new error (0x%X)\n", err); + return err; + } + + /* Creation ok */ + pcm_device = &chip->device_capture[NUM_STREAM_CAPTURE_ANA]; + + /* Set PCM device name */ + strcpy(pcm_capture->name, "Capture"); + /* Associate private data to PCM device */ + pcm_capture->private_data = pcm_device; + /* set operators of PCM device */ + snd_pcm_set_ops(pcm_capture, SNDRV_PCM_STREAM_CAPTURE, + &snd_aw2_capture_ops); + /* store PCM device */ + pcm_device->pcm = pcm_capture; + /* give base chip pointer to our internal pcm device + structure */ + pcm_device->chip = chip; + /* Give stream number to PCM device */ + pcm_device->stream_number = NUM_STREAM_CAPTURE_ANA; + + /* pre-allocation of buffers */ + /* Preallocate continuous pages. */ + err = snd_pcm_lib_preallocate_pages_for_all(pcm_capture, + SNDRV_DMA_TYPE_DEV, + snd_dma_pci_data + (chip->pci), + 64 * 1024, 64 * 1024); + if (err) + printk(KERN_ERR + "aw2: snd_pcm_lib_preallocate_pages_for_all error " + "(0x%X)\n", err); + + + /* Create control */ + err = snd_ctl_add(chip->card, snd_ctl_new1(&aw2_control, chip)); + if (err < 0) { + printk(KERN_ERR "aw2: snd_ctl_add error (0x%X)\n", err); + return err; + } + + return 0; +} + +static int snd_aw2_control_switch_capture_info(struct snd_kcontrol *kcontrol, + struct snd_ctl_elem_info *uinfo) +{ + static char *texts[2] = { + "Analog", "Digital" + }; + uinfo->type = SNDRV_CTL_ELEM_TYPE_ENUMERATED; + uinfo->count = 1; + uinfo->value.enumerated.items = 2; + if (uinfo->value.enumerated.item >= uinfo->value.enumerated.items) { + uinfo->value.enumerated.item = + uinfo->value.enumerated.items - 1; + } + strcpy(uinfo->value.enumerated.name, + texts[uinfo->value.enumerated.item]); + return 0; +} + +static int snd_aw2_control_switch_capture_get(struct snd_kcontrol *kcontrol, + struct snd_ctl_elem_value + *ucontrol) +{ + struct aw2 *chip = snd_kcontrol_chip(kcontrol); + if (snd_aw2_saa7146_is_using_digital_input(&chip->saa7146)) + ucontrol->value.enumerated.item[0] = CTL_ROUTE_DIGITAL; + else + ucontrol->value.enumerated.item[0] = CTL_ROUTE_ANALOG; + return 0; +} + +static int snd_aw2_control_switch_capture_put(struct snd_kcontrol *kcontrol, + struct snd_ctl_elem_value + *ucontrol) +{ + struct aw2 *chip = snd_kcontrol_chip(kcontrol); + int changed = 0; + int is_disgital = + snd_aw2_saa7146_is_using_digital_input(&chip->saa7146); + + if (((ucontrol->value.integer.value[0] == CTL_ROUTE_DIGITAL) + && !is_disgital) + || ((ucontrol->value.integer.value[0] == CTL_ROUTE_ANALOG) + && is_disgital)) { + snd_aw2_saa7146_use_digital_input(&chip->saa7146, !is_disgital); + changed = 1; + } + return changed; +} diff --git a/sound/pci/aw2/aw2-saa7146.c b/sound/pci/aw2/aw2-saa7146.c new file mode 100644 index 0000000..6a3891a --- /dev/null +++ b/sound/pci/aw2/aw2-saa7146.c @@ -0,0 +1,465 @@ +/***************************************************************************** + * + * Copyright (C) 2008 Cedric Bregardis and + * Jean-Christian Hassler + * + * This file is part of the Audiowerk2 ALSA driver + * + * The Audiowerk2 ALSA driver is free software; you can redistribute it and/or + * modify it under the terms of the GNU General Public License as published by + * the Free Software Foundation; version 2. + * + * The Audiowerk2 ALSA driver is distributed in the hope that it will be useful, + * but WITHOUT ANY WARRANTY; without even the implied warranty of + * MERCHANTABILITY or FITNESS FOR A PARTICULAR PURPOSE. See the + * GNU General Public License for more details. + * + * You should have received a copy of the GNU General Public License + * along with the Audiowerk2 ALSA driver; if not, write to the Free Software + * Foundation, Inc., 51 Franklin Street, Fifth Floor, Boston, MA 02110-1301, + * USA. + * + *****************************************************************************/ + +#define AW2_SAA7146_M + +#include +#include +#include +#include +#include +#include +#include +#include +#include +#include +#include + +#include "saa7146.h" +#include "aw2-saa7146.h" + +#include "aw2-tsl.c" + +#define WRITEREG(value, addr) writel((value), chip->base_addr + (addr)) +#define READREG(addr) readl(chip->base_addr + (addr)) + +static struct snd_aw2_saa7146_cb_param + arr_substream_it_playback_cb[NB_STREAM_PLAYBACK]; +static struct snd_aw2_saa7146_cb_param + arr_substream_it_capture_cb[NB_STREAM_CAPTURE]; + +static int snd_aw2_saa7146_get_limit(int size); + +/* chip-specific destructor */ +int snd_aw2_saa7146_free(struct snd_aw2_saa7146 *chip) +{ + /* disable all irqs */ + WRITEREG(0, IER); + + /* reset saa7146 */ + WRITEREG((MRST_N << 16), MC1); + + /* Unset base addr */ + chip->base_addr = NULL; + + return 0; +} + +void snd_aw2_saa7146_setup(struct snd_aw2_saa7146 *chip, + void __iomem *pci_base_addr) +{ + /* set PCI burst/threshold + + Burst length definition + VALUE BURST LENGTH + 000 1 Dword + 001 2 Dwords + 010 4 Dwords + 011 8 Dwords + 100 16 Dwords + 101 32 Dwords + 110 64 Dwords + 111 128 Dwords + + Threshold definition + VALUE WRITE MODE READ MODE + 00 1 Dword of valid data 1 empty Dword + 01 4 Dwords of valid data 4 empty Dwords + 10 8 Dwords of valid data 8 empty Dwords + 11 16 Dwords of valid data 16 empty Dwords */ + + unsigned int acon2; + unsigned int acon1 = 0; + int i; + + /* Set base addr */ + chip->base_addr = pci_base_addr; + + /* disable all irqs */ + WRITEREG(0, IER); + + /* reset saa7146 */ + WRITEREG((MRST_N << 16), MC1); + + /* enable audio interface */ +#ifdef __BIG_ENDIAN + acon1 |= A1_SWAP; + acon1 |= A2_SWAP; +#endif + /* WS0_CTRL, WS0_SYNC: input TSL1, I2S */ + + /* At initialization WS1 and WS2 are disbaled (configured as input */ + acon1 |= 0 * WS1_CTRL; + acon1 |= 0 * WS2_CTRL; + + /* WS4 is not used. So it must not restart A2. + This is why it is configured as output (force to low) */ + acon1 |= 3 * WS4_CTRL; + + /* WS3_CTRL, WS3_SYNC: output TSL2, I2S */ + acon1 |= 2 * WS3_CTRL; + + /* A1 and A2 are active and asynchronous */ + acon1 |= 3 * AUDIO_MODE; + WRITEREG(acon1, ACON1); + + /* The following comes from original windows driver. + It is needed to have a correct behavior of input and output + simultenously, but I don't know why ! */ + WRITEREG(3 * (BurstA1_in) + 3 * (ThreshA1_in) + + 3 * (BurstA1_out) + 3 * (ThreshA1_out) + + 3 * (BurstA2_out) + 3 * (ThreshA2_out), PCI_BT_A); + + /* enable audio port pins */ + WRITEREG((EAP << 16) | EAP, MC1); + + /* enable I2C */ + WRITEREG((EI2C << 16) | EI2C, MC1); + /* enable interrupts */ + WRITEREG(A1_out | A2_out | A1_in | IIC_S | IIC_E, IER); + + /* audio configuration */ + acon2 = A2_CLKSRC | BCLK1_OEN; + WRITEREG(acon2, ACON2); + + /* By default use analog input */ + snd_aw2_saa7146_use_digital_input(chip, 0); + + /* TSL setup */ + for (i = 0; i < 8; ++i) { + WRITEREG(tsl1[i], TSL1 + (i * 4)); + WRITEREG(tsl2[i], TSL2 + (i * 4)); + } + +} + +void snd_aw2_saa7146_pcm_init_playback(struct snd_aw2_saa7146 *chip, + int stream_number, + unsigned long dma_addr, + unsigned long period_size, + unsigned long buffer_size) +{ + unsigned long dw_page, dw_limit; + + /* Configure DMA for substream + Configuration informations: ALSA has allocated continuous memory + pages. So we don't need to use MMU of saa7146. + */ + + /* No MMU -> nothing to do with PageA1, we only configure the limit of + PageAx_out register */ + /* Disable MMU */ + dw_page = (0L << 11); + + /* Configure Limit for DMA access. + The limit register defines an address limit, which generates + an interrupt if passed by the actual PCI address pointer. + '0001' means an interrupt will be generated if the lower + 6 bits (64 bytes) of the PCI address are zero. '0010' + defines a limit of 128 bytes, '0011' one of 256 bytes, and + so on up to 1 Mbyte defined by '1111'. This interrupt range + can be calculated as follows: + Range = 2^(5 + Limit) bytes. + */ + dw_limit = snd_aw2_saa7146_get_limit(period_size); + dw_page |= (dw_limit << 4); + + if (stream_number == 0) { + WRITEREG(dw_page, PageA2_out); + + /* Base address for DMA transfert. */ + /* This address has been reserved by ALSA. */ + /* This is a physical address */ + WRITEREG(dma_addr, BaseA2_out); + + /* Define upper limit for DMA access */ + WRITEREG(dma_addr + buffer_size, ProtA2_out); + + } else if (stream_number == 1) { + WRITEREG(dw_page, PageA1_out); + + /* Base address for DMA transfert. */ + /* This address has been reserved by ALSA. */ + /* This is a physical address */ + WRITEREG(dma_addr, BaseA1_out); + + /* Define upper limit for DMA access */ + WRITEREG(dma_addr + buffer_size, ProtA1_out); + } else { + printk(KERN_ERR + "aw2: snd_aw2_saa7146_pcm_init_playback: " + "Substream number is not 0 or 1 -> not managed\n"); + } +} + +void snd_aw2_saa7146_pcm_init_capture(struct snd_aw2_saa7146 *chip, + int stream_number, unsigned long dma_addr, + unsigned long period_size, + unsigned long buffer_size) +{ + unsigned long dw_page, dw_limit; + + /* Configure DMA for substream + Configuration informations: ALSA has allocated continuous memory + pages. So we don't need to use MMU of saa7146. + */ + + /* No MMU -> nothing to do with PageA1, we only configure the limit of + PageAx_out register */ + /* Disable MMU */ + dw_page = (0L << 11); + + /* Configure Limit for DMA access. + The limit register defines an address limit, which generates + an interrupt if passed by the actual PCI address pointer. + '0001' means an interrupt will be generated if the lower + 6 bits (64 bytes) of the PCI address are zero. '0010' + defines a limit of 128 bytes, '0011' one of 256 bytes, and + so on up to 1 Mbyte defined by '1111'. This interrupt range + can be calculated as follows: + Range = 2^(5 + Limit) bytes. + */ + dw_limit = snd_aw2_saa7146_get_limit(period_size); + dw_page |= (dw_limit << 4); + + if (stream_number == 0) { + WRITEREG(dw_page, PageA1_in); + + /* Base address for DMA transfert. */ + /* This address has been reserved by ALSA. */ + /* This is a physical address */ + WRITEREG(dma_addr, BaseA1_in); + + /* Define upper limit for DMA access */ + WRITEREG(dma_addr + buffer_size, ProtA1_in); + } else { + printk(KERN_ERR + "aw2: snd_aw2_saa7146_pcm_init_capture: " + "Substream number is not 0 -> not managed\n"); + } +} + +void snd_aw2_saa7146_define_it_playback_callback(unsigned int stream_number, + snd_aw2_saa7146_it_cb + p_it_callback, + void *p_callback_param) +{ + if (stream_number < NB_STREAM_PLAYBACK) { + arr_substream_it_playback_cb[stream_number].p_it_callback = + (snd_aw2_saa7146_it_cb) p_it_callback; + arr_substream_it_playback_cb[stream_number].p_callback_param = + (void *)p_callback_param; + } +} + +void snd_aw2_saa7146_define_it_capture_callback(unsigned int stream_number, + snd_aw2_saa7146_it_cb + p_it_callback, + void *p_callback_param) +{ + if (stream_number < NB_STREAM_CAPTURE) { + arr_substream_it_capture_cb[stream_number].p_it_callback = + (snd_aw2_saa7146_it_cb) p_it_callback; + arr_substream_it_capture_cb[stream_number].p_callback_param = + (void *)p_callback_param; + } +} + +void snd_aw2_saa7146_pcm_trigger_start_playback(struct snd_aw2_saa7146 *chip, + int stream_number) +{ + unsigned int acon1 = 0; + /* In aw8 driver, dma transfert is always active. It is + started and stopped in a larger "space" */ + acon1 = READREG(ACON1); + if (stream_number == 0) { + WRITEREG((TR_E_A2_OUT << 16) | TR_E_A2_OUT, MC1); + + /* WS2_CTRL, WS2_SYNC: output TSL2, I2S */ + acon1 |= 2 * WS2_CTRL; + WRITEREG(acon1, ACON1); + + } else if (stream_number == 1) { + WRITEREG((TR_E_A1_OUT << 16) | TR_E_A1_OUT, MC1); + + /* WS1_CTRL, WS1_SYNC: output TSL1, I2S */ + acon1 |= 1 * WS1_CTRL; + WRITEREG(acon1, ACON1); + } +} + +void snd_aw2_saa7146_pcm_trigger_stop_playback(struct snd_aw2_saa7146 *chip, + int stream_number) +{ + unsigned int acon1 = 0; + acon1 = READREG(ACON1); + if (stream_number == 0) { + /* WS2_CTRL, WS2_SYNC: output TSL2, I2S */ + acon1 &= ~(3 * WS2_CTRL); + WRITEREG(acon1, ACON1); + + WRITEREG((TR_E_A2_OUT << 16), MC1); + } else if (stream_number == 1) { + /* WS1_CTRL, WS1_SYNC: output TSL1, I2S */ + acon1 &= ~(3 * WS1_CTRL); + WRITEREG(acon1, ACON1); + + WRITEREG((TR_E_A1_OUT << 16), MC1); + } +} + +void snd_aw2_saa7146_pcm_trigger_start_capture(struct snd_aw2_saa7146 *chip, + int stream_number) +{ + /* In aw8 driver, dma transfert is always active. It is + started and stopped in a larger "space" */ + if (stream_number == 0) + WRITEREG((TR_E_A1_IN << 16) | TR_E_A1_IN, MC1); +} + +void snd_aw2_saa7146_pcm_trigger_stop_capture(struct snd_aw2_saa7146 *chip, + int stream_number) +{ + if (stream_number == 0) + WRITEREG((TR_E_A1_IN << 16), MC1); +} + +irqreturn_t snd_aw2_saa7146_interrupt(int irq, void *dev_id) +{ + unsigned int isr; + unsigned int iicsta; + struct snd_aw2_saa7146 *chip = dev_id; + + isr = READREG(ISR); + if (!isr) + return IRQ_NONE; + + WRITEREG(isr, ISR); + + if (isr & (IIC_S | IIC_E)) { + iicsta = READREG(IICSTA); + WRITEREG(0x100, IICSTA); + } + + if (isr & A1_out) { + if (arr_substream_it_playback_cb[1].p_it_callback != NULL) { + arr_substream_it_playback_cb[1]. + p_it_callback(arr_substream_it_playback_cb[1]. + p_callback_param); + } + } + if (isr & A2_out) { + if (arr_substream_it_playback_cb[0].p_it_callback != NULL) { + arr_substream_it_playback_cb[0]. + p_it_callback(arr_substream_it_playback_cb[0]. + p_callback_param); + } + + } + if (isr & A1_in) { + if (arr_substream_it_capture_cb[0].p_it_callback != NULL) { + arr_substream_it_capture_cb[0]. + p_it_callback(arr_substream_it_capture_cb[0]. + p_callback_param); + } + } + return IRQ_HANDLED; +} + +unsigned int snd_aw2_saa7146_get_hw_ptr_playback(struct snd_aw2_saa7146 *chip, + int stream_number, + unsigned char *start_addr, + unsigned int buffer_size) +{ + long pci_adp = 0; + size_t ptr = 0; + + if (stream_number == 0) { + pci_adp = READREG(PCI_ADP3); + ptr = pci_adp - (long)start_addr; + + if (ptr == buffer_size) + ptr = 0; + } + if (stream_number == 1) { + pci_adp = READREG(PCI_ADP1); + ptr = pci_adp - (size_t) start_addr; + + if (ptr == buffer_size) + ptr = 0; + } + return ptr; +} + +unsigned int snd_aw2_saa7146_get_hw_ptr_capture(struct snd_aw2_saa7146 *chip, + int stream_number, + unsigned char *start_addr, + unsigned int buffer_size) +{ + size_t pci_adp = 0; + size_t ptr = 0; + if (stream_number == 0) { + pci_adp = READREG(PCI_ADP2); + ptr = pci_adp - (size_t) start_addr; + + if (ptr == buffer_size) + ptr = 0; + } + return ptr; +} + +void snd_aw2_saa7146_use_digital_input(struct snd_aw2_saa7146 *chip, + int use_digital) +{ + /* FIXME: switch between analog and digital input does not always work. + It can produce a kind of white noise. It seams that received data + are inverted sometime (endian inversion). Why ? I don't know, maybe + a problem of synchronization... However for the time being I have + not found the problem. Workaround: switch again (and again) between + digital and analog input until it works. */ + if (use_digital) + WRITEREG(0x40, GPIO_CTRL); + else + WRITEREG(0x50, GPIO_CTRL); +} + +int snd_aw2_saa7146_is_using_digital_input(struct snd_aw2_saa7146 *chip) +{ + unsigned int reg_val = READREG(GPIO_CTRL); + if ((reg_val & 0xFF) == 0x40) + return 1; + else + return 0; +} + + +static int snd_aw2_saa7146_get_limit(int size) +{ + int limitsize = 32; + int limit = 0; + while (limitsize < size) { + limitsize *= 2; + limit++; + } + return limit; +} diff --git a/sound/pci/aw2/aw2-saa7146.h b/sound/pci/aw2/aw2-saa7146.h new file mode 100644 index 0000000..5b35e35 --- /dev/null +++ b/sound/pci/aw2/aw2-saa7146.h @@ -0,0 +1,105 @@ +/***************************************************************************** + * + * Copyright (C) 2008 Cedric Bregardis and + * Jean-Christian Hassler + * + * This file is part of the Audiowerk2 ALSA driver + * + * The Audiowerk2 ALSA driver is free software; you can redistribute it and/or + * modify it under the terms of the GNU General Public License as published by + * the Free Software Foundation; version 2. + * + * The Audiowerk2 ALSA driver is distributed in the hope that it will be useful, + * but WITHOUT ANY WARRANTY; without even the implied warranty of + * MERCHANTABILITY or FITNESS FOR A PARTICULAR PURPOSE. See the + * GNU General Public License for more details. + * + * You should have received a copy of the GNU General Public License + * along with the Audiowerk2 ALSA driver; if not, write to the Free Software + * Foundation, Inc., 51 Franklin Street, Fifth Floor, Boston, MA 02110-1301, + * USA. + * + *****************************************************************************/ + +#ifndef AW2_SAA7146_H +#define AW2_SAA7146_H + +#define NB_STREAM_PLAYBACK 2 +#define NB_STREAM_CAPTURE 1 + +#define NUM_STREAM_PLAYBACK_ANA 0 +#define NUM_STREAM_PLAYBACK_DIG 1 + +#define NUM_STREAM_CAPTURE_ANA 0 + +typedef void (*snd_aw2_saa7146_it_cb) (void *); + +struct snd_aw2_saa7146_cb_param { + snd_aw2_saa7146_it_cb p_it_callback; + void *p_callback_param; +}; + +/* definition of the chip-specific record */ + +struct snd_aw2_saa7146 { + void __iomem *base_addr; +}; + +extern void snd_aw2_saa7146_setup(struct snd_aw2_saa7146 *chip, + void __iomem *pci_base_addr); +extern int snd_aw2_saa7146_free(struct snd_aw2_saa7146 *chip); + +extern void snd_aw2_saa7146_pcm_init_playback(struct snd_aw2_saa7146 *chip, + int stream_number, + unsigned long dma_addr, + unsigned long period_size, + unsigned long buffer_size); +extern void snd_aw2_saa7146_pcm_init_capture(struct snd_aw2_saa7146 *chip, + int stream_number, + unsigned long dma_addr, + unsigned long period_size, + unsigned long buffer_size); +extern void snd_aw2_saa7146_define_it_playback_callback(unsigned int + stream_number, + snd_aw2_saa7146_it_cb + p_it_callback, + void *p_callback_param); +extern void snd_aw2_saa7146_define_it_capture_callback(unsigned int + stream_number, + snd_aw2_saa7146_it_cb + p_it_callback, + void *p_callback_param); +extern void snd_aw2_saa7146_pcm_trigger_start_capture(struct snd_aw2_saa7146 + *chip, int stream_number); +extern void snd_aw2_saa7146_pcm_trigger_stop_capture(struct snd_aw2_saa7146 + *chip, int stream_number); + +extern void snd_aw2_saa7146_pcm_trigger_start_playback(struct snd_aw2_saa7146 + *chip, + int stream_number); +extern void snd_aw2_saa7146_pcm_trigger_stop_playback(struct snd_aw2_saa7146 + *chip, int stream_number); + +extern irqreturn_t snd_aw2_saa7146_interrupt(int irq, void *dev_id); +extern unsigned int snd_aw2_saa7146_get_hw_ptr_playback(struct snd_aw2_saa7146 + *chip, + int stream_number, + unsigned char + *start_addr, + unsigned int + buffer_size); +extern unsigned int snd_aw2_saa7146_get_hw_ptr_capture(struct snd_aw2_saa7146 + *chip, + int stream_number, + unsigned char + *start_addr, + unsigned int + buffer_size); + +extern void snd_aw2_saa7146_use_digital_input(struct snd_aw2_saa7146 *chip, + int use_digital); + +extern int snd_aw2_saa7146_is_using_digital_input(struct snd_aw2_saa7146 + *chip); + +#endif diff --git a/sound/pci/aw2/aw2-tsl.c b/sound/pci/aw2/aw2-tsl.c new file mode 100644 index 0000000..459b031 --- /dev/null +++ b/sound/pci/aw2/aw2-tsl.c @@ -0,0 +1,110 @@ +/***************************************************************************** + * + * Copyright (C) 2008 Cedric Bregardis and + * Jean-Christian Hassler + * Copyright 1998 Emagic Soft- und Hardware GmbH + * Copyright 2002 Martijn Sipkema + * + * This file is part of the Audiowerk2 ALSA driver + * + * The Audiowerk2 ALSA driver is free software; you can redistribute it and/or + * modify it under the terms of the GNU General Public License as published by + * the Free Software Foundation; version 2. + * + * The Audiowerk2 ALSA driver is distributed in the hope that it will be useful, + * but WITHOUT ANY WARRANTY; without even the implied warranty of + * MERCHANTABILITY or FITNESS FOR A PARTICULAR PURPOSE. See the + * GNU General Public License for more details. + * + * You should have received a copy of the GNU General Public License + * along with the Audiowerk2 ALSA driver; if not, write to the Free Software + * Foundation, Inc., 51 Franklin Street, Fifth Floor, Boston, MA 02110-1301, + * USA. + * + *****************************************************************************/ + +#define TSL_WS0 (1UL << 31) +#define TSL_WS1 (1UL << 30) +#define TSL_WS2 (1UL << 29) +#define TSL_WS3 (1UL << 28) +#define TSL_WS4 (1UL << 27) +#define TSL_DIS_A1 (1UL << 24) +#define TSL_SDW_A1 (1UL << 23) +#define TSL_SIB_A1 (1UL << 22) +#define TSL_SF_A1 (1UL << 21) +#define TSL_LF_A1 (1UL << 20) +#define TSL_BSEL_A1 (1UL << 17) +#define TSL_DOD_A1 (1UL << 15) +#define TSL_LOW_A1 (1UL << 14) +#define TSL_DIS_A2 (1UL << 11) +#define TSL_SDW_A2 (1UL << 10) +#define TSL_SIB_A2 (1UL << 9) +#define TSL_SF_A2 (1UL << 8) +#define TSL_LF_A2 (1UL << 7) +#define TSL_BSEL_A2 (1UL << 4) +#define TSL_DOD_A2 (1UL << 2) +#define TSL_LOW_A2 (1UL << 1) +#define TSL_EOS (1UL << 0) + + /* Audiowerk8 hardware setup: */ + /* WS0, SD4, TSL1 - Analog/ digital in */ + /* WS1, SD0, TSL1 - Analog out #1, digital out */ + /* WS2, SD2, TSL1 - Analog out #2 */ + /* WS3, SD1, TSL2 - Analog out #3 */ + /* WS4, SD3, TSL2 - Analog out #4 */ + + /* Audiowerk8 timing: */ + /* Timeslot: | 0 | 1 | 2 | 3 | 4 | 5 | 6 | 7 | ... */ + + /* A1_INPUT: */ + /* SD4: <_ADC-L_>-------<_ADC-R_>-------< */ + /* WS0: _______________/---------------\_ */ + + /* A1_OUTPUT: */ + /* SD0: <_1-L___>-------<_1-R___>-------< */ + /* WS1: _______________/---------------\_ */ + /* SD2: >-------<_2-L___>-------<_2-R___> */ + /* WS2: -------\_______________/--------- */ + + /* A2_OUTPUT: */ + /* SD1: <_3-L___>-------<_3-R___>-------< */ + /* WS3: _______________/---------------\_ */ + /* SD3: >-------<_4-L___>-------<_4-R___> */ + /* WS4: -------\_______________/--------- */ + +static int tsl1[8] = { + 1 * TSL_SDW_A1 | 3 * TSL_BSEL_A1 | + 0 * TSL_DIS_A1 | 0 * TSL_DOD_A1 | TSL_LF_A1, + + 1 * TSL_SDW_A1 | 2 * TSL_BSEL_A1 | + 0 * TSL_DIS_A1 | 0 * TSL_DOD_A1, + + 0 * TSL_SDW_A1 | 3 * TSL_BSEL_A1 | + 0 * TSL_DIS_A1 | 0 * TSL_DOD_A1, + + 0 * TSL_SDW_A1 | 2 * TSL_BSEL_A1 | + 0 * TSL_DIS_A1 | 0 * TSL_DOD_A1, + + 1 * TSL_SDW_A1 | 1 * TSL_BSEL_A1 | + 0 * TSL_DIS_A1 | 0 * TSL_DOD_A1 | TSL_WS1 | TSL_WS0, + + 1 * TSL_SDW_A1 | 0 * TSL_BSEL_A1 | + 0 * TSL_DIS_A1 | 0 * TSL_DOD_A1 | TSL_WS1 | TSL_WS0, + + 0 * TSL_SDW_A1 | 1 * TSL_BSEL_A1 | + 0 * TSL_DIS_A1 | 0 * TSL_DOD_A1 | TSL_WS1 | TSL_WS0, + + 0 * TSL_SDW_A1 | 0 * TSL_BSEL_A1 | 0 * TSL_DIS_A1 | + 0 * TSL_DOD_A1 | TSL_WS1 | TSL_WS0 | TSL_SF_A1 | TSL_EOS, +}; + +static int tsl2[8] = { + 0 * TSL_SDW_A2 | 3 * TSL_BSEL_A2 | 2 * TSL_DOD_A2 | TSL_LF_A2, + 0 * TSL_SDW_A2 | 2 * TSL_BSEL_A2 | 2 * TSL_DOD_A2, + 0 * TSL_SDW_A2 | 3 * TSL_BSEL_A2 | 2 * TSL_DOD_A2, + 0 * TSL_SDW_A2 | 2 * TSL_BSEL_A2 | 2 * TSL_DOD_A2, + 0 * TSL_SDW_A2 | 1 * TSL_BSEL_A2 | 2 * TSL_DOD_A2 | TSL_WS2, + 0 * TSL_SDW_A2 | 0 * TSL_BSEL_A2 | 2 * TSL_DOD_A2 | TSL_WS2, + 0 * TSL_SDW_A2 | 1 * TSL_BSEL_A2 | 2 * TSL_DOD_A2 | TSL_WS2, + 0 * TSL_SDW_A2 | 0 * TSL_BSEL_A2 | 2 * TSL_DOD_A2 | TSL_WS2 | TSL_EOS +}; diff --git a/sound/pci/aw2/saa7146.h b/sound/pci/aw2/saa7146.h new file mode 100644 index 0000000..ce0ab5f --- /dev/null +++ b/sound/pci/aw2/saa7146.h @@ -0,0 +1,168 @@ +/***************************************************************************** + * + * Copyright (C) 2008 Cedric Bregardis and + * Jean-Christian Hassler + * + * This file is part of the Audiowerk2 ALSA driver + * + * The Audiowerk2 ALSA driver is free software; you can redistribute it and/or + * modify it under the terms of the GNU General Public License as published by + * the Free Software Foundation; version 2. + * + * The Audiowerk2 ALSA driver is distributed in the hope that it will be useful, + * but WITHOUT ANY WARRANTY; without even the implied warranty of + * MERCHANTABILITY or FITNESS FOR A PARTICULAR PURPOSE. See the + * GNU General Public License for more details. + * + * You should have received a copy of the GNU General Public License + * along with the Audiowerk2 ALSA driver; if not, write to the Free Software + * Foundation, Inc., 51 Franklin Street, Fifth Floor, Boston, MA 02110-1301, + * USA. + * + *****************************************************************************/ + +/* SAA7146 registers */ +#define PCI_BT_A 0x4C +#define IICTFR 0x8C +#define IICSTA 0x90 +#define BaseA1_in 0x94 +#define ProtA1_in 0x98 +#define PageA1_in 0x9C +#define BaseA1_out 0xA0 +#define ProtA1_out 0xA4 +#define PageA1_out 0xA8 +#define BaseA2_in 0xAC +#define ProtA2_in 0xB0 +#define PageA2_in 0xB4 +#define BaseA2_out 0xB8 +#define ProtA2_out 0xBC +#define PageA2_out 0xC0 +#define IER 0xDC +#define GPIO_CTRL 0xE0 +#define ACON1 0xF4 +#define ACON2 0xF8 +#define MC1 0xFC +#define MC2 0x100 +#define ISR 0x10C +#define PSR 0x110 +#define SSR 0x114 +#define PCI_ADP1 0x12C +#define PCI_ADP2 0x130 +#define PCI_ADP3 0x134 +#define PCI_ADP4 0x138 +#define LEVEL_REP 0x140 +#define FB_BUFFER1 0x144 +#define FB_BUFFER2 0x148 +#define TSL1 0x180 +#define TSL2 0x1C0 + +#define ME (1UL << 11) +#define LIMIT (1UL << 4) +#define PV (1UL << 3) + +/* PSR/ISR/IER */ +#define PPEF (1UL << 31) +#define PABO (1UL << 30) +#define IIC_S (1UL << 17) +#define IIC_E (1UL << 16) +#define A2_in (1UL << 15) +#define A2_out (1UL << 14) +#define A1_in (1UL << 13) +#define A1_out (1UL << 12) +#define AFOU (1UL << 11) +#define PIN3 (1UL << 6) +#define PIN2 (1UL << 5) +#define PIN1 (1UL << 4) +#define PIN0 (1UL << 3) +#define ECS (1UL << 2) +#define EC3S (1UL << 1) +#define EC0S (1UL << 0) + +/* SSR */ +#define PRQ (1UL << 31) +#define PMA (1UL << 30) +#define IIC_EA (1UL << 21) +#define IIC_EW (1UL << 20) +#define IIC_ER (1UL << 19) +#define IIC_EL (1UL << 18) +#define IIC_EF (1UL << 17) +#define AF2_in (1UL << 10) +#define AF2_out (1UL << 9) +#define AF1_in (1UL << 8) +#define AF1_out (1UL << 7) +#define EC5S (1UL << 3) +#define EC4S (1UL << 2) +#define EC2S (1UL << 1) +#define EC1S (1UL << 0) + +/* PCI_BT_A */ +#define BurstA1_in (1UL << 26) +#define ThreshA1_in (1UL << 24) +#define BurstA1_out (1UL << 18) +#define ThreshA1_out (1UL << 16) +#define BurstA2_in (1UL << 10) +#define ThreshA2_in (1UL << 8) +#define BurstA2_out (1UL << 2) +#define ThreshA2_out (1UL << 0) + +/* MC1 */ +#define MRST_N (1UL << 15) +#define EAP (1UL << 9) +#define EI2C (1UL << 8) +#define TR_E_A2_OUT (1UL << 3) +#define TR_E_A2_IN (1UL << 2) +#define TR_E_A1_OUT (1UL << 1) +#define TR_E_A1_IN (1UL << 0) + +/* MC2 */ +#define UPLD_IIC (1UL << 0) + +/* ACON1 */ +#define AUDIO_MODE (1UL << 29) +#define MAXLEVEL (1UL << 22) +#define A1_SWAP (1UL << 21) +#define A2_SWAP (1UL << 20) +#define WS0_CTRL (1UL << 18) +#define WS0_SYNC (1UL << 16) +#define WS1_CTRL (1UL << 14) +#define WS1_SYNC (1UL << 12) +#define WS2_CTRL (1UL << 10) +#define WS2_SYNC (1UL << 8) +#define WS3_CTRL (1UL << 6) +#define WS3_SYNC (1UL << 4) +#define WS4_CTRL (1UL << 2) +#define WS4_SYNC (1UL << 0) + +/* ACON2 */ +#define A1_CLKSRC (1UL << 27) +#define A2_CLKSRC (1UL << 22) +#define INVERT_BCLK1 (1UL << 21) +#define INVERT_BCLK2 (1UL << 20) +#define BCLK1_OEN (1UL << 19) +#define BCLK2_OEN (1UL << 18) + +/* IICSTA */ +#define IICCC (1UL << 8) +#define ABORT (1UL << 7) +#define SPERR (1UL << 6) +#define APERR (1UL << 5) +#define DTERR (1UL << 4) +#define DRERR (1UL << 3) +#define AL (1UL << 2) +#define ERR (1UL << 1) +#define BUSY (1UL << 0) + +/* IICTFR */ +#define BYTE2 (1UL << 24) +#define BYTE1 (1UL << 16) +#define BYTE0 (1UL << 8) +#define ATRR2 (1UL << 6) +#define ATRR1 (1UL << 4) +#define ATRR0 (1UL << 2) +#define ERR (1UL << 1) +#define BUSY (1UL << 0) + +#define START 3 +#define CONT 2 +#define STOP 1 +#define NOP 0 diff --git a/sound/pci/azt3328.c b/sound/pci/azt3328.c index 4e71a55..be87d31 100644 --- a/sound/pci/azt3328.c +++ b/sound/pci/azt3328.c @@ -157,8 +157,8 @@ MODULE_SUPPORTED_DEVICE("{{Aztech,AZF3328}}"); #if DEBUG_CALLS #define snd_azf3328_dbgcalls(format, args...) printk(format, ##args) -#define snd_azf3328_dbgcallenter() printk(KERN_ERR "--> %s\n", __FUNCTION__) -#define snd_azf3328_dbgcallleave() printk(KERN_ERR "<-- %s\n", __FUNCTION__) +#define snd_azf3328_dbgcallenter() printk(KERN_ERR "--> %s\n", __func__) +#define snd_azf3328_dbgcallleave() printk(KERN_ERR "<-- %s\n", __func__) #else #define snd_azf3328_dbgcalls(format, args...) #define snd_azf3328_dbgcallenter() diff --git a/sound/pci/ca0106/ca0106_main.c b/sound/pci/ca0106/ca0106_main.c index 176e0f0..3818249 100644 --- a/sound/pci/ca0106/ca0106_main.c +++ b/sound/pci/ca0106/ca0106_main.c @@ -435,22 +435,22 @@ int snd_ca0106_i2c_write(struct snd_ca0106 *emu, static void snd_ca0106_intr_enable(struct snd_ca0106 *emu, unsigned int intrenb) { unsigned long flags; - unsigned int enable; - + unsigned int intr_enable; + spin_lock_irqsave(&emu->emu_lock, flags); - enable = inl(emu->port + INTE) | intrenb; - outl(enable, emu->port + INTE); + intr_enable = inl(emu->port + INTE) | intrenb; + outl(intr_enable, emu->port + INTE); spin_unlock_irqrestore(&emu->emu_lock, flags); } static void snd_ca0106_intr_disable(struct snd_ca0106 *emu, unsigned int intrenb) { unsigned long flags; - unsigned int enable; - + unsigned int intr_enable; + spin_lock_irqsave(&emu->emu_lock, flags); - enable = inl(emu->port + INTE) & ~intrenb; - outl(enable, emu->port + INTE); + intr_enable = inl(emu->port + INTE) & ~intrenb; + outl(intr_enable, emu->port + INTE); spin_unlock_irqrestore(&emu->emu_lock, flags); } diff --git a/sound/pci/ca0106/ca0106_mixer.c b/sound/pci/ca0106/ca0106_mixer.c index af73686..3025ed1 100644 --- a/sound/pci/ca0106/ca0106_mixer.c +++ b/sound/pci/ca0106/ca0106_mixer.c @@ -650,19 +650,55 @@ static int __devinit rename_ctl(struct snd_card *card, const char *src, const ch #define ADD_CTLS(emu, ctls) \ do { \ - int i, err; \ + int i, _err; \ for (i = 0; i < ARRAY_SIZE(ctls); i++) { \ - err = snd_ctl_add(card, snd_ctl_new1(&ctls[i], emu)); \ - if (err < 0) \ - return err; \ + _err = snd_ctl_add(card, snd_ctl_new1(&ctls[i], emu)); \ + if (_err < 0) \ + return _err; \ } \ } while (0) +static __devinitdata +DECLARE_TLV_DB_SCALE(snd_ca0106_master_db_scale, -6375, 50, 1); + +static char *slave_vols[] __devinitdata = { + "Analog Front Playback Volume", + "Analog Rear Playback Volume", + "Analog Center/LFE Playback Volume", + "Analog Side Playback Volume", + "IEC958 Front Playback Volume", + "IEC958 Rear Playback Volume", + "IEC958 Center/LFE Playback Volume", + "IEC958 Unknown Playback Volume", + "CAPTURE feedback Playback Volume", + NULL +}; + +static char *slave_sws[] __devinitdata = { + "Analog Front Playback Switch", + "Analog Rear Playback Switch", + "Analog Center/LFE Playback Switch", + "Analog Side Playback Switch", + "IEC958 Playback Switch", + NULL +}; + +static void __devinit add_slaves(struct snd_card *card, + struct snd_kcontrol *master, char **list) +{ + for (; *list; list++) { + struct snd_kcontrol *slave = ctl_find(card, *list); + if (slave) + snd_ctl_add_slave(master, slave); + } +} + int __devinit snd_ca0106_mixer(struct snd_ca0106 *emu) { int err; struct snd_card *card = emu->card; char **c; + struct snd_kcontrol *vmaster; static char *ca0106_remove_ctls[] = { "Master Mono Playback Switch", "Master Mono Playback Volume", @@ -719,6 +755,21 @@ int __devinit snd_ca0106_mixer(struct snd_ca0106 *emu) } if (emu->details->spi_dac == 1) ADD_CTLS(emu, snd_ca0106_volume_spi_dac_ctls); + + /* Create virtual master controls */ + vmaster = snd_ctl_make_virtual_master("Master Playback Volume", + snd_ca0106_master_db_scale); + if (!vmaster) + return -ENOMEM; + add_slaves(card, vmaster, slave_vols); + + if (emu->details->spi_dac == 1) { + vmaster = snd_ctl_make_virtual_master("Master Playback Switch", + NULL); + if (!vmaster) + return -ENOMEM; + add_slaves(card, vmaster, slave_sws); + } return 0; } diff --git a/sound/pci/cmipci.c b/sound/pci/cmipci.c index 135f308..4074584 100644 --- a/sound/pci/cmipci.c +++ b/sound/pci/cmipci.c @@ -2744,12 +2744,13 @@ static int __devinit snd_cmipci_mixer_new(struct cmipci *cm, int pcm_spdif_devic } for (idx = 0; idx < CM_SAVED_MIXERS; idx++) { - struct snd_ctl_elem_id id; + struct snd_ctl_elem_id elem_id; struct snd_kcontrol *ctl; - memset(&id, 0, sizeof(id)); - id.iface = SNDRV_CTL_ELEM_IFACE_MIXER; - strcpy(id.name, cm_saved_mixer[idx].name); - if ((ctl = snd_ctl_find_id(cm->card, &id)) != NULL) + memset(&elem_id, 0, sizeof(elem_id)); + elem_id.iface = SNDRV_CTL_ELEM_IFACE_MIXER; + strcpy(elem_id.name, cm_saved_mixer[idx].name); + ctl = snd_ctl_find_id(cm->card, &elem_id); + if (ctl) cm->mixer_res_ctl[idx] = ctl; } diff --git a/sound/pci/emu10k1/emu10k1x.c b/sound/pci/emu10k1/emu10k1x.c index 5512abd..341f34e 100644 --- a/sound/pci/emu10k1/emu10k1x.c +++ b/sound/pci/emu10k1/emu10k1x.c @@ -327,22 +327,22 @@ static void snd_emu10k1x_ptr_write(struct emu10k1x *emu, static void snd_emu10k1x_intr_enable(struct emu10k1x *emu, unsigned int intrenb) { unsigned long flags; - unsigned int enable; - + unsigned int intr_enable; + spin_lock_irqsave(&emu->emu_lock, flags); - enable = inl(emu->port + INTE) | intrenb; - outl(enable, emu->port + INTE); + intr_enable = inl(emu->port + INTE) | intrenb; + outl(intr_enable, emu->port + INTE); spin_unlock_irqrestore(&emu->emu_lock, flags); } static void snd_emu10k1x_intr_disable(struct emu10k1x *emu, unsigned int intrenb) { unsigned long flags; - unsigned int enable; - + unsigned int intr_enable; + spin_lock_irqsave(&emu->emu_lock, flags); - enable = inl(emu->port + INTE) & ~intrenb; - outl(enable, emu->port + INTE); + intr_enable = inl(emu->port + INTE) & ~intrenb; + outl(intr_enable, emu->port + INTE); spin_unlock_irqrestore(&emu->emu_lock, flags); } @@ -795,9 +795,9 @@ static irqreturn_t snd_emu10k1x_interrupt(int irq, void *dev_id) // capture interrupt if (status & (IPR_CAP_0_LOOP | IPR_CAP_0_HALF_LOOP)) { - struct emu10k1x_voice *pvoice = &chip->capture_voice; - if (pvoice->use) - snd_emu10k1x_pcm_interrupt(chip, pvoice); + struct emu10k1x_voice *cap_voice = &chip->capture_voice; + if (cap_voice->use) + snd_emu10k1x_pcm_interrupt(chip, cap_voice); else snd_emu10k1x_intr_disable(chip, INTE_CAP_0_LOOP | diff --git a/sound/pci/emu10k1/emuproc.c b/sound/pci/emu10k1/emuproc.c index f3caa3f..216f974 100644 --- a/sound/pci/emu10k1/emuproc.c +++ b/sound/pci/emu10k1/emuproc.c @@ -412,7 +412,7 @@ static void snd_emu_proc_emu1010_reg_read(struct snd_info_entry *entry, struct snd_info_buffer *buffer) { struct snd_emu10k1 *emu = entry->private_data; - int value; + u32 value; unsigned long flags; int i; snd_iprintf(buffer, "EMU1010 Registers:\n\n"); diff --git a/sound/pci/ens1370.c b/sound/pci/ens1370.c index 72d85a5..52fae4a 100644 --- a/sound/pci/ens1370.c +++ b/sound/pci/ens1370.c @@ -1635,20 +1635,20 @@ static int __devinit snd_ensoniq_1371_mixer(struct ensoniq *ensoniq, if (has_spdif > 0 || (!has_spdif && es1371_quirk_lookup(ensoniq, es1371_spdif_present))) { struct snd_kcontrol *kctl; - int i, index = 0; + int i, is_spdif = 0; ensoniq->spdif_default = ensoniq->spdif_stream = SNDRV_PCM_DEFAULT_CON_SPDIF; outl(ensoniq->spdif_default, ES_REG(ensoniq, CHANNEL_STATUS)); if (ensoniq->u.es1371.ac97->ext_id & AC97_EI_SPDIF) - index++; + is_spdif++; for (i = 0; i < ARRAY_SIZE(snd_es1371_mixer_spdif); i++) { kctl = snd_ctl_new1(&snd_es1371_mixer_spdif[i], ensoniq); if (!kctl) return -ENOMEM; - kctl->id.index = index; + kctl->id.index = is_spdif; err = snd_ctl_add(card, kctl); if (err < 0) return err; diff --git a/sound/pci/es1968.c b/sound/pci/es1968.c index 7d911a1..1383798 100644 --- a/sound/pci/es1968.c +++ b/sound/pci/es1968.c @@ -1827,6 +1827,22 @@ snd_es1968_pcm(struct es1968 *chip, int device) return 0; } +/* + * suppress jitter on some maestros when playing stereo + */ +static void snd_es1968_suppress_jitter(struct es1968 *chip, struct esschan *es) +{ + unsigned int cp1; + unsigned int cp2; + unsigned int diff; + + cp1 = __apu_get_register(chip, 0, 5); + cp2 = __apu_get_register(chip, 1, 5); + diff = (cp1 > cp2 ? cp1 - cp2 : cp2 - cp1); + + if (diff > 1) + __maestro_write(chip, IDR0_DATA_PORT, cp1); +} /* * update pointer @@ -1948,8 +1964,11 @@ static irqreturn_t snd_es1968_interrupt(int irq, void *dev_id) struct esschan *es; spin_lock(&chip->substream_lock); list_for_each_entry(es, &chip->substream_list, list) { - if (es->running) + if (es->running) { snd_es1968_update_pcm(chip, es); + if (es->fmt & ESS_FMT_STEREO) + snd_es1968_suppress_jitter(chip, es); + } } spin_unlock(&chip->substream_lock); if (chip->in_measurement) { @@ -1972,7 +1991,7 @@ snd_es1968_mixer(struct es1968 *chip) { struct snd_ac97_bus *pbus; struct snd_ac97_template ac97; - struct snd_ctl_elem_id id; + struct snd_ctl_elem_id elem_id; int err; static struct snd_ac97_bus_ops ops = { .write = snd_es1968_ac97_write, @@ -1989,14 +2008,14 @@ snd_es1968_mixer(struct es1968 *chip) return err; /* attach master switch / volumes for h/w volume control */ - memset(&id, 0, sizeof(id)); - id.iface = SNDRV_CTL_ELEM_IFACE_MIXER; - strcpy(id.name, "Master Playback Switch"); - chip->master_switch = snd_ctl_find_id(chip->card, &id); - memset(&id, 0, sizeof(id)); - id.iface = SNDRV_CTL_ELEM_IFACE_MIXER; - strcpy(id.name, "Master Playback Volume"); - chip->master_volume = snd_ctl_find_id(chip->card, &id); + memset(&elem_id, 0, sizeof(elem_id)); + elem_id.iface = SNDRV_CTL_ELEM_IFACE_MIXER; + strcpy(elem_id.name, "Master Playback Switch"); + chip->master_switch = snd_ctl_find_id(chip->card, &elem_id); + memset(&elem_id, 0, sizeof(elem_id)); + elem_id.iface = SNDRV_CTL_ELEM_IFACE_MIXER; + strcpy(elem_id.name, "Master Playback Volume"); + chip->master_volume = snd_ctl_find_id(chip->card, &elem_id); return 0; } diff --git a/sound/pci/fm801.c b/sound/pci/fm801.c index 4c300e6..c129f9e 100644 --- a/sound/pci/fm801.c +++ b/sound/pci/fm801.c @@ -1285,7 +1285,6 @@ static int wait_for_codec(struct fm801 *chip, unsigned int codec_id, static int snd_fm801_chip_init(struct fm801 *chip, int resume) { - int id; unsigned short cmdw; if (chip->tea575x_tuner & 0x0010) @@ -1310,13 +1309,14 @@ static int snd_fm801_chip_init(struct fm801 *chip, int resume) } else { /* my card has the secondary codec */ /* at address #3, so the loop is inverted */ - for (id = 3; id > 0; id--) { - if (! wait_for_codec(chip, id, AC97_VENDOR_ID1, + int i; + for (i = 3; i > 0; i--) { + if (!wait_for_codec(chip, i, AC97_VENDOR_ID1, msecs_to_jiffies(50))) { cmdw = inw(FM801_REG(chip, AC97_DATA)); if (cmdw != 0xffff && cmdw != 0) { chip->secondary = 1; - chip->secondary_addr = id; + chip->secondary_addr = i; break; } } diff --git a/sound/pci/hda/Makefile b/sound/pci/hda/Makefile index 9e0d8a1..ab0c726 100644 --- a/sound/pci/hda/Makefile +++ b/sound/pci/hda/Makefile @@ -2,7 +2,7 @@ snd-hda-intel-y := hda_intel.o # since snd-hda-intel is the only driver using hda-codec, # merge it into a single module although it was originally # designed to be individual modules -snd-hda-intel-y += hda_codec.o vmaster.o +snd-hda-intel-y += hda_codec.o snd-hda-intel-$(CONFIG_PROC_FS) += hda_proc.o snd-hda-intel-$(CONFIG_SND_HDA_HWDEP) += hda_hwdep.o snd-hda-intel-$(CONFIG_SND_HDA_GENERIC) += hda_generic.o diff --git a/sound/pci/hda/hda_codec.c b/sound/pci/hda/hda_codec.c index 37c4139..a6be6e3 100644 --- a/sound/pci/hda/hda_codec.c +++ b/sound/pci/hda/hda_codec.c @@ -31,6 +31,7 @@ #include #include "hda_local.h" #include +#include "hda_patch.h" /* codec presets */ #ifdef CONFIG_SND_HDA_POWER_SAVE /* define this option here to hide as static */ @@ -51,21 +52,50 @@ struct hda_vendor_id { /* codec vendor labels */ static struct hda_vendor_id hda_vendor_ids[] = { - { 0x10ec, "Realtek" }, + { 0x1002, "ATI" }, { 0x1057, "Motorola" }, + { 0x1095, "Silicon Image" }, + { 0x10ec, "Realtek" }, { 0x1106, "VIA" }, { 0x111d, "IDT" }, + { 0x11c1, "LSI" }, { 0x11d4, "Analog Devices" }, { 0x13f6, "C-Media" }, { 0x14f1, "Conexant" }, + { 0x17e8, "Chrontel" }, + { 0x1854, "LG" }, { 0x434d, "C-Media" }, { 0x8384, "SigmaTel" }, {} /* terminator */ }; -/* codec presets */ -#include "hda_patch.h" - +static const struct hda_codec_preset *hda_preset_tables[] = { +#ifdef CONFIG_SND_HDA_CODEC_REALTEK + snd_hda_preset_realtek, +#endif +#ifdef CONFIG_SND_HDA_CODEC_CMEDIA + snd_hda_preset_cmedia, +#endif +#ifdef CONFIG_SND_HDA_CODEC_ANALOG + snd_hda_preset_analog, +#endif +#ifdef CONFIG_SND_HDA_CODEC_SIGMATEL + snd_hda_preset_sigmatel, +#endif +#ifdef CONFIG_SND_HDA_CODEC_SI3054 + snd_hda_preset_si3054, +#endif +#ifdef CONFIG_SND_HDA_CODEC_ATIHDMI + snd_hda_preset_atihdmi, +#endif +#ifdef CONFIG_SND_HDA_CODEC_CONEXANT + snd_hda_preset_conexant, +#endif +#ifdef CONFIG_SND_HDA_CODEC_VIA + snd_hda_preset_via, +#endif + NULL +}; #ifdef CONFIG_SND_HDA_POWER_SAVE static void hda_power_work(struct work_struct *work); @@ -690,6 +720,19 @@ void snd_hda_codec_setup_stream(struct hda_codec *codec, hda_nid_t nid, snd_hda_codec_write(codec, nid, 0, AC_VERB_SET_STREAM_FORMAT, format); } +void snd_hda_codec_cleanup_stream(struct hda_codec *codec, hda_nid_t nid) +{ + if (!nid) + return; + + snd_printdd("hda_codec_cleanup_stream: NID=0x%x\n", nid); + snd_hda_codec_write(codec, nid, 0, AC_VERB_SET_CHANNEL_STREAMID, 0); +#if 0 /* keep the format */ + msleep(1); + snd_hda_codec_write(codec, nid, 0, AC_VERB_SET_STREAM_FORMAT, 0); +#endif +} + /* * amp access functions */ @@ -1037,16 +1080,24 @@ void snd_hda_set_vmaster_tlv(struct hda_codec *codec, hda_nid_t nid, int dir, } /* find a mixer control element with the given name */ -struct snd_kcontrol *snd_hda_find_mixer_ctl(struct hda_codec *codec, - const char *name) +static struct snd_kcontrol * +_snd_hda_find_mixer_ctl(struct hda_codec *codec, + const char *name, int idx) { struct snd_ctl_elem_id id; memset(&id, 0, sizeof(id)); id.iface = SNDRV_CTL_ELEM_IFACE_MIXER; + id.index = idx; strcpy(id.name, name); return snd_ctl_find_id(codec->bus->card, &id); } +struct snd_kcontrol *snd_hda_find_mixer_ctl(struct hda_codec *codec, + const char *name) +{ + return _snd_hda_find_mixer_ctl(codec, name, 0); +} + /* create a virtual master control and add slaves */ int snd_hda_add_vmaster(struct hda_codec *codec, char *name, unsigned int *tlv, const char **slaves) @@ -1481,6 +1532,8 @@ static struct snd_kcontrol_new dig_mixes[] = { { } /* end */ }; +#define SPDIF_MAX_IDX 4 /* 4 instances should be enough to probe */ + /** * snd_hda_create_spdif_out_ctls - create Output SPDIF-related controls * @codec: the HDA codec @@ -1496,9 +1549,20 @@ int snd_hda_create_spdif_out_ctls(struct hda_codec *codec, hda_nid_t nid) int err; struct snd_kcontrol *kctl; struct snd_kcontrol_new *dig_mix; + int idx; + for (idx = 0; idx < SPDIF_MAX_IDX; idx++) { + if (!_snd_hda_find_mixer_ctl(codec, "IEC958 Playback Switch", + idx)) + break; + } + if (idx >= SPDIF_MAX_IDX) { + printk(KERN_ERR "hda_codec: too many IEC958 outputs\n"); + return -EBUSY; + } for (dig_mix = dig_mixes; dig_mix->name; dig_mix++) { kctl = snd_ctl_new1(dig_mix, codec); + kctl->id.index = idx; kctl->private_value = nid; err = snd_ctl_add(codec->bus->card, kctl); if (err < 0) @@ -1512,6 +1576,43 @@ int snd_hda_create_spdif_out_ctls(struct hda_codec *codec, hda_nid_t nid) } /* + * SPDIF sharing with analog output + */ +static int spdif_share_sw_get(struct snd_kcontrol *kcontrol, + struct snd_ctl_elem_value *ucontrol) +{ + struct hda_multi_out *mout = snd_kcontrol_chip(kcontrol); + ucontrol->value.integer.value[0] = mout->share_spdif; + return 0; +} + +static int spdif_share_sw_put(struct snd_kcontrol *kcontrol, + struct snd_ctl_elem_value *ucontrol) +{ + struct hda_multi_out *mout = snd_kcontrol_chip(kcontrol); + mout->share_spdif = !!ucontrol->value.integer.value[0]; + return 0; +} + +static struct snd_kcontrol_new spdif_share_sw = { + .iface = SNDRV_CTL_ELEM_IFACE_MIXER, + .name = "IEC958 Default PCM Playback Switch", + .info = snd_ctl_boolean_mono_info, + .get = spdif_share_sw_get, + .put = spdif_share_sw_put, +}; + +int snd_hda_create_spdif_share_sw(struct hda_codec *codec, + struct hda_multi_out *mout) +{ + if (!mout->dig_out_nid) + return 0; + /* ATTENTION: here mout is passed as private_data, instead of codec */ + return snd_ctl_add(codec->bus->card, + snd_ctl_new1(&spdif_share_sw, mout)); +} + +/* * SPDIF input */ @@ -1595,7 +1696,17 @@ int snd_hda_create_spdif_in_ctls(struct hda_codec *codec, hda_nid_t nid) int err; struct snd_kcontrol *kctl; struct snd_kcontrol_new *dig_mix; + int idx; + for (idx = 0; idx < SPDIF_MAX_IDX; idx++) { + if (!_snd_hda_find_mixer_ctl(codec, "IEC958 Capture Switch", + idx)) + break; + } + if (idx >= SPDIF_MAX_IDX) { + printk(KERN_ERR "hda_codec: too many IEC958 inputs\n"); + return -EBUSY; + } for (dig_mix = dig_in_ctls; dig_mix->name; dig_mix++) { kctl = snd_ctl_new1(dig_mix, codec); kctl->private_value = nid; @@ -2106,7 +2217,7 @@ static int hda_pcm_default_cleanup(struct hda_pcm_stream *hinfo, struct hda_codec *codec, struct snd_pcm_substream *substream) { - snd_hda_codec_setup_stream(codec, hinfo->nid, 0, 0, 0); + snd_hda_codec_cleanup_stream(codec, hinfo->nid); return 0; } @@ -2491,7 +2602,7 @@ int snd_hda_multi_out_dig_open(struct hda_codec *codec, mutex_lock(&codec->spdif_mutex); if (mout->dig_out_used == HDA_DIG_ANALOG_DUP) /* already opened as analog dup; reset it once */ - snd_hda_codec_setup_stream(codec, mout->dig_out_nid, 0, 0, 0); + snd_hda_codec_cleanup_stream(codec, mout->dig_out_nid); mout->dig_out_used = HDA_DIG_EXCLUSIVE; mutex_unlock(&codec->spdif_mutex); return 0; @@ -2526,9 +2637,36 @@ int snd_hda_multi_out_dig_close(struct hda_codec *codec, */ int snd_hda_multi_out_analog_open(struct hda_codec *codec, struct hda_multi_out *mout, - struct snd_pcm_substream *substream) -{ - substream->runtime->hw.channels_max = mout->max_channels; + struct snd_pcm_substream *substream, + struct hda_pcm_stream *hinfo) +{ + struct snd_pcm_runtime *runtime = substream->runtime; + runtime->hw.channels_max = mout->max_channels; + if (mout->dig_out_nid) { + if (!mout->analog_rates) { + mout->analog_rates = hinfo->rates; + mout->analog_formats = hinfo->formats; + mout->analog_maxbps = hinfo->maxbps; + } else { + runtime->hw.rates = mout->analog_rates; + runtime->hw.formats = mout->analog_formats; + hinfo->maxbps = mout->analog_maxbps; + } + if (!mout->spdif_rates) { + snd_hda_query_supported_pcm(codec, mout->dig_out_nid, + &mout->spdif_rates, + &mout->spdif_formats, + &mout->spdif_maxbps); + } + mutex_lock(&codec->spdif_mutex); + if (mout->share_spdif) { + runtime->hw.rates &= mout->spdif_rates; + runtime->hw.formats &= mout->spdif_formats; + if (mout->spdif_maxbps < hinfo->maxbps) + hinfo->maxbps = mout->spdif_maxbps; + } + mutex_unlock(&codec->spdif_mutex); + } return snd_pcm_hw_constraint_step(substream->runtime, 0, SNDRV_PCM_HW_PARAM_CHANNELS, 2); } @@ -2548,7 +2686,8 @@ int snd_hda_multi_out_analog_prepare(struct hda_codec *codec, int i; mutex_lock(&codec->spdif_mutex); - if (mout->dig_out_nid && mout->dig_out_used != HDA_DIG_EXCLUSIVE) { + if (mout->dig_out_nid && mout->share_spdif && + mout->dig_out_used != HDA_DIG_EXCLUSIVE) { if (chs == 2 && snd_hda_is_supported_format(codec, mout->dig_out_nid, format) && @@ -2558,8 +2697,7 @@ int snd_hda_multi_out_analog_prepare(struct hda_codec *codec, stream_tag, format); } else { mout->dig_out_used = 0; - snd_hda_codec_setup_stream(codec, mout->dig_out_nid, - 0, 0, 0); + snd_hda_codec_cleanup_stream(codec, mout->dig_out_nid); } } mutex_unlock(&codec->spdif_mutex); @@ -2601,17 +2739,16 @@ int snd_hda_multi_out_analog_cleanup(struct hda_codec *codec, int i; for (i = 0; i < mout->num_dacs; i++) - snd_hda_codec_setup_stream(codec, nids[i], 0, 0, 0); + snd_hda_codec_cleanup_stream(codec, nids[i]); if (mout->hp_nid) - snd_hda_codec_setup_stream(codec, mout->hp_nid, 0, 0, 0); + snd_hda_codec_cleanup_stream(codec, mout->hp_nid); for (i = 0; i < ARRAY_SIZE(mout->extra_out_nid); i++) if (mout->extra_out_nid[i]) - snd_hda_codec_setup_stream(codec, - mout->extra_out_nid[i], - 0, 0, 0); + snd_hda_codec_cleanup_stream(codec, + mout->extra_out_nid[i]); mutex_lock(&codec->spdif_mutex); if (mout->dig_out_nid && mout->dig_out_used == HDA_DIG_ANALOG_DUP) { - snd_hda_codec_setup_stream(codec, mout->dig_out_nid, 0, 0, 0); + snd_hda_codec_cleanup_stream(codec, mout->dig_out_nid); mout->dig_out_used = 0; } mutex_unlock(&codec->spdif_mutex); @@ -2790,6 +2927,30 @@ int snd_hda_parse_pin_def_config(struct hda_codec *codec, } } + /* FIX-UP: + * If no line-out is defined but multiple HPs are found, + * some of them might be the real line-outs. + */ + if (!cfg->line_outs && cfg->hp_outs > 1) { + int i = 0; + while (i < cfg->hp_outs) { + /* The real HPs should have the sequence 0x0f */ + if ((sequences_hp[i] & 0x0f) == 0x0f) { + i++; + continue; + } + /* Move it to the line-out table */ + cfg->line_out_pins[cfg->line_outs] = cfg->hp_pins[i]; + sequences_line_out[cfg->line_outs] = sequences_hp[i]; + cfg->line_outs++; + cfg->hp_outs--; + memmove(cfg->hp_pins + i, cfg->hp_pins + i + 1, + sizeof(cfg->hp_pins[0]) * (cfg->hp_outs - i)); + memmove(sequences_hp + i - 1, sequences_hp + i, + sizeof(sequences_hp[0]) * (cfg->hp_outs - i)); + } + } + /* sort by sequence */ sort_pins_by_sequence(cfg->line_out_pins, sequences_line_out, cfg->line_outs); diff --git a/sound/pci/hda/hda_codec.h b/sound/pci/hda/hda_codec.h index f148711..dcd390b 100644 --- a/sound/pci/hda/hda_codec.h +++ b/sound/pci/hda/hda_codec.h @@ -590,11 +590,21 @@ struct hda_pcm_stream { struct hda_pcm_ops ops; }; +/* PCM types */ +enum { + HDA_PCM_TYPE_AUDIO, + HDA_PCM_TYPE_SPDIF, + HDA_PCM_TYPE_HDMI, + HDA_PCM_TYPE_MODEM, + HDA_PCM_NTYPES +}; + /* for PCM creation */ struct hda_pcm { char *name; struct hda_pcm_stream stream[2]; - unsigned int is_modem; /* modem codec? */ + unsigned int pcm_type; /* HDA_PCM_TYPE_XXX */ + int device; /* assigned device number */ }; /* codec information */ @@ -712,6 +722,7 @@ int snd_hda_build_pcms(struct hda_bus *bus); void snd_hda_codec_setup_stream(struct hda_codec *codec, hda_nid_t nid, u32 stream_tag, int channel_id, int format); +void snd_hda_codec_cleanup_stream(struct hda_codec *codec, hda_nid_t nid); unsigned int snd_hda_calc_stream_format(unsigned int rate, unsigned int channels, unsigned int format, diff --git a/sound/pci/hda/hda_generic.c b/sound/pci/hda/hda_generic.c index f9de7c4..59e4389 100644 --- a/sound/pci/hda/hda_generic.c +++ b/sound/pci/hda/hda_generic.c @@ -1007,8 +1007,8 @@ static int generic_pcm2_cleanup(struct hda_pcm_stream *hinfo, { struct hda_gspec *spec = codec->spec; - snd_hda_codec_setup_stream(codec, hinfo->nid, 0, 0, 0); - snd_hda_codec_setup_stream(codec, spec->dac_node[1]->nid, 0, 0, 0); + snd_hda_codec_cleanup_stream(codec, hinfo->nid); + snd_hda_codec_cleanup_stream(codec, spec->dac_node[1]->nid); return 0; } diff --git a/sound/pci/hda/hda_intel.c b/sound/pci/hda/hda_intel.c index 4be36c8..bc3867e 100644 --- a/sound/pci/hda/hda_intel.c +++ b/sound/pci/hda/hda_intel.c @@ -39,6 +39,7 @@ #include #include #include +#include #include #include #include @@ -185,35 +186,28 @@ enum { SDI0, SDI1, SDI2, SDI3, SDO0, SDO1, SDO2, SDO3 }; /* max number of SDs */ /* ICH, ATI and VIA have 4 playback and 4 capture */ -#define ICH6_CAPTURE_INDEX 0 #define ICH6_NUM_CAPTURE 4 -#define ICH6_PLAYBACK_INDEX 4 #define ICH6_NUM_PLAYBACK 4 /* ULI has 6 playback and 5 capture */ -#define ULI_CAPTURE_INDEX 0 #define ULI_NUM_CAPTURE 5 -#define ULI_PLAYBACK_INDEX 5 #define ULI_NUM_PLAYBACK 6 /* ATI HDMI has 1 playback and 0 capture */ -#define ATIHDMI_CAPTURE_INDEX 0 #define ATIHDMI_NUM_CAPTURE 0 -#define ATIHDMI_PLAYBACK_INDEX 0 #define ATIHDMI_NUM_PLAYBACK 1 /* this number is statically defined for simplicity */ #define MAX_AZX_DEV 16 /* max number of fragments - we may use more if allocating more pages for BDL */ -#define BDL_SIZE PAGE_ALIGN(8192) -#define AZX_MAX_FRAG (BDL_SIZE / (MAX_AZX_DEV * 16)) +#define BDL_SIZE 4096 +#define AZX_MAX_BDL_ENTRIES (BDL_SIZE / 16) +#define AZX_MAX_FRAG 32 /* max buffer size - no h/w limit, you can increase as you like */ #define AZX_MAX_BUF_SIZE (1024*1024*1024) /* max number of PCM devics per card */ -#define AZX_MAX_AUDIO_PCMS 6 -#define AZX_MAX_MODEM_PCMS 2 -#define AZX_MAX_PCMS (AZX_MAX_AUDIO_PCMS + AZX_MAX_MODEM_PCMS) +#define AZX_MAX_PCMS 8 /* RIRB int mask: overrun[2], response[0] */ #define RIRB_INT_RESPONSE 0x01 @@ -227,6 +221,9 @@ enum { SDI0, SDI1, SDI2, SDI3, SDO0, SDO1, SDO2, SDO3 }; /* SD_CTL bits */ #define SD_CTL_STREAM_RESET 0x01 /* stream reset bit */ #define SD_CTL_DMA_START 0x02 /* stream DMA start bit */ +#define SD_CTL_STRIPE (3 << 16) /* stripe control */ +#define SD_CTL_TRAFFIC_PRIO (1 << 18) /* traffic priority */ +#define SD_CTL_DIR (1 << 19) /* bi-directional stream */ #define SD_CTL_STREAM_TAG_MASK (0xf << 20) #define SD_CTL_STREAM_TAG_SHIFT 20 @@ -284,12 +281,10 @@ enum { */ struct azx_dev { - u32 *bdl; /* virtual address of the BDL */ - dma_addr_t bdl_addr; /* physical address of the BDL */ + struct snd_dma_buffer bdl; /* BDL buffer */ u32 *posbuf; /* position buffer pointer */ unsigned int bufsize; /* size of the play buffer in bytes */ - unsigned int fragsize; /* size of each period in bytes */ unsigned int frags; /* number for period in the play buffer */ unsigned int fifo_size; /* FIFO size */ @@ -350,7 +345,6 @@ struct azx { struct azx_dev *azx_dev; /* PCM */ - unsigned int pcm_devs; struct snd_pcm *pcm[AZX_MAX_PCMS]; /* HD codec */ @@ -361,8 +355,7 @@ struct azx { struct azx_rb corb; struct azx_rb rirb; - /* BDL, CORB/RIRB and position buffers */ - struct snd_dma_buffer bdl; + /* CORB/RIRB and position buffers */ struct snd_dma_buffer rb; struct snd_dma_buffer posbuf; @@ -546,8 +539,9 @@ static void azx_update_rirb(struct azx *chip) if (res_ex & ICH6_RIRB_EX_UNSOL_EV) snd_hda_queue_unsol_event(chip->bus, res, res_ex); else if (chip->rirb.cmds) { - chip->rirb.cmds--; chip->rirb.res = res; + smp_wmb(); + chip->rirb.cmds--; } } } @@ -566,8 +560,10 @@ static unsigned int azx_rirb_get_response(struct hda_codec *codec) azx_update_rirb(chip); spin_unlock_irq(&chip->reg_lock); } - if (!chip->rirb.cmds) + if (!chip->rirb.cmds) { + smp_rmb(); return chip->rirb.res; /* the last value */ + } if (time_after(jiffies, timeout)) break; if (codec->bus->needs_damn_long_delay) @@ -965,30 +961,57 @@ static irqreturn_t azx_interrupt(int irq, void *dev_id) /* * set up BDL entries */ -static void azx_setup_periods(struct azx_dev *azx_dev) +static int azx_setup_periods(struct snd_pcm_substream *substream, + struct azx_dev *azx_dev) { - u32 *bdl = azx_dev->bdl; - dma_addr_t dma_addr = azx_dev->substream->runtime->dma_addr; - int idx; + struct snd_sg_buf *sgbuf = snd_pcm_substream_sgbuf(substream); + u32 *bdl; + int i, ofs, periods, period_bytes; /* reset BDL address */ azx_sd_writel(azx_dev, SD_BDLPL, 0); azx_sd_writel(azx_dev, SD_BDLPU, 0); + period_bytes = snd_pcm_lib_period_bytes(substream); + periods = azx_dev->bufsize / period_bytes; + /* program the initial BDL entries */ - for (idx = 0; idx < azx_dev->frags; idx++) { - unsigned int off = idx << 2; /* 4 dword step */ - dma_addr_t addr = dma_addr + idx * azx_dev->fragsize; - /* program the address field of the BDL entry */ - bdl[off] = cpu_to_le32((u32)addr); - bdl[off+1] = cpu_to_le32(upper_32bit(addr)); - - /* program the size field of the BDL entry */ - bdl[off+2] = cpu_to_le32(azx_dev->fragsize); - - /* program the IOC to enable interrupt when buffer completes */ - bdl[off+3] = cpu_to_le32(0x01); + bdl = (u32 *)azx_dev->bdl.area; + ofs = 0; + azx_dev->frags = 0; + for (i = 0; i < periods; i++) { + int size, rest; + if (i >= AZX_MAX_BDL_ENTRIES) { + snd_printk(KERN_ERR "Too many BDL entries: " + "buffer=%d, period=%d\n", + azx_dev->bufsize, period_bytes); + /* reset */ + azx_sd_writel(azx_dev, SD_BDLPL, 0); + azx_sd_writel(azx_dev, SD_BDLPU, 0); + return -EINVAL; + } + rest = period_bytes; + do { + dma_addr_t addr = snd_pcm_sgbuf_get_addr(sgbuf, ofs); + /* program the address field of the BDL entry */ + bdl[0] = cpu_to_le32((u32)addr); + bdl[1] = cpu_to_le32(upper_32bit(addr)); + /* program the size field of the BDL entry */ + size = PAGE_SIZE - (ofs % PAGE_SIZE); + if (rest < size) + size = rest; + bdl[2] = cpu_to_le32(size); + /* program the IOC to enable interrupt + * only when the whole fragment is processed + */ + rest -= size; + bdl[3] = rest ? 0 : cpu_to_le32(0x01); + bdl += 4; + azx_dev->frags++; + ofs += size; + } while (rest > 0); } + return 0; } /* @@ -1037,14 +1060,17 @@ static int azx_setup_controller(struct azx *chip, struct azx_dev *azx_dev) /* program the BDL address */ /* lower BDL address */ - azx_sd_writel(azx_dev, SD_BDLPL, (u32)azx_dev->bdl_addr); + azx_sd_writel(azx_dev, SD_BDLPL, (u32)azx_dev->bdl.addr); /* upper BDL address */ - azx_sd_writel(azx_dev, SD_BDLPU, upper_32bit(azx_dev->bdl_addr)); + azx_sd_writel(azx_dev, SD_BDLPU, upper_32bit(azx_dev->bdl.addr)); /* enable the position buffer */ - if (!(azx_readl(chip, DPLBASE) & ICH6_DPLBASE_ENABLE)) - azx_writel(chip, DPLBASE, - (u32)chip->posbuf.addr |ICH6_DPLBASE_ENABLE); + if (chip->position_fix == POS_FIX_POSBUF || + chip->position_fix == POS_FIX_AUTO) { + if (!(azx_readl(chip, DPLBASE) & ICH6_DPLBASE_ENABLE)) + azx_writel(chip, DPLBASE, + (u32)chip->posbuf.addr | ICH6_DPLBASE_ENABLE); + } /* set the interrupt enable bits in the descriptor control register */ azx_sd_writel(azx_dev, SD_CTL, @@ -1157,7 +1183,8 @@ static struct snd_pcm_hardware azx_pcm_hw = { SNDRV_PCM_INFO_MMAP_VALID | /* No full-resume yet implemented */ /* SNDRV_PCM_INFO_RESUME |*/ - SNDRV_PCM_INFO_PAUSE), + SNDRV_PCM_INFO_PAUSE | + SNDRV_PCM_INFO_SYNC_START), .formats = SNDRV_PCM_FMTBIT_S16_LE, .rates = SNDRV_PCM_RATE_48000, .rate_min = 48000, @@ -1219,6 +1246,7 @@ static int azx_pcm_open(struct snd_pcm_substream *substream) spin_unlock_irqrestore(&chip->reg_lock, flags); runtime->private_data = azx_dev; + snd_pcm_set_sync(substream); mutex_unlock(&chip->open_mutex); return 0; } @@ -1275,8 +1303,6 @@ static int azx_pcm_prepare(struct snd_pcm_substream *substream) struct snd_pcm_runtime *runtime = substream->runtime; azx_dev->bufsize = snd_pcm_lib_buffer_bytes(substream); - azx_dev->fragsize = snd_pcm_lib_period_bytes(substream); - azx_dev->frags = azx_dev->bufsize / azx_dev->fragsize; azx_dev->format_val = snd_hda_calc_stream_format(runtime->rate, runtime->channels, runtime->format, @@ -1288,10 +1314,10 @@ static int azx_pcm_prepare(struct snd_pcm_substream *substream) return -EINVAL; } - snd_printdd("azx_pcm_prepare: bufsize=0x%x, fragsize=0x%x, " - "format=0x%x\n", - azx_dev->bufsize, azx_dev->fragsize, azx_dev->format_val); - azx_setup_periods(azx_dev); + snd_printdd("azx_pcm_prepare: bufsize=0x%x, format=0x%x\n", + azx_dev->bufsize, azx_dev->format_val); + if (azx_setup_periods(substream, azx_dev) < 0) + return -EINVAL; azx_setup_controller(chip, azx_dev); if (substream->stream == SNDRV_PCM_STREAM_PLAYBACK) azx_dev->fifo_size = azx_sd_readw(azx_dev, SD_FIFOSIZE) + 1; @@ -1305,37 +1331,94 @@ static int azx_pcm_prepare(struct snd_pcm_substream *substream) static int azx_pcm_trigger(struct snd_pcm_substream *substream, int cmd) { struct azx_pcm *apcm = snd_pcm_substream_chip(substream); - struct azx_dev *azx_dev = get_azx_dev(substream); struct azx *chip = apcm->chip; - int err = 0; + struct azx_dev *azx_dev; + struct snd_pcm_substream *s; + int start, nsync = 0, sbits = 0; + int nwait, timeout; - spin_lock(&chip->reg_lock); switch (cmd) { case SNDRV_PCM_TRIGGER_PAUSE_RELEASE: case SNDRV_PCM_TRIGGER_RESUME: case SNDRV_PCM_TRIGGER_START: - azx_stream_start(chip, azx_dev); - azx_dev->running = 1; + start = 1; break; case SNDRV_PCM_TRIGGER_PAUSE_PUSH: case SNDRV_PCM_TRIGGER_SUSPEND: case SNDRV_PCM_TRIGGER_STOP: - azx_stream_stop(chip, azx_dev); - azx_dev->running = 0; + start = 0; break; default: - err = -EINVAL; + return -EINVAL; + } + + snd_pcm_group_for_each_entry(s, substream) { + if (s->pcm->card != substream->pcm->card) + continue; + azx_dev = get_azx_dev(s); + sbits |= 1 << azx_dev->index; + nsync++; + snd_pcm_trigger_done(s, substream); + } + + spin_lock(&chip->reg_lock); + if (nsync > 1) { + /* first, set SYNC bits of corresponding streams */ + azx_writel(chip, SYNC, azx_readl(chip, SYNC) | sbits); + } + snd_pcm_group_for_each_entry(s, substream) { + if (s->pcm->card != substream->pcm->card) + continue; + azx_dev = get_azx_dev(s); + if (start) + azx_stream_start(chip, azx_dev); + else + azx_stream_stop(chip, azx_dev); + azx_dev->running = start; } spin_unlock(&chip->reg_lock); - if (cmd == SNDRV_PCM_TRIGGER_PAUSE_PUSH || - cmd == SNDRV_PCM_TRIGGER_SUSPEND || - cmd == SNDRV_PCM_TRIGGER_STOP) { - int timeout = 5000; - while ((azx_sd_readb(azx_dev, SD_CTL) & SD_CTL_DMA_START) && - --timeout) - ; + if (start) { + if (nsync == 1) + return 0; + /* wait until all FIFOs get ready */ + for (timeout = 5000; timeout; timeout--) { + nwait = 0; + snd_pcm_group_for_each_entry(s, substream) { + if (s->pcm->card != substream->pcm->card) + continue; + azx_dev = get_azx_dev(s); + if (!(azx_sd_readb(azx_dev, SD_STS) & + SD_STS_FIFO_READY)) + nwait++; + } + if (!nwait) + break; + cpu_relax(); + } + } else { + /* wait until all RUN bits are cleared */ + for (timeout = 5000; timeout; timeout--) { + nwait = 0; + snd_pcm_group_for_each_entry(s, substream) { + if (s->pcm->card != substream->pcm->card) + continue; + azx_dev = get_azx_dev(s); + if (azx_sd_readb(azx_dev, SD_CTL) & + SD_CTL_DMA_START) + nwait++; + } + if (!nwait) + break; + cpu_relax(); + } } - return err; + if (nsync > 1) { + spin_lock(&chip->reg_lock); + /* reset SYNC bits */ + azx_writel(chip, SYNC, azx_readl(chip, SYNC) & ~sbits); + spin_unlock(&chip->reg_lock); + } + return 0; } static snd_pcm_uframes_t azx_pcm_pointer(struct snd_pcm_substream *substream) @@ -1378,6 +1461,7 @@ static struct snd_pcm_ops azx_pcm_ops = { .prepare = azx_pcm_prepare, .trigger = azx_pcm_trigger, .pointer = azx_pcm_pointer, + .page = snd_pcm_sgbuf_ops_page, }; static void azx_pcm_free(struct snd_pcm *pcm) @@ -1386,7 +1470,7 @@ static void azx_pcm_free(struct snd_pcm *pcm) } static int __devinit create_codec_pcm(struct azx *chip, struct hda_codec *codec, - struct hda_pcm *cpcm, int pcm_dev) + struct hda_pcm *cpcm) { int err; struct snd_pcm *pcm; @@ -1400,7 +1484,7 @@ static int __devinit create_codec_pcm(struct azx *chip, struct hda_codec *codec, snd_assert(cpcm->name, return -EINVAL); - err = snd_pcm_new(chip->card, cpcm->name, pcm_dev, + err = snd_pcm_new(chip->card, cpcm->name, cpcm->device, cpcm->stream[0].substreams, cpcm->stream[1].substreams, &pcm); @@ -1420,62 +1504,70 @@ static int __devinit create_codec_pcm(struct azx *chip, struct hda_codec *codec, snd_pcm_set_ops(pcm, SNDRV_PCM_STREAM_PLAYBACK, &azx_pcm_ops); if (cpcm->stream[1].substreams) snd_pcm_set_ops(pcm, SNDRV_PCM_STREAM_CAPTURE, &azx_pcm_ops); - snd_pcm_lib_preallocate_pages_for_all(pcm, SNDRV_DMA_TYPE_DEV, + snd_pcm_lib_preallocate_pages_for_all(pcm, SNDRV_DMA_TYPE_DEV_SG, snd_dma_pci_data(chip->pci), 1024 * 64, 1024 * 1024); - chip->pcm[pcm_dev] = pcm; - if (chip->pcm_devs < pcm_dev + 1) - chip->pcm_devs = pcm_dev + 1; - + chip->pcm[cpcm->device] = pcm; return 0; } static int __devinit azx_pcm_create(struct azx *chip) { + static const char *dev_name[HDA_PCM_NTYPES] = { + "Audio", "SPDIF", "HDMI", "Modem" + }; + /* starting device index for each PCM type */ + static int dev_idx[HDA_PCM_NTYPES] = { + [HDA_PCM_TYPE_AUDIO] = 0, + [HDA_PCM_TYPE_SPDIF] = 1, + [HDA_PCM_TYPE_HDMI] = 3, + [HDA_PCM_TYPE_MODEM] = 6 + }; + /* normal audio device indices; not linear to keep compatibility */ + static int audio_idx[4] = { 0, 2, 4, 5 }; struct hda_codec *codec; int c, err; - int pcm_dev; + int num_devs[HDA_PCM_NTYPES]; err = snd_hda_build_pcms(chip->bus); if (err < 0) return err; /* create audio PCMs */ - pcm_dev = 0; + memset(num_devs, 0, sizeof(num_devs)); list_for_each_entry(codec, &chip->bus->codec_list, list) { for (c = 0; c < codec->num_pcms; c++) { - if (codec->pcm_info[c].is_modem) - continue; /* create later */ - if (pcm_dev >= AZX_MAX_AUDIO_PCMS) { - snd_printk(KERN_ERR SFX - "Too many audio PCMs\n"); - return -EINVAL; - } - err = create_codec_pcm(chip, codec, - &codec->pcm_info[c], pcm_dev); - if (err < 0) - return err; - pcm_dev++; - } - } - - /* create modem PCMs */ - pcm_dev = AZX_MAX_AUDIO_PCMS; - list_for_each_entry(codec, &chip->bus->codec_list, list) { - for (c = 0; c < codec->num_pcms; c++) { - if (!codec->pcm_info[c].is_modem) - continue; /* already created */ - if (pcm_dev >= AZX_MAX_PCMS) { - snd_printk(KERN_ERR SFX - "Too many modem PCMs\n"); - return -EINVAL; + struct hda_pcm *cpcm = &codec->pcm_info[c]; + int type = cpcm->pcm_type; + switch (type) { + case HDA_PCM_TYPE_AUDIO: + if (num_devs[type] >= ARRAY_SIZE(audio_idx)) { + snd_printk(KERN_WARNING + "Too many audio devices\n"); + continue; + } + cpcm->device = audio_idx[num_devs[type]]; + break; + case HDA_PCM_TYPE_SPDIF: + case HDA_PCM_TYPE_HDMI: + case HDA_PCM_TYPE_MODEM: + if (num_devs[type]) { + snd_printk(KERN_WARNING + "%s already defined\n", + dev_name[type]); + continue; + } + cpcm->device = dev_idx[type]; + break; + default: + snd_printk(KERN_WARNING + "Invalid PCM type %d\n", type); + continue; } - err = create_codec_pcm(chip, codec, - &codec->pcm_info[c], pcm_dev); + num_devs[type]++; + err = create_codec_pcm(chip, codec, cpcm); if (err < 0) return err; - chip->pcm[pcm_dev]->dev_class = SNDRV_PCM_CLASS_MODEM; - pcm_dev++; } } return 0; @@ -1502,10 +1594,7 @@ static int __devinit azx_init_stream(struct azx *chip) * and initialize */ for (i = 0; i < chip->num_streams; i++) { - unsigned int off = sizeof(u32) * (i * AZX_MAX_FRAG * 4); struct azx_dev *azx_dev = &chip->azx_dev[i]; - azx_dev->bdl = (u32 *)(chip->bdl.area + off); - azx_dev->bdl_addr = chip->bdl.addr + off; azx_dev->posbuf = (u32 __iomem *)(chip->posbuf.area + i * 8); /* offset: SDI0=0x80, SDI1=0xa0, ... SDO3=0x160 */ azx_dev->sd_addr = chip->remap_addr + (0x20 * i + 0x80); @@ -1587,7 +1676,7 @@ static int azx_suspend(struct pci_dev *pci, pm_message_t state) int i; snd_power_change_state(card, SNDRV_CTL_POWER_D3hot); - for (i = 0; i < chip->pcm_devs; i++) + for (i = 0; i < AZX_MAX_PCMS; i++) snd_pcm_suspend_all(chip->pcm[i]); if (chip->initialized) snd_hda_suspend(chip->bus, state); @@ -1641,8 +1730,9 @@ static int azx_resume(struct pci_dev *pci) */ static int azx_free(struct azx *chip) { + int i; + if (chip->initialized) { - int i; for (i = 0; i < chip->num_streams; i++) azx_stream_stop(chip, &chip->azx_dev[i]); azx_stop_chip(chip); @@ -1657,8 +1747,11 @@ static int azx_free(struct azx *chip) if (chip->remap_addr) iounmap(chip->remap_addr); - if (chip->bdl.area) - snd_dma_free_pages(&chip->bdl); + if (chip->azx_dev) { + for (i = 0; i < chip->num_streams; i++) + if (chip->azx_dev[i].bdl.area) + snd_dma_free_pages(&chip->azx_dev[i].bdl); + } if (chip->rb.area) snd_dma_free_pages(&chip->rb); if (chip->posbuf.area) @@ -1682,6 +1775,7 @@ static int azx_dev_free(struct snd_device *device) static struct snd_pci_quirk position_fix_list[] __devinitdata = { SND_PCI_QUIRK(0x1028, 0x01cc, "Dell D820", POS_FIX_NONE), SND_PCI_QUIRK(0x1028, 0x01de, "Dell Precision 390", POS_FIX_NONE), + SND_PCI_QUIRK(0x1043, 0x813d, "ASUS P5AD2", POS_FIX_NONE), {} }; @@ -1740,7 +1834,7 @@ static int __devinit azx_create(struct snd_card *card, struct pci_dev *pci, struct azx **rchip) { struct azx *chip; - int err; + int i, err; unsigned short gcap; static struct snd_device_ops ops = { .dev_free = azx_dev_free, @@ -1812,38 +1906,35 @@ static int __devinit azx_create(struct snd_card *card, struct pci_dev *pci, gcap = azx_readw(chip, GCAP); snd_printdd("chipset global capabilities = 0x%x\n", gcap); - if (gcap) { - /* read number of streams from GCAP register instead of using - * hardcoded value - */ - chip->playback_streams = (gcap & (0xF << 12)) >> 12; - chip->capture_streams = (gcap & (0xF << 8)) >> 8; - chip->playback_index_offset = chip->capture_streams; - chip->capture_index_offset = 0; - } else { + /* allow 64bit DMA address if supported by H/W */ + if ((gcap & 0x01) && !pci_set_dma_mask(pci, DMA_64BIT_MASK)) + pci_set_consistent_dma_mask(pci, DMA_64BIT_MASK); + + /* read number of streams from GCAP register instead of using + * hardcoded value + */ + chip->capture_streams = (gcap >> 8) & 0x0f; + chip->playback_streams = (gcap >> 12) & 0x0f; + if (!chip->playback_streams && !chip->capture_streams) { /* gcap didn't give any info, switching to old method */ switch (chip->driver_type) { case AZX_DRIVER_ULI: chip->playback_streams = ULI_NUM_PLAYBACK; chip->capture_streams = ULI_NUM_CAPTURE; - chip->playback_index_offset = ULI_PLAYBACK_INDEX; - chip->capture_index_offset = ULI_CAPTURE_INDEX; break; case AZX_DRIVER_ATIHDMI: chip->playback_streams = ATIHDMI_NUM_PLAYBACK; chip->capture_streams = ATIHDMI_NUM_CAPTURE; - chip->playback_index_offset = ATIHDMI_PLAYBACK_INDEX; - chip->capture_index_offset = ATIHDMI_CAPTURE_INDEX; break; default: chip->playback_streams = ICH6_NUM_PLAYBACK; chip->capture_streams = ICH6_NUM_CAPTURE; - chip->playback_index_offset = ICH6_PLAYBACK_INDEX; - chip->capture_index_offset = ICH6_CAPTURE_INDEX; break; } } + chip->capture_index_offset = 0; + chip->playback_index_offset = chip->capture_streams; chip->num_streams = chip->playback_streams + chip->capture_streams; chip->azx_dev = kcalloc(chip->num_streams, sizeof(*chip->azx_dev), GFP_KERNEL); @@ -1852,13 +1943,15 @@ static int __devinit azx_create(struct snd_card *card, struct pci_dev *pci, goto errout; } - /* allocate memory for the BDL for each stream */ - err = snd_dma_alloc_pages(SNDRV_DMA_TYPE_DEV, - snd_dma_pci_data(chip->pci), - BDL_SIZE, &chip->bdl); - if (err < 0) { - snd_printk(KERN_ERR SFX "cannot allocate BDL\n"); - goto errout; + for (i = 0; i < chip->num_streams; i++) { + /* allocate memory for the BDL for each stream */ + err = snd_dma_alloc_pages(SNDRV_DMA_TYPE_DEV, + snd_dma_pci_data(chip->pci), + BDL_SIZE, &chip->azx_dev[i].bdl); + if (err < 0) { + snd_printk(KERN_ERR SFX "cannot allocate BDL\n"); + goto errout; + } } /* allocate memory for the position buffer */ err = snd_dma_alloc_pages(SNDRV_DMA_TYPE_DEV, @@ -1994,48 +2087,63 @@ static void __devexit azx_remove(struct pci_dev *pci) /* PCI IDs */ static struct pci_device_id azx_ids[] = { - { 0x8086, 0x2668, PCI_ANY_ID, PCI_ANY_ID, 0, 0, AZX_DRIVER_ICH }, /* ICH6 */ - { 0x8086, 0x27d8, PCI_ANY_ID, PCI_ANY_ID, 0, 0, AZX_DRIVER_ICH }, /* ICH7 */ - { 0x8086, 0x269a, PCI_ANY_ID, PCI_ANY_ID, 0, 0, AZX_DRIVER_ICH }, /* ESB2 */ - { 0x8086, 0x284b, PCI_ANY_ID, PCI_ANY_ID, 0, 0, AZX_DRIVER_ICH }, /* ICH8 */ - { 0x8086, 0x293e, PCI_ANY_ID, PCI_ANY_ID, 0, 0, AZX_DRIVER_ICH }, /* ICH9 */ - { 0x8086, 0x293f, PCI_ANY_ID, PCI_ANY_ID, 0, 0, AZX_DRIVER_ICH }, /* ICH9 */ - { 0x8086, 0x3a3e, PCI_ANY_ID, PCI_ANY_ID, 0, 0, AZX_DRIVER_ICH }, /* ICH10 */ - { 0x8086, 0x3a6e, PCI_ANY_ID, PCI_ANY_ID, 0, 0, AZX_DRIVER_ICH }, /* ICH10 */ - { 0x8086, 0x811b, PCI_ANY_ID, PCI_ANY_ID, 0, 0, AZX_DRIVER_SCH }, /* SCH*/ - { 0x1002, 0x437b, PCI_ANY_ID, PCI_ANY_ID, 0, 0, AZX_DRIVER_ATI }, /* ATI SB450 */ - { 0x1002, 0x4383, PCI_ANY_ID, PCI_ANY_ID, 0, 0, AZX_DRIVER_ATI }, /* ATI SB600 */ - { 0x1002, 0x793b, PCI_ANY_ID, PCI_ANY_ID, 0, 0, AZX_DRIVER_ATIHDMI }, /* ATI RS600 HDMI */ - { 0x1002, 0x7919, PCI_ANY_ID, PCI_ANY_ID, 0, 0, AZX_DRIVER_ATIHDMI }, /* ATI RS690 HDMI */ - { 0x1002, 0x960f, PCI_ANY_ID, PCI_ANY_ID, 0, 0, AZX_DRIVER_ATIHDMI }, /* ATI RS780 HDMI */ - { 0x1002, 0xaa00, PCI_ANY_ID, PCI_ANY_ID, 0, 0, AZX_DRIVER_ATIHDMI }, /* ATI R600 HDMI */ - { 0x1002, 0xaa08, PCI_ANY_ID, PCI_ANY_ID, 0, 0, AZX_DRIVER_ATIHDMI }, /* ATI RV630 HDMI */ - { 0x1002, 0xaa10, PCI_ANY_ID, PCI_ANY_ID, 0, 0, AZX_DRIVER_ATIHDMI }, /* ATI RV610 HDMI */ - { 0x1002, 0xaa18, PCI_ANY_ID, PCI_ANY_ID, 0, 0, AZX_DRIVER_ATIHDMI }, /* ATI RV670 HDMI */ - { 0x1002, 0xaa20, PCI_ANY_ID, PCI_ANY_ID, 0, 0, AZX_DRIVER_ATIHDMI }, /* ATI RV635 HDMI */ - { 0x1002, 0xaa28, PCI_ANY_ID, PCI_ANY_ID, 0, 0, AZX_DRIVER_ATIHDMI }, /* ATI RV620 HDMI */ - { 0x1002, 0xaa30, PCI_ANY_ID, PCI_ANY_ID, 0, 0, AZX_DRIVER_ATIHDMI }, /* ATI RV770 HDMI */ - { 0x1106, 0x3288, PCI_ANY_ID, PCI_ANY_ID, 0, 0, AZX_DRIVER_VIA }, /* VIA VT8251/VT8237A */ - { 0x1039, 0x7502, PCI_ANY_ID, PCI_ANY_ID, 0, 0, AZX_DRIVER_SIS }, /* SIS966 */ - { 0x10b9, 0x5461, PCI_ANY_ID, PCI_ANY_ID, 0, 0, AZX_DRIVER_ULI }, /* ULI M5461 */ - { 0x10de, 0x026c, PCI_ANY_ID, PCI_ANY_ID, 0, 0, AZX_DRIVER_NVIDIA }, /* NVIDIA MCP51 */ - { 0x10de, 0x0371, PCI_ANY_ID, PCI_ANY_ID, 0, 0, AZX_DRIVER_NVIDIA }, /* NVIDIA MCP55 */ - { 0x10de, 0x03e4, PCI_ANY_ID, PCI_ANY_ID, 0, 0, AZX_DRIVER_NVIDIA }, /* NVIDIA MCP61 */ - { 0x10de, 0x03f0, PCI_ANY_ID, PCI_ANY_ID, 0, 0, AZX_DRIVER_NVIDIA }, /* NVIDIA MCP61 */ - { 0x10de, 0x044a, PCI_ANY_ID, PCI_ANY_ID, 0, 0, AZX_DRIVER_NVIDIA }, /* NVIDIA MCP65 */ - { 0x10de, 0x044b, PCI_ANY_ID, PCI_ANY_ID, 0, 0, AZX_DRIVER_NVIDIA }, /* NVIDIA MCP65 */ - { 0x10de, 0x055c, PCI_ANY_ID, PCI_ANY_ID, 0, 0, AZX_DRIVER_NVIDIA }, /* NVIDIA MCP67 */ - { 0x10de, 0x055d, PCI_ANY_ID, PCI_ANY_ID, 0, 0, AZX_DRIVER_NVIDIA }, /* NVIDIA MCP67 */ - { 0x10de, 0x07fc, PCI_ANY_ID, PCI_ANY_ID, 0, 0, AZX_DRIVER_NVIDIA }, /* NVIDIA MCP73 */ - { 0x10de, 0x07fd, PCI_ANY_ID, PCI_ANY_ID, 0, 0, AZX_DRIVER_NVIDIA }, /* NVIDIA MCP73 */ - { 0x10de, 0x0774, PCI_ANY_ID, PCI_ANY_ID, 0, 0, AZX_DRIVER_NVIDIA }, /* NVIDIA MCP77 */ - { 0x10de, 0x0775, PCI_ANY_ID, PCI_ANY_ID, 0, 0, AZX_DRIVER_NVIDIA }, /* NVIDIA MCP77 */ - { 0x10de, 0x0776, PCI_ANY_ID, PCI_ANY_ID, 0, 0, AZX_DRIVER_NVIDIA }, /* NVIDIA MCP77 */ - { 0x10de, 0x0777, PCI_ANY_ID, PCI_ANY_ID, 0, 0, AZX_DRIVER_NVIDIA }, /* NVIDIA MCP77 */ - { 0x10de, 0x0ac0, PCI_ANY_ID, PCI_ANY_ID, 0, 0, AZX_DRIVER_NVIDIA }, /* NVIDIA MCP79 */ - { 0x10de, 0x0ac1, PCI_ANY_ID, PCI_ANY_ID, 0, 0, AZX_DRIVER_NVIDIA }, /* NVIDIA MCP79 */ - { 0x10de, 0x0ac2, PCI_ANY_ID, PCI_ANY_ID, 0, 0, AZX_DRIVER_NVIDIA }, /* NVIDIA MCP79 */ - { 0x10de, 0x0ac3, PCI_ANY_ID, PCI_ANY_ID, 0, 0, AZX_DRIVER_NVIDIA }, /* NVIDIA MCP79 */ + /* ICH 6..10 */ + { PCI_DEVICE(0x8086, 0x2668), .driver_data = AZX_DRIVER_ICH }, + { PCI_DEVICE(0x8086, 0x27d8), .driver_data = AZX_DRIVER_ICH }, + { PCI_DEVICE(0x8086, 0x269a), .driver_data = AZX_DRIVER_ICH }, + { PCI_DEVICE(0x8086, 0x284b), .driver_data = AZX_DRIVER_ICH }, + { PCI_DEVICE(0x8086, 0x293e), .driver_data = AZX_DRIVER_ICH }, + { PCI_DEVICE(0x8086, 0x293f), .driver_data = AZX_DRIVER_ICH }, + { PCI_DEVICE(0x8086, 0x3a3e), .driver_data = AZX_DRIVER_ICH }, + { PCI_DEVICE(0x8086, 0x3a6e), .driver_data = AZX_DRIVER_ICH }, + /* SCH */ + { PCI_DEVICE(0x8086, 0x811b), .driver_data = AZX_DRIVER_SCH }, + /* ATI SB 450/600 */ + { PCI_DEVICE(0x1002, 0x437b), .driver_data = AZX_DRIVER_ATI }, + { PCI_DEVICE(0x1002, 0x4383), .driver_data = AZX_DRIVER_ATI }, + /* ATI HDMI */ + { PCI_DEVICE(0x1002, 0x793b), .driver_data = AZX_DRIVER_ATIHDMI }, + { PCI_DEVICE(0x1002, 0x7919), .driver_data = AZX_DRIVER_ATIHDMI }, + { PCI_DEVICE(0x1002, 0x960f), .driver_data = AZX_DRIVER_ATIHDMI }, + { PCI_DEVICE(0x1002, 0xaa00), .driver_data = AZX_DRIVER_ATIHDMI }, + { PCI_DEVICE(0x1002, 0xaa08), .driver_data = AZX_DRIVER_ATIHDMI }, + { PCI_DEVICE(0x1002, 0xaa10), .driver_data = AZX_DRIVER_ATIHDMI }, + { PCI_DEVICE(0x1002, 0xaa18), .driver_data = AZX_DRIVER_ATIHDMI }, + { PCI_DEVICE(0x1002, 0xaa20), .driver_data = AZX_DRIVER_ATIHDMI }, + { PCI_DEVICE(0x1002, 0xaa28), .driver_data = AZX_DRIVER_ATIHDMI }, + { PCI_DEVICE(0x1002, 0xaa30), .driver_data = AZX_DRIVER_ATIHDMI }, + { PCI_DEVICE(0x1002, 0xaa38), .driver_data = AZX_DRIVER_ATIHDMI }, + { PCI_DEVICE(0x1002, 0xaa40), .driver_data = AZX_DRIVER_ATIHDMI }, + { PCI_DEVICE(0x1002, 0xaa48), .driver_data = AZX_DRIVER_ATIHDMI }, + /* VIA VT8251/VT8237A */ + { PCI_DEVICE(0x1106, 0x3288), .driver_data = AZX_DRIVER_VIA }, + /* SIS966 */ + { PCI_DEVICE(0x1039, 0x7502), .driver_data = AZX_DRIVER_SIS }, + /* ULI M5461 */ + { PCI_DEVICE(0x10b9, 0x5461), .driver_data = AZX_DRIVER_ULI }, + /* NVIDIA MCP */ + { PCI_DEVICE(0x10de, 0x026c), .driver_data = AZX_DRIVER_NVIDIA }, + { PCI_DEVICE(0x10de, 0x0371), .driver_data = AZX_DRIVER_NVIDIA }, + { PCI_DEVICE(0x10de, 0x03e4), .driver_data = AZX_DRIVER_NVIDIA }, + { PCI_DEVICE(0x10de, 0x03f0), .driver_data = AZX_DRIVER_NVIDIA }, + { PCI_DEVICE(0x10de, 0x044a), .driver_data = AZX_DRIVER_NVIDIA }, + { PCI_DEVICE(0x10de, 0x044b), .driver_data = AZX_DRIVER_NVIDIA }, + { PCI_DEVICE(0x10de, 0x055c), .driver_data = AZX_DRIVER_NVIDIA }, + { PCI_DEVICE(0x10de, 0x055d), .driver_data = AZX_DRIVER_NVIDIA }, + { PCI_DEVICE(0x10de, 0x0774), .driver_data = AZX_DRIVER_NVIDIA }, + { PCI_DEVICE(0x10de, 0x0775), .driver_data = AZX_DRIVER_NVIDIA }, + { PCI_DEVICE(0x10de, 0x0776), .driver_data = AZX_DRIVER_NVIDIA }, + { PCI_DEVICE(0x10de, 0x0777), .driver_data = AZX_DRIVER_NVIDIA }, + { PCI_DEVICE(0x10de, 0x07fc), .driver_data = AZX_DRIVER_NVIDIA }, + { PCI_DEVICE(0x10de, 0x07fd), .driver_data = AZX_DRIVER_NVIDIA }, + { PCI_DEVICE(0x10de, 0x0ac0), .driver_data = AZX_DRIVER_NVIDIA }, + { PCI_DEVICE(0x10de, 0x0ac1), .driver_data = AZX_DRIVER_NVIDIA }, + { PCI_DEVICE(0x10de, 0x0ac2), .driver_data = AZX_DRIVER_NVIDIA }, + { PCI_DEVICE(0x10de, 0x0ac3), .driver_data = AZX_DRIVER_NVIDIA }, + { PCI_DEVICE(0x10de, 0x0bd4), .driver_data = AZX_DRIVER_NVIDIA }, + { PCI_DEVICE(0x10de, 0x0bd5), .driver_data = AZX_DRIVER_NVIDIA }, + { PCI_DEVICE(0x10de, 0x0bd6), .driver_data = AZX_DRIVER_NVIDIA }, + { PCI_DEVICE(0x10de, 0x0bd7), .driver_data = AZX_DRIVER_NVIDIA }, { 0, } }; MODULE_DEVICE_TABLE(pci, azx_ids); diff --git a/sound/pci/hda/hda_local.h b/sound/pci/hda/hda_local.h index ad0014a..5c9e578 100644 --- a/sound/pci/hda/hda_local.h +++ b/sound/pci/hda/hda_local.h @@ -228,8 +228,18 @@ struct hda_multi_out { int max_channels; /* currently supported analog channels */ int dig_out_used; /* current usage of digital out (HDA_DIG_XXX) */ int no_share_stream; /* don't share a stream with multiple pins */ + int share_spdif; /* share SPDIF pin */ + /* PCM information for both analog and SPDIF DACs */ + unsigned int analog_rates; + unsigned int analog_maxbps; + u64 analog_formats; + unsigned int spdif_rates; + unsigned int spdif_maxbps; + u64 spdif_formats; }; +int snd_hda_create_spdif_share_sw(struct hda_codec *codec, + struct hda_multi_out *mout); int snd_hda_multi_out_dig_open(struct hda_codec *codec, struct hda_multi_out *mout); int snd_hda_multi_out_dig_close(struct hda_codec *codec, @@ -241,7 +251,8 @@ int snd_hda_multi_out_dig_prepare(struct hda_codec *codec, struct snd_pcm_substream *substream); int snd_hda_multi_out_analog_open(struct hda_codec *codec, struct hda_multi_out *mout, - struct snd_pcm_substream *substream); + struct snd_pcm_substream *substream, + struct hda_pcm_stream *hinfo); int snd_hda_multi_out_analog_prepare(struct hda_codec *codec, struct hda_multi_out *mout, unsigned int stream_tag, @@ -407,11 +418,4 @@ int snd_hda_check_amp_list_power(struct hda_codec *codec, hda_nid_t nid); #endif /* CONFIG_SND_HDA_POWER_SAVE */ -/* - * virtual master control - */ -struct snd_kcontrol *snd_ctl_make_virtual_master(char *name, - const unsigned int *tlv); -int snd_ctl_add_slave(struct snd_kcontrol *master, struct snd_kcontrol *slave); - #endif /* __SOUND_HDA_LOCAL_H */ diff --git a/sound/pci/hda/hda_patch.h b/sound/pci/hda/hda_patch.h index f5c23bb..2fdf235 100644 --- a/sound/pci/hda/hda_patch.h +++ b/sound/pci/hda/hda_patch.h @@ -18,31 +18,3 @@ extern struct hda_codec_preset snd_hda_preset_atihdmi[]; extern struct hda_codec_preset snd_hda_preset_conexant[]; /* VIA codecs */ extern struct hda_codec_preset snd_hda_preset_via[]; - -static const struct hda_codec_preset *hda_preset_tables[] = { -#ifdef CONFIG_SND_HDA_CODEC_REALTEK - snd_hda_preset_realtek, -#endif -#ifdef CONFIG_SND_HDA_CODEC_CMEDIA - snd_hda_preset_cmedia, -#endif -#ifdef CONFIG_SND_HDA_CODEC_ANALOG - snd_hda_preset_analog, -#endif -#ifdef CONFIG_SND_HDA_CODEC_SIGMATEL - snd_hda_preset_sigmatel, -#endif -#ifdef CONFIG_SND_HDA_CODEC_SI3054 - snd_hda_preset_si3054, -#endif -#ifdef CONFIG_SND_HDA_CODEC_ATIHDMI - snd_hda_preset_atihdmi, -#endif -#ifdef CONFIG_SND_HDA_CODEC_CONEXANT - snd_hda_preset_conexant, -#endif -#ifdef CONFIG_SND_HDA_CODEC_VIA - snd_hda_preset_via, -#endif - NULL -}; diff --git a/sound/pci/hda/patch_analog.c b/sound/pci/hda/patch_analog.c index c864928..f486eb1 100644 --- a/sound/pci/hda/patch_analog.c +++ b/sound/pci/hda/patch_analog.c @@ -28,6 +28,7 @@ #include #include "hda_codec.h" #include "hda_local.h" +#include "hda_patch.h" struct ad198x_spec { struct snd_kcontrol_new *mixers[5]; @@ -80,7 +81,6 @@ struct ad198x_spec { #endif /* for virtual master */ hda_nid_t vmaster_nid; - u32 vmaster_tlv[4]; const char **slave_vols; const char **slave_sws; }; @@ -171,6 +171,11 @@ static int ad198x_build_controls(struct hda_codec *codec) err = snd_hda_create_spdif_out_ctls(codec, spec->multiout.dig_out_nid); if (err < 0) return err; + err = snd_hda_create_spdif_share_sw(codec, + &spec->multiout); + if (err < 0) + return err; + spec->multiout.share_spdif = 1; } if (spec->dig_in_nid) { err = snd_hda_create_spdif_in_ctls(codec, spec->dig_in_nid); @@ -180,10 +185,11 @@ static int ad198x_build_controls(struct hda_codec *codec) /* if we have no master control, let's create it */ if (!snd_hda_find_mixer_ctl(codec, "Master Playback Volume")) { + unsigned int vmaster_tlv[4]; snd_hda_set_vmaster_tlv(codec, spec->vmaster_nid, - HDA_OUTPUT, spec->vmaster_tlv); + HDA_OUTPUT, vmaster_tlv); err = snd_hda_add_vmaster(codec, "Master Playback Volume", - spec->vmaster_tlv, + vmaster_tlv, (spec->slave_vols ? spec->slave_vols : ad_slave_vols)); if (err < 0) @@ -217,7 +223,8 @@ static int ad198x_playback_pcm_open(struct hda_pcm_stream *hinfo, struct snd_pcm_substream *substream) { struct ad198x_spec *spec = codec->spec; - return snd_hda_multi_out_analog_open(codec, &spec->multiout, substream); + return snd_hda_multi_out_analog_open(codec, &spec->multiout, substream, + hinfo); } static int ad198x_playback_pcm_prepare(struct hda_pcm_stream *hinfo, @@ -289,8 +296,7 @@ static int ad198x_capture_pcm_cleanup(struct hda_pcm_stream *hinfo, struct snd_pcm_substream *substream) { struct ad198x_spec *spec = codec->spec; - snd_hda_codec_setup_stream(codec, spec->adc_nids[substream->number], - 0, 0, 0); + snd_hda_codec_cleanup_stream(codec, spec->adc_nids[substream->number]); return 0; } @@ -359,6 +365,7 @@ static int ad198x_build_pcms(struct hda_codec *codec) info++; codec->num_pcms++; info->name = "AD198x Digital"; + info->pcm_type = HDA_PCM_TYPE_SPDIF; info->stream[SNDRV_PCM_STREAM_PLAYBACK] = ad198x_pcm_digital_playback; info->stream[SNDRV_PCM_STREAM_PLAYBACK].nid = spec->multiout.dig_out_nid; if (spec->dig_in_nid) { @@ -611,13 +618,19 @@ static struct hda_input_mux ad1986a_laptop_eapd_capture_source = { }, }; +static struct hda_input_mux ad1986a_automic_capture_source = { + .num_items = 2, + .items = { + { "Mic", 0x0 }, + { "Mix", 0x5 }, + }, +}; + static struct snd_kcontrol_new ad1986a_laptop_eapd_mixers[] = { HDA_BIND_VOL("Master Playback Volume", &ad1986a_laptop_master_vol), HDA_BIND_SW("Master Playback Switch", &ad1986a_laptop_master_sw), HDA_CODEC_VOLUME("PCM Playback Volume", 0x03, 0x0, HDA_OUTPUT), HDA_CODEC_MUTE("PCM Playback Switch", 0x03, 0x0, HDA_OUTPUT), - HDA_CODEC_VOLUME("Internal Mic Playback Volume", 0x17, 0x0, HDA_OUTPUT), - HDA_CODEC_MUTE("Internal Mic Playback Switch", 0x17, 0x0, HDA_OUTPUT), HDA_CODEC_VOLUME("Mic Playback Volume", 0x13, 0x0, HDA_OUTPUT), HDA_CODEC_MUTE("Mic Playback Switch", 0x13, 0x0, HDA_OUTPUT), HDA_CODEC_VOLUME("Mic Boost", 0x0f, 0x0, HDA_OUTPUT), @@ -641,6 +654,33 @@ static struct snd_kcontrol_new ad1986a_laptop_eapd_mixers[] = { { } /* end */ }; +/* re-connect the mic boost input according to the jack sensing */ +static void ad1986a_automic(struct hda_codec *codec) +{ + unsigned int present; + present = snd_hda_codec_read(codec, 0x1f, 0, AC_VERB_GET_PIN_SENSE, 0); + /* 0 = 0x1f, 2 = 0x1d, 4 = mixed */ + snd_hda_codec_write(codec, 0x0f, 0, AC_VERB_SET_CONNECT_SEL, + (present & AC_PINSENSE_PRESENCE) ? 0 : 2); +} + +#define AD1986A_MIC_EVENT 0x36 + +static void ad1986a_automic_unsol_event(struct hda_codec *codec, + unsigned int res) +{ + if ((res >> 26) != AD1986A_MIC_EVENT) + return; + ad1986a_automic(codec); +} + +static int ad1986a_automic_init(struct hda_codec *codec) +{ + ad198x_init(codec); + ad1986a_automic(codec); + return 0; +} + /* laptop-automute - 2ch only */ static void ad1986a_update_hp(struct hda_codec *codec) @@ -844,6 +884,15 @@ static struct hda_verb ad1986a_eapd_init_verbs[] = { {} }; +static struct hda_verb ad1986a_automic_verbs[] = { + {0x1d, AC_VERB_SET_PIN_WIDGET_CONTROL, PIN_VREF80}, + {0x1f, AC_VERB_SET_PIN_WIDGET_CONTROL, PIN_VREF80}, + /*{0x20, AC_VERB_SET_PIN_WIDGET_CONTROL, PIN_VREF80},*/ + {0x0f, AC_VERB_SET_CONNECT_SEL, 0x0}, + {0x1f, AC_VERB_SET_UNSOLICITED_ENABLE, AC_USRSP_EN | AD1986A_MIC_EVENT}, + {} +}; + /* Ultra initialization */ static struct hda_verb ad1986a_ultra_init[] = { /* eapd initialization */ @@ -986,14 +1035,17 @@ static int patch_ad1986a(struct hda_codec *codec) break; case AD1986A_LAPTOP_EAPD: spec->mixers[0] = ad1986a_laptop_eapd_mixers; - spec->num_init_verbs = 2; + spec->num_init_verbs = 3; spec->init_verbs[1] = ad1986a_eapd_init_verbs; + spec->init_verbs[2] = ad1986a_automic_verbs; spec->multiout.max_channels = 2; spec->multiout.num_dacs = 1; spec->multiout.dac_nids = ad1986a_laptop_dac_nids; if (!is_jack_available(codec, 0x25)) spec->multiout.dig_out_nid = 0; - spec->input_mux = &ad1986a_laptop_eapd_capture_source; + spec->input_mux = &ad1986a_automic_capture_source; + codec->patch_ops.unsol_event = ad1986a_automic_unsol_event; + codec->patch_ops.init = ad1986a_automic_init; break; case AD1986A_LAPTOP_AUTOMUTE: spec->mixers[0] = ad1986a_laptop_automute_mixers; @@ -1365,7 +1417,10 @@ static int ad1981_hp_master_sw_put(struct snd_kcontrol *kcontrol, if (! ad198x_eapd_put(kcontrol, ucontrol)) return 0; - + /* change speaker pin appropriately */ + snd_hda_codec_write(codec, 0x05, 0, + AC_VERB_SET_PIN_WIDGET_CONTROL, + spec->cur_eapd ? PIN_OUT : 0); /* toggle HP mute appropriately */ snd_hda_codec_amp_stereo(codec, 0x06, HDA_OUTPUT, 0, HDA_AMP_MUTE, @@ -2087,6 +2142,10 @@ static struct snd_kcontrol_new ad1988_spdif_in_mixers[] = { { } /* end */ }; +static struct snd_kcontrol_new ad1989_spdif_out_mixers[] = { + HDA_CODEC_VOLUME("IEC958 Playback Volume", 0x1b, 0x0, HDA_OUTPUT), + { } /* end */ +}; /* * initialization verbs @@ -2187,6 +2246,13 @@ static struct hda_verb ad1988_spdif_init_verbs[] = { { } }; +/* AD1989 has no ADC -> SPDIF route */ +static struct hda_verb ad1989_spdif_init_verbs[] = { + /* SPDIF out pin */ + {0x1b, AC_VERB_SET_AMP_GAIN_MUTE, AMP_OUT_UNMUTE | 0x27}, /* 0dB */ + { } +}; + /* * verbs for 3stack (+dig) */ @@ -2894,10 +2960,19 @@ static int patch_ad1988(struct hda_codec *codec) spec->mixers[spec->num_mixers++] = ad1988_capture_mixers; spec->init_verbs[spec->num_init_verbs++] = ad1988_capture_init_verbs; if (spec->multiout.dig_out_nid) { - spec->mixers[spec->num_mixers++] = ad1988_spdif_out_mixers; - spec->init_verbs[spec->num_init_verbs++] = ad1988_spdif_init_verbs; + if (codec->vendor_id >= 0x11d4989a) { + spec->mixers[spec->num_mixers++] = + ad1989_spdif_out_mixers; + spec->init_verbs[spec->num_init_verbs++] = + ad1989_spdif_init_verbs; + } else { + spec->mixers[spec->num_mixers++] = + ad1988_spdif_out_mixers; + spec->init_verbs[spec->num_init_verbs++] = + ad1988_spdif_init_verbs; + } } - if (spec->dig_in_nid) + if (spec->dig_in_nid && codec->vendor_id < 0x11d4989a) spec->mixers[spec->num_mixers++] = ad1988_spdif_in_mixers; codec->patch_ops = ad198x_patch_ops; @@ -3133,11 +3208,12 @@ static int patch_ad1884(struct hda_codec *codec) * Lenovo Thinkpad T61/X61 */ static struct hda_input_mux ad1984_thinkpad_capture_source = { - .num_items = 3, + .num_items = 4, .items = { { "Mic", 0x0 }, { "Internal Mic", 0x1 }, { "Mix", 0x3 }, + { "Docking-Station", 0x4 }, }, }; @@ -3268,8 +3344,7 @@ static int ad1984_pcm_dmic_cleanup(struct hda_pcm_stream *hinfo, struct hda_codec *codec, struct snd_pcm_substream *substream) { - snd_hda_codec_setup_stream(codec, 0x05 + substream->number, - 0, 0, 0); + snd_hda_codec_cleanup_stream(codec, 0x05 + substream->number); return 0; } @@ -3356,6 +3431,469 @@ static int patch_ad1984(struct hda_codec *codec) /* + * AD1883 / AD1884A / AD1984A / AD1984B + * + * port-B (0x14) - front mic-in + * port-E (0x1c) - rear mic-in + * port-F (0x16) - CD / ext out + * port-C (0x15) - rear line-in + * port-D (0x12) - rear line-out + * port-A (0x11) - front hp-out + * + * AD1984A = AD1884A + digital-mic + * AD1883 = equivalent with AD1984A + * AD1984B = AD1984A + extra SPDIF-out + * + * FIXME: + * We share the single DAC for both HP and line-outs (see AD1884/1984). + */ + +static hda_nid_t ad1884a_dac_nids[1] = { + 0x03, +}; + +#define ad1884a_adc_nids ad1884_adc_nids +#define ad1884a_capsrc_nids ad1884_capsrc_nids + +#define AD1884A_SPDIF_OUT 0x02 + +static struct hda_input_mux ad1884a_capture_source = { + .num_items = 5, + .items = { + { "Front Mic", 0x0 }, + { "Mic", 0x4 }, + { "Line", 0x1 }, + { "CD", 0x2 }, + { "Mix", 0x3 }, + }, +}; + +static struct snd_kcontrol_new ad1884a_base_mixers[] = { + HDA_CODEC_VOLUME("Master Playback Volume", 0x21, 0x0, HDA_OUTPUT), + HDA_CODEC_MUTE("Master Playback Switch", 0x21, 0x0, HDA_OUTPUT), + HDA_CODEC_MUTE("Headphone Playback Switch", 0x11, 0x0, HDA_OUTPUT), + HDA_CODEC_MUTE("Front Playback Switch", 0x12, 0x0, HDA_OUTPUT), + HDA_CODEC_VOLUME_MONO("Mono Playback Volume", 0x13, 1, 0x0, HDA_OUTPUT), + HDA_CODEC_MUTE_MONO("Mono Playback Switch", 0x13, 1, 0x0, HDA_OUTPUT), + HDA_CODEC_VOLUME("PCM Playback Volume", 0x20, 0x5, HDA_INPUT), + HDA_CODEC_MUTE("PCM Playback Switch", 0x20, 0x5, HDA_INPUT), + HDA_CODEC_VOLUME("Front Mic Playback Volume", 0x20, 0x00, HDA_INPUT), + HDA_CODEC_MUTE("Front Mic Playback Switch", 0x20, 0x00, HDA_INPUT), + HDA_CODEC_VOLUME("Line Playback Volume", 0x20, 0x01, HDA_INPUT), + HDA_CODEC_MUTE("Line Playback Switch", 0x20, 0x01, HDA_INPUT), + HDA_CODEC_VOLUME("Mic Playback Volume", 0x20, 0x04, HDA_INPUT), + HDA_CODEC_MUTE("Mic Playback Switch", 0x20, 0x04, HDA_INPUT), + HDA_CODEC_VOLUME("CD Playback Volume", 0x20, 0x02, HDA_INPUT), + HDA_CODEC_MUTE("CD Playback Switch", 0x20, 0x02, HDA_INPUT), + HDA_CODEC_VOLUME("Beep Playback Volume", 0x20, 0x03, HDA_INPUT), + HDA_CODEC_MUTE("Beep Playback Switch", 0x20, 0x03, HDA_INPUT), + HDA_CODEC_VOLUME("Front Mic Boost", 0x14, 0x0, HDA_INPUT), + HDA_CODEC_VOLUME("Line Boost", 0x15, 0x0, HDA_INPUT), + HDA_CODEC_VOLUME("Mic Boost", 0x25, 0x0, HDA_OUTPUT), + HDA_CODEC_VOLUME("Capture Volume", 0x0c, 0x0, HDA_OUTPUT), + HDA_CODEC_MUTE("Capture Switch", 0x0c, 0x0, HDA_OUTPUT), + HDA_CODEC_VOLUME_IDX("Capture Volume", 1, 0x0d, 0x0, HDA_OUTPUT), + HDA_CODEC_MUTE_IDX("Capture Switch", 1, 0x0d, 0x0, HDA_OUTPUT), + { + .iface = SNDRV_CTL_ELEM_IFACE_MIXER, + /* The multiple "Capture Source" controls confuse alsamixer + * So call somewhat different.. + */ + /* .name = "Capture Source", */ + .name = "Input Source", + .count = 2, + .info = ad198x_mux_enum_info, + .get = ad198x_mux_enum_get, + .put = ad198x_mux_enum_put, + }, + /* SPDIF controls */ + HDA_CODEC_VOLUME("IEC958 Playback Volume", 0x1b, 0x0, HDA_OUTPUT), + { + .iface = SNDRV_CTL_ELEM_IFACE_MIXER, + .name = SNDRV_CTL_NAME_IEC958("",PLAYBACK,NONE) "Source", + /* identical with ad1983 */ + .info = ad1983_spdif_route_info, + .get = ad1983_spdif_route_get, + .put = ad1983_spdif_route_put, + }, + { } /* end */ +}; + +/* + * initialization verbs + */ +static struct hda_verb ad1884a_init_verbs[] = { + /* DACs; unmute as default */ + {0x03, AC_VERB_SET_AMP_GAIN_MUTE, 0x27}, /* 0dB */ + {0x04, AC_VERB_SET_AMP_GAIN_MUTE, 0x27}, /* 0dB */ + /* Port-A (HP) mixer - route only from analog mixer */ + {0x07, AC_VERB_SET_AMP_GAIN_MUTE, AMP_IN_MUTE(0)}, + {0x07, AC_VERB_SET_AMP_GAIN_MUTE, AMP_IN_UNMUTE(1)}, + /* Port-A pin */ + {0x11, AC_VERB_SET_PIN_WIDGET_CONTROL, PIN_HP}, + {0x11, AC_VERB_SET_AMP_GAIN_MUTE, AMP_OUT_MUTE}, + /* Port-D (Line-out) mixer - route only from analog mixer */ + {0x0a, AC_VERB_SET_AMP_GAIN_MUTE, AMP_IN_MUTE(0)}, + {0x0a, AC_VERB_SET_AMP_GAIN_MUTE, AMP_IN_UNMUTE(1)}, + /* Port-D pin */ + {0x12, AC_VERB_SET_PIN_WIDGET_CONTROL, PIN_HP}, + {0x12, AC_VERB_SET_AMP_GAIN_MUTE, AMP_OUT_MUTE}, + /* Mono-out mixer - route only from analog mixer */ + {0x1e, AC_VERB_SET_AMP_GAIN_MUTE, AMP_IN_MUTE(0)}, + {0x1e, AC_VERB_SET_AMP_GAIN_MUTE, AMP_IN_UNMUTE(1)}, + /* Mono-out pin */ + {0x13, AC_VERB_SET_PIN_WIDGET_CONTROL, PIN_HP}, + {0x13, AC_VERB_SET_AMP_GAIN_MUTE, AMP_OUT_MUTE}, + /* Port-B (front mic) pin */ + {0x14, AC_VERB_SET_PIN_WIDGET_CONTROL, PIN_VREF80}, + {0x14, AC_VERB_SET_AMP_GAIN_MUTE, AMP_OUT_MUTE}, + /* Port-C (rear line-in) pin */ + {0x15, AC_VERB_SET_PIN_WIDGET_CONTROL, PIN_IN}, + {0x15, AC_VERB_SET_AMP_GAIN_MUTE, AMP_OUT_MUTE}, + /* Port-E (rear mic) pin */ + {0x1c, AC_VERB_SET_PIN_WIDGET_CONTROL, PIN_VREF80}, + {0x1c, AC_VERB_SET_AMP_GAIN_MUTE, AMP_OUT_MUTE}, + {0x25, AC_VERB_SET_AMP_GAIN_MUTE, AMP_OUT_ZERO}, /* no boost */ + /* Port-F (CD) pin */ + {0x16, AC_VERB_SET_PIN_WIDGET_CONTROL, PIN_IN}, + {0x16, AC_VERB_SET_AMP_GAIN_MUTE, AMP_OUT_MUTE}, + /* Analog mixer; mute as default */ + {0x20, AC_VERB_SET_AMP_GAIN_MUTE, AMP_IN_MUTE(0)}, + {0x20, AC_VERB_SET_AMP_GAIN_MUTE, AMP_IN_MUTE(1)}, + {0x20, AC_VERB_SET_AMP_GAIN_MUTE, AMP_IN_MUTE(2)}, + {0x20, AC_VERB_SET_AMP_GAIN_MUTE, AMP_IN_MUTE(3)}, + {0x20, AC_VERB_SET_AMP_GAIN_MUTE, AMP_IN_UNMUTE(4)}, /* aux */ + {0x20, AC_VERB_SET_AMP_GAIN_MUTE, AMP_IN_MUTE(5)}, + /* Analog Mix output amp */ + {0x21, AC_VERB_SET_AMP_GAIN_MUTE, AMP_OUT_MUTE}, + /* capture sources */ + {0x0c, AC_VERB_SET_CONNECT_SEL, 0x0}, + {0x0c, AC_VERB_SET_AMP_GAIN_MUTE, AMP_OUT_MUTE}, + {0x0d, AC_VERB_SET_CONNECT_SEL, 0x0}, + {0x0d, AC_VERB_SET_AMP_GAIN_MUTE, AMP_OUT_MUTE}, + /* SPDIF output amp */ + {0x1b, AC_VERB_SET_AMP_GAIN_MUTE, AMP_OUT_UNMUTE | 0x27}, /* 0dB */ + { } /* end */ +}; + +#ifdef CONFIG_SND_HDA_POWER_SAVE +static struct hda_amp_list ad1884a_loopbacks[] = { + { 0x20, HDA_INPUT, 0 }, /* Front Mic */ + { 0x20, HDA_INPUT, 1 }, /* Mic */ + { 0x20, HDA_INPUT, 2 }, /* CD */ + { 0x20, HDA_INPUT, 4 }, /* Docking */ + { } /* end */ +}; +#endif + +/* + * Laptop model + * + * Port A: Headphone jack + * Port B: MIC jack + * Port C: Internal MIC + * Port D: Dock Line Out (if enabled) + * Port E: Dock Line In (if enabled) + * Port F: Internal speakers + */ + +static struct hda_input_mux ad1884a_laptop_capture_source = { + .num_items = 4, + .items = { + { "Mic", 0x0 }, /* port-B */ + { "Internal Mic", 0x1 }, /* port-C */ + { "Dock Mic", 0x4 }, /* port-E */ + { "Mix", 0x3 }, + }, +}; + +static struct snd_kcontrol_new ad1884a_laptop_mixers[] = { + HDA_CODEC_VOLUME("Master Playback Volume", 0x21, 0x0, HDA_OUTPUT), + HDA_CODEC_MUTE("Master Playback Switch", 0x21, 0x0, HDA_OUTPUT), + HDA_CODEC_MUTE("Dock Playback Switch", 0x12, 0x0, HDA_OUTPUT), + HDA_CODEC_VOLUME("PCM Playback Volume", 0x20, 0x5, HDA_INPUT), + HDA_CODEC_MUTE("PCM Playback Switch", 0x20, 0x5, HDA_INPUT), + HDA_CODEC_VOLUME("Mic Playback Volume", 0x20, 0x00, HDA_INPUT), + HDA_CODEC_MUTE("Mic Playback Switch", 0x20, 0x00, HDA_INPUT), + HDA_CODEC_VOLUME("Internal Mic Playback Volume", 0x20, 0x01, HDA_INPUT), + HDA_CODEC_MUTE("Internal Mic Playback Switch", 0x20, 0x01, HDA_INPUT), + HDA_CODEC_VOLUME("Dock Mic Playback Volume", 0x20, 0x04, HDA_INPUT), + HDA_CODEC_MUTE("Dock Mic Playback Switch", 0x20, 0x04, HDA_INPUT), + HDA_CODEC_VOLUME("Beep Playback Volume", 0x20, 0x03, HDA_INPUT), + HDA_CODEC_MUTE("Beep Playback Switch", 0x20, 0x03, HDA_INPUT), + HDA_CODEC_VOLUME("Mic Boost", 0x14, 0x0, HDA_INPUT), + HDA_CODEC_VOLUME("Internal Mic Boost", 0x15, 0x0, HDA_INPUT), + HDA_CODEC_VOLUME("Dock Mic Boost", 0x25, 0x0, HDA_OUTPUT), + HDA_CODEC_VOLUME("Capture Volume", 0x0c, 0x0, HDA_OUTPUT), + HDA_CODEC_MUTE("Capture Switch", 0x0c, 0x0, HDA_OUTPUT), + HDA_CODEC_VOLUME_IDX("Capture Volume", 1, 0x0d, 0x0, HDA_OUTPUT), + HDA_CODEC_MUTE_IDX("Capture Switch", 1, 0x0d, 0x0, HDA_OUTPUT), + { + .iface = SNDRV_CTL_ELEM_IFACE_MIXER, + /* The multiple "Capture Source" controls confuse alsamixer + * So call somewhat different.. + */ + /* .name = "Capture Source", */ + .name = "Input Source", + .count = 2, + .info = ad198x_mux_enum_info, + .get = ad198x_mux_enum_get, + .put = ad198x_mux_enum_put, + }, + { } /* end */ +}; + +static struct hda_input_mux ad1884a_mobile_capture_source = { + .num_items = 2, + .items = { + { "Mic", 0x1 }, /* port-C */ + { "Mix", 0x3 }, + }, +}; + +static struct snd_kcontrol_new ad1884a_mobile_mixers[] = { + HDA_CODEC_VOLUME("Master Playback Volume", 0x21, 0x0, HDA_OUTPUT), + HDA_CODEC_MUTE("Master Playback Switch", 0x21, 0x0, HDA_OUTPUT), + HDA_CODEC_VOLUME("PCM Playback Volume", 0x20, 0x5, HDA_INPUT), + HDA_CODEC_MUTE("PCM Playback Switch", 0x20, 0x5, HDA_INPUT), + HDA_CODEC_VOLUME("Mic Playback Volume", 0x20, 0x01, HDA_INPUT), + HDA_CODEC_MUTE("Mic Playback Switch", 0x20, 0x01, HDA_INPUT), + HDA_CODEC_VOLUME("Beep Playback Volume", 0x20, 0x03, HDA_INPUT), + HDA_CODEC_MUTE("Beep Playback Switch", 0x20, 0x03, HDA_INPUT), + HDA_CODEC_VOLUME("Mic Boost", 0x15, 0x0, HDA_INPUT), + HDA_CODEC_VOLUME("Capture Volume", 0x0c, 0x0, HDA_OUTPUT), + HDA_CODEC_MUTE("Capture Switch", 0x0c, 0x0, HDA_OUTPUT), + { + .iface = SNDRV_CTL_ELEM_IFACE_MIXER, + .name = "Capture Source", + .info = ad198x_mux_enum_info, + .get = ad198x_mux_enum_get, + .put = ad198x_mux_enum_put, + }, + { } /* end */ +}; + +/* mute internal speaker if HP is plugged */ +static void ad1884a_hp_automute(struct hda_codec *codec) +{ + unsigned int present; + + present = snd_hda_codec_read(codec, 0x11, 0, + AC_VERB_GET_PIN_SENSE, 0) & 0x80000000; + snd_hda_codec_amp_stereo(codec, 0x16, HDA_OUTPUT, 0, + HDA_AMP_MUTE, present ? HDA_AMP_MUTE : 0); + snd_hda_codec_write(codec, 0x16, 0, AC_VERB_SET_EAPD_BTLENABLE, + present ? 0x00 : 0x02); +} + +#define AD1884A_HP_EVENT 0x37 + +/* unsolicited event for HP jack sensing */ +static void ad1884a_hp_unsol_event(struct hda_codec *codec, unsigned int res) +{ + if ((res >> 26) != AD1884A_HP_EVENT) + return; + ad1884a_hp_automute(codec); +} + +/* initialize jack-sensing, too */ +static int ad1884a_hp_init(struct hda_codec *codec) +{ + ad198x_init(codec); + ad1884a_hp_automute(codec); + return 0; +} + +/* additional verbs for laptop model */ +static struct hda_verb ad1884a_laptop_verbs[] = { + /* Port-A (HP) pin - always unmuted */ + {0x11, AC_VERB_SET_AMP_GAIN_MUTE, AMP_OUT_UNMUTE}, + /* Port-F (int speaker) mixer - route only from analog mixer */ + {0x0b, AC_VERB_SET_AMP_GAIN_MUTE, AMP_IN_MUTE(0)}, + {0x0b, AC_VERB_SET_AMP_GAIN_MUTE, AMP_IN_UNMUTE(1)}, + /* Port-F pin */ + {0x16, AC_VERB_SET_PIN_WIDGET_CONTROL, PIN_HP}, + {0x16, AC_VERB_SET_AMP_GAIN_MUTE, AMP_OUT_MUTE}, + /* analog mix */ + {0x20, AC_VERB_SET_AMP_GAIN_MUTE, AMP_IN_MUTE(4)}, + /* unsolicited event for pin-sense */ + {0x11, AC_VERB_SET_UNSOLICITED_ENABLE, AC_USRSP_EN | AD1884A_HP_EVENT}, + { } /* end */ +}; + +/* + * Thinkpad X300 + * 0x11 - HP + * 0x12 - speaker + * 0x14 - mic-in + * 0x17 - built-in mic + */ + +static struct hda_verb ad1984a_thinkpad_verbs[] = { + /* HP unmute */ + {0x11, AC_VERB_SET_AMP_GAIN_MUTE, AMP_OUT_UNMUTE}, + /* analog mix */ + {0x20, AC_VERB_SET_AMP_GAIN_MUTE, AMP_IN_MUTE(4)}, + /* turn on EAPD */ + {0x12, AC_VERB_SET_EAPD_BTLENABLE, 0x02}, + /* unsolicited event for pin-sense */ + {0x11, AC_VERB_SET_UNSOLICITED_ENABLE, AC_USRSP_EN | AD1884A_HP_EVENT}, + /* internal mic - dmic */ + {0x17, AC_VERB_SET_PIN_WIDGET_CONTROL, PIN_IN}, + { } /* end */ +}; + +static struct snd_kcontrol_new ad1984a_thinkpad_mixers[] = { + HDA_CODEC_VOLUME("Master Playback Volume", 0x21, 0x0, HDA_OUTPUT), + HDA_CODEC_MUTE("Master Playback Switch", 0x21, 0x0, HDA_OUTPUT), + HDA_CODEC_VOLUME("PCM Playback Volume", 0x20, 0x5, HDA_INPUT), + HDA_CODEC_MUTE("PCM Playback Switch", 0x20, 0x5, HDA_INPUT), + HDA_CODEC_VOLUME("Mic Playback Volume", 0x20, 0x00, HDA_INPUT), + HDA_CODEC_MUTE("Mic Playback Switch", 0x20, 0x00, HDA_INPUT), + HDA_CODEC_VOLUME("Beep Playback Volume", 0x20, 0x03, HDA_INPUT), + HDA_CODEC_MUTE("Beep Playback Switch", 0x20, 0x03, HDA_INPUT), + HDA_CODEC_VOLUME("Mic Boost", 0x14, 0x0, HDA_INPUT), + HDA_CODEC_VOLUME("Internal Mic Boost", 0x17, 0x0, HDA_INPUT), + HDA_CODEC_VOLUME("Capture Volume", 0x0c, 0x0, HDA_OUTPUT), + HDA_CODEC_MUTE("Capture Switch", 0x0c, 0x0, HDA_OUTPUT), + { + .iface = SNDRV_CTL_ELEM_IFACE_MIXER, + .name = "Capture Source", + .info = ad198x_mux_enum_info, + .get = ad198x_mux_enum_get, + .put = ad198x_mux_enum_put, + }, + { } /* end */ +}; + +static struct hda_input_mux ad1984a_thinkpad_capture_source = { + .num_items = 3, + .items = { + { "Mic", 0x0 }, + { "Internal Mic", 0x5 }, + { "Mix", 0x3 }, + }, +}; + +/* mute internal speaker if HP is plugged */ +static void ad1984a_thinkpad_automute(struct hda_codec *codec) +{ + unsigned int present; + + present = snd_hda_codec_read(codec, 0x11, 0, AC_VERB_GET_PIN_SENSE, 0) + & AC_PINSENSE_PRESENCE; + snd_hda_codec_amp_stereo(codec, 0x12, HDA_OUTPUT, 0, + HDA_AMP_MUTE, present ? HDA_AMP_MUTE : 0); +} + +/* unsolicited event for HP jack sensing */ +static void ad1984a_thinkpad_unsol_event(struct hda_codec *codec, + unsigned int res) +{ + if ((res >> 26) != AD1884A_HP_EVENT) + return; + ad1984a_thinkpad_automute(codec); +} + +/* initialize jack-sensing, too */ +static int ad1984a_thinkpad_init(struct hda_codec *codec) +{ + ad198x_init(codec); + ad1984a_thinkpad_automute(codec); + return 0; +} + +/* + */ + +enum { + AD1884A_DESKTOP, + AD1884A_LAPTOP, + AD1884A_MOBILE, + AD1884A_THINKPAD, + AD1884A_MODELS +}; + +static const char *ad1884a_models[AD1884A_MODELS] = { + [AD1884A_DESKTOP] = "desktop", + [AD1884A_LAPTOP] = "laptop", + [AD1884A_MOBILE] = "mobile", + [AD1884A_THINKPAD] = "thinkpad", +}; + +static struct snd_pci_quirk ad1884a_cfg_tbl[] = { + SND_PCI_QUIRK(0x103c, 0x3030, "HP", AD1884A_MOBILE), + SND_PCI_QUIRK(0x17aa, 0x20ac, "Thinkpad X300", AD1884A_THINKPAD), + {} +}; + +static int patch_ad1884a(struct hda_codec *codec) +{ + struct ad198x_spec *spec; + int board_config; + + spec = kzalloc(sizeof(*spec), GFP_KERNEL); + if (spec == NULL) + return -ENOMEM; + + mutex_init(&spec->amp_mutex); + codec->spec = spec; + + spec->multiout.max_channels = 2; + spec->multiout.num_dacs = ARRAY_SIZE(ad1884a_dac_nids); + spec->multiout.dac_nids = ad1884a_dac_nids; + spec->multiout.dig_out_nid = AD1884A_SPDIF_OUT; + spec->num_adc_nids = ARRAY_SIZE(ad1884a_adc_nids); + spec->adc_nids = ad1884a_adc_nids; + spec->capsrc_nids = ad1884a_capsrc_nids; + spec->input_mux = &ad1884a_capture_source; + spec->num_mixers = 1; + spec->mixers[0] = ad1884a_base_mixers; + spec->num_init_verbs = 1; + spec->init_verbs[0] = ad1884a_init_verbs; + spec->spdif_route = 0; +#ifdef CONFIG_SND_HDA_POWER_SAVE + spec->loopback.amplist = ad1884a_loopbacks; +#endif + codec->patch_ops = ad198x_patch_ops; + + /* override some parameters */ + board_config = snd_hda_check_board_config(codec, AD1884A_MODELS, + ad1884a_models, + ad1884a_cfg_tbl); + switch (board_config) { + case AD1884A_LAPTOP: + spec->mixers[0] = ad1884a_laptop_mixers; + spec->init_verbs[spec->num_init_verbs++] = ad1884a_laptop_verbs; + spec->multiout.dig_out_nid = 0; + spec->input_mux = &ad1884a_laptop_capture_source; + codec->patch_ops.unsol_event = ad1884a_hp_unsol_event; + codec->patch_ops.init = ad1884a_hp_init; + break; + case AD1884A_MOBILE: + spec->mixers[0] = ad1884a_mobile_mixers; + spec->init_verbs[spec->num_init_verbs++] = ad1884a_laptop_verbs; + spec->multiout.dig_out_nid = 0; + spec->input_mux = &ad1884a_mobile_capture_source; + codec->patch_ops.unsol_event = ad1884a_hp_unsol_event; + codec->patch_ops.init = ad1884a_hp_init; + break; + case AD1884A_THINKPAD: + spec->mixers[0] = ad1984a_thinkpad_mixers; + spec->init_verbs[spec->num_init_verbs++] = + ad1984a_thinkpad_verbs; + spec->multiout.dig_out_nid = 0; + spec->input_mux = &ad1984a_thinkpad_capture_source; + codec->patch_ops.unsol_event = ad1984a_thinkpad_unsol_event; + codec->patch_ops.init = ad1984a_thinkpad_init; + break; + } + + return 0; +} + + +/* * AD1882 * * port-A - front hp-out @@ -3654,13 +4192,19 @@ static int patch_ad1882(struct hda_codec *codec) * patch entries */ struct hda_codec_preset snd_hda_preset_analog[] = { + { .id = 0x11d4184a, .name = "AD1884A", .patch = patch_ad1884a }, { .id = 0x11d41882, .name = "AD1882", .patch = patch_ad1882 }, + { .id = 0x11d41883, .name = "AD1883", .patch = patch_ad1884a }, { .id = 0x11d41884, .name = "AD1884", .patch = patch_ad1884 }, + { .id = 0x11d4194a, .name = "AD1984A", .patch = patch_ad1884a }, + { .id = 0x11d4194b, .name = "AD1984B", .patch = patch_ad1884a }, { .id = 0x11d41981, .name = "AD1981", .patch = patch_ad1981 }, { .id = 0x11d41983, .name = "AD1983", .patch = patch_ad1983 }, { .id = 0x11d41984, .name = "AD1984", .patch = patch_ad1984 }, { .id = 0x11d41986, .name = "AD1986A", .patch = patch_ad1986a }, { .id = 0x11d41988, .name = "AD1988", .patch = patch_ad1988 }, { .id = 0x11d4198b, .name = "AD1988B", .patch = patch_ad1988 }, + { .id = 0x11d4989a, .name = "AD1989A", .patch = patch_ad1988 }, + { .id = 0x11d4989b, .name = "AD1989B", .patch = patch_ad1988 }, {} /* terminator */ }; diff --git a/sound/pci/hda/patch_atihdmi.c b/sound/pci/hda/patch_atihdmi.c index 9a8bb4c..1227250 100644 --- a/sound/pci/hda/patch_atihdmi.c +++ b/sound/pci/hda/patch_atihdmi.c @@ -27,6 +27,7 @@ #include #include "hda_codec.h" #include "hda_local.h" +#include "hda_patch.h" struct atihdmi_spec { struct hda_multi_out multiout; @@ -58,6 +59,10 @@ static int atihdmi_build_controls(struct hda_codec *codec) static int atihdmi_init(struct hda_codec *codec) { snd_hda_sequence_write(codec, atihdmi_basic_init); + /* SI codec requires to unmute the pin */ + if (get_wcaps(codec, 0x03) & AC_WCAP_OUT_AMP) + snd_hda_codec_write(codec, 0x03, 0, AC_VERB_SET_AMP_GAIN_MUTE, + AMP_OUT_UNMUTE); return 0; } @@ -112,6 +117,7 @@ static int atihdmi_build_pcms(struct hda_codec *codec) codec->pcm_info = info; info->name = "ATI HDMI"; + info->pcm_type = HDA_PCM_TYPE_HDMI; info->stream[SNDRV_PCM_STREAM_PLAYBACK] = atihdmi_pcm_digital_playback; return 0; @@ -158,5 +164,7 @@ struct hda_codec_preset snd_hda_preset_atihdmi[] = { { .id = 0x10027919, .name = "ATI RS600 HDMI", .patch = patch_atihdmi }, { .id = 0x1002791a, .name = "ATI RS690/780 HDMI", .patch = patch_atihdmi }, { .id = 0x1002aa01, .name = "ATI R6xx HDMI", .patch = patch_atihdmi }, + { .id = 0x10951392, .name = "SiI1392 HDMI", .patch = patch_atihdmi }, + { .id = 0x17e80047, .name = "Chrontel HDMI", .patch = patch_atihdmi }, {} /* terminator */ }; diff --git a/sound/pci/hda/patch_cmedia.c b/sound/pci/hda/patch_cmedia.c index 3d6097b..c73ce07 100644 --- a/sound/pci/hda/patch_cmedia.c +++ b/sound/pci/hda/patch_cmedia.c @@ -28,6 +28,7 @@ #include #include "hda_codec.h" #include "hda_local.h" +#include "hda_patch.h" #define NUM_PINS 11 @@ -329,6 +330,11 @@ static int cmi9880_build_controls(struct hda_codec *codec) err = snd_hda_create_spdif_out_ctls(codec, spec->multiout.dig_out_nid); if (err < 0) return err; + err = snd_hda_create_spdif_share_sw(codec, + &spec->multiout); + if (err < 0) + return err; + spec->multiout.share_spdif = 1; } if (spec->dig_in_nid) { err = snd_hda_create_spdif_in_ctls(codec, spec->dig_in_nid); @@ -432,7 +438,8 @@ static int cmi9880_playback_pcm_open(struct hda_pcm_stream *hinfo, struct snd_pcm_substream *substream) { struct cmi_spec *spec = codec->spec; - return snd_hda_multi_out_analog_open(codec, &spec->multiout, substream); + return snd_hda_multi_out_analog_open(codec, &spec->multiout, substream, + hinfo); } static int cmi9880_playback_pcm_prepare(struct hda_pcm_stream *hinfo, @@ -506,7 +513,7 @@ static int cmi9880_capture_pcm_cleanup(struct hda_pcm_stream *hinfo, { struct cmi_spec *spec = codec->spec; - snd_hda_codec_setup_stream(codec, spec->adc_nids[substream->number], 0, 0, 0); + snd_hda_codec_cleanup_stream(codec, spec->adc_nids[substream->number]); return 0; } @@ -571,6 +578,7 @@ static int cmi9880_build_pcms(struct hda_codec *codec) codec->num_pcms++; info++; info->name = "CMI9880 Digital"; + info->pcm_type = HDA_PCM_TYPE_SPDIF; if (spec->multiout.dig_out_nid) { info->stream[SNDRV_PCM_STREAM_PLAYBACK] = cmi9880_pcm_digital_playback; info->stream[SNDRV_PCM_STREAM_PLAYBACK].nid = spec->multiout.dig_out_nid; @@ -603,6 +611,7 @@ static const char *cmi9880_models[CMI_MODELS] = { static struct snd_pci_quirk cmi9880_cfg_tbl[] = { SND_PCI_QUIRK(0x1043, 0x813d, "ASUS P5AD2", CMI_FULL_DIG), + SND_PCI_QUIRK(0x1854, 0x0032, "LG", CMI_FULL_DIG), {} /* terminator */ }; diff --git a/sound/pci/hda/patch_conexant.c b/sound/pci/hda/patch_conexant.c index 7206b30..36fd852 100644 --- a/sound/pci/hda/patch_conexant.c +++ b/sound/pci/hda/patch_conexant.c @@ -27,6 +27,7 @@ #include #include "hda_codec.h" #include "hda_local.h" +#include "hda_patch.h" #define CXT_PIN_DIR_IN 0x00 #define CXT_PIN_DIR_OUT 0x01 @@ -98,7 +99,8 @@ static int conexant_playback_pcm_open(struct hda_pcm_stream *hinfo, struct snd_pcm_substream *substream) { struct conexant_spec *spec = codec->spec; - return snd_hda_multi_out_analog_open(codec, &spec->multiout, substream); + return snd_hda_multi_out_analog_open(codec, &spec->multiout, substream, + hinfo); } static int conexant_playback_pcm_prepare(struct hda_pcm_stream *hinfo, @@ -172,8 +174,7 @@ static int conexant_capture_pcm_cleanup(struct hda_pcm_stream *hinfo, struct snd_pcm_substream *substream) { struct conexant_spec *spec = codec->spec; - snd_hda_codec_setup_stream(codec, spec->adc_nids[substream->number], - 0, 0, 0); + snd_hda_codec_cleanup_stream(codec, spec->adc_nids[substream->number]); return 0; } @@ -241,7 +242,7 @@ static int cx5051_capture_pcm_cleanup(struct hda_pcm_stream *hinfo, struct snd_pcm_substream *substream) { struct conexant_spec *spec = codec->spec; - snd_hda_codec_setup_stream(codec, spec->cur_adc, 0, 0, 0); + snd_hda_codec_cleanup_stream(codec, spec->cur_adc); spec->cur_adc = 0; return 0; } @@ -284,6 +285,7 @@ static int conexant_build_pcms(struct hda_codec *codec) info++; codec->num_pcms++; info->name = "Conexant Digital"; + info->pcm_type = HDA_PCM_TYPE_SPDIF; info->stream[SNDRV_PCM_STREAM_PLAYBACK] = conexant_pcm_digital_playback; info->stream[SNDRV_PCM_STREAM_PLAYBACK].nid = @@ -371,6 +373,11 @@ static int conexant_build_controls(struct hda_codec *codec) spec->multiout.dig_out_nid); if (err < 0) return err; + err = snd_hda_create_spdif_share_sw(codec, + &spec->multiout); + if (err < 0) + return err; + spec->multiout.share_spdif = 1; } if (spec->dig_in_nid) { err = snd_hda_create_spdif_in_ctls(codec,spec->dig_in_nid); @@ -511,6 +518,14 @@ static struct hda_input_mux cxt5045_capture_source_benq = { } }; +static struct hda_input_mux cxt5045_capture_source_hp530 = { + .num_items = 2, + .items = { + { "ExtMic", 0x1 }, + { "IntMic", 0x2 }, + } +}; + /* turn on/off EAPD (+ mute HP) as a master switch */ static int cxt5045_hp_master_sw_put(struct snd_kcontrol *kcontrol, struct snd_ctl_elem_value *ucontrol) @@ -639,6 +654,37 @@ static struct snd_kcontrol_new cxt5045_benq_mixers[] = { {} }; +static struct snd_kcontrol_new cxt5045_mixers_hp530[] = { + { + .iface = SNDRV_CTL_ELEM_IFACE_MIXER, + .name = "Capture Source", + .info = conexant_mux_enum_info, + .get = conexant_mux_enum_get, + .put = conexant_mux_enum_put + }, + HDA_CODEC_VOLUME("Int Mic Capture Volume", 0x1a, 0x02, HDA_INPUT), + HDA_CODEC_MUTE("Int Mic Capture Switch", 0x1a, 0x02, HDA_INPUT), + HDA_CODEC_VOLUME("Ext Mic Capture Volume", 0x1a, 0x01, HDA_INPUT), + HDA_CODEC_MUTE("Ext Mic Capture Switch", 0x1a, 0x01, HDA_INPUT), + HDA_CODEC_VOLUME("PCM Playback Volume", 0x17, 0x0, HDA_INPUT), + HDA_CODEC_MUTE("PCM Playback Switch", 0x17, 0x0, HDA_INPUT), + HDA_CODEC_VOLUME("Int Mic Playback Volume", 0x17, 0x2, HDA_INPUT), + HDA_CODEC_MUTE("Int Mic Playback Switch", 0x17, 0x2, HDA_INPUT), + HDA_CODEC_VOLUME("Ext Mic Playback Volume", 0x17, 0x1, HDA_INPUT), + HDA_CODEC_MUTE("Ext Mic Playback Switch", 0x17, 0x1, HDA_INPUT), + HDA_BIND_VOL("Master Playback Volume", &cxt5045_hp_bind_master_vol), + { + .iface = SNDRV_CTL_ELEM_IFACE_MIXER, + .name = "Master Playback Switch", + .info = cxt_eapd_info, + .get = cxt_eapd_get, + .put = cxt5045_hp_master_sw_put, + .private_value = 0x10, + }, + + {} +}; + static struct hda_verb cxt5045_init_verbs[] = { /* Line in, Mic */ {0x12, AC_VERB_SET_PIN_WIDGET_CONTROL, PIN_IN }, @@ -833,6 +879,7 @@ enum { CXT5045_LAPTOP_MICSENSE, CXT5045_LAPTOP_HPMICSENSE, CXT5045_BENQ, + CXT5045_LAPTOP_HP530, #ifdef CONFIG_SND_DEBUG CXT5045_TEST, #endif @@ -844,6 +891,7 @@ static const char *cxt5045_models[CXT5045_MODELS] = { [CXT5045_LAPTOP_MICSENSE] = "laptop-micsense", [CXT5045_LAPTOP_HPMICSENSE] = "laptop-hpmicsense", [CXT5045_BENQ] = "benq", + [CXT5045_LAPTOP_HP530] = "laptop-hp530", #ifdef CONFIG_SND_DEBUG [CXT5045_TEST] = "test", #endif @@ -857,7 +905,7 @@ static struct snd_pci_quirk cxt5045_cfg_tbl[] = { SND_PCI_QUIRK(0x103c, 0x30bb, "HP DV8000", CXT5045_LAPTOP_HPSENSE), SND_PCI_QUIRK(0x103c, 0x30cd, "HP DV Series", CXT5045_LAPTOP_HPSENSE), SND_PCI_QUIRK(0x103c, 0x30cf, "HP DV9533EG", CXT5045_LAPTOP_HPSENSE), - SND_PCI_QUIRK(0x103c, 0x30d5, "HP 530", CXT5045_LAPTOP_HPSENSE), + SND_PCI_QUIRK(0x103c, 0x30d5, "HP 530", CXT5045_LAPTOP_HP530), SND_PCI_QUIRK(0x103c, 0x30d9, "HP Spartan", CXT5045_LAPTOP_HPSENSE), SND_PCI_QUIRK(0x152d, 0x0753, "Benq R55E", CXT5045_BENQ), SND_PCI_QUIRK(0x1734, 0x10ad, "Fujitsu Si1520", CXT5045_LAPTOP_MICSENSE), @@ -941,6 +989,14 @@ static int patch_cxt5045(struct hda_codec *codec) spec->num_mixers = 2; codec->patch_ops.init = cxt5045_init; break; + case CXT5045_LAPTOP_HP530: + codec->patch_ops.unsol_event = cxt5045_hp_unsol_event; + spec->input_mux = &cxt5045_capture_source_hp530; + spec->num_init_verbs = 2; + spec->init_verbs[1] = cxt5045_hp_sense_init_verbs; + spec->mixers[0] = cxt5045_mixers_hp530; + codec->patch_ops.init = cxt5045_init; + break; #ifdef CONFIG_SND_DEBUG case CXT5045_TEST: spec->input_mux = &cxt5045_test_capture_source; @@ -1537,7 +1593,7 @@ static void cxt5051_portc_automic(struct hda_codec *codec) new_adc = spec->adc_nids[spec->cur_adc_idx]; if (spec->cur_adc && spec->cur_adc != new_adc) { /* stream is running, let's swap the current ADC */ - snd_hda_codec_setup_stream(codec, spec->cur_adc, 0, 0, 0); + snd_hda_codec_cleanup_stream(codec, spec->cur_adc); spec->cur_adc = new_adc; snd_hda_codec_setup_stream(codec, new_adc, spec->cur_adc_stream_tag, 0, diff --git a/sound/pci/hda/patch_realtek.c b/sound/pci/hda/patch_realtek.c index 33282f9..732515d 100644 --- a/sound/pci/hda/patch_realtek.c +++ b/sound/pci/hda/patch_realtek.c @@ -30,6 +30,7 @@ #include #include "hda_codec.h" #include "hda_local.h" +#include "hda_patch.h" #define ALC880_FRONT_EVENT 0x01 #define ALC880_DCVOL_EVENT 0x02 @@ -97,16 +98,19 @@ enum { ALC262_SONY_ASSAMD, ALC262_BENQ_T31, ALC262_ULTRA, + ALC262_LENOVO_3000, ALC262_AUTO, ALC262_MODEL_LAST /* last tag */ }; /* ALC268 models */ enum { + ALC267_QUANTA_IL1, ALC268_3ST, ALC268_TOSHIBA, ALC268_ACER, ALC268_DELL, + ALC268_ZEPTO, #ifdef CONFIG_SND_DEBUG ALC268_TEST, #endif @@ -195,10 +199,11 @@ enum { ALC883_LENOVO_NB0763, ALC888_LENOVO_MS7195_DIG, ALC883_HAIER_W66, - ALC888_6ST_HP, ALC888_3ST_HP, ALC888_6ST_DELL, ALC883_MITAC, + ALC883_CLEVO_M720, + ALC883_FUJITSU_PI2515, ALC883_AUTO, ALC883_MODEL_LAST, }; @@ -237,6 +242,7 @@ struct alc_spec { /* capture */ unsigned int num_adc_nids; hda_nid_t *adc_nids; + hda_nid_t *capsrc_nids; hda_nid_t dig_in_nid; /* digital-in NID; optional */ /* capture source */ @@ -270,7 +276,6 @@ struct alc_spec { /* for virtual master */ hda_nid_t vmaster_nid; - u32 vmaster_tlv[4]; #ifdef CONFIG_SND_HDA_POWER_SAVE struct hda_loopback_check loopback; #endif @@ -290,6 +295,7 @@ struct alc_config_preset { hda_nid_t hp_nid; /* optional */ unsigned int num_adc_nids; hda_nid_t *adc_nids; + hda_nid_t *capsrc_nids; hda_nid_t dig_in_nid; unsigned int num_channel_mode; const struct hda_channel_mode *channel_mode; @@ -336,9 +342,10 @@ static int alc_mux_enum_put(struct snd_kcontrol *kcontrol, struct alc_spec *spec = codec->spec; unsigned int adc_idx = snd_ctl_get_ioffidx(kcontrol, &ucontrol->id); unsigned int mux_idx = adc_idx >= spec->num_mux_defs ? 0 : adc_idx; + hda_nid_t nid = spec->capsrc_nids ? + spec->capsrc_nids[adc_idx] : spec->adc_nids[adc_idx]; return snd_hda_input_mux_put(codec, &spec->input_mux[mux_idx], ucontrol, - spec->adc_nids[adc_idx], - &spec->cur_mux[adc_idx]); + nid, &spec->cur_mux[adc_idx]); } @@ -707,6 +714,7 @@ static void setup_preset(struct alc_spec *spec, spec->num_adc_nids = preset->num_adc_nids; spec->adc_nids = preset->adc_nids; + spec->capsrc_nids = preset->capsrc_nids; spec->dig_in_nid = preset->dig_in_nid; spec->unsol_event = preset->unsol_event; @@ -741,7 +749,6 @@ static struct hda_verb alc_gpio3_init_verbs[] = { static void alc_sku_automute(struct hda_codec *codec) { struct alc_spec *spec = codec->spec; - unsigned int mute; unsigned int present; unsigned int hp_nid = spec->autocfg.hp_pins[0]; unsigned int sp_nid = spec->autocfg.speaker_pins[0]; @@ -751,16 +758,8 @@ static void alc_sku_automute(struct hda_codec *codec) present = snd_hda_codec_read(codec, hp_nid, 0, AC_VERB_GET_PIN_SENSE, 0); spec->jack_present = (present & 0x80000000) != 0; - if (spec->jack_present) { - /* mute internal speaker */ - snd_hda_codec_amp_stereo(codec, sp_nid, HDA_OUTPUT, 0, - HDA_AMP_MUTE, HDA_AMP_MUTE); - } else { - /* unmute internal speaker if necessary */ - mute = snd_hda_codec_amp_read(codec, hp_nid, 0, HDA_OUTPUT, 0); - snd_hda_codec_amp_stereo(codec, sp_nid, HDA_OUTPUT, 0, - HDA_AMP_MUTE, mute); - } + snd_hda_codec_write(codec, sp_nid, 0, AC_VERB_SET_PIN_WIDGET_CONTROL, + spec->jack_present ? 0 : PIN_OUT); } /* unsolicited event for HP jack sensing */ @@ -1319,11 +1318,19 @@ static struct snd_kcontrol_new alc880_f1734_mixer[] = { HDA_BIND_MUTE("Speaker Playback Switch", 0x0d, 2, HDA_INPUT), HDA_CODEC_VOLUME("CD Playback Volume", 0x0b, 0x04, HDA_INPUT), HDA_CODEC_MUTE("CD Playback Switch", 0x0b, 0x04, HDA_INPUT), - HDA_CODEC_VOLUME("Mic Playback Volume", 0x0b, 0x0, HDA_INPUT), - HDA_CODEC_MUTE("Mic Playback Switch", 0x0b, 0x0, HDA_INPUT), + HDA_CODEC_VOLUME("Mic Playback Volume", 0x0b, 0x1, HDA_INPUT), + HDA_CODEC_MUTE("Mic Playback Switch", 0x0b, 0x1, HDA_INPUT), { } /* end */ }; +static struct hda_input_mux alc880_f1734_capture_source = { + .num_items = 2, + .items = { + { "Mic", 0x1 }, + { "CD", 0x4 }, + }, +}; + /* * ALC880 ASUS model @@ -1516,6 +1523,11 @@ static int alc_build_controls(struct hda_codec *codec) spec->multiout.dig_out_nid); if (err < 0) return err; + err = snd_hda_create_spdif_share_sw(codec, + &spec->multiout); + if (err < 0) + return err; + spec->multiout.share_spdif = 1; } if (spec->dig_in_nid) { err = snd_hda_create_spdif_in_ctls(codec, spec->dig_in_nid); @@ -1525,10 +1537,11 @@ static int alc_build_controls(struct hda_codec *codec) /* if we have no master control, let's create it */ if (!snd_hda_find_mixer_ctl(codec, "Master Playback Volume")) { + unsigned int vmaster_tlv[4]; snd_hda_set_vmaster_tlv(codec, spec->vmaster_nid, - HDA_OUTPUT, spec->vmaster_tlv); + HDA_OUTPUT, vmaster_tlv); err = snd_hda_add_vmaster(codec, "Master Playback Volume", - spec->vmaster_tlv, alc_slave_vols); + vmaster_tlv, alc_slave_vols); if (err < 0) return err; } @@ -1882,7 +1895,7 @@ static void alc880_uniwill_p53_hp_automute(struct hda_codec *codec) present = snd_hda_codec_read(codec, 0x14, 0, AC_VERB_GET_PIN_SENSE, 0) & 0x80000000; bits = present ? HDA_AMP_MUTE : 0; - snd_hda_codec_amp_stereo(codec, 0x15, HDA_INPUT, 0, HDA_AMP_MUTE, bits); + snd_hda_codec_amp_stereo(codec, 0x15, HDA_OUTPUT, 0, HDA_AMP_MUTE, bits); } static void alc880_uniwill_p53_dcvol_automute(struct hda_codec *codec) @@ -1915,6 +1928,7 @@ static void alc880_uniwill_p53_unsol_event(struct hda_codec *codec, * HP = 0x14, speaker-out = 0x15, mic = 0x18 */ static struct hda_verb alc880_pin_f1734_init_verbs[] = { + {0x07, AC_VERB_SET_CONNECT_SEL, 0x01}, {0x10, AC_VERB_SET_CONNECT_SEL, 0x02}, {0x11, AC_VERB_SET_CONNECT_SEL, 0x00}, {0x12, AC_VERB_SET_CONNECT_SEL, 0x01}, @@ -1927,7 +1941,7 @@ static struct hda_verb alc880_pin_f1734_init_verbs[] = { {0x18, AC_VERB_SET_PIN_WIDGET_CONTROL, PIN_VREF80}, {0x18, AC_VERB_SET_AMP_GAIN_MUTE, AMP_OUT_MUTE}, - {0x19, AC_VERB_SET_PIN_WIDGET_CONTROL, PIN_VREF80}, + {0x19, AC_VERB_SET_PIN_WIDGET_CONTROL, PIN_VREF50}, {0x19, AC_VERB_SET_AMP_GAIN_MUTE, AMP_OUT_MUTE}, {0x1a, AC_VERB_SET_PIN_WIDGET_CONTROL, PIN_OUT}, {0x1a, AC_VERB_SET_AMP_GAIN_MUTE, AMP_OUT_UNMUTE}, @@ -1935,6 +1949,9 @@ static struct hda_verb alc880_pin_f1734_init_verbs[] = { {0x1b, AC_VERB_SET_AMP_GAIN_MUTE, AMP_OUT_UNMUTE}, {0x1c, AC_VERB_SET_PIN_WIDGET_CONTROL, PIN_IN}, + {0x14, AC_VERB_SET_UNSOLICITED_ENABLE, AC_USRSP_EN|ALC880_HP_EVENT}, + {0x21, AC_VERB_SET_UNSOLICITED_ENABLE, AC_USRSP_EN|ALC880_DCVOL_EVENT}, + { } }; @@ -2318,7 +2335,8 @@ static int alc880_playback_pcm_open(struct hda_pcm_stream *hinfo, struct snd_pcm_substream *substream) { struct alc_spec *spec = codec->spec; - return snd_hda_multi_out_analog_open(codec, &spec->multiout, substream); + return snd_hda_multi_out_analog_open(codec, &spec->multiout, substream, + hinfo); } static int alc880_playback_pcm_prepare(struct hda_pcm_stream *hinfo, @@ -2392,8 +2410,8 @@ static int alc880_alt_capture_pcm_cleanup(struct hda_pcm_stream *hinfo, { struct alc_spec *spec = codec->spec; - snd_hda_codec_setup_stream(codec, spec->adc_nids[substream->number + 1], - 0, 0, 0); + snd_hda_codec_cleanup_stream(codec, + spec->adc_nids[substream->number + 1]); return 0; } @@ -2498,6 +2516,7 @@ static int alc_build_pcms(struct hda_codec *codec) codec->num_pcms = 2; info = spec->pcm_rec + 1; info->name = spec->stream_name_digital; + info->pcm_type = HDA_PCM_TYPE_SPDIF; if (spec->multiout.dig_out_nid && spec->stream_digital_playback) { info->stream[SNDRV_PCM_STREAM_PLAYBACK] = *(spec->stream_digital_playback); @@ -2560,6 +2579,7 @@ static void alc_free(struct hda_codec *codec) kfree(spec->kctl_alloc); } kfree(spec); + codec->spec = NULL; /* to be sure */ } /* @@ -3057,7 +3077,9 @@ static struct alc_config_preset alc880_presets[] = { .hp_nid = 0x02, .num_channel_mode = ARRAY_SIZE(alc880_2_jack_modes), .channel_mode = alc880_2_jack_modes, - .input_mux = &alc880_capture_source, + .input_mux = &alc880_f1734_capture_source, + .unsol_event = alc880_uniwill_p53_unsol_event, + .init_hook = alc880_uniwill_p53_hp_automute, }, [ALC880_ASUS] = { .mixers = { alc880_asus_mixer }, @@ -3467,15 +3489,21 @@ static int alc880_auto_create_analog_input_ctls(struct alc_spec *spec, return 0; } -static void alc880_auto_set_output_and_unmute(struct hda_codec *codec, - hda_nid_t nid, int pin_type, - int dac_idx) +static void alc_set_pin_output(struct hda_codec *codec, hda_nid_t nid, + unsigned int pin_type) { - /* set as output */ snd_hda_codec_write(codec, nid, 0, AC_VERB_SET_PIN_WIDGET_CONTROL, pin_type); + /* unmute pin */ snd_hda_codec_write(codec, nid, 0, AC_VERB_SET_AMP_GAIN_MUTE, AMP_OUT_UNMUTE); +} + +static void alc880_auto_set_output_and_unmute(struct hda_codec *codec, + hda_nid_t nid, int pin_type, + int dac_idx) +{ + alc_set_pin_output(codec, nid, pin_type); /* need the manual connection? */ if (alc880_is_multi_pin(nid)) { struct alc_spec *spec = codec->spec; @@ -3597,9 +3625,12 @@ static int alc880_parse_auto_config(struct hda_codec *codec) /* additional initialization for auto-configuration model */ static void alc880_auto_init(struct hda_codec *codec) { + struct alc_spec *spec = codec->spec; alc880_auto_init_multi_out(codec); alc880_auto_init_extra_out(codec); alc880_auto_init_analog_input(codec); + if (spec->unsol_event) + alc_sku_automute(codec); } /* @@ -4795,11 +4826,7 @@ static void alc260_auto_set_output_and_unmute(struct hda_codec *codec, hda_nid_t nid, int pin_type, int sel_idx) { - /* set as output */ - snd_hda_codec_write(codec, nid, 0, AC_VERB_SET_PIN_WIDGET_CONTROL, - pin_type); - snd_hda_codec_write(codec, nid, 0, AC_VERB_SET_AMP_GAIN_MUTE, - AMP_OUT_UNMUTE); + alc_set_pin_output(codec, nid, pin_type); /* need the manual connection? */ if (nid >= 0x12) { int idx = nid - 0x12; @@ -4929,7 +4956,7 @@ static int alc260_parse_auto_config(struct hda_codec *codec) /* check whether NID 0x04 is valid */ wcap = get_wcaps(codec, 0x04); wcap = (wcap & AC_WCAP_TYPE) >> AC_WCAP_TYPE_SHIFT; /* get type */ - if (wcap != AC_WID_AUD_IN) { + if (wcap != AC_WID_AUD_IN || spec->input_mux->num_items == 1) { spec->adc_nids = alc260_adc_nids_alt; spec->num_adc_nids = ARRAY_SIZE(alc260_adc_nids_alt); spec->mixers[spec->num_mixers] = alc260_capture_alt_mixer; @@ -4946,8 +4973,11 @@ static int alc260_parse_auto_config(struct hda_codec *codec) /* additional initialization for auto-configuration model */ static void alc260_auto_init(struct hda_codec *codec) { + struct alc_spec *spec = codec->spec; alc260_auto_init_multi_out(codec); alc260_auto_init_analog_input(codec); + if (spec->unsol_event) + alc_sku_automute(codec); } #ifdef CONFIG_SND_HDA_POWER_SAVE @@ -5204,6 +5234,9 @@ static hda_nid_t alc882_dac_nids[4] = { #define alc882_adc_nids alc880_adc_nids #define alc882_adc_nids_alt alc880_adc_nids_alt +static hda_nid_t alc882_capsrc_nids[3] = { 0x24, 0x23, 0x22 }; +static hda_nid_t alc882_capsrc_nids_alt[2] = { 0x23, 0x22 }; + /* input MUX */ /* FIXME: should be a matrix-type input source selection */ @@ -5226,15 +5259,11 @@ static int alc882_mux_enum_put(struct snd_kcontrol *kcontrol, struct alc_spec *spec = codec->spec; const struct hda_input_mux *imux = spec->input_mux; unsigned int adc_idx = snd_ctl_get_ioffidx(kcontrol, &ucontrol->id); - static hda_nid_t capture_mixers[3] = { 0x24, 0x23, 0x22 }; - hda_nid_t nid; + hda_nid_t nid = spec->capsrc_nids ? + spec->capsrc_nids[adc_idx] : spec->adc_nids[adc_idx]; unsigned int *cur_val = &spec->cur_mux[adc_idx]; unsigned int i, idx; - if (spec->num_adc_nids < 3) - nid = capture_mixers[adc_idx + 1]; - else - nid = capture_mixers[adc_idx]; idx = ucontrol->value.enumerated.item[0]; if (idx >= imux->num_items) idx = imux->num_items - 1; @@ -6111,6 +6140,7 @@ static struct alc_config_preset alc882_presets[] = { .dig_out_nid = ALC882_DIGOUT_NID, .num_adc_nids = ARRAY_SIZE(alc882_adc_nids), .adc_nids = alc882_adc_nids, + .capsrc_nids = alc882_capsrc_nids, .num_channel_mode = ARRAY_SIZE(alc882_3ST_6ch_modes), .channel_mode = alc882_3ST_6ch_modes, .need_dac_fix = 1, @@ -6127,6 +6157,7 @@ static struct alc_config_preset alc882_presets[] = { .dig_out_nid = ALC882_DIGOUT_NID, .num_adc_nids = ARRAY_SIZE(alc882_adc_nids), .adc_nids = alc882_adc_nids, + .capsrc_nids = alc882_capsrc_nids, .num_channel_mode = ARRAY_SIZE(alc882_3ST_6ch_modes), .channel_mode = alc882_3ST_6ch_modes, .need_dac_fix = 1, @@ -6182,15 +6213,11 @@ static void alc882_auto_set_output_and_unmute(struct hda_codec *codec, struct alc_spec *spec = codec->spec; int idx; + alc_set_pin_output(codec, nid, pin_type); if (spec->multiout.dac_nids[dac_idx] == 0x25) idx = 4; else idx = spec->multiout.dac_nids[dac_idx] - 2; - - snd_hda_codec_write(codec, nid, 0, AC_VERB_SET_PIN_WIDGET_CONTROL, - pin_type); - snd_hda_codec_write(codec, nid, 0, AC_VERB_SET_AMP_GAIN_MUTE, - AMP_OUT_UNMUTE); snd_hda_codec_write(codec, nid, 0, AC_VERB_SET_CONNECT_SEL, idx); } @@ -6219,6 +6246,9 @@ static void alc882_auto_init_hp_out(struct hda_codec *codec) if (pin) /* connect to front */ /* use dac 0 */ alc882_auto_set_output_and_unmute(codec, pin, PIN_HP, 0); + pin = spec->autocfg.speaker_pins[0]; + if (pin) + alc882_auto_set_output_and_unmute(codec, pin, PIN_OUT, 0); } #define alc882_is_input_pin(nid) alc880_is_input_pin(nid) @@ -6231,16 +6261,21 @@ static void alc882_auto_init_analog_input(struct hda_codec *codec) for (i = 0; i < AUTO_PIN_LAST; i++) { hda_nid_t nid = spec->autocfg.input_pins[i]; - if (alc882_is_input_pin(nid)) { - snd_hda_codec_write(codec, nid, 0, - AC_VERB_SET_PIN_WIDGET_CONTROL, - i <= AUTO_PIN_FRONT_MIC ? - PIN_VREF80 : PIN_IN); - if (nid != ALC882_PIN_CD_NID) - snd_hda_codec_write(codec, nid, 0, - AC_VERB_SET_AMP_GAIN_MUTE, - AMP_OUT_MUTE); + unsigned int vref; + if (!nid) + continue; + vref = PIN_IN; + if (1 /*i <= AUTO_PIN_FRONT_MIC*/) { + if (snd_hda_param_read(codec, nid, AC_PAR_PIN_CAP) & + AC_PINCAP_VREF_80) + vref = PIN_VREF80; } + snd_hda_codec_write(codec, nid, 0, + AC_VERB_SET_PIN_WIDGET_CONTROL, vref); + if (get_wcaps(codec, nid) & AC_WCAP_OUT_AMP) + snd_hda_codec_write(codec, nid, 0, + AC_VERB_SET_AMP_GAIN_MUTE, + AMP_OUT_MUTE); } } @@ -6294,11 +6329,16 @@ static int alc882_parse_auto_config(struct hda_codec *codec) /* additional initialization for auto-configuration model */ static void alc882_auto_init(struct hda_codec *codec) { + struct alc_spec *spec = codec->spec; alc882_auto_init_multi_out(codec); alc882_auto_init_hp_out(codec); alc882_auto_init_analog_input(codec); + if (spec->unsol_event) + alc_sku_automute(codec); } +static int patch_alc883(struct hda_codec *codec); /* called in patch_alc882() */ + static int patch_alc882(struct hda_codec *codec) { struct alc_spec *spec; @@ -6328,6 +6368,11 @@ static int patch_alc882(struct hda_codec *codec) board_config = ALC885_MBP3; break; default: + /* ALC889A is handled better as ALC888-compatible */ + if (codec->revision_id == 0x100103) { + alc_free(codec); + return patch_alc883(codec); + } printk(KERN_INFO "hda_codec: Unknown model for ALC882, " "trying auto-probe from BIOS...\n"); board_config = ALC882_AUTO; @@ -6372,12 +6417,14 @@ static int patch_alc882(struct hda_codec *codec) if (wcap != AC_WID_AUD_IN) { spec->adc_nids = alc882_adc_nids_alt; spec->num_adc_nids = ARRAY_SIZE(alc882_adc_nids_alt); + spec->capsrc_nids = alc882_capsrc_nids_alt; spec->mixers[spec->num_mixers] = alc882_capture_alt_mixer; spec->num_mixers++; } else { spec->adc_nids = alc882_adc_nids; spec->num_adc_nids = ARRAY_SIZE(alc882_adc_nids); + spec->capsrc_nids = alc882_capsrc_nids; spec->mixers[spec->num_mixers] = alc882_capture_mixer; spec->num_mixers++; } @@ -6412,7 +6459,7 @@ static int patch_alc882(struct hda_codec *codec) static hda_nid_t alc883_dac_nids[4] = { /* front, rear, clfe, rear_surr */ - 0x02, 0x04, 0x03, 0x05 + 0x02, 0x03, 0x04, 0x05 }; static hda_nid_t alc883_adc_nids[2] = { @@ -6420,6 +6467,8 @@ static hda_nid_t alc883_adc_nids[2] = { 0x08, 0x09, }; +static hda_nid_t alc883_capsrc_nids[2] = { 0x23, 0x22 }; + /* input MUX */ /* FIXME: should be a matrix-type input source selection */ @@ -6451,35 +6500,18 @@ static struct hda_input_mux alc883_lenovo_nb0763_capture_source = { }, }; +static struct hda_input_mux alc883_fujitsu_pi2515_capture_source = { + .num_items = 2, + .items = { + { "Mic", 0x0 }, + { "Int Mic", 0x1 }, + }, +}; + #define alc883_mux_enum_info alc_mux_enum_info #define alc883_mux_enum_get alc_mux_enum_get - -static int alc883_mux_enum_put(struct snd_kcontrol *kcontrol, - struct snd_ctl_elem_value *ucontrol) -{ - struct hda_codec *codec = snd_kcontrol_chip(kcontrol); - struct alc_spec *spec = codec->spec; - const struct hda_input_mux *imux = spec->input_mux; - unsigned int adc_idx = snd_ctl_get_ioffidx(kcontrol, &ucontrol->id); - static hda_nid_t capture_mixers[2] = { 0x23, 0x22 }; - hda_nid_t nid = capture_mixers[adc_idx]; - unsigned int *cur_val = &spec->cur_mux[adc_idx]; - unsigned int i, idx; - - idx = ucontrol->value.enumerated.item[0]; - if (idx >= imux->num_items) - idx = imux->num_items - 1; - if (*cur_val == idx) - return 0; - for (i = 0; i < imux->num_items; i++) { - unsigned int v = (i == idx) ? 0 : HDA_AMP_MUTE; - snd_hda_codec_amp_stereo(codec, nid, HDA_INPUT, - imux->items[i].index, - HDA_AMP_MUTE, v); - } - *cur_val = idx; - return 1; -} +/* ALC883 has the ALC882-type input selection */ +#define alc883_mux_enum_put alc882_mux_enum_put /* * 2ch mode @@ -6638,6 +6670,60 @@ static struct snd_kcontrol_new alc883_mitac_mixer[] = { { } /* end */ }; +static struct snd_kcontrol_new alc883_clevo_m720_mixer[] = { + HDA_CODEC_VOLUME("Headphone Playback Volume", 0x0c, 0x0, HDA_OUTPUT), + HDA_BIND_MUTE("Headphone Playback Switch", 0x0c, 2, HDA_INPUT), + HDA_CODEC_VOLUME("Speaker Playback Volume", 0x0d, 0x0, HDA_OUTPUT), + HDA_BIND_MUTE("Speaker Playback Switch", 0x0d, 2, HDA_INPUT), + HDA_CODEC_VOLUME("Mic Playback Volume", 0x0b, 0x0, HDA_INPUT), + HDA_CODEC_VOLUME("Mic Boost", 0x18, 0, HDA_INPUT), + HDA_CODEC_MUTE("Mic Playback Switch", 0x0b, 0x0, HDA_INPUT), + HDA_CODEC_VOLUME("Int Mic Playback Volume", 0x0b, 0x1, HDA_INPUT), + HDA_CODEC_VOLUME("Int Mic Boost", 0x19, 0, HDA_INPUT), + HDA_CODEC_MUTE("Int Mic Playback Switch", 0x0b, 0x1, HDA_INPUT), + HDA_CODEC_VOLUME("Capture Volume", 0x08, 0x0, HDA_INPUT), + HDA_CODEC_MUTE("Capture Switch", 0x08, 0x0, HDA_INPUT), + HDA_CODEC_VOLUME_IDX("Capture Volume", 1, 0x09, 0x0, HDA_INPUT), + HDA_CODEC_MUTE_IDX("Capture Switch", 1, 0x09, 0x0, HDA_INPUT), + { + .iface = SNDRV_CTL_ELEM_IFACE_MIXER, + /* .name = "Capture Source", */ + .name = "Input Source", + .count = 2, + .info = alc883_mux_enum_info, + .get = alc883_mux_enum_get, + .put = alc883_mux_enum_put, + }, + { } /* end */ +}; + +static struct snd_kcontrol_new alc883_2ch_fujitsu_pi2515_mixer[] = { + HDA_CODEC_VOLUME("Headphone Playback Volume", 0x0c, 0x0, HDA_OUTPUT), + HDA_BIND_MUTE("Headphone Playback Switch", 0x0c, 2, HDA_INPUT), + HDA_CODEC_VOLUME("Speaker Playback Volume", 0x0d, 0x0, HDA_OUTPUT), + HDA_BIND_MUTE("Speaker Playback Switch", 0x0d, 2, HDA_INPUT), + HDA_CODEC_VOLUME("Mic Playback Volume", 0x0b, 0x0, HDA_INPUT), + HDA_CODEC_VOLUME("Mic Boost", 0x18, 0, HDA_INPUT), + HDA_CODEC_MUTE("Mic Playback Switch", 0x0b, 0x0, HDA_INPUT), + HDA_CODEC_VOLUME("Int Mic Playback Volume", 0x0b, 0x1, HDA_INPUT), + HDA_CODEC_VOLUME("Int Mic Boost", 0x19, 0, HDA_INPUT), + HDA_CODEC_MUTE("Int Mic Playback Switch", 0x0b, 0x1, HDA_INPUT), + HDA_CODEC_VOLUME("Capture Volume", 0x08, 0x0, HDA_INPUT), + HDA_CODEC_MUTE("Capture Switch", 0x08, 0x0, HDA_INPUT), + HDA_CODEC_VOLUME_IDX("Capture Volume", 1, 0x09, 0x0, HDA_INPUT), + HDA_CODEC_MUTE_IDX("Capture Switch", 1, 0x09, 0x0, HDA_INPUT), + { + .iface = SNDRV_CTL_ELEM_IFACE_MIXER, + /* .name = "Capture Source", */ + .name = "Input Source", + .count = 2, + .info = alc883_mux_enum_info, + .get = alc883_mux_enum_get, + .put = alc883_mux_enum_put, + }, + { } /* end */ +}; + static struct snd_kcontrol_new alc883_3ST_2ch_mixer[] = { HDA_CODEC_VOLUME("Front Playback Volume", 0x0c, 0x0, HDA_OUTPUT), HDA_BIND_MUTE("Front Playback Switch", 0x0c, 2, HDA_INPUT), @@ -6787,6 +6873,9 @@ static struct snd_kcontrol_new alc883_tagra_2ch_mixer[] = { HDA_CODEC_VOLUME("Mic Playback Volume", 0x0b, 0x0, HDA_INPUT), HDA_CODEC_VOLUME("Mic Boost", 0x18, 0, HDA_INPUT), HDA_CODEC_MUTE("Mic Playback Switch", 0x0b, 0x0, HDA_INPUT), + HDA_CODEC_VOLUME("Int Mic Playback Volume", 0x0b, 0x1, HDA_INPUT), + HDA_CODEC_VOLUME("Int Mic Boost", 0x19, 0, HDA_INPUT), + HDA_CODEC_MUTE("Int Mic Playback Switch", 0x0b, 0x1, HDA_INPUT), HDA_CODEC_VOLUME("Capture Volume", 0x08, 0x0, HDA_INPUT), HDA_CODEC_MUTE("Capture Switch", 0x08, 0x0, HDA_INPUT), HDA_CODEC_VOLUME_IDX("Capture Volume", 1, 0x09, 0x0, HDA_INPUT), @@ -6878,124 +6967,6 @@ static struct snd_kcontrol_new alc883_medion_md2_mixer[] = { { } /* end */ }; -static struct snd_kcontrol_new alc888_6st_hp_mixer[] = { - HDA_CODEC_VOLUME("Front Playback Volume", 0x0c, 0x0, HDA_OUTPUT), - HDA_BIND_MUTE("Front Playback Switch", 0x0c, 2, HDA_INPUT), - HDA_CODEC_VOLUME("Surround Playback Volume", 0x0e, 0x0, HDA_OUTPUT), - HDA_BIND_MUTE("Surround Playback Switch", 0x0e, 2, HDA_INPUT), - HDA_CODEC_VOLUME_MONO("Center Playback Volume", 0x0d, 1, 0x0, HDA_OUTPUT), - HDA_CODEC_VOLUME_MONO("LFE Playback Volume", 0x0d, 2, 0x0, HDA_OUTPUT), - HDA_BIND_MUTE_MONO("Center Playback Switch", 0x0d, 1, 2, HDA_INPUT), - HDA_BIND_MUTE_MONO("LFE Playback Switch", 0x0d, 2, 2, HDA_INPUT), - HDA_CODEC_VOLUME("Side Playback Volume", 0x0f, 0x0, HDA_OUTPUT), - HDA_BIND_MUTE("Side Playback Switch", 0x0f, 2, HDA_INPUT), - HDA_CODEC_MUTE("Headphone Playback Switch", 0x1b, 0x0, HDA_OUTPUT), - HDA_CODEC_VOLUME("CD Playback Volume", 0x0b, 0x04, HDA_INPUT), - HDA_CODEC_MUTE("CD Playback Switch", 0x0b, 0x04, HDA_INPUT), - HDA_CODEC_VOLUME("Line Playback Volume", 0x0b, 0x02, HDA_INPUT), - HDA_CODEC_MUTE("Line Playback Switch", 0x0b, 0x02, HDA_INPUT), - HDA_CODEC_VOLUME("Mic Playback Volume", 0x0b, 0x0, HDA_INPUT), - HDA_CODEC_VOLUME("Mic Boost", 0x18, 0, HDA_INPUT), - HDA_CODEC_MUTE("Mic Playback Switch", 0x0b, 0x0, HDA_INPUT), - HDA_CODEC_VOLUME("Front Mic Playback Volume", 0x0b, 0x1, HDA_INPUT), - HDA_CODEC_VOLUME("Front Mic Boost", 0x19, 0, HDA_INPUT), - HDA_CODEC_MUTE("Front Mic Playback Switch", 0x0b, 0x1, HDA_INPUT), - HDA_CODEC_VOLUME("PC Speaker Playback Volume", 0x0b, 0x05, HDA_INPUT), - HDA_CODEC_MUTE("PC Speaker Playback Switch", 0x0b, 0x05, HDA_INPUT), - HDA_CODEC_VOLUME("Capture Volume", 0x08, 0x0, HDA_INPUT), - HDA_CODEC_MUTE("Capture Switch", 0x08, 0x0, HDA_INPUT), - HDA_CODEC_VOLUME_IDX("Capture Volume", 1, 0x09, 0x0, HDA_INPUT), - HDA_CODEC_MUTE_IDX("Capture Switch", 1, 0x09, 0x0, HDA_INPUT), - { - .iface = SNDRV_CTL_ELEM_IFACE_MIXER, - /* .name = "Capture Source", */ - .name = "Input Source", - .count = 2, - .info = alc883_mux_enum_info, - .get = alc883_mux_enum_get, - .put = alc883_mux_enum_put, - }, - { } /* end */ -}; - -static struct snd_kcontrol_new alc888_3st_hp_mixer[] = { - HDA_CODEC_VOLUME("Front Playback Volume", 0x0c, 0x0, HDA_OUTPUT), - HDA_BIND_MUTE("Front Playback Switch", 0x0c, 2, HDA_INPUT), - HDA_CODEC_VOLUME("Surround Playback Volume", 0x0e, 0x0, HDA_OUTPUT), - HDA_BIND_MUTE("Surround Playback Switch", 0x0e, 2, HDA_INPUT), - HDA_CODEC_VOLUME_MONO("Center Playback Volume", 0x0d, 1, 0x0, HDA_OUTPUT), - HDA_CODEC_VOLUME_MONO("LFE Playback Volume", 0x0d, 2, 0x0, HDA_OUTPUT), - HDA_BIND_MUTE_MONO("Center Playback Switch", 0x0d, 1, 2, HDA_INPUT), - HDA_BIND_MUTE_MONO("LFE Playback Switch", 0x0d, 2, 2, HDA_INPUT), - HDA_CODEC_MUTE("Headphone Playback Switch", 0x1b, 0x0, HDA_OUTPUT), - HDA_CODEC_VOLUME("CD Playback Volume", 0x0b, 0x04, HDA_INPUT), - HDA_CODEC_MUTE("CD Playback Switch", 0x0b, 0x04, HDA_INPUT), - HDA_CODEC_VOLUME("Line Playback Volume", 0x0b, 0x02, HDA_INPUT), - HDA_CODEC_MUTE("Line Playback Switch", 0x0b, 0x02, HDA_INPUT), - HDA_CODEC_VOLUME("Mic Playback Volume", 0x0b, 0x0, HDA_INPUT), - HDA_CODEC_VOLUME("Mic Boost", 0x18, 0, HDA_INPUT), - HDA_CODEC_MUTE("Mic Playback Switch", 0x0b, 0x0, HDA_INPUT), - HDA_CODEC_VOLUME("Front Mic Playback Volume", 0x0b, 0x1, HDA_INPUT), - HDA_CODEC_VOLUME("Front Mic Boost", 0x19, 0, HDA_INPUT), - HDA_CODEC_MUTE("Front Mic Playback Switch", 0x0b, 0x1, HDA_INPUT), - HDA_CODEC_VOLUME("PC Speaker Playback Volume", 0x0b, 0x05, HDA_INPUT), - HDA_CODEC_MUTE("PC Speaker Playback Switch", 0x0b, 0x05, HDA_INPUT), - HDA_CODEC_VOLUME("Capture Volume", 0x08, 0x0, HDA_INPUT), - HDA_CODEC_MUTE("Capture Switch", 0x08, 0x0, HDA_INPUT), - HDA_CODEC_VOLUME_IDX("Capture Volume", 1, 0x09, 0x0, HDA_INPUT), - HDA_CODEC_MUTE_IDX("Capture Switch", 1, 0x09, 0x0, HDA_INPUT), - { - .iface = SNDRV_CTL_ELEM_IFACE_MIXER, - /* .name = "Capture Source", */ - .name = "Input Source", - .count = 2, - .info = alc883_mux_enum_info, - .get = alc883_mux_enum_get, - .put = alc883_mux_enum_put, - }, - { } /* end */ -}; - -static struct snd_kcontrol_new alc888_6st_dell_mixer[] = { - HDA_CODEC_VOLUME("Front Playback Volume", 0x0c, 0x0, HDA_OUTPUT), - HDA_BIND_MUTE("Front Playback Switch", 0x0c, 2, HDA_INPUT), - HDA_CODEC_VOLUME("Surround Playback Volume", 0x0e, 0x0, HDA_OUTPUT), - HDA_BIND_MUTE("Surround Playback Switch", 0x0e, 2, HDA_INPUT), - HDA_CODEC_VOLUME_MONO("Center Playback Volume", 0x0d, 1, 0x0, HDA_OUTPUT), - HDA_CODEC_VOLUME_MONO("LFE Playback Volume", 0x0d, 2, 0x0, HDA_OUTPUT), - HDA_BIND_MUTE_MONO("Center Playback Switch", 0x0d, 1, 2, HDA_INPUT), - HDA_BIND_MUTE_MONO("LFE Playback Switch", 0x0d, 2, 2, HDA_INPUT), - HDA_CODEC_VOLUME("Side Playback Volume", 0x0f, 0x0, HDA_OUTPUT), - HDA_BIND_MUTE("Side Playback Switch", 0x0f, 2, HDA_INPUT), - HDA_CODEC_MUTE("Headphone Playback Switch", 0x1b, 0x0, HDA_OUTPUT), - HDA_CODEC_VOLUME("CD Playback Volume", 0x0b, 0x04, HDA_INPUT), - HDA_CODEC_MUTE("CD Playback Switch", 0x0b, 0x04, HDA_INPUT), - HDA_CODEC_VOLUME("Line Playback Volume", 0x0b, 0x02, HDA_INPUT), - HDA_CODEC_MUTE("Line Playback Switch", 0x0b, 0x02, HDA_INPUT), - HDA_CODEC_VOLUME("Mic Playback Volume", 0x0b, 0x0, HDA_INPUT), - HDA_CODEC_VOLUME("Mic Boost", 0x18, 0, HDA_INPUT), - HDA_CODEC_MUTE("Mic Playback Switch", 0x0b, 0x0, HDA_INPUT), - HDA_CODEC_VOLUME("Front Mic Playback Volume", 0x0b, 0x1, HDA_INPUT), - HDA_CODEC_VOLUME("Front Mic Boost", 0x19, 0, HDA_INPUT), - HDA_CODEC_MUTE("Front Mic Playback Switch", 0x0b, 0x1, HDA_INPUT), - HDA_CODEC_VOLUME("PC Speaker Playback Volume", 0x0b, 0x05, HDA_INPUT), - HDA_CODEC_MUTE("PC Speaker Playback Switch", 0x0b, 0x05, HDA_INPUT), - HDA_CODEC_VOLUME("Capture Volume", 0x08, 0x0, HDA_INPUT), - HDA_CODEC_MUTE("Capture Switch", 0x08, 0x0, HDA_INPUT), - HDA_CODEC_VOLUME_IDX("Capture Volume", 1, 0x09, 0x0, HDA_INPUT), - HDA_CODEC_MUTE_IDX("Capture Switch", 1, 0x09, 0x0, HDA_INPUT), - { - .iface = SNDRV_CTL_ELEM_IFACE_MIXER, - /* .name = "Capture Source", */ - .name = "Input Source", - .count = 2, - .info = alc883_mux_enum_info, - .get = alc883_mux_enum_get, - .put = alc883_mux_enum_put, - }, - { } /* end */ -}; - static struct snd_kcontrol_new alc883_acer_aspire_mixer[] = { HDA_CODEC_VOLUME("Front Playback Volume", 0x0c, 0x0, HDA_OUTPUT), HDA_BIND_MUTE("Front Playback Switch", 0x0c, 2, HDA_INPUT), @@ -7171,6 +7142,35 @@ static struct hda_verb alc883_mitac_verbs[] = { { } /* end */ }; +static struct hda_verb alc883_clevo_m720_verbs[] = { + /* HP */ + {0x15, AC_VERB_SET_CONNECT_SEL, 0x00}, + {0x15, AC_VERB_SET_PIN_WIDGET_CONTROL, PIN_HP}, + /* Int speaker */ + {0x14, AC_VERB_SET_CONNECT_SEL, 0x01}, + {0x14, AC_VERB_SET_PIN_WIDGET_CONTROL, PIN_OUT}, + + /* enable unsolicited event */ + {0x15, AC_VERB_SET_UNSOLICITED_ENABLE, ALC880_HP_EVENT | AC_USRSP_EN}, + {0x18, AC_VERB_SET_UNSOLICITED_ENABLE, ALC880_MIC_EVENT | AC_USRSP_EN}, + + { } /* end */ +}; + +static struct hda_verb alc883_2ch_fujitsu_pi2515_verbs[] = { + /* HP */ + {0x14, AC_VERB_SET_CONNECT_SEL, 0x00}, + {0x14, AC_VERB_SET_PIN_WIDGET_CONTROL, PIN_HP}, + /* Subwoofer */ + {0x15, AC_VERB_SET_CONNECT_SEL, 0x01}, + {0x15, AC_VERB_SET_PIN_WIDGET_CONTROL, PIN_OUT}, + + /* enable unsolicited event */ + {0x14, AC_VERB_SET_UNSOLICITED_ENABLE, ALC880_HP_EVENT | AC_USRSP_EN}, + + { } /* end */ +}; + static struct hda_verb alc883_tagra_verbs[] = { {0x0c, AC_VERB_SET_AMP_GAIN_MUTE, AMP_IN_UNMUTE(0)}, {0x0c, AC_VERB_SET_AMP_GAIN_MUTE, AMP_IN_UNMUTE(1)}, @@ -7227,26 +7227,14 @@ static struct hda_verb alc883_haier_w66_verbs[] = { { } /* end */ }; -static struct hda_verb alc888_6st_hp_verbs[] = { - {0x14, AC_VERB_SET_CONNECT_SEL, 0x00}, /* Front: output 0 (0x0c) */ - {0x15, AC_VERB_SET_CONNECT_SEL, 0x02}, /* Rear : output 2 (0x0e) */ - {0x16, AC_VERB_SET_CONNECT_SEL, 0x01}, /* CLFE : output 1 (0x0d) */ - {0x17, AC_VERB_SET_CONNECT_SEL, 0x03}, /* Side : output 3 (0x0f) */ - { } -}; - static struct hda_verb alc888_3st_hp_verbs[] = { {0x14, AC_VERB_SET_CONNECT_SEL, 0x00}, /* Front: output 0 (0x0c) */ - {0x18, AC_VERB_SET_CONNECT_SEL, 0x01}, /* Rear : output 1 (0x0d) */ - {0x16, AC_VERB_SET_CONNECT_SEL, 0x02}, /* CLFE : output 2 (0x0e) */ + {0x16, AC_VERB_SET_CONNECT_SEL, 0x01}, /* Rear : output 1 (0x0d) */ + {0x18, AC_VERB_SET_CONNECT_SEL, 0x02}, /* CLFE : output 2 (0x0e) */ { } }; static struct hda_verb alc888_6st_dell_verbs[] = { - {0x14, AC_VERB_SET_CONNECT_SEL, 0x00}, /* Front: output 0 (0x0c) */ - {0x15, AC_VERB_SET_CONNECT_SEL, 0x02}, /* Rear : output 1 (0x0e) */ - {0x16, AC_VERB_SET_CONNECT_SEL, 0x01}, /* CLFE : output 2 (0x0d) */ - {0x17, AC_VERB_SET_CONNECT_SEL, 0x03}, /* Side : output 3 (0x0f) */ {0x1b, AC_VERB_SET_UNSOLICITED_ENABLE, ALC880_HP_EVENT | AC_USRSP_EN}, { } }; @@ -7354,6 +7342,68 @@ static void alc883_tagra_unsol_event(struct hda_codec *codec, unsigned int res) alc883_tagra_automute(codec); } +/* toggle speaker-output according to the hp-jack state */ +static void alc883_clevo_m720_hp_automute(struct hda_codec *codec) +{ + unsigned int present; + unsigned char bits; + + present = snd_hda_codec_read(codec, 0x15, 0, AC_VERB_GET_PIN_SENSE, 0) + & AC_PINSENSE_PRESENCE; + bits = present ? HDA_AMP_MUTE : 0; + snd_hda_codec_amp_stereo(codec, 0x14, HDA_OUTPUT, 0, + HDA_AMP_MUTE, bits); +} + +static void alc883_clevo_m720_mic_automute(struct hda_codec *codec) +{ + unsigned int present; + + present = snd_hda_codec_read(codec, 0x18, 0, + AC_VERB_GET_PIN_SENSE, 0) & 0x80000000; + snd_hda_codec_amp_stereo(codec, 0x0b, HDA_INPUT, 1, + HDA_AMP_MUTE, present ? HDA_AMP_MUTE : 0); +} + +static void alc883_clevo_m720_automute(struct hda_codec *codec) +{ + alc883_clevo_m720_hp_automute(codec); + alc883_clevo_m720_mic_automute(codec); +} + +static void alc883_clevo_m720_unsol_event(struct hda_codec *codec, + unsigned int res) +{ + switch (res >> 26) { + case ALC880_HP_EVENT: + alc883_clevo_m720_hp_automute(codec); + break; + case ALC880_MIC_EVENT: + alc883_clevo_m720_mic_automute(codec); + break; + } +} + +/* toggle speaker-output according to the hp-jack state */ +static void alc883_2ch_fujitsu_pi2515_automute(struct hda_codec *codec) +{ + unsigned int present; + unsigned char bits; + + present = snd_hda_codec_read(codec, 0x14, 0, AC_VERB_GET_PIN_SENSE, 0) + & AC_PINSENSE_PRESENCE; + bits = present ? HDA_AMP_MUTE : 0; + snd_hda_codec_amp_stereo(codec, 0x15, HDA_OUTPUT, 0, + HDA_AMP_MUTE, bits); +} + +static void alc883_2ch_fujitsu_pi2515_unsol_event(struct hda_codec *codec, + unsigned int res) +{ + if ((res >> 26) == ALC880_HP_EVENT) + alc883_2ch_fujitsu_pi2515_automute(codec); +} + static void alc883_haier_w66_automute(struct hda_codec *codec) { unsigned int present; @@ -7587,10 +7637,11 @@ static const char *alc883_models[ALC883_MODEL_LAST] = { [ALC883_LENOVO_NB0763] = "lenovo-nb0763", [ALC888_LENOVO_MS7195_DIG] = "lenovo-ms7195-dig", [ALC883_HAIER_W66] = "haier-w66", - [ALC888_6ST_HP] = "6stack-hp", [ALC888_3ST_HP] = "3stack-hp", [ALC888_6ST_DELL] = "6stack-dell", [ALC883_MITAC] = "mitac", + [ALC883_CLEVO_M720] = "clevo-m720", + [ALC883_FUJITSU_PI2515] = "fujitsu-pi2515", [ALC883_AUTO] = "auto", }; @@ -7604,7 +7655,7 @@ static struct snd_pci_quirk alc883_cfg_tbl[] = { SND_PCI_QUIRK(0x103c, 0x2a3d, "HP Pavillion", ALC883_6ST_DIG), SND_PCI_QUIRK(0x103c, 0x2a4f, "HP Samba", ALC888_3ST_HP), SND_PCI_QUIRK(0x103c, 0x2a60, "HP Lucknow", ALC888_3ST_HP), - SND_PCI_QUIRK(0x103c, 0x2a61, "HP Nettle", ALC888_6ST_HP), + SND_PCI_QUIRK(0x103c, 0x2a61, "HP Nettle", ALC883_6ST_DIG), SND_PCI_QUIRK(0x1043, 0x8249, "Asus M2A-VM HDMI", ALC883_3ST_6ch_DIG), SND_PCI_QUIRK(0x105b, 0x6668, "Foxconn", ALC883_6ST_DIG), SND_PCI_QUIRK(0x1071, 0x8253, "Mitac 8252d", ALC883_MITAC), @@ -7614,7 +7665,9 @@ static struct snd_pci_quirk alc883_cfg_tbl[] = { SND_PCI_QUIRK(0x1462, 0x0349, "MSI", ALC883_TARGA_2ch_DIG), SND_PCI_QUIRK(0x1462, 0x040d, "MSI", ALC883_TARGA_2ch_DIG), SND_PCI_QUIRK(0x1462, 0x0579, "MSI", ALC883_TARGA_2ch_DIG), + SND_PCI_QUIRK(0x1462, 0x2fb3, "MSI", ALC883_TARGA_2ch_DIG), SND_PCI_QUIRK(0x1462, 0x3729, "MSI S420", ALC883_TARGA_DIG), + SND_PCI_QUIRK(0x1462, 0x3783, "NEC S970", ALC883_TARGA_DIG), SND_PCI_QUIRK(0x1462, 0x3b7f, "MSI", ALC883_TARGA_2ch_DIG), SND_PCI_QUIRK(0x1462, 0x3ef9, "MSI", ALC883_TARGA_DIG), SND_PCI_QUIRK(0x1462, 0x3fc1, "MSI", ALC883_TARGA_DIG), @@ -7627,13 +7680,17 @@ static struct snd_pci_quirk alc883_cfg_tbl[] = { SND_PCI_QUIRK(0x1462, 0x6668, "MSI", ALC883_6ST_DIG), SND_PCI_QUIRK(0x1462, 0x7187, "MSI", ALC883_6ST_DIG), SND_PCI_QUIRK(0x1462, 0x7250, "MSI", ALC883_6ST_DIG), + SND_PCI_QUIRK(0x1462, 0x7267, "MSI", ALC883_3ST_6ch_DIG), SND_PCI_QUIRK(0x1462, 0x7280, "MSI", ALC883_6ST_DIG), SND_PCI_QUIRK(0x1462, 0x7327, "MSI", ALC883_6ST_DIG), SND_PCI_QUIRK(0x1462, 0xa422, "MSI", ALC883_TARGA_2ch_DIG), SND_PCI_QUIRK(0x147b, 0x1083, "Abit IP35-PRO", ALC883_6ST_DIG), + SND_PCI_QUIRK(0x1558, 0x0721, "Clevo laptop M720R", ALC883_CLEVO_M720), + SND_PCI_QUIRK(0x1558, 0x0722, "Clevo laptop M720SR", ALC883_CLEVO_M720), SND_PCI_QUIRK(0x1558, 0, "Clevo laptop", ALC883_LAPTOP_EAPD), SND_PCI_QUIRK(0x15d9, 0x8780, "Supermicro PDSBA", ALC883_3ST_6ch), SND_PCI_QUIRK(0x161f, 0x2054, "Medion laptop", ALC883_MEDION), + SND_PCI_QUIRK(0x1734, 0x1108, "Fujitsu AMILO Pi2515", ALC883_FUJITSU_PI2515), SND_PCI_QUIRK(0x17aa, 0x101e, "Lenovo 101e", ALC883_LENOVO_101E_2ch), SND_PCI_QUIRK(0x17aa, 0x2085, "Lenovo NB0763", ALC883_LENOVO_NB0763), SND_PCI_QUIRK(0x17aa, 0x3bfc, "Lenovo NB0763", ALC883_LENOVO_NB0763), @@ -7652,8 +7709,6 @@ static struct alc_config_preset alc883_presets[] = { .num_dacs = ARRAY_SIZE(alc883_dac_nids), .dac_nids = alc883_dac_nids, .dig_out_nid = ALC883_DIGOUT_NID, - .num_adc_nids = ARRAY_SIZE(alc883_adc_nids), - .adc_nids = alc883_adc_nids, .dig_in_nid = ALC883_DIGIN_NID, .num_channel_mode = ARRAY_SIZE(alc883_3ST_2ch_modes), .channel_mode = alc883_3ST_2ch_modes, @@ -7665,8 +7720,6 @@ static struct alc_config_preset alc883_presets[] = { .num_dacs = ARRAY_SIZE(alc883_dac_nids), .dac_nids = alc883_dac_nids, .dig_out_nid = ALC883_DIGOUT_NID, - .num_adc_nids = ARRAY_SIZE(alc883_adc_nids), - .adc_nids = alc883_adc_nids, .dig_in_nid = ALC883_DIGIN_NID, .num_channel_mode = ARRAY_SIZE(alc883_3ST_6ch_modes), .channel_mode = alc883_3ST_6ch_modes, @@ -7678,8 +7731,6 @@ static struct alc_config_preset alc883_presets[] = { .init_verbs = { alc883_init_verbs }, .num_dacs = ARRAY_SIZE(alc883_dac_nids), .dac_nids = alc883_dac_nids, - .num_adc_nids = ARRAY_SIZE(alc883_adc_nids), - .adc_nids = alc883_adc_nids, .num_channel_mode = ARRAY_SIZE(alc883_3ST_6ch_modes), .channel_mode = alc883_3ST_6ch_modes, .need_dac_fix = 1, @@ -7691,8 +7742,6 @@ static struct alc_config_preset alc883_presets[] = { .num_dacs = ARRAY_SIZE(alc883_dac_nids), .dac_nids = alc883_dac_nids, .dig_out_nid = ALC883_DIGOUT_NID, - .num_adc_nids = ARRAY_SIZE(alc883_adc_nids), - .adc_nids = alc883_adc_nids, .dig_in_nid = ALC883_DIGIN_NID, .num_channel_mode = ARRAY_SIZE(alc883_sixstack_modes), .channel_mode = alc883_sixstack_modes, @@ -7704,8 +7753,6 @@ static struct alc_config_preset alc883_presets[] = { .num_dacs = ARRAY_SIZE(alc883_dac_nids), .dac_nids = alc883_dac_nids, .dig_out_nid = ALC883_DIGOUT_NID, - .num_adc_nids = ARRAY_SIZE(alc883_adc_nids), - .adc_nids = alc883_adc_nids, .num_channel_mode = ARRAY_SIZE(alc883_3ST_6ch_modes), .channel_mode = alc883_3ST_6ch_modes, .need_dac_fix = 1, @@ -7719,8 +7766,6 @@ static struct alc_config_preset alc883_presets[] = { .num_dacs = ARRAY_SIZE(alc883_dac_nids), .dac_nids = alc883_dac_nids, .dig_out_nid = ALC883_DIGOUT_NID, - .num_adc_nids = ARRAY_SIZE(alc883_adc_nids), - .adc_nids = alc883_adc_nids, .num_channel_mode = ARRAY_SIZE(alc883_3ST_2ch_modes), .channel_mode = alc883_3ST_2ch_modes, .input_mux = &alc883_capture_source, @@ -7737,8 +7782,6 @@ static struct alc_config_preset alc883_presets[] = { .init_verbs = { alc883_init_verbs, alc880_gpio1_init_verbs }, .num_dacs = ARRAY_SIZE(alc883_dac_nids), .dac_nids = alc883_dac_nids, - .num_adc_nids = ARRAY_SIZE(alc883_adc_nids), - .adc_nids = alc883_adc_nids, .num_channel_mode = ARRAY_SIZE(alc883_3ST_2ch_modes), .channel_mode = alc883_3ST_2ch_modes, .input_mux = &alc883_capture_source, @@ -7749,8 +7792,6 @@ static struct alc_config_preset alc883_presets[] = { .num_dacs = ARRAY_SIZE(alc883_dac_nids), .dac_nids = alc883_dac_nids, .dig_out_nid = ALC883_DIGOUT_NID, - .num_adc_nids = ARRAY_SIZE(alc883_adc_nids), - .adc_nids = alc883_adc_nids, .num_channel_mode = ARRAY_SIZE(alc883_3ST_2ch_modes), .channel_mode = alc883_3ST_2ch_modes, .input_mux = &alc883_capture_source, @@ -7764,8 +7805,6 @@ static struct alc_config_preset alc883_presets[] = { alc883_medion_eapd_verbs }, .num_dacs = ARRAY_SIZE(alc883_dac_nids), .dac_nids = alc883_dac_nids, - .num_adc_nids = ARRAY_SIZE(alc883_adc_nids), - .adc_nids = alc883_adc_nids, .num_channel_mode = ARRAY_SIZE(alc883_sixstack_modes), .channel_mode = alc883_sixstack_modes, .input_mux = &alc883_capture_source, @@ -7776,8 +7815,6 @@ static struct alc_config_preset alc883_presets[] = { .num_dacs = ARRAY_SIZE(alc883_dac_nids), .dac_nids = alc883_dac_nids, .dig_out_nid = ALC883_DIGOUT_NID, - .num_adc_nids = ARRAY_SIZE(alc883_adc_nids), - .adc_nids = alc883_adc_nids, .num_channel_mode = ARRAY_SIZE(alc883_3ST_2ch_modes), .channel_mode = alc883_3ST_2ch_modes, .input_mux = &alc883_capture_source, @@ -7789,19 +7826,27 @@ static struct alc_config_preset alc883_presets[] = { .init_verbs = { alc883_init_verbs, alc882_eapd_verbs }, .num_dacs = ARRAY_SIZE(alc883_dac_nids), .dac_nids = alc883_dac_nids, - .num_adc_nids = ARRAY_SIZE(alc883_adc_nids), - .adc_nids = alc883_adc_nids, .num_channel_mode = ARRAY_SIZE(alc883_3ST_2ch_modes), .channel_mode = alc883_3ST_2ch_modes, .input_mux = &alc883_capture_source, }, + [ALC883_CLEVO_M720] = { + .mixers = { alc883_clevo_m720_mixer }, + .init_verbs = { alc883_init_verbs, alc883_clevo_m720_verbs }, + .num_dacs = ARRAY_SIZE(alc883_dac_nids), + .dac_nids = alc883_dac_nids, + .dig_out_nid = ALC883_DIGOUT_NID, + .num_channel_mode = ARRAY_SIZE(alc883_3ST_2ch_modes), + .channel_mode = alc883_3ST_2ch_modes, + .input_mux = &alc883_capture_source, + .unsol_event = alc883_clevo_m720_unsol_event, + .init_hook = alc883_clevo_m720_automute, + }, [ALC883_LENOVO_101E_2ch] = { .mixers = { alc883_lenovo_101e_2ch_mixer}, .init_verbs = { alc883_init_verbs, alc883_lenovo_101e_verbs}, .num_dacs = ARRAY_SIZE(alc883_dac_nids), .dac_nids = alc883_dac_nids, - .num_adc_nids = ARRAY_SIZE(alc883_adc_nids), - .adc_nids = alc883_adc_nids, .num_channel_mode = ARRAY_SIZE(alc883_3ST_2ch_modes), .channel_mode = alc883_3ST_2ch_modes, .input_mux = &alc883_lenovo_101e_capture_source, @@ -7813,8 +7858,6 @@ static struct alc_config_preset alc883_presets[] = { .init_verbs = { alc883_init_verbs, alc883_lenovo_nb0763_verbs}, .num_dacs = ARRAY_SIZE(alc883_dac_nids), .dac_nids = alc883_dac_nids, - .num_adc_nids = ARRAY_SIZE(alc883_adc_nids), - .adc_nids = alc883_adc_nids, .num_channel_mode = ARRAY_SIZE(alc883_3ST_2ch_modes), .channel_mode = alc883_3ST_2ch_modes, .need_dac_fix = 1, @@ -7828,8 +7871,6 @@ static struct alc_config_preset alc883_presets[] = { .num_dacs = ARRAY_SIZE(alc883_dac_nids), .dac_nids = alc883_dac_nids, .dig_out_nid = ALC883_DIGOUT_NID, - .num_adc_nids = ARRAY_SIZE(alc883_adc_nids), - .adc_nids = alc883_adc_nids, .num_channel_mode = ARRAY_SIZE(alc883_3ST_6ch_modes), .channel_mode = alc883_3ST_6ch_modes, .need_dac_fix = 1, @@ -7843,47 +7884,28 @@ static struct alc_config_preset alc883_presets[] = { .num_dacs = ARRAY_SIZE(alc883_dac_nids), .dac_nids = alc883_dac_nids, .dig_out_nid = ALC883_DIGOUT_NID, - .num_adc_nids = ARRAY_SIZE(alc883_adc_nids), - .adc_nids = alc883_adc_nids, .num_channel_mode = ARRAY_SIZE(alc883_3ST_2ch_modes), .channel_mode = alc883_3ST_2ch_modes, .input_mux = &alc883_capture_source, .unsol_event = alc883_haier_w66_unsol_event, .init_hook = alc883_haier_w66_automute, - }, - [ALC888_6ST_HP] = { - .mixers = { alc888_6st_hp_mixer, alc883_chmode_mixer }, - .init_verbs = { alc883_init_verbs, alc888_6st_hp_verbs }, - .num_dacs = ARRAY_SIZE(alc883_dac_nids), - .dac_nids = alc883_dac_nids, - .dig_out_nid = ALC883_DIGOUT_NID, - .num_adc_nids = ARRAY_SIZE(alc883_adc_nids), - .adc_nids = alc883_adc_nids, - .dig_in_nid = ALC883_DIGIN_NID, - .num_channel_mode = ARRAY_SIZE(alc883_sixstack_modes), - .channel_mode = alc883_sixstack_modes, - .input_mux = &alc883_capture_source, }, [ALC888_3ST_HP] = { - .mixers = { alc888_3st_hp_mixer, alc883_chmode_mixer }, + .mixers = { alc883_3ST_6ch_mixer, alc883_chmode_mixer }, .init_verbs = { alc883_init_verbs, alc888_3st_hp_verbs }, .num_dacs = ARRAY_SIZE(alc883_dac_nids), .dac_nids = alc883_dac_nids, - .num_adc_nids = ARRAY_SIZE(alc883_adc_nids), - .adc_nids = alc883_adc_nids, .num_channel_mode = ARRAY_SIZE(alc888_3st_hp_modes), .channel_mode = alc888_3st_hp_modes, .need_dac_fix = 1, .input_mux = &alc883_capture_source, }, [ALC888_6ST_DELL] = { - .mixers = { alc888_6st_dell_mixer, alc883_chmode_mixer }, + .mixers = { alc883_base_mixer, alc883_chmode_mixer }, .init_verbs = { alc883_init_verbs, alc888_6st_dell_verbs }, .num_dacs = ARRAY_SIZE(alc883_dac_nids), .dac_nids = alc883_dac_nids, .dig_out_nid = ALC883_DIGOUT_NID, - .num_adc_nids = ARRAY_SIZE(alc883_adc_nids), - .adc_nids = alc883_adc_nids, .dig_in_nid = ALC883_DIGIN_NID, .num_channel_mode = ARRAY_SIZE(alc883_sixstack_modes), .channel_mode = alc883_sixstack_modes, @@ -7896,14 +7918,25 @@ static struct alc_config_preset alc883_presets[] = { .init_verbs = { alc883_init_verbs, alc883_mitac_verbs }, .num_dacs = ARRAY_SIZE(alc883_dac_nids), .dac_nids = alc883_dac_nids, - .num_adc_nids = ARRAY_SIZE(alc883_adc_nids), - .adc_nids = alc883_adc_nids, .num_channel_mode = ARRAY_SIZE(alc883_3ST_2ch_modes), .channel_mode = alc883_3ST_2ch_modes, .input_mux = &alc883_capture_source, .unsol_event = alc883_mitac_unsol_event, .init_hook = alc883_mitac_automute, }, + [ALC883_FUJITSU_PI2515] = { + .mixers = { alc883_2ch_fujitsu_pi2515_mixer }, + .init_verbs = { alc883_init_verbs, + alc883_2ch_fujitsu_pi2515_verbs}, + .num_dacs = ARRAY_SIZE(alc883_dac_nids), + .dac_nids = alc883_dac_nids, + .dig_out_nid = ALC883_DIGOUT_NID, + .num_channel_mode = ARRAY_SIZE(alc883_3ST_2ch_modes), + .channel_mode = alc883_3ST_2ch_modes, + .input_mux = &alc883_fujitsu_pi2515_capture_source, + .unsol_event = alc883_2ch_fujitsu_pi2515_unsol_event, + .init_hook = alc883_2ch_fujitsu_pi2515_automute, + }, }; @@ -7918,15 +7951,11 @@ static void alc883_auto_set_output_and_unmute(struct hda_codec *codec, struct alc_spec *spec = codec->spec; int idx; + alc_set_pin_output(codec, nid, pin_type); if (spec->multiout.dac_nids[dac_idx] == 0x25) idx = 4; else idx = spec->multiout.dac_nids[dac_idx] - 2; - - snd_hda_codec_write(codec, nid, 0, AC_VERB_SET_PIN_WIDGET_CONTROL, - pin_type); - snd_hda_codec_write(codec, nid, 0, AC_VERB_SET_AMP_GAIN_MUTE, - AMP_OUT_UNMUTE); snd_hda_codec_write(codec, nid, 0, AC_VERB_SET_CONNECT_SEL, idx); } @@ -7955,6 +7984,9 @@ static void alc883_auto_init_hp_out(struct hda_codec *codec) if (pin) /* connect to front */ /* use dac 0 */ alc883_auto_set_output_and_unmute(codec, pin, PIN_HP, 0); + pin = spec->autocfg.speaker_pins[0]; + if (pin) + alc883_auto_set_output_and_unmute(codec, pin, PIN_OUT, 0); } #define alc883_is_input_pin(nid) alc880_is_input_pin(nid) @@ -8006,9 +8038,12 @@ static int alc883_parse_auto_config(struct hda_codec *codec) /* additional initialization for auto-configuration model */ static void alc883_auto_init(struct hda_codec *codec) { + struct alc_spec *spec = codec->spec; alc883_auto_init_multi_out(codec); alc883_auto_init_hp_out(codec); alc883_auto_init_analog_input(codec); + if (spec->unsol_event) + alc_sku_automute(codec); } static int patch_alc883(struct hda_codec *codec) @@ -8057,10 +8092,9 @@ static int patch_alc883(struct hda_codec *codec) spec->stream_digital_playback = &alc883_pcm_digital_playback; spec->stream_digital_capture = &alc883_pcm_digital_capture; - if (!spec->adc_nids && spec->input_mux) { - spec->adc_nids = alc883_adc_nids; - spec->num_adc_nids = ARRAY_SIZE(alc883_adc_nids); - } + spec->num_adc_nids = ARRAY_SIZE(alc883_adc_nids); + spec->adc_nids = alc883_adc_nids; + spec->capsrc_nids = alc883_capsrc_nids; spec->vmaster_nid = 0x0c; @@ -8085,6 +8119,8 @@ static int patch_alc883(struct hda_codec *codec) #define alc262_dac_nids alc260_dac_nids #define alc262_adc_nids alc882_adc_nids #define alc262_adc_nids_alt alc882_adc_nids_alt +#define alc262_capsrc_nids alc882_capsrc_nids +#define alc262_capsrc_nids_alt alc882_capsrc_nids_alt #define alc262_modes alc260_modes #define alc262_capture_source alc882_capture_source @@ -8585,7 +8621,8 @@ static void alc262_hippo1_unsol_event(struct hda_codec *codec, /* * fujitsu model - * 0x14 = headphone/spdif-out, 0x15 = internal speaker + * 0x14 = headphone/spdif-out, 0x15 = internal speaker, + * 0x1b = port replicator headphone out */ #define ALC_HP_EVENT 0x37 @@ -8593,6 +8630,14 @@ static void alc262_hippo1_unsol_event(struct hda_codec *codec, static struct hda_verb alc262_fujitsu_unsol_verbs[] = { {0x14, AC_VERB_SET_UNSOLICITED_ENABLE, AC_USRSP_EN | ALC_HP_EVENT}, {0x14, AC_VERB_SET_PIN_WIDGET_CONTROL, PIN_HP}, + {0x1b, AC_VERB_SET_UNSOLICITED_ENABLE, AC_USRSP_EN | ALC_HP_EVENT}, + {0x1b, AC_VERB_SET_PIN_WIDGET_CONTROL, PIN_HP}, + {} +}; + +static struct hda_verb alc262_lenovo_3000_unsol_verbs[] = { + {0x1b, AC_VERB_SET_UNSOLICITED_ENABLE, AC_USRSP_EN | ALC_HP_EVENT}, + {0x1b, AC_VERB_SET_PIN_WIDGET_CONTROL, PIN_HP}, {} }; @@ -8633,12 +8678,16 @@ static void alc262_fujitsu_automute(struct hda_codec *codec, int force) unsigned int mute; if (force || !spec->sense_updated) { - unsigned int present; + unsigned int present_int_hp, present_dock_hp; /* need to execute and sync at first */ snd_hda_codec_read(codec, 0x14, 0, AC_VERB_SET_PIN_SENSE, 0); - present = snd_hda_codec_read(codec, 0x14, 0, - AC_VERB_GET_PIN_SENSE, 0); - spec->jack_present = (present & 0x80000000) != 0; + present_int_hp = snd_hda_codec_read(codec, 0x14, 0, + AC_VERB_GET_PIN_SENSE, 0); + snd_hda_codec_read(codec, 0x1B, 0, AC_VERB_SET_PIN_SENSE, 0); + present_dock_hp = snd_hda_codec_read(codec, 0x1b, 0, + AC_VERB_GET_PIN_SENSE, 0); + spec->jack_present = (present_int_hp & 0x80000000) != 0; + spec->jack_present |= (present_dock_hp & 0x80000000) != 0; spec->sense_updated = 1; } if (spec->jack_present) { @@ -8672,6 +8721,46 @@ static struct hda_bind_ctls alc262_fujitsu_bind_master_vol = { }, }; +/* mute/unmute internal speaker according to the hp jack and mute state */ +static void alc262_lenovo_3000_automute(struct hda_codec *codec, int force) +{ + struct alc_spec *spec = codec->spec; + unsigned int mute; + + if (force || !spec->sense_updated) { + unsigned int present_int_hp; + /* need to execute and sync at first */ + snd_hda_codec_read(codec, 0x1b, 0, AC_VERB_SET_PIN_SENSE, 0); + present_int_hp = snd_hda_codec_read(codec, 0x1b, 0, + AC_VERB_GET_PIN_SENSE, 0); + spec->jack_present = (present_int_hp & 0x80000000) != 0; + spec->sense_updated = 1; + } + if (spec->jack_present) { + /* mute internal speaker */ + snd_hda_codec_amp_stereo(codec, 0x14, HDA_OUTPUT, 0, + HDA_AMP_MUTE, HDA_AMP_MUTE); + snd_hda_codec_amp_stereo(codec, 0x16, HDA_OUTPUT, 0, + HDA_AMP_MUTE, HDA_AMP_MUTE); + } else { + /* unmute internal speaker if necessary */ + mute = snd_hda_codec_amp_read(codec, 0x1b, 0, HDA_OUTPUT, 0); + snd_hda_codec_amp_stereo(codec, 0x14, HDA_OUTPUT, 0, + HDA_AMP_MUTE, mute); + snd_hda_codec_amp_stereo(codec, 0x16, HDA_OUTPUT, 0, + HDA_AMP_MUTE, mute); + } +} + +/* unsolicited event for HP jack sensing */ +static void alc262_lenovo_3000_unsol_event(struct hda_codec *codec, + unsigned int res) +{ + if ((res >> 26) != ALC_HP_EVENT) + return; + alc262_lenovo_3000_automute(codec, 1); +} + /* bind hp and internal speaker mute (with plug check) */ static int alc262_fujitsu_master_sw_put(struct snd_kcontrol *kcontrol, struct snd_ctl_elem_value *ucontrol) @@ -8680,12 +8769,13 @@ static int alc262_fujitsu_master_sw_put(struct snd_kcontrol *kcontrol, long *valp = ucontrol->value.integer.value; int change; - change = snd_hda_codec_amp_update(codec, 0x14, 0, HDA_OUTPUT, 0, - HDA_AMP_MUTE, - valp[0] ? 0 : HDA_AMP_MUTE); - change |= snd_hda_codec_amp_update(codec, 0x14, 1, HDA_OUTPUT, 0, - HDA_AMP_MUTE, - valp[1] ? 0 : HDA_AMP_MUTE); + change = snd_hda_codec_amp_stereo(codec, 0x14, HDA_OUTPUT, 0, + HDA_AMP_MUTE, + valp ? 0 : HDA_AMP_MUTE); + change |= snd_hda_codec_amp_stereo(codec, 0x1b, HDA_OUTPUT, 0, + HDA_AMP_MUTE, + valp ? 0 : HDA_AMP_MUTE); + if (change) alc262_fujitsu_automute(codec, 0); return change; @@ -8703,6 +8793,46 @@ static struct snd_kcontrol_new alc262_fujitsu_mixer[] = { }, HDA_CODEC_VOLUME("CD Playback Volume", 0x0b, 0x04, HDA_INPUT), HDA_CODEC_MUTE("CD Playback Switch", 0x0b, 0x04, HDA_INPUT), + HDA_CODEC_VOLUME("PC Speaker Volume", 0x0b, 0x05, HDA_INPUT), + HDA_CODEC_MUTE("PC Speaker Switch", 0x0b, 0x05, HDA_INPUT), + HDA_CODEC_VOLUME("Mic Boost", 0x18, 0, HDA_INPUT), + HDA_CODEC_VOLUME("Mic Playback Volume", 0x0b, 0x0, HDA_INPUT), + HDA_CODEC_MUTE("Mic Playback Switch", 0x0b, 0x0, HDA_INPUT), + HDA_CODEC_VOLUME("Int Mic Boost", 0x19, 0, HDA_INPUT), + HDA_CODEC_VOLUME("Int Mic Playback Volume", 0x0b, 0x1, HDA_INPUT), + HDA_CODEC_MUTE("Int Mic Playback Switch", 0x0b, 0x1, HDA_INPUT), + { } /* end */ +}; + +/* bind hp and internal speaker mute (with plug check) */ +static int alc262_lenovo_3000_master_sw_put(struct snd_kcontrol *kcontrol, + struct snd_ctl_elem_value *ucontrol) +{ + struct hda_codec *codec = snd_kcontrol_chip(kcontrol); + long *valp = ucontrol->value.integer.value; + int change; + + change = snd_hda_codec_amp_stereo(codec, 0x1b, HDA_OUTPUT, 0, + HDA_AMP_MUTE, + valp ? 0 : HDA_AMP_MUTE); + + if (change) + alc262_lenovo_3000_automute(codec, 0); + return change; +} + +static struct snd_kcontrol_new alc262_lenovo_3000_mixer[] = { + HDA_BIND_VOL("Master Playback Volume", &alc262_fujitsu_bind_master_vol), + { + .iface = SNDRV_CTL_ELEM_IFACE_MIXER, + .name = "Master Playback Switch", + .info = snd_hda_mixer_amp_switch_info, + .get = snd_hda_mixer_amp_switch_get, + .put = alc262_lenovo_3000_master_sw_put, + .private_value = HDA_COMPOSE_AMP_VAL(0x1b, 3, 0, HDA_OUTPUT), + }, + HDA_CODEC_VOLUME("CD Playback Volume", 0x0b, 0x04, HDA_INPUT), + HDA_CODEC_MUTE("CD Playback Switch", 0x0b, 0x04, HDA_INPUT), HDA_CODEC_VOLUME("Mic Boost", 0x18, 0, HDA_INPUT), HDA_CODEC_VOLUME("Mic Playback Volume", 0x0b, 0x0, HDA_INPUT), HDA_CODEC_MUTE("Mic Playback Switch", 0x0b, 0x0, HDA_INPUT), @@ -8730,59 +8860,72 @@ static struct hda_verb alc262_benq_t31_EAPD_verbs[] = { /* Samsung Q1 Ultra Vista model setup */ static struct snd_kcontrol_new alc262_ultra_mixer[] = { - HDA_CODEC_VOLUME("Front Playback Volume", 0x0c, 0x0, HDA_OUTPUT), - HDA_CODEC_MUTE("Front Playback Switch", 0x14, 0x0, HDA_OUTPUT), - HDA_CODEC_MUTE("Headphone Playback Switch", 0x15, 0x0, HDA_OUTPUT), + HDA_CODEC_VOLUME("Master Playback Volume", 0x0c, 0x0, HDA_OUTPUT), + HDA_BIND_MUTE("Master Playback Switch", 0x0c, 2, HDA_INPUT), HDA_CODEC_VOLUME("Mic Playback Volume", 0x0b, 0x01, HDA_INPUT), HDA_CODEC_MUTE("Mic Playback Switch", 0x0b, 0x01, HDA_INPUT), HDA_CODEC_VOLUME("Mic Boost", 0x19, 0, HDA_INPUT), + HDA_CODEC_VOLUME("Headphone Mic Boost", 0x15, 0, HDA_INPUT), { } /* end */ }; static struct hda_verb alc262_ultra_verbs[] = { - {0x15, AC_VERB_SET_UNSOLICITED_ENABLE, AC_USRSP_EN | ALC880_HP_EVENT}, + /* output mixer */ + {0x0c, AC_VERB_SET_AMP_GAIN_MUTE, AMP_IN_MUTE(0)}, + {0x0c, AC_VERB_SET_AMP_GAIN_MUTE, AMP_IN_MUTE(1)}, + {0x0c, AC_VERB_SET_AMP_GAIN_MUTE, AMP_OUT_ZERO}, + /* speaker */ + {0x14, AC_VERB_SET_PIN_WIDGET_CONTROL, PIN_OUT}, + {0x14, AC_VERB_SET_AMP_GAIN_MUTE, AMP_OUT_MUTE}, + {0x14, AC_VERB_SET_AMP_GAIN_MUTE, AMP_IN_UNMUTE(0)}, + {0x14, AC_VERB_SET_CONNECT_SEL, 0x00}, + /* HP */ {0x15, AC_VERB_SET_PIN_WIDGET_CONTROL, PIN_HP}, - {0x18, AC_VERB_SET_PIN_WIDGET_CONTROL, 0x40}, - /* Mic is on Node 0x19 */ - {0x19, AC_VERB_SET_PIN_WIDGET_CONTROL, 0x24}, - {0x22, AC_VERB_SET_CONNECT_SEL, 0x01}, - {0x22, AC_VERB_SET_AMP_GAIN_MUTE, (0x7000 | (0x01 << 8))}, - {0x23, AC_VERB_SET_CONNECT_SEL, 0x01}, - {0x23, AC_VERB_SET_AMP_GAIN_MUTE, (0x7000 | (0x01 << 8))}, - {0x24, AC_VERB_SET_CONNECT_SEL, 0x01}, - {0x24, AC_VERB_SET_AMP_GAIN_MUTE, (0x7000 | (0x01 << 8))}, + {0x15, AC_VERB_SET_AMP_GAIN_MUTE, AMP_OUT_MUTE}, + {0x15, AC_VERB_SET_AMP_GAIN_MUTE, AMP_IN_UNMUTE(0)}, + {0x15, AC_VERB_SET_CONNECT_SEL, 0x00}, + {0x15, AC_VERB_SET_UNSOLICITED_ENABLE, AC_USRSP_EN | ALC880_HP_EVENT}, + /* internal mic */ + {0x19, AC_VERB_SET_PIN_WIDGET_CONTROL, PIN_VREF80}, + {0x0b, AC_VERB_SET_AMP_GAIN_MUTE, AMP_IN_MUTE(1)}, + /* ADC, choose mic */ + {0x07, AC_VERB_SET_AMP_GAIN_MUTE, AMP_IN_MUTE(0)}, + {0x24, AC_VERB_SET_AMP_GAIN_MUTE, AMP_IN_MUTE(0)}, + {0x24, AC_VERB_SET_AMP_GAIN_MUTE, AMP_IN_UNMUTE(1)}, + {0x24, AC_VERB_SET_AMP_GAIN_MUTE, AMP_IN_MUTE(2)}, + {0x24, AC_VERB_SET_AMP_GAIN_MUTE, AMP_IN_MUTE(3)}, + {0x24, AC_VERB_SET_AMP_GAIN_MUTE, AMP_IN_MUTE(4)}, + {0x24, AC_VERB_SET_AMP_GAIN_MUTE, AMP_IN_MUTE(5)}, + {0x24, AC_VERB_SET_AMP_GAIN_MUTE, AMP_IN_MUTE(6)}, + {0x24, AC_VERB_SET_AMP_GAIN_MUTE, AMP_IN_MUTE(7)}, + {0x24, AC_VERB_SET_AMP_GAIN_MUTE, AMP_IN_MUTE(8)}, {} }; -static struct hda_input_mux alc262_ultra_capture_source = { - .num_items = 1, - .items = { - { "Mic", 0x1 }, - }, -}; - /* mute/unmute internal speaker according to the hp jack and mute state */ static void alc262_ultra_automute(struct hda_codec *codec) { struct alc_spec *spec = codec->spec; unsigned int mute; - unsigned int present; - /* need to execute and sync at first */ - snd_hda_codec_read(codec, 0x15, 0, AC_VERB_SET_PIN_SENSE, 0); - present = snd_hda_codec_read(codec, 0x15, 0, - AC_VERB_GET_PIN_SENSE, 0); - spec->jack_present = (present & 0x80000000) != 0; - if (spec->jack_present) { - /* mute internal speaker */ - snd_hda_codec_amp_stereo(codec, 0x14, HDA_OUTPUT, 0, - HDA_AMP_MUTE, HDA_AMP_MUTE); - } else { - /* unmute internal speaker if necessary */ - mute = snd_hda_codec_amp_read(codec, 0x15, 0, HDA_OUTPUT, 0); - snd_hda_codec_amp_stereo(codec, 0x14, HDA_OUTPUT, 0, - HDA_AMP_MUTE, mute); + mute = 0; + /* auto-mute only when HP is used as HP */ + if (!spec->cur_mux[0]) { + unsigned int present; + /* need to execute and sync at first */ + snd_hda_codec_read(codec, 0x15, 0, AC_VERB_SET_PIN_SENSE, 0); + present = snd_hda_codec_read(codec, 0x15, 0, + AC_VERB_GET_PIN_SENSE, 0); + spec->jack_present = (present & AC_PINSENSE_PRESENCE) != 0; + if (spec->jack_present) + mute = HDA_AMP_MUTE; } + /* mute/unmute internal speaker */ + snd_hda_codec_amp_stereo(codec, 0x14, HDA_OUTPUT, 0, + HDA_AMP_MUTE, mute); + /* mute/unmute HP */ + snd_hda_codec_amp_stereo(codec, 0x15, HDA_OUTPUT, 0, + HDA_AMP_MUTE, mute ? 0 : HDA_AMP_MUTE); } /* unsolicited event for HP jack sensing */ @@ -8794,6 +8937,45 @@ static void alc262_ultra_unsol_event(struct hda_codec *codec, alc262_ultra_automute(codec); } +static struct hda_input_mux alc262_ultra_capture_source = { + .num_items = 2, + .items = { + { "Mic", 0x1 }, + { "Headphone", 0x7 }, + }, +}; + +static int alc262_ultra_mux_enum_put(struct snd_kcontrol *kcontrol, + struct snd_ctl_elem_value *ucontrol) +{ + struct hda_codec *codec = snd_kcontrol_chip(kcontrol); + struct alc_spec *spec = codec->spec; + int ret; + + ret = alc882_mux_enum_put(kcontrol, ucontrol); + if (!ret) + return 0; + /* reprogram the HP pin as mic or HP according to the input source */ + snd_hda_codec_write_cache(codec, 0x15, 0, + AC_VERB_SET_PIN_WIDGET_CONTROL, + spec->cur_mux[0] ? PIN_VREF80 : PIN_HP); + alc262_ultra_automute(codec); /* mute/unmute HP */ + return ret; +} + +static struct snd_kcontrol_new alc262_ultra_capture_mixer[] = { + HDA_CODEC_VOLUME("Capture Volume", 0x07, 0x0, HDA_INPUT), + HDA_CODEC_MUTE("Capture Switch", 0x07, 0x0, HDA_INPUT), + { + .iface = SNDRV_CTL_ELEM_IFACE_MIXER, + .name = "Capture Source", + .info = alc882_mux_enum_info, + .get = alc882_mux_enum_get, + .put = alc262_ultra_mux_enum_put, + }, + { } /* end */ +}; + /* add playback controls from the parsed DAC table */ static int alc262_auto_create_multi_out_ctls(struct alc_spec *spec, const struct auto_pin_cfg *cfg) @@ -9185,9 +9367,12 @@ static int alc262_parse_auto_config(struct hda_codec *codec) /* init callback for auto-configuration model -- overriding the default init */ static void alc262_auto_init(struct hda_codec *codec) { + struct alc_spec *spec = codec->spec; alc262_auto_init_multi_out(codec); alc262_auto_init_hp_out(codec); alc262_auto_init_analog_input(codec); + if (spec->unsol_event) + alc_sku_automute(codec); } /* @@ -9206,6 +9391,7 @@ static const char *alc262_models[ALC262_MODEL_LAST] = { [ALC262_BENQ_T31] = "benq-t31", [ALC262_SONY_ASSAMD] = "sony-assamd", [ALC262_ULTRA] = "ultra", + [ALC262_LENOVO_3000] = "lenovo-3000", [ALC262_AUTO] = "auto", }; @@ -9241,6 +9427,8 @@ static struct snd_pci_quirk alc262_cfg_tbl[] = { SND_PCI_QUIRK(0x10cf, 0x1397, "Fujitsu", ALC262_FUJITSU), SND_PCI_QUIRK(0x10cf, 0x142d, "Fujitsu Lifebook E8410", ALC262_FUJITSU), SND_PCI_QUIRK(0x144d, 0xc032, "Samsung Q1 Ultra", ALC262_ULTRA), + SND_PCI_QUIRK(0x144d, 0xc039, "Samsung Q1U EL", ALC262_ULTRA), + SND_PCI_QUIRK(0x17aa, 0x384e, "Lenovo 3000 y410", ALC262_LENOVO_3000), SND_PCI_QUIRK(0x17ff, 0x0560, "Benq ED8", ALC262_BENQ_ED8), SND_PCI_QUIRK(0x17ff, 0x058d, "Benq T31-16", ALC262_BENQ_T31), SND_PCI_QUIRK(0x17ff, 0x058f, "Benq Hippo", ALC262_HIPPO_1), @@ -9390,18 +9578,32 @@ static struct alc_config_preset alc262_presets[] = { .init_hook = alc262_hippo_automute, }, [ALC262_ULTRA] = { - .mixers = { alc262_ultra_mixer }, - .init_verbs = { alc262_init_verbs, alc262_ultra_verbs }, + .mixers = { alc262_ultra_mixer, alc262_ultra_capture_mixer }, + .init_verbs = { alc262_ultra_verbs }, .num_dacs = ARRAY_SIZE(alc262_dac_nids), .dac_nids = alc262_dac_nids, - .hp_nid = 0x03, - .dig_out_nid = ALC262_DIGOUT_NID, .num_channel_mode = ARRAY_SIZE(alc262_modes), .channel_mode = alc262_modes, .input_mux = &alc262_ultra_capture_source, + .adc_nids = alc262_adc_nids, /* ADC0 */ + .capsrc_nids = alc262_capsrc_nids, + .num_adc_nids = 1, /* single ADC */ .unsol_event = alc262_ultra_unsol_event, .init_hook = alc262_ultra_automute, }, + [ALC262_LENOVO_3000] = { + .mixers = { alc262_lenovo_3000_mixer }, + .init_verbs = { alc262_init_verbs, alc262_EAPD_verbs, + alc262_lenovo_3000_unsol_verbs }, + .num_dacs = ARRAY_SIZE(alc262_dac_nids), + .dac_nids = alc262_dac_nids, + .hp_nid = 0x03, + .dig_out_nid = ALC262_DIGOUT_NID, + .num_channel_mode = ARRAY_SIZE(alc262_modes), + .channel_mode = alc262_modes, + .input_mux = &alc262_fujitsu_capture_source, + .unsol_event = alc262_lenovo_3000_unsol_event, + }, }; static int patch_alc262(struct hda_codec *codec) @@ -9472,12 +9674,14 @@ static int patch_alc262(struct hda_codec *codec) if (wcap != AC_WID_AUD_IN) { spec->adc_nids = alc262_adc_nids_alt; spec->num_adc_nids = ARRAY_SIZE(alc262_adc_nids_alt); + spec->capsrc_nids = alc262_capsrc_nids_alt; spec->mixers[spec->num_mixers] = alc262_capture_alt_mixer; spec->num_mixers++; } else { spec->adc_nids = alc262_adc_nids; spec->num_adc_nids = ARRAY_SIZE(alc262_adc_nids); + spec->capsrc_nids = alc262_capsrc_nids; spec->mixers[spec->num_mixers] = alc262_capture_mixer; spec->num_mixers++; } @@ -9517,6 +9721,8 @@ static hda_nid_t alc268_adc_nids_alt[1] = { 0x08 }; +static hda_nid_t alc268_capsrc_nids[2] = { 0x23, 0x24 }; + static struct snd_kcontrol_new alc268_base_mixer[] = { /* output mixer control */ HDA_CODEC_VOLUME("Front Playback Volume", 0x2, 0x0, HDA_OUTPUT), @@ -9529,6 +9735,22 @@ static struct snd_kcontrol_new alc268_base_mixer[] = { { } }; +/* bind Beep switches of both NID 0x0f and 0x10 */ +static struct hda_bind_ctls alc268_bind_beep_sw = { + .ops = &snd_hda_bind_sw, + .values = { + HDA_COMPOSE_AMP_VAL(0x0f, 3, 1, HDA_INPUT), + HDA_COMPOSE_AMP_VAL(0x10, 3, 1, HDA_INPUT), + 0 + }, +}; + +static struct snd_kcontrol_new alc268_beep_mixer[] = { + HDA_CODEC_VOLUME("Beep Playback Volume", 0x1d, 0x0, HDA_INPUT), + HDA_BIND_SW("Beep Playback Switch", &alc268_bind_beep_sw), + { } +}; + static struct hda_verb alc268_eapd_verbs[] = { {0x14, AC_VERB_SET_EAPD_BTLENABLE, 2}, {0x15, AC_VERB_SET_EAPD_BTLENABLE, 2}, @@ -9613,8 +9835,12 @@ static struct snd_kcontrol_new alc268_acer_mixer[] = { }; static struct hda_verb alc268_acer_verbs[] = { + {0x12, AC_VERB_SET_PIN_WIDGET_CONTROL, PIN_IN}, /* internal dmic? */ + {0x13, AC_VERB_SET_PIN_WIDGET_CONTROL, PIN_IN}, {0x14, AC_VERB_SET_PIN_WIDGET_CONTROL, PIN_HP}, {0x15, AC_VERB_SET_PIN_WIDGET_CONTROL, PIN_OUT}, + {0x18, AC_VERB_SET_PIN_WIDGET_CONTROL, PIN_VREF80}, + {0x1a, AC_VERB_SET_PIN_WIDGET_CONTROL, PIN_VREF80}, {0x14, AC_VERB_SET_UNSOLICITED_ENABLE, ALC880_HP_EVENT | AC_USRSP_EN}, { } @@ -9685,6 +9911,64 @@ static void alc268_dell_unsol_event(struct hda_codec *codec, #define alc268_dell_init_hook alc268_dell_automute +static struct snd_kcontrol_new alc267_quanta_il1_mixer[] = { + HDA_CODEC_VOLUME("Speaker Playback Volume", 0x2, 0x0, HDA_OUTPUT), + HDA_CODEC_MUTE("Speaker Playback Switch", 0x14, 0x0, HDA_OUTPUT), + HDA_CODEC_VOLUME("Headphone Playback Volume", 0x3, 0x0, HDA_OUTPUT), + HDA_CODEC_MUTE("Headphone Playback Switch", 0x15, 0x0, HDA_OUTPUT), + HDA_CODEC_VOLUME("Mic Capture Volume", 0x23, 0x0, HDA_OUTPUT), + HDA_BIND_MUTE("Mic Capture Switch", 0x23, 2, HDA_OUTPUT), + HDA_CODEC_VOLUME("Ext Mic Boost", 0x18, 0, HDA_INPUT), + HDA_CODEC_VOLUME("Int Mic Boost", 0x19, 0, HDA_INPUT), + { } +}; + +static struct hda_verb alc267_quanta_il1_verbs[] = { + {0x15, AC_VERB_SET_UNSOLICITED_ENABLE, ALC880_HP_EVENT | AC_USRSP_EN}, + {0x18, AC_VERB_SET_UNSOLICITED_ENABLE, ALC880_MIC_EVENT | AC_USRSP_EN}, + { } +}; + +static void alc267_quanta_il1_hp_automute(struct hda_codec *codec) +{ + unsigned int present; + + present = snd_hda_codec_read(codec, 0x15, 0, AC_VERB_GET_PIN_SENSE, 0) + & AC_PINSENSE_PRESENCE; + snd_hda_codec_write(codec, 0x14, 0, AC_VERB_SET_PIN_WIDGET_CONTROL, + present ? 0 : PIN_OUT); +} + +static void alc267_quanta_il1_mic_automute(struct hda_codec *codec) +{ + unsigned int present; + + present = snd_hda_codec_read(codec, 0x18, 0, + AC_VERB_GET_PIN_SENSE, 0) & 0x80000000; + snd_hda_codec_write(codec, 0x23, 0, + AC_VERB_SET_CONNECT_SEL, + present ? 0x00 : 0x01); +} + +static void alc267_quanta_il1_automute(struct hda_codec *codec) +{ + alc267_quanta_il1_hp_automute(codec); + alc267_quanta_il1_mic_automute(codec); +} + +static void alc267_quanta_il1_unsol_event(struct hda_codec *codec, + unsigned int res) +{ + switch (res >> 26) { + case ALC880_HP_EVENT: + alc267_quanta_il1_hp_automute(codec); + break; + case ALC880_MIC_EVENT: + alc267_quanta_il1_mic_automute(codec); + break; + } +} + /* * generic initialization of ADC, input mixers and output mixers */ @@ -9725,7 +10009,11 @@ static struct hda_verb alc268_base_init_verbs[] = { {0x19, AC_VERB_SET_AMP_GAIN_MUTE, AMP_OUT_MUTE}, {0x1a, AC_VERB_SET_AMP_GAIN_MUTE, AMP_OUT_MUTE}, {0x1c, AC_VERB_SET_AMP_GAIN_MUTE, AMP_OUT_MUTE}, - {0x1d, AC_VERB_SET_AMP_GAIN_MUTE, (0x7000 | (0x00 << 8))}, + + /* set PCBEEP vol = 0, mute connections */ + {0x1d, AC_VERB_SET_AMP_GAIN_MUTE, AMP_IN_UNMUTE(0)}, + {0x0f, AC_VERB_SET_AMP_GAIN_MUTE, AMP_IN_MUTE(1)}, + {0x10, AC_VERB_SET_AMP_GAIN_MUTE, AMP_IN_MUTE(1)}, /* Unmute Selector 23h,24h and set the default input to mic-in */ @@ -9764,29 +10052,17 @@ static struct hda_verb alc268_volume_init_verbs[] = { {0x1a, AC_VERB_SET_AMP_GAIN_MUTE, AMP_OUT_MUTE}, {0x1c, AC_VERB_SET_AMP_GAIN_MUTE, AMP_OUT_MUTE}, - /* set PCBEEP vol = 0 */ - {0x1d, AC_VERB_SET_AMP_GAIN_MUTE, (0xb000 | (0x00 << 8))}, + /* set PCBEEP vol = 0, mute connections */ + {0x1d, AC_VERB_SET_AMP_GAIN_MUTE, AMP_IN_UNMUTE(0)}, + {0x0f, AC_VERB_SET_AMP_GAIN_MUTE, AMP_IN_MUTE(1)}, + {0x10, AC_VERB_SET_AMP_GAIN_MUTE, AMP_IN_MUTE(1)}, { } }; #define alc268_mux_enum_info alc_mux_enum_info #define alc268_mux_enum_get alc_mux_enum_get - -static int alc268_mux_enum_put(struct snd_kcontrol *kcontrol, - struct snd_ctl_elem_value *ucontrol) -{ - struct hda_codec *codec = snd_kcontrol_chip(kcontrol); - struct alc_spec *spec = codec->spec; - - unsigned int adc_idx = snd_ctl_get_ioffidx(kcontrol, &ucontrol->id); - static hda_nid_t capture_mixers[3] = { 0x23, 0x24 }; - hda_nid_t nid = capture_mixers[adc_idx]; - - return snd_hda_input_mux_put(codec, spec->input_mux, ucontrol, - nid, - &spec->cur_mux[adc_idx]); -} +#define alc268_mux_enum_put alc_mux_enum_put static struct snd_kcontrol_new alc268_capture_alt_mixer[] = { HDA_CODEC_VOLUME("Capture Volume", 0x23, 0x0, HDA_OUTPUT), @@ -9836,13 +10112,17 @@ static struct hda_input_mux alc268_capture_source = { }, }; +static struct hda_input_mux alc268_acer_capture_source = { + .num_items = 3, + .items = { + { "Mic", 0x0 }, + { "Internal Mic", 0x6 }, + { "Line", 0x2 }, + }, +}; + #ifdef CONFIG_SND_DEBUG static struct snd_kcontrol_new alc268_test_mixer[] = { - HDA_CODEC_VOLUME("Front Playback Volume", 0x2, 0x0, HDA_OUTPUT), - HDA_CODEC_MUTE("Front Playback Switch", 0x14, 0x0, HDA_OUTPUT), - HDA_CODEC_VOLUME("Headphone Playback Volume", 0x3, 0x0, HDA_OUTPUT), - HDA_CODEC_MUTE("Headphone Playback Switch", 0x15, 0x0, HDA_OUTPUT), - /* Volume widgets */ HDA_CODEC_VOLUME("LOUT1 Playback Volume", 0x02, 0x0, HDA_OUTPUT), HDA_CODEC_VOLUME("LOUT2 Playback Volume", 0x03, 0x0, HDA_OUTPUT), @@ -9981,6 +10261,10 @@ static int alc268_auto_create_analog_input_ctls(struct alc_spec *spec, case 0x1c: idx1 = 3; /* CD */ break; + case 0x12: + case 0x13: + idx1 = 6; /* digital mics */ + break; default: continue; } @@ -10073,6 +10357,9 @@ static int alc268_parse_auto_config(struct hda_codec *codec) if (spec->kctl_alloc) spec->mixers[spec->num_mixers++] = spec->kctl_alloc; + if (spec->autocfg.speaker_pins[0] != 0x1d) + spec->mixers[spec->num_mixers++] = alc268_beep_mixer; + spec->init_verbs[spec->num_init_verbs++] = alc268_volume_init_verbs; spec->num_mux_defs = 1; spec->input_mux = &spec->private_imux; @@ -10091,20 +10378,25 @@ static int alc268_parse_auto_config(struct hda_codec *codec) /* init callback for auto-configuration model -- overriding the default init */ static void alc268_auto_init(struct hda_codec *codec) { + struct alc_spec *spec = codec->spec; alc268_auto_init_multi_out(codec); alc268_auto_init_hp_out(codec); alc268_auto_init_mono_speaker_out(codec); alc268_auto_init_analog_input(codec); + if (spec->unsol_event) + alc_sku_automute(codec); } /* * configuration and preset */ static const char *alc268_models[ALC268_MODEL_LAST] = { + [ALC267_QUANTA_IL1] = "quanta-il1", [ALC268_3ST] = "3stack", [ALC268_TOSHIBA] = "toshiba", [ALC268_ACER] = "acer", [ALC268_DELL] = "dell", + [ALC268_ZEPTO] = "zepto", #ifdef CONFIG_SND_DEBUG [ALC268_TEST] = "test", #endif @@ -10122,17 +10414,36 @@ static struct snd_pci_quirk alc268_cfg_tbl[] = { SND_PCI_QUIRK(0x1179, 0xff10, "TOSHIBA A205", ALC268_TOSHIBA), SND_PCI_QUIRK(0x1179, 0xff50, "TOSHIBA A305", ALC268_TOSHIBA), SND_PCI_QUIRK(0x152d, 0x0763, "Diverse (CPR2000)", ALC268_ACER), + SND_PCI_QUIRK(0x152d, 0x0771, "Quanta IL1", ALC267_QUANTA_IL1), + SND_PCI_QUIRK(0x1170, 0x0040, "ZEPTO", ALC268_ZEPTO), {} }; static struct alc_config_preset alc268_presets[] = { + [ALC267_QUANTA_IL1] = { + .mixers = { alc267_quanta_il1_mixer }, + .init_verbs = { alc268_base_init_verbs, alc268_eapd_verbs, + alc267_quanta_il1_verbs }, + .num_dacs = ARRAY_SIZE(alc268_dac_nids), + .dac_nids = alc268_dac_nids, + .num_adc_nids = ARRAY_SIZE(alc268_adc_nids_alt), + .adc_nids = alc268_adc_nids_alt, + .hp_nid = 0x03, + .num_channel_mode = ARRAY_SIZE(alc268_modes), + .channel_mode = alc268_modes, + .input_mux = &alc268_capture_source, + .unsol_event = alc267_quanta_il1_unsol_event, + .init_hook = alc267_quanta_il1_automute, + }, [ALC268_3ST] = { - .mixers = { alc268_base_mixer, alc268_capture_alt_mixer }, + .mixers = { alc268_base_mixer, alc268_capture_alt_mixer, + alc268_beep_mixer }, .init_verbs = { alc268_base_init_verbs }, .num_dacs = ARRAY_SIZE(alc268_dac_nids), .dac_nids = alc268_dac_nids, .num_adc_nids = ARRAY_SIZE(alc268_adc_nids_alt), .adc_nids = alc268_adc_nids_alt, + .capsrc_nids = alc268_capsrc_nids, .hp_nid = 0x03, .dig_out_nid = ALC268_DIGOUT_NID, .num_channel_mode = ARRAY_SIZE(alc268_modes), @@ -10140,13 +10451,15 @@ static struct alc_config_preset alc268_presets[] = { .input_mux = &alc268_capture_source, }, [ALC268_TOSHIBA] = { - .mixers = { alc268_base_mixer, alc268_capture_alt_mixer }, + .mixers = { alc268_base_mixer, alc268_capture_alt_mixer, + alc268_beep_mixer }, .init_verbs = { alc268_base_init_verbs, alc268_eapd_verbs, alc268_toshiba_verbs }, .num_dacs = ARRAY_SIZE(alc268_dac_nids), .dac_nids = alc268_dac_nids, .num_adc_nids = ARRAY_SIZE(alc268_adc_nids_alt), .adc_nids = alc268_adc_nids_alt, + .capsrc_nids = alc268_capsrc_nids, .hp_nid = 0x03, .num_channel_mode = ARRAY_SIZE(alc268_modes), .channel_mode = alc268_modes, @@ -10155,22 +10468,24 @@ static struct alc_config_preset alc268_presets[] = { .init_hook = alc268_toshiba_automute, }, [ALC268_ACER] = { - .mixers = { alc268_acer_mixer, alc268_capture_alt_mixer }, + .mixers = { alc268_acer_mixer, alc268_capture_alt_mixer, + alc268_beep_mixer }, .init_verbs = { alc268_base_init_verbs, alc268_eapd_verbs, alc268_acer_verbs }, .num_dacs = ARRAY_SIZE(alc268_dac_nids), .dac_nids = alc268_dac_nids, .num_adc_nids = ARRAY_SIZE(alc268_adc_nids_alt), .adc_nids = alc268_adc_nids_alt, + .capsrc_nids = alc268_capsrc_nids, .hp_nid = 0x02, .num_channel_mode = ARRAY_SIZE(alc268_modes), .channel_mode = alc268_modes, - .input_mux = &alc268_capture_source, + .input_mux = &alc268_acer_capture_source, .unsol_event = alc268_acer_unsol_event, .init_hook = alc268_acer_init_hook, }, [ALC268_DELL] = { - .mixers = { alc268_dell_mixer }, + .mixers = { alc268_dell_mixer, alc268_beep_mixer }, .init_verbs = { alc268_base_init_verbs, alc268_eapd_verbs, alc268_dell_verbs }, .num_dacs = ARRAY_SIZE(alc268_dac_nids), @@ -10182,6 +10497,24 @@ static struct alc_config_preset alc268_presets[] = { .init_hook = alc268_dell_init_hook, .input_mux = &alc268_capture_source, }, + [ALC268_ZEPTO] = { + .mixers = { alc268_base_mixer, alc268_capture_alt_mixer, + alc268_beep_mixer }, + .init_verbs = { alc268_base_init_verbs, alc268_eapd_verbs, + alc268_toshiba_verbs }, + .num_dacs = ARRAY_SIZE(alc268_dac_nids), + .dac_nids = alc268_dac_nids, + .num_adc_nids = ARRAY_SIZE(alc268_adc_nids_alt), + .adc_nids = alc268_adc_nids_alt, + .capsrc_nids = alc268_capsrc_nids, + .hp_nid = 0x03, + .dig_out_nid = ALC268_DIGOUT_NID, + .num_channel_mode = ARRAY_SIZE(alc268_modes), + .channel_mode = alc268_modes, + .input_mux = &alc268_capture_source, + .unsol_event = alc268_toshiba_unsol_event, + .init_hook = alc268_toshiba_automute + }, #ifdef CONFIG_SND_DEBUG [ALC268_TEST] = { .mixers = { alc268_test_mixer, alc268_capture_mixer }, @@ -10191,6 +10524,7 @@ static struct alc_config_preset alc268_presets[] = { .dac_nids = alc268_dac_nids, .num_adc_nids = ARRAY_SIZE(alc268_adc_nids_alt), .adc_nids = alc268_adc_nids_alt, + .capsrc_nids = alc268_capsrc_nids, .hp_nid = 0x03, .dig_out_nid = ALC268_DIGOUT_NID, .num_channel_mode = ARRAY_SIZE(alc268_modes), @@ -10247,13 +10581,22 @@ static int patch_alc268(struct hda_codec *codec) spec->stream_name_digital = "ALC268 Digital"; spec->stream_digital_playback = &alc268_pcm_digital_playback; + if (!query_amp_caps(codec, 0x1d, HDA_INPUT)) + /* override the amp caps for beep generator */ + snd_hda_override_amp_caps(codec, 0x1d, HDA_INPUT, + (0x0c << AC_AMPCAP_OFFSET_SHIFT) | + (0x0c << AC_AMPCAP_NUM_STEPS_SHIFT) | + (0x07 << AC_AMPCAP_STEP_SIZE_SHIFT) | + (0 << AC_AMPCAP_MUTE_SHIFT)); + if (!spec->adc_nids && spec->input_mux) { /* check whether NID 0x07 is valid */ unsigned int wcap = get_wcaps(codec, 0x07); + int i; /* get type */ wcap = (wcap & AC_WCAP_TYPE) >> AC_WCAP_TYPE_SHIFT; - if (wcap != AC_WID_AUD_IN) { + if (wcap != AC_WID_AUD_IN || spec->input_mux->num_items == 1) { spec->adc_nids = alc268_adc_nids_alt; spec->num_adc_nids = ARRAY_SIZE(alc268_adc_nids_alt); spec->mixers[spec->num_mixers] = @@ -10266,6 +10609,12 @@ static int patch_alc268(struct hda_codec *codec) alc268_capture_mixer; spec->num_mixers++; } + spec->capsrc_nids = alc268_capsrc_nids; + /* set default input source */ + for (i = 0; i < spec->num_adc_nids; i++) + snd_hda_codec_write_cache(codec, alc268_capsrc_nids[i], + 0, AC_VERB_SET_CONNECT_SEL, + spec->input_mux->items[0].index); } spec->vmaster_nid = 0x02; @@ -10539,9 +10888,12 @@ static int alc269_parse_auto_config(struct hda_codec *codec) /* init callback for auto-configuration model -- overriding the default init */ static void alc269_auto_init(struct hda_codec *codec) { + struct alc_spec *spec = codec->spec; alc269_auto_init_multi_out(codec); alc269_auto_init_hp_out(codec); alc269_auto_init_analog_input(codec); + if (spec->unsol_event) + alc_sku_automute(codec); } /* @@ -11463,13 +11815,7 @@ static void alc861_auto_set_output_and_unmute(struct hda_codec *codec, hda_nid_t nid, int pin_type, int dac_idx) { - /* set as output */ - - snd_hda_codec_write(codec, nid, 0, AC_VERB_SET_PIN_WIDGET_CONTROL, - pin_type); - snd_hda_codec_write(codec, dac_idx, 0, AC_VERB_SET_AMP_GAIN_MUTE, - AMP_OUT_UNMUTE); - + alc_set_pin_output(codec, nid, pin_type); } static void alc861_auto_init_multi_out(struct hda_codec *codec) @@ -11496,6 +11842,9 @@ static void alc861_auto_init_hp_out(struct hda_codec *codec) if (pin) /* connect to front */ alc861_auto_set_output_and_unmute(codec, pin, PIN_HP, spec->multiout.dac_nids[0]); + pin = spec->autocfg.speaker_pins[0]; + if (pin) + alc861_auto_set_output_and_unmute(codec, pin, PIN_OUT, 0); } static void alc861_auto_init_analog_input(struct hda_codec *codec) @@ -11568,9 +11917,12 @@ static int alc861_parse_auto_config(struct hda_codec *codec) /* additional initialization for auto-configuration model */ static void alc861_auto_init(struct hda_codec *codec) { + struct alc_spec *spec = codec->spec; alc861_auto_init_multi_out(codec); alc861_auto_init_hp_out(codec); alc861_auto_init_analog_input(codec); + if (spec->unsol_event) + alc_sku_automute(codec); } #ifdef CONFIG_SND_HDA_POWER_SAVE @@ -11822,6 +12174,8 @@ static hda_nid_t alc861vd_adc_nids[1] = { 0x09, }; +static hda_nid_t alc861vd_capsrc_nids[1] = { 0x22 }; + /* input MUX */ /* FIXME: should be a matrix-type input source selection */ static struct hda_input_mux alc861vd_capture_source = { @@ -11835,11 +12189,10 @@ static struct hda_input_mux alc861vd_capture_source = { }; static struct hda_input_mux alc861vd_dallas_capture_source = { - .num_items = 3, + .num_items = 2, .items = { - { "Front Mic", 0x0 }, - { "ATAPI Mic", 0x1 }, - { "Line In", 0x5 }, + { "Ext Mic", 0x0 }, + { "Int Mic", 0x1 }, }, }; @@ -11853,33 +12206,8 @@ static struct hda_input_mux alc861vd_hp_capture_source = { #define alc861vd_mux_enum_info alc_mux_enum_info #define alc861vd_mux_enum_get alc_mux_enum_get - -static int alc861vd_mux_enum_put(struct snd_kcontrol *kcontrol, - struct snd_ctl_elem_value *ucontrol) -{ - struct hda_codec *codec = snd_kcontrol_chip(kcontrol); - struct alc_spec *spec = codec->spec; - const struct hda_input_mux *imux = spec->input_mux; - unsigned int adc_idx = snd_ctl_get_ioffidx(kcontrol, &ucontrol->id); - static hda_nid_t capture_mixers[1] = { 0x22 }; - hda_nid_t nid = capture_mixers[adc_idx]; - unsigned int *cur_val = &spec->cur_mux[adc_idx]; - unsigned int i, idx; - - idx = ucontrol->value.enumerated.item[0]; - if (idx >= imux->num_items) - idx = imux->num_items - 1; - if (*cur_val == idx) - return 0; - for (i = 0; i < imux->num_items; i++) { - unsigned int v = (i == idx) ? 0 : HDA_AMP_MUTE; - snd_hda_codec_amp_stereo(codec, nid, HDA_INPUT, - imux->items[i].index, - HDA_AMP_MUTE, v); - } - *cur_val = idx; - return 1; -} +/* ALC861VD has the ALC882-type input selection (but has only one ADC) */ +#define alc861vd_mux_enum_put alc882_mux_enum_put /* * 2ch mode @@ -12034,20 +12362,22 @@ static struct snd_kcontrol_new alc861vd_lenovo_mixer[] = { { } /* end */ }; -/* Pin assignment: Front=0x14, HP = 0x15, - * Front Mic=0x18, ATAPI Mic = 0x19, Line In = 0x1d +/* Pin assignment: Speaker=0x14, HP = 0x15, + * Ext Mic=0x18, Int Mic = 0x19, CD = 0x1c, PC Beep = 0x1d */ static struct snd_kcontrol_new alc861vd_dallas_mixer[] = { - HDA_CODEC_VOLUME("Front Playback Volume", 0x02, 0x0, HDA_OUTPUT), - HDA_BIND_MUTE("Front Playback Switch", 0x0c, 2, HDA_INPUT), + HDA_CODEC_VOLUME("Speaker Playback Volume", 0x02, 0x0, HDA_OUTPUT), + HDA_BIND_MUTE("Speaker Playback Switch", 0x0c, 2, HDA_INPUT), HDA_CODEC_VOLUME("Headphone Playback Volume", 0x03, 0x0, HDA_OUTPUT), HDA_BIND_MUTE("Headphone Playback Switch", 0x0d, 2, HDA_INPUT), - HDA_CODEC_VOLUME("Front Mic Playback Volume", 0x0b, 0x0, HDA_INPUT), - HDA_CODEC_MUTE("Front Mic Playback Switch", 0x0b, 0x0, HDA_INPUT), - HDA_CODEC_VOLUME("ATAPI Mic Playback Volume", 0x0b, 0x1, HDA_INPUT), - HDA_CODEC_MUTE("ATAPI Mic Playback Switch", 0x0b, 0x1, HDA_INPUT), - HDA_CODEC_VOLUME("Line Playback Volume", 0x0b, 0x05, HDA_INPUT), - HDA_CODEC_MUTE("Line Playback Switch", 0x0b, 0x05, HDA_INPUT), + HDA_CODEC_VOLUME("Ext Mic Boost", 0x18, 0, HDA_INPUT), + HDA_CODEC_VOLUME("Ext Mic Playback Volume", 0x0b, 0x0, HDA_INPUT), + HDA_CODEC_MUTE("Ext Mic Playback Switch", 0x0b, 0x0, HDA_INPUT), + HDA_CODEC_VOLUME("Int Mic Boost", 0x19, 0, HDA_INPUT), + HDA_CODEC_VOLUME("Int Mic Playback Volume", 0x0b, 0x1, HDA_INPUT), + HDA_CODEC_MUTE("Int Mic Playback Switch", 0x0b, 0x1, HDA_INPUT), + HDA_CODEC_VOLUME("PC Beep Volume", 0x0b, 0x05, HDA_INPUT), + HDA_CODEC_MUTE("PC Beep Switch", 0x0b, 0x05, HDA_INPUT), { } /* end */ }; @@ -12348,6 +12678,7 @@ static struct snd_pci_quirk alc861vd_cfg_tbl[] = { /*SND_PCI_QUIRK(0x1179, 0xff00, "DALLAS", ALC861VD_DALLAS),*/ /*lenovo*/ SND_PCI_QUIRK(0x1179, 0xff01, "DALLAS", ALC861VD_DALLAS), SND_PCI_QUIRK(0x1179, 0xff03, "Toshiba P205", ALC861VD_LENOVO), + SND_PCI_QUIRK(0x1179, 0xff31, "Toshiba L30-149", ALC861VD_DALLAS), SND_PCI_QUIRK(0x1565, 0x820d, "Biostar NF61S SE", ALC861VD_6ST_DIG), SND_PCI_QUIRK(0x17aa, 0x2066, "Lenovo", ALC861VD_LENOVO), SND_PCI_QUIRK(0x17aa, 0x3802, "Lenovo 3000 C200", ALC861VD_LENOVO), @@ -12362,8 +12693,6 @@ static struct alc_config_preset alc861vd_presets[] = { alc861vd_3stack_init_verbs }, .num_dacs = ARRAY_SIZE(alc660vd_dac_nids), .dac_nids = alc660vd_dac_nids, - .num_adc_nids = ARRAY_SIZE(alc861vd_adc_nids), - .adc_nids = alc861vd_adc_nids, .num_channel_mode = ARRAY_SIZE(alc861vd_3stack_2ch_modes), .channel_mode = alc861vd_3stack_2ch_modes, .input_mux = &alc861vd_capture_source, @@ -12375,8 +12704,6 @@ static struct alc_config_preset alc861vd_presets[] = { .num_dacs = ARRAY_SIZE(alc660vd_dac_nids), .dac_nids = alc660vd_dac_nids, .dig_out_nid = ALC861VD_DIGOUT_NID, - .num_adc_nids = ARRAY_SIZE(alc861vd_adc_nids), - .adc_nids = alc861vd_adc_nids, .num_channel_mode = ARRAY_SIZE(alc861vd_3stack_2ch_modes), .channel_mode = alc861vd_3stack_2ch_modes, .input_mux = &alc861vd_capture_source, @@ -12421,8 +12748,6 @@ static struct alc_config_preset alc861vd_presets[] = { alc861vd_lenovo_unsol_verbs }, .num_dacs = ARRAY_SIZE(alc660vd_dac_nids), .dac_nids = alc660vd_dac_nids, - .num_adc_nids = ARRAY_SIZE(alc861vd_adc_nids), - .adc_nids = alc861vd_adc_nids, .num_channel_mode = ARRAY_SIZE(alc861vd_3stack_2ch_modes), .channel_mode = alc861vd_3stack_2ch_modes, .input_mux = &alc861vd_capture_source, @@ -12434,8 +12759,6 @@ static struct alc_config_preset alc861vd_presets[] = { .init_verbs = { alc861vd_dallas_verbs }, .num_dacs = ARRAY_SIZE(alc861vd_dac_nids), .dac_nids = alc861vd_dac_nids, - .num_adc_nids = ARRAY_SIZE(alc861vd_adc_nids), - .adc_nids = alc861vd_adc_nids, .num_channel_mode = ARRAY_SIZE(alc861vd_3stack_2ch_modes), .channel_mode = alc861vd_3stack_2ch_modes, .input_mux = &alc861vd_dallas_capture_source, @@ -12447,9 +12770,7 @@ static struct alc_config_preset alc861vd_presets[] = { .init_verbs = { alc861vd_dallas_verbs, alc861vd_eapd_verbs }, .num_dacs = ARRAY_SIZE(alc861vd_dac_nids), .dac_nids = alc861vd_dac_nids, - .num_adc_nids = ARRAY_SIZE(alc861vd_adc_nids), .dig_out_nid = ALC861VD_DIGOUT_NID, - .adc_nids = alc861vd_adc_nids, .num_channel_mode = ARRAY_SIZE(alc861vd_3stack_2ch_modes), .channel_mode = alc861vd_3stack_2ch_modes, .input_mux = &alc861vd_hp_capture_source, @@ -12464,11 +12785,7 @@ static struct alc_config_preset alc861vd_presets[] = { static void alc861vd_auto_set_output_and_unmute(struct hda_codec *codec, hda_nid_t nid, int pin_type, int dac_idx) { - /* set as output */ - snd_hda_codec_write(codec, nid, 0, - AC_VERB_SET_PIN_WIDGET_CONTROL, pin_type); - snd_hda_codec_write(codec, nid, 0, - AC_VERB_SET_AMP_GAIN_MUTE, AMP_OUT_UNMUTE); + alc_set_pin_output(codec, nid, pin_type); } static void alc861vd_auto_init_multi_out(struct hda_codec *codec) @@ -12495,6 +12812,9 @@ static void alc861vd_auto_init_hp_out(struct hda_codec *codec) pin = spec->autocfg.hp_pins[0]; if (pin) /* connect to front and use dac 0 */ alc861vd_auto_set_output_and_unmute(codec, pin, PIN_HP, 0); + pin = spec->autocfg.speaker_pins[0]; + if (pin) + alc861vd_auto_set_output_and_unmute(codec, pin, PIN_OUT, 0); } #define alc861vd_is_input_pin(nid) alc880_is_input_pin(nid) @@ -12698,9 +13018,12 @@ static int alc861vd_parse_auto_config(struct hda_codec *codec) /* additional initialization for auto-configuration model */ static void alc861vd_auto_init(struct hda_codec *codec) { + struct alc_spec *spec = codec->spec; alc861vd_auto_init_multi_out(codec); alc861vd_auto_init_hp_out(codec); alc861vd_auto_init_analog_input(codec); + if (spec->unsol_event) + alc_sku_automute(codec); } static int patch_alc861vd(struct hda_codec *codec) @@ -12751,6 +13074,7 @@ static int patch_alc861vd(struct hda_codec *codec) spec->adc_nids = alc861vd_adc_nids; spec->num_adc_nids = ARRAY_SIZE(alc861vd_adc_nids); + spec->capsrc_nids = alc861vd_capsrc_nids; spec->mixers[spec->num_mixers] = alc861vd_capture_mixer; spec->num_mixers++; @@ -12792,9 +13116,11 @@ static hda_nid_t alc662_adc_nids[1] = { /* ADC1-2 */ 0x09, }; + +static hda_nid_t alc662_capsrc_nids[1] = { 0x22 }; + /* input MUX */ /* FIXME: should be a matrix-type input source selection */ - static struct hda_input_mux alc662_capture_source = { .num_items = 4, .items = { @@ -12823,33 +13149,8 @@ static struct hda_input_mux alc662_eeepc_capture_source = { #define alc662_mux_enum_info alc_mux_enum_info #define alc662_mux_enum_get alc_mux_enum_get +#define alc662_mux_enum_put alc882_mux_enum_put -static int alc662_mux_enum_put(struct snd_kcontrol *kcontrol, - struct snd_ctl_elem_value *ucontrol) -{ - struct hda_codec *codec = snd_kcontrol_chip(kcontrol); - struct alc_spec *spec = codec->spec; - const struct hda_input_mux *imux = spec->input_mux; - unsigned int adc_idx = snd_ctl_get_ioffidx(kcontrol, &ucontrol->id); - static hda_nid_t capture_mixers[2] = { 0x23, 0x22 }; - hda_nid_t nid = capture_mixers[adc_idx]; - unsigned int *cur_val = &spec->cur_mux[adc_idx]; - unsigned int i, idx; - - idx = ucontrol->value.enumerated.item[0]; - if (idx >= imux->num_items) - idx = imux->num_items - 1; - if (*cur_val == idx) - return 0; - for (i = 0; i < imux->num_items; i++) { - unsigned int v = (i == idx) ? 0 : HDA_AMP_MUTE; - snd_hda_codec_amp_stereo(codec, nid, HDA_INPUT, - imux->items[i].index, - HDA_AMP_MUTE, v); - } - *cur_val = idx; - return 1; -} /* * 2ch mode */ @@ -12918,13 +13219,13 @@ static struct hda_channel_mode alc662_5stack_modes[2] = { static struct snd_kcontrol_new alc662_base_mixer[] = { /* output mixer control */ HDA_CODEC_VOLUME("Front Playback Volume", 0x2, 0x0, HDA_OUTPUT), - HDA_CODEC_MUTE("Front Playback Switch", 0x02, 0x0, HDA_OUTPUT), + HDA_CODEC_MUTE("Front Playback Switch", 0x0c, 0x0, HDA_INPUT), HDA_CODEC_VOLUME("Surround Playback Volume", 0x3, 0x0, HDA_OUTPUT), - HDA_CODEC_MUTE("Surround Playback Switch", 0x03, 0x0, HDA_OUTPUT), + HDA_CODEC_MUTE("Surround Playback Switch", 0x0d, 0x0, HDA_INPUT), HDA_CODEC_VOLUME_MONO("Center Playback Volume", 0x04, 1, 0x0, HDA_OUTPUT), HDA_CODEC_VOLUME_MONO("LFE Playback Volume", 0x04, 2, 0x0, HDA_OUTPUT), - HDA_BIND_MUTE_MONO("Center Playback Switch", 0x04, 1, 2, HDA_INPUT), - HDA_BIND_MUTE_MONO("LFE Playback Switch", 0x04, 2, 2, HDA_INPUT), + HDA_CODEC_MUTE_MONO("Center Playback Switch", 0x0e, 1, 0x0, HDA_INPUT), + HDA_CODEC_MUTE_MONO("LFE Playback Switch", 0x0e, 2, 0x0, HDA_INPUT), HDA_CODEC_MUTE("Headphone Playback Switch", 0x1b, 0x0, HDA_OUTPUT), /*Input mixer control */ @@ -12941,7 +13242,7 @@ static struct snd_kcontrol_new alc662_base_mixer[] = { static struct snd_kcontrol_new alc662_3ST_2ch_mixer[] = { HDA_CODEC_VOLUME("Front Playback Volume", 0x02, 0x0, HDA_OUTPUT), - HDA_BIND_MUTE("Front Playback Switch", 0x02, 2, HDA_INPUT), + HDA_CODEC_MUTE("Front Playback Switch", 0x0c, 0x0, HDA_INPUT), HDA_CODEC_MUTE("Headphone Playback Switch", 0x1b, 0x0, HDA_OUTPUT), HDA_CODEC_VOLUME("CD Playback Volume", 0x0b, 0x04, HDA_INPUT), HDA_CODEC_MUTE("CD Playback Switch", 0x0b, 0x04, HDA_INPUT), @@ -12958,13 +13259,13 @@ static struct snd_kcontrol_new alc662_3ST_2ch_mixer[] = { static struct snd_kcontrol_new alc662_3ST_6ch_mixer[] = { HDA_CODEC_VOLUME("Front Playback Volume", 0x02, 0x0, HDA_OUTPUT), - HDA_BIND_MUTE("Front Playback Switch", 0x02, 2, HDA_INPUT), + HDA_CODEC_MUTE("Front Playback Switch", 0x0c, 0x0, HDA_INPUT), HDA_CODEC_VOLUME("Surround Playback Volume", 0x03, 0x0, HDA_OUTPUT), - HDA_BIND_MUTE("Surround Playback Switch", 0x03, 2, HDA_INPUT), + HDA_CODEC_MUTE("Surround Playback Switch", 0x0d, 0x0, HDA_INPUT), HDA_CODEC_VOLUME_MONO("Center Playback Volume", 0x04, 1, 0x0, HDA_OUTPUT), HDA_CODEC_VOLUME_MONO("LFE Playback Volume", 0x04, 2, 0x0, HDA_OUTPUT), - HDA_BIND_MUTE_MONO("Center Playback Switch", 0x04, 1, 2, HDA_INPUT), - HDA_BIND_MUTE_MONO("LFE Playback Switch", 0x04, 2, 2, HDA_INPUT), + HDA_CODEC_MUTE_MONO("Center Playback Switch", 0x0e, 1, 0x0, HDA_INPUT), + HDA_CODEC_MUTE_MONO("LFE Playback Switch", 0x0e, 2, 0x0, HDA_INPUT), HDA_CODEC_MUTE("Headphone Playback Switch", 0x1b, 0x0, HDA_OUTPUT), HDA_CODEC_VOLUME("CD Playback Volume", 0x0b, 0x04, HDA_INPUT), HDA_CODEC_MUTE("CD Playback Switch", 0x0b, 0x04, HDA_INPUT), @@ -13313,6 +13614,7 @@ static const char *alc662_models[ALC662_MODEL_LAST] = { }; static struct snd_pci_quirk alc662_cfg_tbl[] = { + SND_PCI_QUIRK(0x1043, 0x8290, "ASUS P5GC-MX", ALC662_3ST_6ch_DIG), SND_PCI_QUIRK(0x1043, 0x82a1, "ASUS Eeepc", ALC662_ASUS_EEEPC_P701), SND_PCI_QUIRK(0x1043, 0x82d1, "ASUS Eeepc EP20", ALC662_ASUS_EEEPC_EP20), SND_PCI_QUIRK(0x17aa, 0x101e, "Lenovo", ALC662_LENOVO_101E), @@ -13326,8 +13628,6 @@ static struct alc_config_preset alc662_presets[] = { .num_dacs = ARRAY_SIZE(alc662_dac_nids), .dac_nids = alc662_dac_nids, .dig_out_nid = ALC662_DIGOUT_NID, - .num_adc_nids = ARRAY_SIZE(alc662_adc_nids), - .adc_nids = alc662_adc_nids, .dig_in_nid = ALC662_DIGIN_NID, .num_channel_mode = ARRAY_SIZE(alc662_3ST_2ch_modes), .channel_mode = alc662_3ST_2ch_modes, @@ -13340,8 +13640,6 @@ static struct alc_config_preset alc662_presets[] = { .num_dacs = ARRAY_SIZE(alc662_dac_nids), .dac_nids = alc662_dac_nids, .dig_out_nid = ALC662_DIGOUT_NID, - .num_adc_nids = ARRAY_SIZE(alc662_adc_nids), - .adc_nids = alc662_adc_nids, .dig_in_nid = ALC662_DIGIN_NID, .num_channel_mode = ARRAY_SIZE(alc662_3ST_6ch_modes), .channel_mode = alc662_3ST_6ch_modes, @@ -13354,8 +13652,6 @@ static struct alc_config_preset alc662_presets[] = { .init_verbs = { alc662_init_verbs }, .num_dacs = ARRAY_SIZE(alc662_dac_nids), .dac_nids = alc662_dac_nids, - .num_adc_nids = ARRAY_SIZE(alc662_adc_nids), - .adc_nids = alc662_adc_nids, .num_channel_mode = ARRAY_SIZE(alc662_3ST_6ch_modes), .channel_mode = alc662_3ST_6ch_modes, .need_dac_fix = 1, @@ -13368,8 +13664,6 @@ static struct alc_config_preset alc662_presets[] = { .num_dacs = ARRAY_SIZE(alc662_dac_nids), .dac_nids = alc662_dac_nids, .dig_out_nid = ALC662_DIGOUT_NID, - .num_adc_nids = ARRAY_SIZE(alc662_adc_nids), - .adc_nids = alc662_adc_nids, .dig_in_nid = ALC662_DIGIN_NID, .num_channel_mode = ARRAY_SIZE(alc662_5stack_modes), .channel_mode = alc662_5stack_modes, @@ -13380,8 +13674,6 @@ static struct alc_config_preset alc662_presets[] = { .init_verbs = { alc662_init_verbs, alc662_sue_init_verbs }, .num_dacs = ARRAY_SIZE(alc662_dac_nids), .dac_nids = alc662_dac_nids, - .num_adc_nids = ARRAY_SIZE(alc662_adc_nids), - .adc_nids = alc662_adc_nids, .num_channel_mode = ARRAY_SIZE(alc662_3ST_2ch_modes), .channel_mode = alc662_3ST_2ch_modes, .input_mux = &alc662_lenovo_101e_capture_source, @@ -13394,8 +13686,6 @@ static struct alc_config_preset alc662_presets[] = { alc662_eeepc_sue_init_verbs }, .num_dacs = ARRAY_SIZE(alc662_dac_nids), .dac_nids = alc662_dac_nids, - .num_adc_nids = ARRAY_SIZE(alc861vd_adc_nids), - .adc_nids = alc662_adc_nids, .num_channel_mode = ARRAY_SIZE(alc662_3ST_2ch_modes), .channel_mode = alc662_3ST_2ch_modes, .input_mux = &alc662_eeepc_capture_source, @@ -13409,8 +13699,6 @@ static struct alc_config_preset alc662_presets[] = { alc662_eeepc_ep20_sue_init_verbs }, .num_dacs = ARRAY_SIZE(alc662_dac_nids), .dac_nids = alc662_dac_nids, - .num_adc_nids = ARRAY_SIZE(alc662_adc_nids), - .adc_nids = alc662_adc_nids, .num_channel_mode = ARRAY_SIZE(alc662_3ST_6ch_modes), .channel_mode = alc662_3ST_6ch_modes, .input_mux = &alc662_lenovo_101e_capture_source, @@ -13556,11 +13844,7 @@ static void alc662_auto_set_output_and_unmute(struct hda_codec *codec, hda_nid_t nid, int pin_type, int dac_idx) { - /* set as output */ - snd_hda_codec_write(codec, nid, 0, - AC_VERB_SET_PIN_WIDGET_CONTROL, pin_type); - snd_hda_codec_write(codec, nid, 0, - AC_VERB_SET_AMP_GAIN_MUTE, AMP_OUT_UNMUTE); + alc_set_pin_output(codec, nid, pin_type); /* need the manual connection? */ if (alc880_is_multi_pin(nid)) { struct alc_spec *spec = codec->spec; @@ -13595,6 +13879,9 @@ static void alc662_auto_init_hp_out(struct hda_codec *codec) if (pin) /* connect to front */ /* use dac 0 */ alc662_auto_set_output_and_unmute(codec, pin, PIN_HP, 0); + pin = spec->autocfg.speaker_pins[0]; + if (pin) + alc662_auto_set_output_and_unmute(codec, pin, PIN_OUT, 0); } #define alc662_is_input_pin(nid) alc880_is_input_pin(nid) @@ -13672,9 +13959,12 @@ static int alc662_parse_auto_config(struct hda_codec *codec) /* additional initialization for auto-configuration model */ static void alc662_auto_init(struct hda_codec *codec) { + struct alc_spec *spec = codec->spec; alc662_auto_init_multi_out(codec); alc662_auto_init_hp_out(codec); alc662_auto_init_analog_input(codec); + if (spec->unsol_event) + alc_sku_automute(codec); } static int patch_alc662(struct hda_codec *codec) @@ -13722,10 +14012,9 @@ static int patch_alc662(struct hda_codec *codec) spec->stream_digital_playback = &alc662_pcm_digital_playback; spec->stream_digital_capture = &alc662_pcm_digital_capture; - if (!spec->adc_nids && spec->input_mux) { - spec->adc_nids = alc662_adc_nids; - spec->num_adc_nids = ARRAY_SIZE(alc662_adc_nids); - } + spec->adc_nids = alc662_adc_nids; + spec->num_adc_nids = ARRAY_SIZE(alc662_adc_nids); + spec->capsrc_nids = alc662_capsrc_nids; spec->vmaster_nid = 0x02; @@ -13761,6 +14050,8 @@ struct hda_codec_preset snd_hda_preset_realtek[] = { { .id = 0x10ec0880, .name = "ALC880", .patch = patch_alc880 }, { .id = 0x10ec0882, .name = "ALC882", .patch = patch_alc882 }, { .id = 0x10ec0883, .name = "ALC883", .patch = patch_alc883 }, + { .id = 0x10ec0885, .rev = 0x100103, .name = "ALC889A", + .patch = patch_alc882 }, /* should be patch_alc883() in future */ { .id = 0x10ec0885, .name = "ALC885", .patch = patch_alc882 }, { .id = 0x10ec0888, .name = "ALC888", .patch = patch_alc883 }, { .id = 0x10ec0889, .name = "ALC889", .patch = patch_alc883 }, diff --git a/sound/pci/hda/patch_si3054.c b/sound/pci/hda/patch_si3054.c index d22f5a6..9332b63 100644 --- a/sound/pci/hda/patch_si3054.c +++ b/sound/pci/hda/patch_si3054.c @@ -28,7 +28,7 @@ #include #include "hda_codec.h" #include "hda_local.h" - +#include "hda_patch.h" /* si3054 verbs */ #define SI3054_VERB_READ_NODE 0x900 @@ -206,7 +206,7 @@ static int si3054_build_pcms(struct hda_codec *codec) info->name = "Si3054 Modem"; info->stream[SNDRV_PCM_STREAM_PLAYBACK] = si3054_pcm; info->stream[SNDRV_PCM_STREAM_CAPTURE] = si3054_pcm; - info->is_modem = 1; + info->pcm_type = HDA_PCM_TYPE_MODEM; return 0; } diff --git a/sound/pci/hda/patch_sigmatel.c b/sound/pci/hda/patch_sigmatel.c index caf48ed..b3a15d6 100644 --- a/sound/pci/hda/patch_sigmatel.c +++ b/sound/pci/hda/patch_sigmatel.c @@ -32,6 +32,7 @@ #include #include "hda_codec.h" #include "hda_local.h" +#include "hda_patch.h" #define NUM_CONTROL_ALLOC 32 #define STAC_PWR_EVENT 0x20 @@ -39,6 +40,7 @@ enum { STAC_REF, + STAC_9200_OQO, STAC_9200_DELL_D21, STAC_9200_DELL_D22, STAC_9200_DELL_D23, @@ -50,6 +52,7 @@ enum { STAC_9200_DELL_M26, STAC_9200_DELL_M27, STAC_9200_GATEWAY, + STAC_9200_PANASONIC, STAC_9200_MODELS }; @@ -63,11 +66,14 @@ enum { enum { STAC_92HD73XX_REF, + STAC_DELL_M6, STAC_92HD73XX_MODELS }; enum { STAC_92HD71BXX_REF, + STAC_DELL_M4_1, + STAC_DELL_M4_2, STAC_92HD71BXX_MODELS }; @@ -123,6 +129,7 @@ struct sigmatel_spec { unsigned int hp_detect: 1; /* gpio lines */ + unsigned int eapd_mask; unsigned int gpio_mask; unsigned int gpio_dir; unsigned int gpio_data; @@ -135,6 +142,7 @@ struct sigmatel_spec { /* power management */ unsigned int num_pwrs; hda_nid_t *pwr_nids; + hda_nid_t *dac_list; /* playback */ struct hda_input_mux *mono_mux; @@ -173,6 +181,7 @@ struct sigmatel_spec { /* i/o switches */ unsigned int io_switch[2]; unsigned int clfe_swap; + unsigned int hp_switch; unsigned int aloopback; struct hda_pcm pcm_rec[2]; /* PCM information */ @@ -184,9 +193,6 @@ struct sigmatel_spec { struct hda_input_mux private_dimux; struct hda_input_mux private_imux; struct hda_input_mux private_mono_mux; - - /* virtual master */ - unsigned int vmaster_tlv[4]; }; static hda_nid_t stac9200_adc_nids[1] = { @@ -244,7 +250,7 @@ static hda_nid_t stac92hd71bxx_dmux_nids[1] = { 0x1c, }; -static hda_nid_t stac92hd71bxx_dac_nids[2] = { +static hda_nid_t stac92hd71bxx_dac_nids[1] = { 0x10, /*0x11, */ }; @@ -290,6 +296,10 @@ static hda_nid_t stac927x_mux_nids[3] = { 0x15, 0x16, 0x17 }; +static hda_nid_t stac927x_dac_nids[6] = { + 0x02, 0x03, 0x04, 0x05, 0x06, 0 +}; + static hda_nid_t stac927x_dmux_nids[1] = { 0x1b, }; @@ -331,10 +341,10 @@ static hda_nid_t stac922x_pin_nids[10] = { 0x0f, 0x10, 0x11, 0x15, 0x1b, }; -static hda_nid_t stac92hd73xx_pin_nids[12] = { +static hda_nid_t stac92hd73xx_pin_nids[13] = { 0x0a, 0x0b, 0x0c, 0x0d, 0x0e, 0x0f, 0x10, 0x11, 0x12, 0x13, - 0x14, 0x22 + 0x14, 0x1e, 0x22 }; static hda_nid_t stac92hd71bxx_pin_nids[10] = { @@ -527,6 +537,43 @@ static struct hda_verb stac92hd73xx_6ch_core_init[] = { {} }; +static struct hda_verb dell_eq_core_init[] = { + /* set master volume to max value without distortion + * and direct control */ + { 0x1f, AC_VERB_SET_VOLUME_KNOB_CONTROL, 0xec}, + /* setup audio connections */ + { 0x0d, AC_VERB_SET_CONNECT_SEL, 0x00}, + { 0x0a, AC_VERB_SET_CONNECT_SEL, 0x01}, + { 0x0f, AC_VERB_SET_CONNECT_SEL, 0x02}, + /* setup adcs to point to mixer */ + { 0x20, AC_VERB_SET_CONNECT_SEL, 0x0b}, + { 0x21, AC_VERB_SET_CONNECT_SEL, 0x0b}, + /* setup import muxs */ + { 0x28, AC_VERB_SET_CONNECT_SEL, 0x01}, + { 0x29, AC_VERB_SET_CONNECT_SEL, 0x01}, + { 0x2a, AC_VERB_SET_CONNECT_SEL, 0x01}, + { 0x2b, AC_VERB_SET_CONNECT_SEL, 0x00}, + {} +}; + +static struct hda_verb dell_m6_core_init[] = { + /* set master volume and direct control */ + { 0x1f, AC_VERB_SET_VOLUME_KNOB_CONTROL, 0xff}, + /* setup audio connections */ + { 0x0d, AC_VERB_SET_CONNECT_SEL, 0x00}, + { 0x0a, AC_VERB_SET_CONNECT_SEL, 0x01}, + { 0x0f, AC_VERB_SET_CONNECT_SEL, 0x02}, + /* setup adcs to point to mixer */ + { 0x20, AC_VERB_SET_CONNECT_SEL, 0x0b}, + { 0x21, AC_VERB_SET_CONNECT_SEL, 0x0b}, + /* setup import muxs */ + { 0x28, AC_VERB_SET_CONNECT_SEL, 0x01}, + { 0x29, AC_VERB_SET_CONNECT_SEL, 0x01}, + { 0x2a, AC_VERB_SET_CONNECT_SEL, 0x01}, + { 0x2b, AC_VERB_SET_CONNECT_SEL, 0x00}, + {} +}; + static struct hda_verb stac92hd73xx_8ch_core_init[] = { /* set master volume and direct control */ { 0x1f, AC_VERB_SET_VOLUME_KNOB_CONTROL, 0xff}, @@ -910,6 +957,11 @@ static int stac92xx_build_controls(struct hda_codec *codec) err = snd_hda_create_spdif_out_ctls(codec, spec->multiout.dig_out_nid); if (err < 0) return err; + err = snd_hda_create_spdif_share_sw(codec, + &spec->multiout); + if (err < 0) + return err; + spec->multiout.share_spdif = 1; } if (spec->dig_in_nid) { err = snd_hda_create_spdif_in_ctls(codec, spec->dig_in_nid); @@ -919,10 +971,11 @@ static int stac92xx_build_controls(struct hda_codec *codec) /* if we have no master control, let's create it */ if (!snd_hda_find_mixer_ctl(codec, "Master Playback Volume")) { + unsigned int vmaster_tlv[4]; snd_hda_set_vmaster_tlv(codec, spec->multiout.dac_nids[0], - HDA_OUTPUT, spec->vmaster_tlv); + HDA_OUTPUT, vmaster_tlv); err = snd_hda_add_vmaster(codec, "Master Playback Volume", - spec->vmaster_tlv, slave_vols); + vmaster_tlv, slave_vols); if (err < 0) return err; } @@ -1052,9 +1105,15 @@ static unsigned int dell9200_m27_pin_configs[8] = { 0x90170310, 0x04a11020, 0x90170310, 0x40f003fc, }; +static unsigned int oqo9200_pin_configs[8] = { + 0x40c000f0, 0x404000f1, 0x0221121f, 0x02211210, + 0x90170111, 0x90a70120, 0x400000f2, 0x400000f3, +}; + static unsigned int *stac9200_brd_tbl[STAC_9200_MODELS] = { [STAC_REF] = ref9200_pin_configs, + [STAC_9200_OQO] = oqo9200_pin_configs, [STAC_9200_DELL_D21] = dell9200_d21_pin_configs, [STAC_9200_DELL_D22] = dell9200_d22_pin_configs, [STAC_9200_DELL_D23] = dell9200_d23_pin_configs, @@ -1065,10 +1124,12 @@ static unsigned int *stac9200_brd_tbl[STAC_9200_MODELS] = { [STAC_9200_DELL_M25] = dell9200_m25_pin_configs, [STAC_9200_DELL_M26] = dell9200_m26_pin_configs, [STAC_9200_DELL_M27] = dell9200_m27_pin_configs, + [STAC_9200_PANASONIC] = ref9200_pin_configs, }; static const char *stac9200_models[STAC_9200_MODELS] = { [STAC_REF] = "ref", + [STAC_9200_OQO] = "oqo", [STAC_9200_DELL_D21] = "dell-d21", [STAC_9200_DELL_D22] = "dell-d22", [STAC_9200_DELL_D23] = "dell-d23", @@ -1080,6 +1141,7 @@ static const char *stac9200_models[STAC_9200_MODELS] = { [STAC_9200_DELL_M26] = "dell-m26", [STAC_9200_DELL_M27] = "dell-m27", [STAC_9200_GATEWAY] = "gateway", + [STAC_9200_PANASONIC] = "panasonic", }; static struct snd_pci_quirk stac9200_cfg_tbl[] = { @@ -1146,13 +1208,15 @@ static struct snd_pci_quirk stac9200_cfg_tbl[] = { SND_PCI_QUIRK(PCI_VENDOR_ID_DELL, 0x01f6, "unknown Dell", STAC_9200_DELL_M26), /* Panasonic */ - SND_PCI_QUIRK(0x10f7, 0x8338, "Panasonic CF-74", STAC_REF), + SND_PCI_QUIRK(0x10f7, 0x8338, "Panasonic CF-74", STAC_9200_PANASONIC), /* Gateway machines needs EAPD to be set on resume */ SND_PCI_QUIRK(0x107b, 0x0205, "Gateway S-7110M", STAC_9200_GATEWAY), SND_PCI_QUIRK(0x107b, 0x0317, "Gateway MT3423, MX341*", STAC_9200_GATEWAY), SND_PCI_QUIRK(0x107b, 0x0318, "Gateway ML3019, MT3707", STAC_9200_GATEWAY), + /* OQO Mobile */ + SND_PCI_QUIRK(0x1106, 0x3288, "OQO Model 2", STAC_9200_OQO), {} /* terminator */ }; @@ -1202,24 +1266,48 @@ static struct snd_pci_quirk stac925x_cfg_tbl[] = { {} /* terminator */ }; -static unsigned int ref92hd73xx_pin_configs[12] = { +static unsigned int ref92hd73xx_pin_configs[13] = { 0x02214030, 0x02a19040, 0x01a19020, 0x02214030, 0x0181302e, 0x01014010, 0x01014020, 0x01014030, 0x02319040, 0x90a000f0, 0x90a000f0, 0x01452050, + 0x01452050, +}; + +static unsigned int dell_m6_pin_configs[13] = { + 0x0321101f, 0x4f00000f, 0x4f0000f0, 0x90170110, + 0x03a11020, 0x0321101f, 0x4f0000f0, 0x4f0000f0, + 0x4f0000f0, 0x90a60160, 0x4f0000f0, 0x4f0000f0, + 0x4f0000f0, }; static unsigned int *stac92hd73xx_brd_tbl[STAC_92HD73XX_MODELS] = { - [STAC_92HD73XX_REF] = ref92hd73xx_pin_configs, + [STAC_92HD73XX_REF] = ref92hd73xx_pin_configs, + [STAC_DELL_M6] = dell_m6_pin_configs, }; static const char *stac92hd73xx_models[STAC_92HD73XX_MODELS] = { [STAC_92HD73XX_REF] = "ref", + [STAC_DELL_M6] = "dell-m6", }; static struct snd_pci_quirk stac92hd73xx_cfg_tbl[] = { /* SigmaTel reference board */ SND_PCI_QUIRK(PCI_VENDOR_ID_INTEL, 0x2668, - "DFI LanParty", STAC_92HD73XX_REF), + "DFI LanParty", STAC_92HD73XX_REF), + SND_PCI_QUIRK(PCI_VENDOR_ID_DELL, 0x0254, + "unknown Dell", STAC_DELL_M6), + SND_PCI_QUIRK(PCI_VENDOR_ID_DELL, 0x0255, + "unknown Dell", STAC_DELL_M6), + SND_PCI_QUIRK(PCI_VENDOR_ID_DELL, 0x0256, + "unknown Dell", STAC_DELL_M6), + SND_PCI_QUIRK(PCI_VENDOR_ID_DELL, 0x0257, + "unknown Dell", STAC_DELL_M6), + SND_PCI_QUIRK(PCI_VENDOR_ID_DELL, 0x025e, + "unknown Dell", STAC_DELL_M6), + SND_PCI_QUIRK(PCI_VENDOR_ID_DELL, 0x025f, + "unknown Dell", STAC_DELL_M6), + SND_PCI_QUIRK(PCI_VENDOR_ID_DELL, 0x0271, + "unknown Dell", STAC_DELL_M6), {} /* terminator */ }; @@ -1229,18 +1317,56 @@ static unsigned int ref92hd71bxx_pin_configs[10] = { 0x90a000f0, 0x01452050, }; +static unsigned int dell_m4_1_pin_configs[13] = { + 0x0421101f, 0x04a11221, 0x40f000f0, 0x90170110, + 0x23a1902e, 0x23014250, 0x40f000f0, 0x90a000f0, + 0x40f000f0, 0x4f0000f0, +}; + +static unsigned int dell_m4_2_pin_configs[13] = { + 0x0421101f, 0x04a11221, 0x90a70330, 0x90170110, + 0x23a1902e, 0x23014250, 0x40f000f0, 0x40f000f0, + 0x40f000f0, 0x044413b0, +}; + static unsigned int *stac92hd71bxx_brd_tbl[STAC_92HD71BXX_MODELS] = { [STAC_92HD71BXX_REF] = ref92hd71bxx_pin_configs, + [STAC_DELL_M4_1] = dell_m4_1_pin_configs, + [STAC_DELL_M4_2] = dell_m4_2_pin_configs, }; static const char *stac92hd71bxx_models[STAC_92HD71BXX_MODELS] = { [STAC_92HD71BXX_REF] = "ref", + [STAC_DELL_M4_1] = "dell-m4-1", + [STAC_DELL_M4_2] = "dell-m4-2", }; static struct snd_pci_quirk stac92hd71bxx_cfg_tbl[] = { /* SigmaTel reference board */ SND_PCI_QUIRK(PCI_VENDOR_ID_INTEL, 0x2668, "DFI LanParty", STAC_92HD71BXX_REF), + SND_PCI_QUIRK(PCI_VENDOR_ID_DELL, 0x0233, + "unknown Dell", STAC_DELL_M4_1), + SND_PCI_QUIRK(PCI_VENDOR_ID_DELL, 0x0234, + "unknown Dell", STAC_DELL_M4_1), + SND_PCI_QUIRK(PCI_VENDOR_ID_DELL, 0x0250, + "unknown Dell", STAC_DELL_M4_1), + SND_PCI_QUIRK(PCI_VENDOR_ID_DELL, 0x024f, + "unknown Dell", STAC_DELL_M4_1), + SND_PCI_QUIRK(PCI_VENDOR_ID_DELL, 0x024d, + "unknown Dell", STAC_DELL_M4_1), + SND_PCI_QUIRK(PCI_VENDOR_ID_DELL, 0x0251, + "unknown Dell", STAC_DELL_M4_1), + SND_PCI_QUIRK(PCI_VENDOR_ID_DELL, 0x0277, + "unknown Dell", STAC_DELL_M4_1), + SND_PCI_QUIRK(PCI_VENDOR_ID_DELL, 0x0263, + "unknown Dell", STAC_DELL_M4_2), + SND_PCI_QUIRK(PCI_VENDOR_ID_DELL, 0x0265, + "unknown Dell", STAC_DELL_M4_2), + SND_PCI_QUIRK(PCI_VENDOR_ID_DELL, 0x0262, + "unknown Dell", STAC_DELL_M4_2), + SND_PCI_QUIRK(PCI_VENDOR_ID_DELL, 0x0264, + "unknown Dell", STAC_DELL_M4_2), {} /* terminator */ }; @@ -1733,7 +1859,8 @@ static int stac92xx_playback_pcm_open(struct hda_pcm_stream *hinfo, struct snd_pcm_substream *substream) { struct sigmatel_spec *spec = codec->spec; - return snd_hda_multi_out_analog_open(codec, &spec->multiout, substream); + return snd_hda_multi_out_analog_open(codec, &spec->multiout, substream, + hinfo); } static int stac92xx_playback_pcm_prepare(struct hda_pcm_stream *hinfo, @@ -1807,7 +1934,7 @@ static int stac92xx_capture_pcm_cleanup(struct hda_pcm_stream *hinfo, { struct sigmatel_spec *spec = codec->spec; - snd_hda_codec_setup_stream(codec, spec->adc_nids[substream->number], 0, 0, 0); + snd_hda_codec_cleanup_stream(codec, spec->adc_nids[substream->number]); return 0; } @@ -1889,6 +2016,7 @@ static int stac92xx_build_pcms(struct hda_codec *codec) codec->num_pcms++; info++; info->name = "STAC92xx Digital"; + info->pcm_type = HDA_PCM_TYPE_SPDIF; if (spec->multiout.dig_out_nid) { info->stream[SNDRV_PCM_STREAM_PLAYBACK] = stac92xx_pcm_digital_playback; info->stream[SNDRV_PCM_STREAM_PLAYBACK].nid = spec->multiout.dig_out_nid; @@ -1925,6 +2053,34 @@ static void stac92xx_auto_set_pinctl(struct hda_codec *codec, hda_nid_t nid, int AC_VERB_SET_PIN_WIDGET_CONTROL, pin_type); } +#define stac92xx_hp_switch_info snd_ctl_boolean_mono_info + +static int stac92xx_hp_switch_get(struct snd_kcontrol *kcontrol, + struct snd_ctl_elem_value *ucontrol) +{ + struct hda_codec *codec = snd_kcontrol_chip(kcontrol); + struct sigmatel_spec *spec = codec->spec; + + ucontrol->value.integer.value[0] = spec->hp_switch; + return 0; +} + +static int stac92xx_hp_switch_put(struct snd_kcontrol *kcontrol, + struct snd_ctl_elem_value *ucontrol) +{ + struct hda_codec *codec = snd_kcontrol_chip(kcontrol); + struct sigmatel_spec *spec = codec->spec; + + spec->hp_switch = ucontrol->value.integer.value[0]; + + /* check to be sure that the ports are upto date with + * switch changes + */ + codec->patch_ops.unsol_event(codec, STAC_HP_EVENT << 26); + + return 1; +} + #define stac92xx_io_switch_info snd_ctl_boolean_mono_info static int stac92xx_io_switch_get(struct snd_kcontrol *kcontrol, struct snd_ctl_elem_value *ucontrol) @@ -1996,6 +2152,15 @@ static int stac92xx_clfe_switch_put(struct snd_kcontrol *kcontrol, return 1; } +#define STAC_CODEC_HP_SWITCH(xname) \ + { .iface = SNDRV_CTL_ELEM_IFACE_MIXER, \ + .name = xname, \ + .index = 0, \ + .info = stac92xx_hp_switch_info, \ + .get = stac92xx_hp_switch_get, \ + .put = stac92xx_hp_switch_put, \ + } + #define STAC_CODEC_IO_SWITCH(xname, xpval) \ { .iface = SNDRV_CTL_ELEM_IFACE_MIXER, \ .name = xname, \ @@ -2020,6 +2185,7 @@ enum { STAC_CTL_WIDGET_VOL, STAC_CTL_WIDGET_MUTE, STAC_CTL_WIDGET_MONO_MUX, + STAC_CTL_WIDGET_HP_SWITCH, STAC_CTL_WIDGET_IO_SWITCH, STAC_CTL_WIDGET_CLFE_SWITCH }; @@ -2028,6 +2194,7 @@ static struct snd_kcontrol_new stac92xx_control_templates[] = { HDA_CODEC_VOLUME(NULL, 0, 0, 0), HDA_CODEC_MUTE(NULL, 0, 0, 0), STAC_MONO_MUX, + STAC_CODEC_HP_SWITCH(NULL), STAC_CODEC_IO_SWITCH(NULL, 0), STAC_CODEC_CLFE_SWITCH(NULL, 0), }; @@ -2222,6 +2389,29 @@ static int create_controls(struct sigmatel_spec *spec, const char *pfx, hda_nid_ return 0; } +static int add_spec_dacs(struct sigmatel_spec *spec, hda_nid_t nid) +{ + if (!spec->multiout.hp_nid) + spec->multiout.hp_nid = nid; + else if (spec->multiout.num_dacs > 4) { + printk(KERN_WARNING "stac92xx: No space for DAC 0x%x\n", nid); + return 1; + } else { + spec->multiout.dac_nids[spec->multiout.num_dacs] = nid; + spec->multiout.num_dacs++; + } + return 0; +} + +static int check_in_dac_nids(struct sigmatel_spec *spec, hda_nid_t nid) +{ + if (is_in_dac_nids(spec, nid)) + return 1; + if (spec->multiout.hp_nid == nid) + return 1; + return 0; +} + /* add playback controls from the parsed DAC table */ static int stac92xx_auto_create_multi_out_ctls(struct hda_codec *codec, const struct auto_pin_cfg *cfg) @@ -2236,7 +2426,7 @@ static int stac92xx_auto_create_multi_out_ctls(struct hda_codec *codec, unsigned int wid_caps, pincap; - for (i = 0; i < cfg->line_outs; i++) { + for (i = 0; i < cfg->line_outs && i < spec->multiout.num_dacs; i++) { if (!spec->multiout.dac_nids[i]) continue; @@ -2269,6 +2459,14 @@ static int stac92xx_auto_create_multi_out_ctls(struct hda_codec *codec, } } + if (cfg->hp_outs > 1) { + err = stac92xx_add_control(spec, + STAC_CTL_WIDGET_HP_SWITCH, + "Headphone as Line Out Switch", 0); + if (err < 0) + return err; + } + if (spec->line_switch) { nid = cfg->input_pins[AUTO_PIN_LINE]; pincap = snd_hda_param_read(codec, nid, @@ -2284,10 +2482,11 @@ static int stac92xx_auto_create_multi_out_ctls(struct hda_codec *codec, if (spec->mic_switch) { unsigned int def_conf; - nid = cfg->input_pins[AUTO_PIN_MIC]; + unsigned int mic_pin = AUTO_PIN_MIC; +again: + nid = cfg->input_pins[mic_pin]; def_conf = snd_hda_codec_read(codec, nid, 0, AC_VERB_GET_CONFIG_DEFAULT, 0); - /* some laptops have an internal analog microphone * which can't be used as a output */ if (get_defcfg_connect(def_conf) != AC_JACK_PORT_FIXED) { @@ -2297,38 +2496,22 @@ static int stac92xx_auto_create_multi_out_ctls(struct hda_codec *codec, err = stac92xx_add_control(spec, STAC_CTL_WIDGET_IO_SWITCH, "Mic as Output Switch", (nid << 8) | 1); + nid = snd_hda_codec_read(codec, nid, 0, + AC_VERB_GET_CONNECT_LIST, 0) & 0xff; + if (!check_in_dac_nids(spec, nid)) + add_spec_dacs(spec, nid); if (err < 0) return err; } + } else if (mic_pin == AUTO_PIN_MIC) { + mic_pin = AUTO_PIN_FRONT_MIC; + goto again; } } return 0; } -static int check_in_dac_nids(struct sigmatel_spec *spec, hda_nid_t nid) -{ - if (is_in_dac_nids(spec, nid)) - return 1; - if (spec->multiout.hp_nid == nid) - return 1; - return 0; -} - -static int add_spec_dacs(struct sigmatel_spec *spec, hda_nid_t nid) -{ - if (!spec->multiout.hp_nid) - spec->multiout.hp_nid = nid; - else if (spec->multiout.num_dacs > 4) { - printk(KERN_WARNING "stac92xx: No space for DAC 0x%x\n", nid); - return 1; - } else { - spec->multiout.dac_nids[spec->multiout.num_dacs] = nid; - spec->multiout.num_dacs++; - } - return 0; -} - /* add playback controls for Speaker and HP outputs */ static int stac92xx_auto_create_hp_ctls(struct hda_codec *codec, struct auto_pin_cfg *cfg) @@ -2378,12 +2561,8 @@ static int stac92xx_auto_create_hp_ctls(struct hda_codec *codec, return err; } if (spec->multiout.hp_nid) { - const char *pfx; - if (old_num_dacs == spec->multiout.num_dacs) - pfx = "Master"; - else - pfx = "Headphone"; - err = create_controls(spec, pfx, spec->multiout.hp_nid, 3); + err = create_controls(spec, "Headphone", + spec->multiout.hp_nid, 3); if (err < 0) return err; } @@ -2745,7 +2924,7 @@ static int stac9200_auto_create_lfe_ctls(struct hda_codec *codec, */ for (i = 0; i < spec->autocfg.speaker_outs && lfe_pin == 0x0; i++) { hda_nid_t pin = spec->autocfg.speaker_pins[i]; - unsigned long wcaps = get_wcaps(codec, pin); + unsigned int wcaps = get_wcaps(codec, pin); wcaps &= (AC_WCAP_STEREO | AC_WCAP_OUT_AMP); if (wcaps == AC_WCAP_OUT_AMP) /* found a mono speaker with an amp, must be lfe */ @@ -2756,12 +2935,12 @@ static int stac9200_auto_create_lfe_ctls(struct hda_codec *codec, if (lfe_pin == 0 && spec->autocfg.speaker_outs == 0) { for (i = 0; i < spec->autocfg.line_outs && lfe_pin == 0x0; i++) { hda_nid_t pin = spec->autocfg.line_out_pins[i]; - unsigned long cfg; - cfg = snd_hda_codec_read(codec, pin, 0, + unsigned int defcfg; + defcfg = snd_hda_codec_read(codec, pin, 0, AC_VERB_GET_CONFIG_DEFAULT, 0x00); - if (get_defcfg_device(cfg) == AC_JACK_SPEAKER) { - unsigned long wcaps = get_wcaps(codec, pin); + if (get_defcfg_device(defcfg) == AC_JACK_SPEAKER) { + unsigned int wcaps = get_wcaps(codec, pin); wcaps &= (AC_WCAP_STEREO | AC_WCAP_OUT_AMP); if (wcaps == AC_WCAP_OUT_AMP) /* found a mono speaker with an amp, @@ -2866,6 +3045,19 @@ static int is_nid_hp_pin(struct auto_pin_cfg *cfg, hda_nid_t nid) return 0; /* nid is not a HP-Out */ }; +static void stac92xx_power_down(struct hda_codec *codec) +{ + struct sigmatel_spec *spec = codec->spec; + + /* power down inactive DACs */ + hda_nid_t *dac; + for (dac = spec->dac_list; *dac; dac++) + if (!is_in_dac_nids(spec, *dac) && + spec->multiout.hp_nid != *dac) + snd_hda_codec_write_cache(codec, *dac, 0, + AC_VERB_SET_POWER_STATE, AC_PWRST_D3); +} + static int stac92xx_init(struct hda_codec *codec) { struct sigmatel_spec *spec = codec->spec; @@ -2909,16 +3101,21 @@ static int stac92xx_init(struct hda_codec *codec) ? STAC_HP_EVENT : STAC_PWR_EVENT; int pinctl = snd_hda_codec_read(codec, spec->pwr_nids[i], 0, AC_VERB_GET_PIN_WIDGET_CONTROL, 0); + int def_conf = snd_hda_codec_read(codec, spec->pwr_nids[i], + 0, AC_VERB_GET_CONFIG_DEFAULT, 0); /* outputs are only ports capable of power management * any attempts on powering down a input port cause the * referenced VREF to act quirky. */ if (pinctl & AC_PINCTL_IN_EN) continue; + if (get_defcfg_connect(def_conf) != AC_JACK_PORT_FIXED) + continue; enable_pin_detect(codec, spec->pwr_nids[i], event | i); codec->patch_ops.unsol_event(codec, (event | i) << 26); } - + if (spec->dac_list) + stac92xx_power_down(codec); if (cfg->dig_out_pin) stac92xx_auto_set_pinctl(codec, cfg->dig_out_pin, AC_PINCTL_OUT_EN); @@ -3014,6 +3211,7 @@ static void stac92xx_hp_detect(struct hda_codec *codec, unsigned int res) { struct sigmatel_spec *spec = codec->spec; struct auto_pin_cfg *cfg = &spec->autocfg; + int nid = cfg->hp_pins[cfg->hp_outs - 1]; int i, presence; presence = 0; @@ -3024,26 +3222,42 @@ static void stac92xx_hp_detect(struct hda_codec *codec, unsigned int res) for (i = 0; i < cfg->hp_outs; i++) { if (presence) break; + if (spec->hp_switch && cfg->hp_pins[i] == nid) + break; presence = get_hp_pin_presence(codec, cfg->hp_pins[i]); } if (presence) { /* disable lineouts, enable hp */ + if (spec->hp_switch) + stac92xx_reset_pinctl(codec, nid, AC_PINCTL_OUT_EN); for (i = 0; i < cfg->line_outs; i++) stac92xx_reset_pinctl(codec, cfg->line_out_pins[i], AC_PINCTL_OUT_EN); for (i = 0; i < cfg->speaker_outs; i++) stac92xx_reset_pinctl(codec, cfg->speaker_pins[i], AC_PINCTL_OUT_EN); + if (spec->eapd_mask) + stac_gpio_set(codec, spec->gpio_mask, + spec->gpio_dir, spec->gpio_data & + ~spec->eapd_mask); } else { /* enable lineouts, disable hp */ + if (spec->hp_switch) + stac92xx_set_pinctl(codec, nid, AC_PINCTL_OUT_EN); for (i = 0; i < cfg->line_outs; i++) stac92xx_set_pinctl(codec, cfg->line_out_pins[i], AC_PINCTL_OUT_EN); for (i = 0; i < cfg->speaker_outs; i++) stac92xx_set_pinctl(codec, cfg->speaker_pins[i], AC_PINCTL_OUT_EN); + if (spec->eapd_mask) + stac_gpio_set(codec, spec->gpio_mask, + spec->gpio_dir, spec->gpio_data | + spec->eapd_mask); } + if (!spec->hp_switch && cfg->hp_outs > 1 && presence) + stac92xx_set_pinctl(codec, nid, AC_PINCTL_OUT_EN); } static void stac92xx_pin_sense(struct hda_codec *codec, int idx) @@ -3091,6 +3305,9 @@ static int stac92xx_resume(struct hda_codec *codec) spec->gpio_dir, spec->gpio_data); snd_hda_codec_resume_amp(codec); snd_hda_codec_resume_cache(codec); + /* power down inactive DACs */ + if (spec->dac_list) + stac92xx_power_down(codec); /* invoke unsolicited event to reset the HP state */ if (spec->hp_detect) codec->patch_ops.unsol_event(codec, STAC_HP_EVENT << 26); @@ -3147,12 +3364,18 @@ static int patch_stac9200(struct hda_codec *codec) spec->num_adcs = 1; spec->num_pwrs = 0; - if (spec->board_config == STAC_9200_GATEWAY) + if (spec->board_config == STAC_9200_GATEWAY || + spec->board_config == STAC_9200_OQO) spec->init = stac9200_eapd_init; else spec->init = stac9200_core_init; spec->mixer = stac9200_mixer; + if (spec->board_config == STAC_9200_PANASONIC) { + spec->gpio_mask = spec->gpio_dir = 0x09; + spec->gpio_data = 0x00; + } + err = stac9200_parse_auto_config(codec); if (err < 0) { stac92xx_free(codec); @@ -3293,6 +3516,7 @@ again: switch (spec->multiout.num_dacs) { case 0x3: /* 6 Channel */ + spec->multiout.hp_nid = 0x17; spec->mixer = stac92hd73xx_6ch_mixer; spec->init = stac92hd73xx_6ch_core_init; break; @@ -3318,13 +3542,42 @@ again: spec->num_muxes = ARRAY_SIZE(stac92hd73xx_mux_nids); spec->num_adcs = ARRAY_SIZE(stac92hd73xx_adc_nids); - spec->num_dmics = STAC92HD73XX_NUM_DMICS; spec->num_dmuxes = ARRAY_SIZE(stac92hd73xx_dmux_nids); spec->dinput_mux = &stac92hd73xx_dmux; /* GPIO0 High = Enable EAPD */ - spec->gpio_mask = spec->gpio_dir = 0x1; + spec->eapd_mask = spec->gpio_mask = spec->gpio_dir = 0x1; spec->gpio_data = 0x01; + switch (spec->board_config) { + case STAC_DELL_M6: + spec->init = dell_eq_core_init; + switch (codec->subsystem_id) { + case 0x1028025e: /* Analog Mics */ + case 0x1028025f: + stac92xx_set_config_reg(codec, 0x0b, 0x90A70170); + spec->num_dmics = 0; + break; + case 0x10280271: /* Digital Mics */ + case 0x10280272: + spec->init = dell_m6_core_init; + /* fall-through */ + case 0x10280254: + case 0x10280255: + stac92xx_set_config_reg(codec, 0x13, 0x90A60160); + spec->num_dmics = 1; + break; + case 0x10280256: /* Both */ + case 0x10280057: + stac92xx_set_config_reg(codec, 0x0b, 0x90A70170); + stac92xx_set_config_reg(codec, 0x13, 0x90A60160); + spec->num_dmics = 1; + break; + } + break; + default: + spec->num_dmics = STAC92HD73XX_NUM_DMICS; + } + spec->num_pwrs = ARRAY_SIZE(stac92hd73xx_pwr_nids); spec->pwr_nids = stac92hd73xx_pwr_nids; @@ -3398,7 +3651,10 @@ again: spec->aloopback_shift = 0; /* GPIO0 High = EAPD */ - spec->gpio_mask = spec->gpio_dir = spec->gpio_data = 0x1; + spec->gpio_mask = 0x01; + spec->gpio_dir = 0x01; + spec->gpio_mask = 0x01; + spec->gpio_data = 0x01; spec->mux_nids = stac92hd71bxx_mux_nids; spec->adc_nids = stac92hd71bxx_adc_nids; @@ -3413,7 +3669,7 @@ again: spec->num_pwrs = ARRAY_SIZE(stac92hd71bxx_pwr_nids); spec->pwr_nids = stac92hd71bxx_pwr_nids; - spec->multiout.num_dacs = 2; + spec->multiout.num_dacs = 1; spec->multiout.hp_nid = 0x11; spec->multiout.dac_nids = stac92hd71bxx_dac_nids; @@ -3577,13 +3833,14 @@ static int patch_stac927x(struct hda_codec *codec) spec->num_adcs = ARRAY_SIZE(stac927x_adc_nids); spec->mux_nids = stac927x_mux_nids; spec->num_muxes = ARRAY_SIZE(stac927x_mux_nids); + spec->dac_list = stac927x_dac_nids; spec->multiout.dac_nids = spec->dac_nids; switch (spec->board_config) { case STAC_D965_3ST: case STAC_D965_5ST: /* GPIO0 High = Enable EAPD */ - spec->gpio_mask = spec->gpio_dir = 0x01; + spec->eapd_mask = spec->gpio_mask = spec->gpio_dir = 0x01; spec->gpio_data = 0x01; spec->num_dmics = 0; @@ -3591,14 +3848,23 @@ static int patch_stac927x(struct hda_codec *codec) spec->mixer = stac927x_mixer; break; case STAC_DELL_BIOS: + switch (codec->subsystem_id) { + case 0x10280209: + case 0x1028022e: + /* correct the device field to SPDIF out */ + stac92xx_set_config_reg(codec, 0x21, 0x01442070); + break; + }; + /* configure the analog microphone on some laptops */ + stac92xx_set_config_reg(codec, 0x0c, 0x90a79130); /* correct the front output jack as a hp out */ - stac92xx_set_config_reg(codec, 0x0f, 0x02270110); + stac92xx_set_config_reg(codec, 0x0f, 0x0227011f); /* correct the front input jack as a mic */ stac92xx_set_config_reg(codec, 0x0e, 0x02a79130); /* fallthru */ case STAC_DELL_3ST: /* GPIO2 High = Enable EAPD */ - spec->gpio_mask = spec->gpio_dir = 0x04; + spec->eapd_mask = spec->gpio_mask = spec->gpio_dir = 0x04; spec->gpio_data = 0x04; spec->dmic_nids = stac927x_dmic_nids; spec->num_dmics = STAC927X_NUM_DMICS; @@ -3610,7 +3876,7 @@ static int patch_stac927x(struct hda_codec *codec) break; default: /* GPIO0 High = Enable EAPD */ - spec->gpio_mask = spec->gpio_dir = 0x1; + spec->eapd_mask = spec->gpio_mask = spec->gpio_dir = 0x1; spec->gpio_data = 0x01; spec->num_dmics = 0; @@ -3714,6 +3980,7 @@ static int patch_stac9205(struct hda_codec *codec) (AC_USRSP_EN | STAC_HP_EVENT)); spec->gpio_dir = 0x0b; + spec->eapd_mask = 0x01; spec->gpio_mask = 0x1b; spec->gpio_mute = 0x10; /* GPIO0 High = EAPD, GPIO1 Low = Headphone Mute, @@ -3723,7 +3990,7 @@ static int patch_stac9205(struct hda_codec *codec) break; default: /* GPIO0 High = EAPD */ - spec->gpio_mask = spec->gpio_dir = 0x1; + spec->eapd_mask = spec->gpio_mask = spec->gpio_dir = 0x1; spec->gpio_data = 0x01; break; } diff --git a/sound/pci/hda/patch_via.c b/sound/pci/hda/patch_via.c index 4e5dd4c..52b1d81 100644 --- a/sound/pci/hda/patch_via.c +++ b/sound/pci/hda/patch_via.c @@ -39,7 +39,7 @@ #include #include "hda_codec.h" #include "hda_local.h" - +#include "hda_patch.h" /* amp values */ #define AMP_VAL_IDX_SHIFT 19 @@ -357,7 +357,8 @@ static int via_playback_pcm_open(struct hda_pcm_stream *hinfo, struct snd_pcm_substream *substream) { struct via_spec *spec = codec->spec; - return snd_hda_multi_out_analog_open(codec, &spec->multiout, substream); + return snd_hda_multi_out_analog_open(codec, &spec->multiout, substream, + hinfo); } static int via_playback_pcm_prepare(struct hda_pcm_stream *hinfo, @@ -430,8 +431,7 @@ static int via_capture_pcm_cleanup(struct hda_pcm_stream *hinfo, struct snd_pcm_substream *substream) { struct via_spec *spec = codec->spec; - snd_hda_codec_setup_stream(codec, spec->adc_nids[substream->number], - 0, 0, 0); + snd_hda_codec_cleanup_stream(codec, spec->adc_nids[substream->number]); return 0; } @@ -493,6 +493,11 @@ static int via_build_controls(struct hda_codec *codec) spec->multiout.dig_out_nid); if (err < 0) return err; + err = snd_hda_create_spdif_share_sw(codec, + &spec->multiout); + if (err < 0) + return err; + spec->multiout.share_spdif = 1; } if (spec->dig_in_nid) { err = snd_hda_create_spdif_in_ctls(codec, spec->dig_in_nid); @@ -523,6 +528,7 @@ static int via_build_pcms(struct hda_codec *codec) codec->num_pcms++; info++; info->name = spec->stream_name_digital; + info->pcm_type = HDA_PCM_TYPE_SPDIF; if (spec->multiout.dig_out_nid) { info->stream[SNDRV_PCM_STREAM_PLAYBACK] = *(spec->stream_digital_playback); diff --git a/sound/pci/hda/vmaster.c b/sound/pci/hda/vmaster.c deleted file mode 100644 index 2da49d2..0000000 --- a/sound/pci/hda/vmaster.c +++ /dev/null @@ -1,364 +0,0 @@ -/* - * Virtual master and slave controls - * - * Copyright (c) 2008 by Takashi Iwai - * - * This program is free software; you can redistribute it and/or - * modify it under the terms of the GNU General Public License as - * published by the Free Software Foundation, version 2. - * - */ - -#include -#include -#include - -/* - * a subset of information returned via ctl info callback - */ -struct link_ctl_info { - int type; /* value type */ - int count; /* item count */ - int min_val, max_val; /* min, max values */ -}; - -/* - * link master - this contains a list of slave controls that are - * identical types, i.e. info returns the same value type and value - * ranges, but may have different number of counts. - * - * The master control is so far only mono volume/switch for simplicity. - * The same value will be applied to all slaves. - */ -struct link_master { - struct list_head slaves; - struct link_ctl_info info; - int val; /* the master value */ -}; - -/* - * link slave - this contains a slave control element - * - * It fakes the control callbacsk with additional attenuation by the - * master control. A slave may have either one or two channels. - */ - -struct link_slave { - struct list_head list; - struct link_master *master; - struct link_ctl_info info; - int vals[2]; /* current values */ - struct snd_kcontrol slave; /* the copy of original control entry */ -}; - -/* get the slave ctl info and save the initial values */ -static int slave_init(struct link_slave *slave) -{ - struct snd_ctl_elem_info *uinfo; - struct snd_ctl_elem_value *uctl; - int err, ch; - - if (slave->info.count) - return 0; /* already initialized */ - - uinfo = kmalloc(sizeof(*uinfo), GFP_KERNEL); - if (!uinfo) - return -ENOMEM; - uinfo->id = slave->slave.id; - err = slave->slave.info(&slave->slave, uinfo); - if (err < 0) { - kfree(uinfo); - return err; - } - slave->info.type = uinfo->type; - slave->info.count = uinfo->count; - if (slave->info.count > 2 || - (slave->info.type != SNDRV_CTL_ELEM_TYPE_INTEGER && - slave->info.type != SNDRV_CTL_ELEM_TYPE_BOOLEAN)) { - snd_printk(KERN_ERR "invalid slave element\n"); - kfree(uinfo); - return -EINVAL; - } - slave->info.min_val = uinfo->value.integer.min; - slave->info.max_val = uinfo->value.integer.max; - kfree(uinfo); - - uctl = kmalloc(sizeof(*uctl), GFP_KERNEL); - if (!uctl) - return -ENOMEM; - uctl->id = slave->slave.id; - err = slave->slave.get(&slave->slave, uctl); - for (ch = 0; ch < slave->info.count; ch++) - slave->vals[ch] = uctl->value.integer.value[ch]; - kfree(uctl); - return 0; -} - -/* initialize master volume */ -static int master_init(struct link_master *master) -{ - struct link_slave *slave; - - if (master->info.count) - return 0; /* already initialized */ - - list_for_each_entry(slave, &master->slaves, list) { - int err = slave_init(slave); - if (err < 0) - return err; - master->info = slave->info; - master->info.count = 1; /* always mono */ - /* set full volume as default (= no attenuation) */ - master->val = master->info.max_val; - return 0; - } - return -ENOENT; -} - -static int slave_get_val(struct link_slave *slave, - struct snd_ctl_elem_value *ucontrol) -{ - int err, ch; - - err = slave_init(slave); - if (err < 0) - return err; - for (ch = 0; ch < slave->info.count; ch++) - ucontrol->value.integer.value[ch] = slave->vals[ch]; - return 0; -} - -static int slave_put_val(struct link_slave *slave, - struct snd_ctl_elem_value *ucontrol) -{ - int err, ch, vol; - - err = master_init(slave->master); - if (err < 0) - return err; - - switch (slave->info.type) { - case SNDRV_CTL_ELEM_TYPE_BOOLEAN: - for (ch = 0; ch < slave->info.count; ch++) - ucontrol->value.integer.value[ch] &= - !!slave->master->val; - break; - case SNDRV_CTL_ELEM_TYPE_INTEGER: - for (ch = 0; ch < slave->info.count; ch++) { - /* max master volume is supposed to be 0 dB */ - vol = ucontrol->value.integer.value[ch]; - vol += slave->master->val - slave->master->info.max_val; - if (vol < slave->info.min_val) - vol = slave->info.min_val; - else if (vol > slave->info.max_val) - vol = slave->info.max_val; - ucontrol->value.integer.value[ch] = vol; - } - break; - } - return slave->slave.put(&slave->slave, ucontrol); -} - -/* - * ctl callbacks for slaves - */ -static int slave_info(struct snd_kcontrol *kcontrol, - struct snd_ctl_elem_info *uinfo) -{ - struct link_slave *slave = snd_kcontrol_chip(kcontrol); - return slave->slave.info(&slave->slave, uinfo); -} - -static int slave_get(struct snd_kcontrol *kcontrol, - struct snd_ctl_elem_value *ucontrol) -{ - struct link_slave *slave = snd_kcontrol_chip(kcontrol); - return slave_get_val(slave, ucontrol); -} - -static int slave_put(struct snd_kcontrol *kcontrol, - struct snd_ctl_elem_value *ucontrol) -{ - struct link_slave *slave = snd_kcontrol_chip(kcontrol); - int err, ch, changed = 0; - - err = slave_init(slave); - if (err < 0) - return err; - for (ch = 0; ch < slave->info.count; ch++) { - if (slave->vals[ch] != ucontrol->value.integer.value[ch]) { - changed = 1; - slave->vals[ch] = ucontrol->value.integer.value[ch]; - } - } - if (!changed) - return 0; - return slave_put_val(slave, ucontrol); -} - -static int slave_tlv_cmd(struct snd_kcontrol *kcontrol, - int op_flag, unsigned int size, - unsigned int __user *tlv) -{ - struct link_slave *slave = snd_kcontrol_chip(kcontrol); - /* FIXME: this assumes that the max volume is 0 dB */ - return slave->slave.tlv.c(&slave->slave, op_flag, size, tlv); -} - -static void slave_free(struct snd_kcontrol *kcontrol) -{ - struct link_slave *slave = snd_kcontrol_chip(kcontrol); - if (slave->slave.private_free) - slave->slave.private_free(&slave->slave); - if (slave->master) - list_del(&slave->list); - kfree(slave); -} - -/* - * Add a slave control to the group with the given master control - * - * All slaves must be the same type (returning the same information - * via info callback). The fucntion doesn't check it, so it's your - * responsibility. - * - * Also, some additional limitations: - * - at most two channels - * - logarithmic volume control (dB level), no linear volume - * - master can only attenuate the volume, no gain - */ -int snd_ctl_add_slave(struct snd_kcontrol *master, struct snd_kcontrol *slave) -{ - struct link_master *master_link = snd_kcontrol_chip(master); - struct link_slave *srec; - - srec = kzalloc(sizeof(*srec) + - slave->count * sizeof(*slave->vd), GFP_KERNEL); - if (!srec) - return -ENOMEM; - srec->slave = *slave; - memcpy(srec->slave.vd, slave->vd, slave->count * sizeof(*slave->vd)); - srec->master = master_link; - - /* override callbacks */ - slave->info = slave_info; - slave->get = slave_get; - slave->put = slave_put; - if (slave->vd[0].access & SNDRV_CTL_ELEM_ACCESS_TLV_CALLBACK) - slave->tlv.c = slave_tlv_cmd; - slave->private_data = srec; - slave->private_free = slave_free; - - list_add_tail(&srec->list, &master_link->slaves); - return 0; -} - -/* - * ctl callbacks for master controls - */ -static int master_info(struct snd_kcontrol *kcontrol, - struct snd_ctl_elem_info *uinfo) -{ - struct link_master *master = snd_kcontrol_chip(kcontrol); - int ret; - - ret = master_init(master); - if (ret < 0) - return ret; - uinfo->type = master->info.type; - uinfo->count = master->info.count; - uinfo->value.integer.min = master->info.min_val; - uinfo->value.integer.max = master->info.max_val; - return 0; -} - -static int master_get(struct snd_kcontrol *kcontrol, - struct snd_ctl_elem_value *ucontrol) -{ - struct link_master *master = snd_kcontrol_chip(kcontrol); - int err = master_init(master); - if (err < 0) - return err; - ucontrol->value.integer.value[0] = master->val; - return 0; -} - -static int master_put(struct snd_kcontrol *kcontrol, - struct snd_ctl_elem_value *ucontrol) -{ - struct link_master *master = snd_kcontrol_chip(kcontrol); - struct link_slave *slave; - struct snd_ctl_elem_value *uval; - int err, old_val; - - err = master_init(master); - if (err < 0) - return err; - old_val = master->val; - if (ucontrol->value.integer.value[0] == old_val) - return 0; - - uval = kmalloc(sizeof(*uval), GFP_KERNEL); - if (!uval) - return -ENOMEM; - list_for_each_entry(slave, &master->slaves, list) { - master->val = old_val; - uval->id = slave->slave.id; - slave_get_val(slave, uval); - master->val = ucontrol->value.integer.value[0]; - slave_put_val(slave, uval); - } - kfree(uval); - return 1; -} - -static void master_free(struct snd_kcontrol *kcontrol) -{ - struct link_master *master = snd_kcontrol_chip(kcontrol); - struct link_slave *slave; - - list_for_each_entry(slave, &master->slaves, list) - slave->master = NULL; - kfree(master); -} - - -/* - * Create a virtual master control with the given name - */ -struct snd_kcontrol *snd_ctl_make_virtual_master(char *name, - const unsigned int *tlv) -{ - struct link_master *master; - struct snd_kcontrol *kctl; - struct snd_kcontrol_new knew; - - memset(&knew, 0, sizeof(knew)); - knew.iface = SNDRV_CTL_ELEM_IFACE_MIXER; - knew.name = name; - knew.info = master_info; - - master = kzalloc(sizeof(*master), GFP_KERNEL); - if (!master) - return NULL; - INIT_LIST_HEAD(&master->slaves); - - kctl = snd_ctl_new1(&knew, master); - if (!kctl) { - kfree(master); - return NULL; - } - /* override some callbacks */ - kctl->info = master_info; - kctl->get = master_get; - kctl->put = master_put; - kctl->private_free = master_free; - - /* additional (constant) TLV read */ - if (tlv) { - /* FIXME: this assumes that the max volume is 0 dB */ - kctl->vd[0].access |= SNDRV_CTL_ELEM_ACCESS_TLV_READ; - kctl->tlv.p = tlv; - } - return kctl; -} diff --git a/sound/pci/ice1712/delta.c b/sound/pci/ice1712/delta.c index efd180b..0ed96c1 100644 --- a/sound/pci/ice1712/delta.c +++ b/sound/pci/ice1712/delta.c @@ -1,8 +1,8 @@ /* * ALSA driver for ICEnsemble ICE1712 (Envy24) * - * Lowlevel functions for M-Audio Delta 1010, 44, 66, Dio2496, Audiophile - * Digigram VX442 + * Lowlevel functions for M-Audio Delta 1010, 1010E, 44, 66, 66E, Dio2496, + * Audiophile, Digigram VX442 * * Copyright (c) 2000 Jaroslav Kysela * @@ -86,6 +86,7 @@ static unsigned char ap_cs8427_codec_select(struct snd_ice1712 *ice) unsigned char tmp; tmp = snd_ice1712_read(ice, ICE1712_IREG_GPIO_DATA); switch (ice->eeprom.subvendor) { + case ICE1712_SUBDEVICE_DELTA1010E: case ICE1712_SUBDEVICE_DELTA1010LT: tmp &= ~ICE1712_DELTA_1010LT_CS; tmp |= ICE1712_DELTA_1010LT_CCLK | ICE1712_DELTA_1010LT_CS_CS8427; @@ -109,6 +110,7 @@ static unsigned char ap_cs8427_codec_select(struct snd_ice1712 *ice) static void ap_cs8427_codec_deassert(struct snd_ice1712 *ice, unsigned char tmp) { switch (ice->eeprom.subvendor) { + case ICE1712_SUBDEVICE_DELTA1010E: case ICE1712_SUBDEVICE_DELTA1010LT: tmp &= ~ICE1712_DELTA_1010LT_CS; tmp |= ICE1712_DELTA_1010LT_CS_NONE; @@ -534,6 +536,14 @@ static int __devinit snd_ice1712_delta_init(struct snd_ice1712 *ice) int err; struct snd_akm4xxx *ak; + if (ice->eeprom.subvendor == ICE1712_SUBDEVICE_DELTA1010 && + ice->eeprom.gpiodir == 0x7b) + ice->eeprom.subvendor = ICE1712_SUBDEVICE_DELTA1010E; + + if (ice->eeprom.subvendor == ICE1712_SUBDEVICE_DELTA66 && + ice->eeprom.gpiodir == 0xfb) + ice->eeprom.subvendor = ICE1712_SUBDEVICE_DELTA66E; + /* determine I2C, DACs and ADCs */ switch (ice->eeprom.subvendor) { case ICE1712_SUBDEVICE_AUDIOPHILE: @@ -550,6 +560,7 @@ static int __devinit snd_ice1712_delta_init(struct snd_ice1712 *ice) ice->num_total_adcs = ice->omni ? 8 : 4; break; case ICE1712_SUBDEVICE_DELTA1010: + case ICE1712_SUBDEVICE_DELTA1010E: case ICE1712_SUBDEVICE_DELTA1010LT: case ICE1712_SUBDEVICE_MEDIASTATION: ice->num_total_dacs = 8; @@ -559,6 +570,7 @@ static int __devinit snd_ice1712_delta_init(struct snd_ice1712 *ice) ice->num_total_dacs = 4; /* two AK4324 codecs */ break; case ICE1712_SUBDEVICE_VX442: + case ICE1712_SUBDEVICE_DELTA66E: /* omni not suported yet */ ice->num_total_dacs = 4; ice->num_total_adcs = 4; break; @@ -568,8 +580,10 @@ static int __devinit snd_ice1712_delta_init(struct snd_ice1712 *ice) switch (ice->eeprom.subvendor) { case ICE1712_SUBDEVICE_AUDIOPHILE: case ICE1712_SUBDEVICE_DELTA410: + case ICE1712_SUBDEVICE_DELTA1010E: case ICE1712_SUBDEVICE_DELTA1010LT: case ICE1712_SUBDEVICE_VX442: + case ICE1712_SUBDEVICE_DELTA66E: if ((err = snd_i2c_bus_create(ice->card, "ICE1712 GPIO 1", NULL, &ice->i2c)) < 0) { snd_printk(KERN_ERR "unable to create I2C bus\n"); return err; @@ -601,6 +615,7 @@ static int __devinit snd_ice1712_delta_init(struct snd_ice1712 *ice) /* no analog? */ switch (ice->eeprom.subvendor) { case ICE1712_SUBDEVICE_DELTA1010: + case ICE1712_SUBDEVICE_DELTA1010E: case ICE1712_SUBDEVICE_DELTADIO2496: case ICE1712_SUBDEVICE_MEDIASTATION: return 0; @@ -627,6 +642,7 @@ static int __devinit snd_ice1712_delta_init(struct snd_ice1712 *ice) err = snd_ice1712_akm4xxx_init(ak, &akm_delta44, &akm_delta44_priv, ice); break; case ICE1712_SUBDEVICE_VX442: + case ICE1712_SUBDEVICE_DELTA66E: err = snd_ice1712_akm4xxx_init(ak, &akm_vx442, &akm_vx442_priv, ice); break; default: @@ -674,6 +690,7 @@ static int __devinit snd_ice1712_delta_add_controls(struct snd_ice1712 *ice) if (err < 0) return err; break; + case ICE1712_SUBDEVICE_DELTA1010E: case ICE1712_SUBDEVICE_DELTA1010LT: err = snd_ctl_add(ice->card, snd_ctl_new1(&snd_ice1712_delta1010lt_wordclock_select, ice)); if (err < 0) @@ -716,6 +733,7 @@ static int __devinit snd_ice1712_delta_add_controls(struct snd_ice1712 *ice) case ICE1712_SUBDEVICE_DELTA44: case ICE1712_SUBDEVICE_DELTA66: case ICE1712_SUBDEVICE_VX442: + case ICE1712_SUBDEVICE_DELTA66E: err = snd_ice1712_akm4xxx_build_controls(ice); if (err < 0) return err; diff --git a/sound/pci/ice1712/delta.h b/sound/pci/ice1712/delta.h index 26ea05a..ea7116c 100644 --- a/sound/pci/ice1712/delta.h +++ b/sound/pci/ice1712/delta.h @@ -36,8 +36,10 @@ "{Lionstracs,Mediastation}," #define ICE1712_SUBDEVICE_DELTA1010 0x121430d6 +#define ICE1712_SUBDEVICE_DELTA1010E 0xff1430d6 #define ICE1712_SUBDEVICE_DELTADIO2496 0x121431d6 #define ICE1712_SUBDEVICE_DELTA66 0x121432d6 +#define ICE1712_SUBDEVICE_DELTA66E 0xff1432d6 #define ICE1712_SUBDEVICE_DELTA44 0x121433d6 #define ICE1712_SUBDEVICE_AUDIOPHILE 0x121434d6 #define ICE1712_SUBDEVICE_DELTA410 0x121438d6 diff --git a/sound/pci/ice1712/hoontech.c b/sound/pci/ice1712/hoontech.c index cf5c7c0..6914189 100644 --- a/sound/pci/ice1712/hoontech.c +++ b/sound/pci/ice1712/hoontech.c @@ -208,6 +208,19 @@ static int __devinit snd_ice1712_hoontech_init(struct snd_ice1712 *ice) /* ICE1712_STDSP24_MUTE | ICE1712_STDSP24_INSEL | ICE1712_STDSP24_DAREAR; */ + /* These boxconfigs have caused problems in the past. + * The code is not optimal, but should now enable a working config to + * be achieved. + * ** MIDI IN can only be configured on one box ** + * ICE1712_STDSP24_BOX_MIDI1 needs to be set for that box. + * Tests on a ADAC2000 box suggest the box config flags do not + * work as would be expected, and the inputs are crossed. + * Setting ICE1712_STDSP24_BOX_MIDI1 and ICE1712_STDSP24_BOX_MIDI2 + * on the same box connects MIDI-In to both 401 uarts; both outputs + * are then active on all boxes. + * The default config here sets up everything on the first box. + * Alan Horstmann 5.2.2008 + */ spec->boxconfig[0] = ICE1712_STDSP24_BOX_CHN1 | ICE1712_STDSP24_BOX_CHN2 | ICE1712_STDSP24_BOX_CHN3 | @@ -223,14 +236,14 @@ static int __devinit snd_ice1712_hoontech_init(struct snd_ice1712 *ice) (spec->config & ICE1712_STDSP24_MUTE) ? 1 : 0); snd_ice1712_stdsp24_insel(ice, (spec->config & ICE1712_STDSP24_INSEL) ? 1 : 0); - for (box = 0; box < 1; box++) { + for (box = 0; box < 4; box++) { if (spec->boxconfig[box] & ICE1712_STDSP24_BOX_MIDI2) snd_ice1712_stdsp24_midi2(ice, 1); for (chn = 0; chn < 4; chn++) snd_ice1712_stdsp24_box_channel(ice, box, chn, (spec->boxconfig[box] & (1 << chn)) ? 1 : 0); - snd_ice1712_stdsp24_box_midi(ice, box, - (spec->boxconfig[box] & ICE1712_STDSP24_BOX_MIDI1) ? 1 : 0); + if (spec->boxconfig[box] & ICE1712_STDSP24_BOX_MIDI1) + snd_ice1712_stdsp24_box_midi(ice, box, 1); } return 0; @@ -322,6 +335,8 @@ struct snd_ice1712_card_info snd_ice1712_hoontech_cards[] __devinitdata = { .name = "Hoontech SoundTrack Audio DSP24", .model = "dsp24", .chip_init = snd_ice1712_hoontech_init, + .mpu401_1_name = "MIDI-1 Hoontech/STA DSP24", + .mpu401_2_name = "MIDI-2 Hoontech/STA DSP24", }, { .subvendor = ICE1712_SUBDEVICE_STDSP24_VALUE, /* a dummy id */ diff --git a/sound/pci/ice1712/ice1712.c b/sound/pci/ice1712/ice1712.c index df292af..38e93ca 100644 --- a/sound/pci/ice1712/ice1712.c +++ b/sound/pci/ice1712/ice1712.c @@ -1297,11 +1297,14 @@ static void snd_ice1712_update_volume(struct snd_ice1712 *ice, int index) static int snd_ice1712_pro_mixer_switch_get(struct snd_kcontrol *kcontrol, struct snd_ctl_elem_value *ucontrol) { struct snd_ice1712 *ice = snd_kcontrol_chip(kcontrol); - int index = snd_ctl_get_ioffidx(kcontrol, &ucontrol->id) + kcontrol->private_value; + int priv_idx = snd_ctl_get_ioffidx(kcontrol, &ucontrol->id) + + kcontrol->private_value; spin_lock_irq(&ice->reg_lock); - ucontrol->value.integer.value[0] = !((ice->pro_volumes[index] >> 15) & 1); - ucontrol->value.integer.value[1] = !((ice->pro_volumes[index] >> 31) & 1); + ucontrol->value.integer.value[0] = + !((ice->pro_volumes[priv_idx] >> 15) & 1); + ucontrol->value.integer.value[1] = + !((ice->pro_volumes[priv_idx] >> 31) & 1); spin_unlock_irq(&ice->reg_lock); return 0; } @@ -1309,16 +1312,17 @@ static int snd_ice1712_pro_mixer_switch_get(struct snd_kcontrol *kcontrol, struc static int snd_ice1712_pro_mixer_switch_put(struct snd_kcontrol *kcontrol, struct snd_ctl_elem_value *ucontrol) { struct snd_ice1712 *ice = snd_kcontrol_chip(kcontrol); - int index = snd_ctl_get_ioffidx(kcontrol, &ucontrol->id) + kcontrol->private_value; + int priv_idx = snd_ctl_get_ioffidx(kcontrol, &ucontrol->id) + + kcontrol->private_value; unsigned int nval, change; nval = (ucontrol->value.integer.value[0] ? 0 : 0x00008000) | (ucontrol->value.integer.value[1] ? 0 : 0x80000000); spin_lock_irq(&ice->reg_lock); - nval |= ice->pro_volumes[index] & ~0x80008000; - change = nval != ice->pro_volumes[index]; - ice->pro_volumes[index] = nval; - snd_ice1712_update_volume(ice, index); + nval |= ice->pro_volumes[priv_idx] & ~0x80008000; + change = nval != ice->pro_volumes[priv_idx]; + ice->pro_volumes[priv_idx] = nval; + snd_ice1712_update_volume(ice, priv_idx); spin_unlock_irq(&ice->reg_lock); return change; } @@ -1335,11 +1339,14 @@ static int snd_ice1712_pro_mixer_volume_info(struct snd_kcontrol *kcontrol, stru static int snd_ice1712_pro_mixer_volume_get(struct snd_kcontrol *kcontrol, struct snd_ctl_elem_value *ucontrol) { struct snd_ice1712 *ice = snd_kcontrol_chip(kcontrol); - int index = snd_ctl_get_ioffidx(kcontrol, &ucontrol->id) + kcontrol->private_value; + int priv_idx = snd_ctl_get_ioffidx(kcontrol, &ucontrol->id) + + kcontrol->private_value; spin_lock_irq(&ice->reg_lock); - ucontrol->value.integer.value[0] = (ice->pro_volumes[index] >> 0) & 127; - ucontrol->value.integer.value[1] = (ice->pro_volumes[index] >> 16) & 127; + ucontrol->value.integer.value[0] = + (ice->pro_volumes[priv_idx] >> 0) & 127; + ucontrol->value.integer.value[1] = + (ice->pro_volumes[priv_idx] >> 16) & 127; spin_unlock_irq(&ice->reg_lock); return 0; } @@ -1347,16 +1354,17 @@ static int snd_ice1712_pro_mixer_volume_get(struct snd_kcontrol *kcontrol, struc static int snd_ice1712_pro_mixer_volume_put(struct snd_kcontrol *kcontrol, struct snd_ctl_elem_value *ucontrol) { struct snd_ice1712 *ice = snd_kcontrol_chip(kcontrol); - int index = snd_ctl_get_ioffidx(kcontrol, &ucontrol->id) + kcontrol->private_value; + int priv_idx = snd_ctl_get_ioffidx(kcontrol, &ucontrol->id) + + kcontrol->private_value; unsigned int nval, change; nval = (ucontrol->value.integer.value[0] & 127) | ((ucontrol->value.integer.value[1] & 127) << 16); spin_lock_irq(&ice->reg_lock); - nval |= ice->pro_volumes[index] & ~0x007f007f; - change = nval != ice->pro_volumes[index]; - ice->pro_volumes[index] = nval; - snd_ice1712_update_volume(ice, index); + nval |= ice->pro_volumes[priv_idx] & ~0x007f007f; + change = nval != ice->pro_volumes[priv_idx]; + ice->pro_volumes[priv_idx] = nval; + snd_ice1712_update_volume(ice, priv_idx); spin_unlock_irq(&ice->reg_lock); return change; } diff --git a/sound/pci/ice1712/ice1712.h b/sound/pci/ice1712/ice1712.h index 303cffe..a3bea22 100644 --- a/sound/pci/ice1712/ice1712.h +++ b/sound/pci/ice1712/ice1712.h @@ -367,6 +367,15 @@ struct snd_ice1712 { /* other board-specific data */ void *spec; + + /* VT172x specific */ + int pro_rate_default; + int (*is_spdif_master)(struct snd_ice1712 *ice); + unsigned int (*get_rate)(struct snd_ice1712 *ice); + void (*set_rate)(struct snd_ice1712 *ice, unsigned int rate); + unsigned char (*set_mclk)(struct snd_ice1712 *ice, unsigned int rate); + void (*set_spdif_clock)(struct snd_ice1712 *ice); + }; diff --git a/sound/pci/ice1712/ice1724.c b/sound/pci/ice1712/ice1724.c index f533850..ceac870 100644 --- a/sound/pci/ice1712/ice1724.c +++ b/sound/pci/ice1712/ice1724.c @@ -106,15 +106,19 @@ static unsigned int PRO_RATE_DEFAULT = 44100; * Basic I/O */ +/* + * default rates, default clock routines + */ + /* check whether the clock mode is spdif-in */ -static inline int is_spdif_master(struct snd_ice1712 *ice) +static inline int stdclock_is_spdif_master(struct snd_ice1712 *ice) { return (inb(ICEMT1724(ice, RATE)) & VT1724_SPDIF_MASTER) ? 1 : 0; } static inline int is_pro_rate_locked(struct snd_ice1712 *ice) { - return is_spdif_master(ice) || PRO_RATE_LOCKED; + return ice->is_spdif_master(ice) || PRO_RATE_LOCKED; } /* @@ -391,51 +395,61 @@ static int snd_vt1724_pcm_trigger(struct snd_pcm_substream *substream, int cmd) #define DMA_PAUSES (VT1724_RDMA0_PAUSE|VT1724_PDMA0_PAUSE|VT1724_RDMA1_PAUSE|\ VT1724_PDMA1_PAUSE|VT1724_PDMA2_PAUSE|VT1724_PDMA3_PAUSE|VT1724_PDMA4_PAUSE) -static int get_max_rate(struct snd_ice1712 *ice) +static const unsigned int stdclock_rate_list[16] = { + 48000, 24000, 12000, 9600, 32000, 16000, 8000, 96000, 44100, + 22050, 11025, 88200, 176400, 0, 192000, 64000 +}; + +static unsigned int stdclock_get_rate(struct snd_ice1712 *ice) { + unsigned int rate; + rate = stdclock_rate_list[inb(ICEMT1724(ice, RATE)) & 15]; + return rate; +} + +static void stdclock_set_rate(struct snd_ice1712 *ice, unsigned int rate) +{ + int i; + for (i = 0; i < ARRAY_SIZE(stdclock_rate_list); i++) { + if (stdclock_rate_list[i] == rate) { + outb(i, ICEMT1724(ice, RATE)); + return; + } + } +} + +static unsigned char stdclock_set_mclk(struct snd_ice1712 *ice, + unsigned int rate) +{ + unsigned char val, old; + /* check MT02 */ if (ice->eeprom.data[ICE_EEP2_ACLINK] & VT1724_CFG_PRO_I2S) { - if ((ice->eeprom.data[ICE_EEP2_I2S] & 0x08) && !ice->vt1720) - return 192000; + val = old = inb(ICEMT1724(ice, I2S_FORMAT)); + if (rate > 96000) + val |= VT1724_MT_I2S_MCLK_128X; /* 128x MCLK */ else - return 96000; - } else - return 48000; + val &= ~VT1724_MT_I2S_MCLK_128X; /* 256x MCLK */ + if (val != old) { + outb(val, ICEMT1724(ice, I2S_FORMAT)); + /* master clock changed */ + return 1; + } + } + /* no change in master clock */ + return 0; } static void snd_vt1724_set_pro_rate(struct snd_ice1712 *ice, unsigned int rate, int force) { unsigned long flags; - unsigned char val, old; - unsigned int i, mclk_change; + unsigned char mclk_change; + unsigned int i, old_rate; - if (rate > get_max_rate(ice)) + if (rate > ice->hw_rates->list[ice->hw_rates->count - 1]) return; - - switch (rate) { - case 8000: val = 6; break; - case 9600: val = 3; break; - case 11025: val = 10; break; - case 12000: val = 2; break; - case 16000: val = 5; break; - case 22050: val = 9; break; - case 24000: val = 1; break; - case 32000: val = 4; break; - case 44100: val = 8; break; - case 48000: val = 0; break; - case 64000: val = 15; break; - case 88200: val = 11; break; - case 96000: val = 7; break; - case 176400: val = 12; break; - case 192000: val = 14; break; - default: - snd_BUG(); - val = 0; - break; - } - spin_lock_irqsave(&ice->reg_lock, flags); - if ((inb(ICEMT1724(ice, DMA_CONTROL)) & DMA_STARTS) || + if ((inb(ICEMT1724(ice, DMA_CONTROL)) & DMA_STARTS) || (inb(ICEMT1724(ice, DMA_PAUSE)) & DMA_PAUSES)) { /* running? we cannot change the rate now... */ spin_unlock_irqrestore(&ice->reg_lock, flags); @@ -446,9 +460,9 @@ static void snd_vt1724_set_pro_rate(struct snd_ice1712 *ice, unsigned int rate, return; } - old = inb(ICEMT1724(ice, RATE)); - if (force || old != val) - outb(val, ICEMT1724(ice, RATE)); + old_rate = ice->get_rate(ice); + if (force || (old_rate != rate)) + ice->set_rate(ice, rate); else if (rate == ice->cur_rate) { spin_unlock_irqrestore(&ice->reg_lock, flags); return; @@ -456,19 +470,9 @@ static void snd_vt1724_set_pro_rate(struct snd_ice1712 *ice, unsigned int rate, ice->cur_rate = rate; - /* check MT02 */ - mclk_change = 0; - if (ice->eeprom.data[ICE_EEP2_ACLINK] & VT1724_CFG_PRO_I2S) { - val = old = inb(ICEMT1724(ice, I2S_FORMAT)); - if (rate > 96000) - val |= VT1724_MT_I2S_MCLK_128X; /* 128x MCLK */ - else - val &= ~VT1724_MT_I2S_MCLK_128X; /* 256x MCLK */ - if (val != old) { - outb(val, ICEMT1724(ice, I2S_FORMAT)); - mclk_change = 1; - } - } + /* setting master clock */ + mclk_change = ice->set_mclk(ice, rate); + spin_unlock_irqrestore(&ice->reg_lock, flags); if (mclk_change && ice->gpio.i2s_mclk_changed) @@ -727,43 +731,32 @@ static const struct snd_pcm_hardware snd_vt1724_2ch_stereo = /* * set rate constraints */ -static int set_rate_constraints(struct snd_ice1712 *ice, - struct snd_pcm_substream *substream) +static void set_std_hw_rates(struct snd_ice1712 *ice) { - struct snd_pcm_runtime *runtime = substream->runtime; - if (ice->hw_rates) { - /* hardware specific */ - runtime->hw.rate_min = ice->hw_rates->list[0]; - runtime->hw.rate_max = ice->hw_rates->list[ice->hw_rates->count - 1]; - runtime->hw.rates = SNDRV_PCM_RATE_KNOT; - return snd_pcm_hw_constraint_list(runtime, 0, - SNDRV_PCM_HW_PARAM_RATE, - ice->hw_rates); - } if (ice->eeprom.data[ICE_EEP2_ACLINK] & VT1724_CFG_PRO_I2S) { /* I2S */ /* VT1720 doesn't support more than 96kHz */ if ((ice->eeprom.data[ICE_EEP2_I2S] & 0x08) && !ice->vt1720) - return snd_pcm_hw_constraint_list(runtime, 0, - SNDRV_PCM_HW_PARAM_RATE, - &hw_constraints_rates_192); - else { - runtime->hw.rates = SNDRV_PCM_RATE_KNOT | - SNDRV_PCM_RATE_8000_96000; - runtime->hw.rate_max = 96000; - return snd_pcm_hw_constraint_list(runtime, 0, - SNDRV_PCM_HW_PARAM_RATE, - &hw_constraints_rates_96); - } - } else if (ice->ac97) { + ice->hw_rates = &hw_constraints_rates_192; + else + ice->hw_rates = &hw_constraints_rates_96; + } else { /* ACLINK */ - runtime->hw.rate_max = 48000; - runtime->hw.rates = SNDRV_PCM_RATE_KNOT | SNDRV_PCM_RATE_8000_48000; - return snd_pcm_hw_constraint_list(runtime, 0, - SNDRV_PCM_HW_PARAM_RATE, - &hw_constraints_rates_48); + ice->hw_rates = &hw_constraints_rates_48; } - return 0; +} + +static int set_rate_constraints(struct snd_ice1712 *ice, + struct snd_pcm_substream *substream) +{ + struct snd_pcm_runtime *runtime = substream->runtime; + + runtime->hw.rate_min = ice->hw_rates->list[0]; + runtime->hw.rate_max = ice->hw_rates->list[ice->hw_rates->count - 1]; + runtime->hw.rates = SNDRV_PCM_RATE_KNOT; + return snd_pcm_hw_constraint_list(runtime, 0, + SNDRV_PCM_HW_PARAM_RATE, + ice->hw_rates); } /* multi-channel playback needs alignment 8x32bit regardless of the channels @@ -824,7 +817,7 @@ static int snd_vt1724_playback_pro_close(struct snd_pcm_substream *substream) struct snd_ice1712 *ice = snd_pcm_substream_chip(substream); if (PRO_RATE_RESET) - snd_vt1724_set_pro_rate(ice, PRO_RATE_DEFAULT, 0); + snd_vt1724_set_pro_rate(ice, ice->pro_rate_default, 0); ice->playback_pro_substream = NULL; return 0; @@ -835,7 +828,7 @@ static int snd_vt1724_capture_pro_close(struct snd_pcm_substream *substream) struct snd_ice1712 *ice = snd_pcm_substream_chip(substream); if (PRO_RATE_RESET) - snd_vt1724_set_pro_rate(ice, PRO_RATE_DEFAULT, 0); + snd_vt1724_set_pro_rate(ice, ice->pro_rate_default, 0); ice->capture_pro_substream = NULL; return 0; } @@ -970,6 +963,8 @@ static int snd_vt1724_playback_spdif_open(struct snd_pcm_substream *substream) VT1724_BUFFER_ALIGN); snd_pcm_hw_constraint_step(runtime, 0, SNDRV_PCM_HW_PARAM_BUFFER_BYTES, VT1724_BUFFER_ALIGN); + if (ice->spdif.ops.open) + ice->spdif.ops.open(ice, substream); return 0; } @@ -978,8 +973,10 @@ static int snd_vt1724_playback_spdif_close(struct snd_pcm_substream *substream) struct snd_ice1712 *ice = snd_pcm_substream_chip(substream); if (PRO_RATE_RESET) - snd_vt1724_set_pro_rate(ice, PRO_RATE_DEFAULT, 0); + snd_vt1724_set_pro_rate(ice, ice->pro_rate_default, 0); ice->playback_con_substream = NULL; + if (ice->spdif.ops.close) + ice->spdif.ops.close(ice, substream); return 0; } @@ -1002,6 +999,8 @@ static int snd_vt1724_capture_spdif_open(struct snd_pcm_substream *substream) VT1724_BUFFER_ALIGN); snd_pcm_hw_constraint_step(runtime, 0, SNDRV_PCM_HW_PARAM_BUFFER_BYTES, VT1724_BUFFER_ALIGN); + if (ice->spdif.ops.open) + ice->spdif.ops.open(ice, substream); return 0; } @@ -1010,8 +1009,10 @@ static int snd_vt1724_capture_spdif_close(struct snd_pcm_substream *substream) struct snd_ice1712 *ice = snd_pcm_substream_chip(substream); if (PRO_RATE_RESET) - snd_vt1724_set_pro_rate(ice, PRO_RATE_DEFAULT, 0); + snd_vt1724_set_pro_rate(ice, ice->pro_rate_default, 0); ice->capture_con_substream = NULL; + if (ice->spdif.ops.close) + ice->spdif.ops.close(ice, substream); return 0; } @@ -1154,7 +1155,7 @@ static int snd_vt1724_playback_indep_close(struct snd_pcm_substream *substream) struct snd_ice1712 *ice = snd_pcm_substream_chip(substream); if (PRO_RATE_RESET) - snd_vt1724_set_pro_rate(ice, PRO_RATE_DEFAULT, 0); + snd_vt1724_set_pro_rate(ice, ice->pro_rate_default, 0); ice->playback_con_substream_ds[substream->number] = NULL; ice->pcm_reserved[substream->number] = NULL; @@ -1572,50 +1573,18 @@ int snd_ice1712_gpio_put(struct snd_kcontrol *kcontrol, static int snd_vt1724_pro_internal_clock_info(struct snd_kcontrol *kcontrol, struct snd_ctl_elem_info *uinfo) { - static const char * const texts_1724[] = { - "8000", /* 0: 6 */ - "9600", /* 1: 3 */ - "11025", /* 2: 10 */ - "12000", /* 3: 2 */ - "16000", /* 4: 5 */ - "22050", /* 5: 9 */ - "24000", /* 6: 1 */ - "32000", /* 7: 4 */ - "44100", /* 8: 8 */ - "48000", /* 9: 0 */ - "64000", /* 10: 15 */ - "88200", /* 11: 11 */ - "96000", /* 12: 7 */ - "176400", /* 13: 12 */ - "192000", /* 14: 14 */ - "IEC958 Input", /* 15: -- */ - }; - static const char * const texts_1720[] = { - "8000", /* 0: 6 */ - "9600", /* 1: 3 */ - "11025", /* 2: 10 */ - "12000", /* 3: 2 */ - "16000", /* 4: 5 */ - "22050", /* 5: 9 */ - "24000", /* 6: 1 */ - "32000", /* 7: 4 */ - "44100", /* 8: 8 */ - "48000", /* 9: 0 */ - "64000", /* 10: 15 */ - "88200", /* 11: 11 */ - "96000", /* 12: 7 */ - "IEC958 Input", /* 13: -- */ - }; struct snd_ice1712 *ice = snd_kcontrol_chip(kcontrol); uinfo->type = SNDRV_CTL_ELEM_TYPE_ENUMERATED; uinfo->count = 1; - uinfo->value.enumerated.items = ice->vt1720 ? 14 : 16; + uinfo->value.enumerated.items = ice->hw_rates->count + 1; if (uinfo->value.enumerated.item >= uinfo->value.enumerated.items) uinfo->value.enumerated.item = uinfo->value.enumerated.items - 1; - strcpy(uinfo->value.enumerated.name, - ice->vt1720 ? texts_1720[uinfo->value.enumerated.item] : - texts_1724[uinfo->value.enumerated.item]); + if (uinfo->value.enumerated.item == uinfo->value.enumerated.items - 1) + strcpy(uinfo->value.enumerated.name, "IEC958 Input"); + else + sprintf(uinfo->value.enumerated.name, "%d", + ice->hw_rates->list[uinfo->value.enumerated.item]); return 0; } @@ -1623,68 +1592,79 @@ static int snd_vt1724_pro_internal_clock_get(struct snd_kcontrol *kcontrol, struct snd_ctl_elem_value *ucontrol) { struct snd_ice1712 *ice = snd_kcontrol_chip(kcontrol); - static const unsigned char xlate[16] = { - 9, 6, 3, 1, 7, 4, 0, 12, 8, 5, 2, 11, 13, 255, 14, 10 - }; - unsigned char val; + unsigned int i, rate; spin_lock_irq(&ice->reg_lock); - if (is_spdif_master(ice)) { - ucontrol->value.enumerated.item[0] = ice->vt1720 ? 13 : 15; + if (ice->is_spdif_master(ice)) { + ucontrol->value.enumerated.item[0] = ice->hw_rates->count; } else { - val = xlate[inb(ICEMT1724(ice, RATE)) & 15]; - if (val == 255) { - snd_BUG(); - val = 0; + rate = ice->get_rate(ice); + ucontrol->value.enumerated.item[0] = 0; + for (i = 0; i < ice->hw_rates->count; i++) { + if (ice->hw_rates->list[i] == rate) { + ucontrol->value.enumerated.item[0] = i; + break; + } } - ucontrol->value.enumerated.item[0] = val; } spin_unlock_irq(&ice->reg_lock); return 0; } +/* setting clock to external - SPDIF */ +static void stdclock_set_spdif_clock(struct snd_ice1712 *ice) +{ + unsigned char oval; + unsigned char i2s_oval; + oval = inb(ICEMT1724(ice, RATE)); + outb(oval | VT1724_SPDIF_MASTER, ICEMT1724(ice, RATE)); + /* setting 256fs */ + i2s_oval = inb(ICEMT1724(ice, I2S_FORMAT)); + outb(i2s_oval & ~VT1724_MT_I2S_MCLK_128X, ICEMT1724(ice, I2S_FORMAT)); +} + static int snd_vt1724_pro_internal_clock_put(struct snd_kcontrol *kcontrol, struct snd_ctl_elem_value *ucontrol) { struct snd_ice1712 *ice = snd_kcontrol_chip(kcontrol); - unsigned char oval; - int rate; - int change = 0; - int spdif = ice->vt1720 ? 13 : 15; + unsigned int old_rate, new_rate; + unsigned int item = ucontrol->value.enumerated.item[0]; + unsigned int spdif = ice->hw_rates->count; + + if (item > spdif) + return -EINVAL; spin_lock_irq(&ice->reg_lock); - oval = inb(ICEMT1724(ice, RATE)); - if (ucontrol->value.enumerated.item[0] == spdif) { - unsigned char i2s_oval; - outb(oval | VT1724_SPDIF_MASTER, ICEMT1724(ice, RATE)); - /* setting 256fs */ - i2s_oval = inb(ICEMT1724(ice, I2S_FORMAT)); - outb(i2s_oval & ~VT1724_MT_I2S_MCLK_128X, - ICEMT1724(ice, I2S_FORMAT)); + if (ice->is_spdif_master(ice)) + old_rate = 0; + else + old_rate = ice->get_rate(ice); + if (item == spdif) { + /* switching to external clock via SPDIF */ + ice->set_spdif_clock(ice); + new_rate = 0; } else { - rate = rates[ucontrol->value.integer.value[0] % 15]; - if (rate <= get_max_rate(ice)) { - PRO_RATE_DEFAULT = rate; - spin_unlock_irq(&ice->reg_lock); - snd_vt1724_set_pro_rate(ice, PRO_RATE_DEFAULT, 1); - spin_lock_irq(&ice->reg_lock); - } + /* internal on-card clock */ + new_rate = ice->hw_rates->list[item]; + ice->pro_rate_default = new_rate; + spin_unlock_irq(&ice->reg_lock); + snd_vt1724_set_pro_rate(ice, ice->pro_rate_default, 1); + spin_lock_irq(&ice->reg_lock); } - change = inb(ICEMT1724(ice, RATE)) != oval; spin_unlock_irq(&ice->reg_lock); - if ((oval & VT1724_SPDIF_MASTER) != - (inb(ICEMT1724(ice, RATE)) & VT1724_SPDIF_MASTER)) { + /* the first reset to the SPDIF master mode? */ + if (old_rate != new_rate && !new_rate) { /* notify akm chips as well */ - if (is_spdif_master(ice)) { - unsigned int i; - for (i = 0; i < ice->akm_codecs; i++) { - if (ice->akm[i].ops.set_rate_val) - ice->akm[i].ops.set_rate_val(&ice->akm[i], 0); - } + unsigned int i; + if (ice->gpio.set_pro_rate) + ice->gpio.set_pro_rate(ice, 0); + for (i = 0; i < ice->akm_codecs; i++) { + if (ice->akm[i].ops.set_rate_val) + ice->akm[i].ops.set_rate_val(&ice->akm[i], 0); } } - return change; + return old_rate != new_rate; } static struct snd_kcontrol_new snd_vt1724_pro_internal_clock __devinitdata = { @@ -2335,6 +2315,19 @@ static int __devinit snd_vt1724_probe(struct pci_dev *pci, * was called so in ice1712 driver, and vt1724 driver is derived from * ice1712 driver. */ + ice->pro_rate_default = PRO_RATE_DEFAULT; + if (!ice->is_spdif_master) + ice->is_spdif_master = stdclock_is_spdif_master; + if (!ice->get_rate) + ice->get_rate = stdclock_get_rate; + if (!ice->set_rate) + ice->set_rate = stdclock_set_rate; + if (!ice->set_mclk) + ice->set_mclk = stdclock_set_mclk; + if (!ice->set_spdif_clock) + ice->set_spdif_clock = stdclock_set_spdif_clock; + if (!ice->hw_rates) + set_std_hw_rates(ice); if ((err = snd_vt1724_pcm_profi(ice, pcm_dev++)) < 0) { snd_card_free(card); diff --git a/sound/pci/ice1712/juli.c b/sound/pci/ice1712/juli.c index e8038c0..b4e0c16 100644 --- a/sound/pci/ice1712/juli.c +++ b/sound/pci/ice1712/juli.c @@ -4,6 +4,8 @@ * Lowlevel functions for ESI Juli@ cards * * Copyright (c) 2004 Jaroslav Kysela + * 2008 Pavel Hofman + * * * This program is free software; you can redistribute it and/or modify * it under the terms of the GNU General Public License as published by @@ -27,11 +29,11 @@ #include #include #include +#include #include "ice1712.h" #include "envy24ht.h" #include "juli.h" - struct juli_spec { struct ak4114 *ak4114; unsigned int analog: 1; @@ -44,6 +46,32 @@ struct juli_spec { #define AK4358_ADDR 0x22 /* DAC */ /* + * Juli does not use the standard ICE1724 clock scheme. Juli's ice1724 chip is + * supplied by external clock provided by Xilinx array and MK73-1 PLL frequency + * multiplier. Actual frequency is set by ice1724 GPIOs hooked to the Xilinx. + * + * The clock circuitry is supplied by the two ice1724 crystals. This + * arrangement allows to generate independent clock signal for AK4114's input + * rate detection circuit. As a result, Juli, unlike most other + * ice1724+ak4114-based cards, detects spdif input rate correctly. + * This fact is applied in the driver, allowing to modify PCM stream rate + * parameter according to the actual input rate. + * + * Juli uses the remaining three stereo-channels of its DAC to optionally + * monitor analog input, digital input, and digital output. The corresponding + * I2S signals are routed by Xilinx, controlled by GPIOs. + * + * The master mute is implemented using output muting transistors (GPIO) in + * combination with smuting the DAC. + * + * The card itself has no HW master volume control, implemented using the + * vmaster control. + * + * TODO: + * researching and fixing the input monitors + */ + +/* * GPIO pins */ #define GPIO_FREQ_MASK (3<<0) @@ -55,17 +83,82 @@ struct juli_spec { #define GPIO_MULTI_2X (1<<2) #define GPIO_MULTI_1X (2<<2) /* also external */ #define GPIO_MULTI_HALF (3<<2) -#define GPIO_INTERNAL_CLOCK (1<<4) +#define GPIO_INTERNAL_CLOCK (1<<4) /* 0 = external, 1 = internal */ +#define GPIO_CLOCK_MASK (1<<4) #define GPIO_ANALOG_PRESENT (1<<5) /* RO only: 0 = present */ #define GPIO_RXMCLK_SEL (1<<7) /* must be 0 */ #define GPIO_AK5385A_CKS0 (1<<8) -#define GPIO_AK5385A_DFS0 (1<<9) /* swapped with DFS1 according doc? */ -#define GPIO_AK5385A_DFS1 (1<<10) +#define GPIO_AK5385A_DFS1 (1<<9) +#define GPIO_AK5385A_DFS0 (1<<10) #define GPIO_DIGOUT_MONITOR (1<<11) /* 1 = active */ #define GPIO_DIGIN_MONITOR (1<<12) /* 1 = active */ #define GPIO_ANAIN_MONITOR (1<<13) /* 1 = active */ -#define GPIO_AK5385A_MCLK (1<<14) /* must be 0 */ -#define GPIO_MUTE_CONTROL (1<<15) /* 0 = off, 1 = on */ +#define GPIO_AK5385A_CKS1 (1<<14) /* must be 0 */ +#define GPIO_MUTE_CONTROL (1<<15) /* output mute, 1 = muted */ + +#define GPIO_RATE_MASK (GPIO_FREQ_MASK | GPIO_MULTI_MASK | \ + GPIO_CLOCK_MASK) +#define GPIO_AK5385A_MASK (GPIO_AK5385A_CKS0 | GPIO_AK5385A_DFS0 | \ + GPIO_AK5385A_DFS1 | GPIO_AK5385A_CKS1) + +#define JULI_PCM_RATE (SNDRV_PCM_RATE_16000 | SNDRV_PCM_RATE_22050 | \ + SNDRV_PCM_RATE_32000 | SNDRV_PCM_RATE_44100 | \ + SNDRV_PCM_RATE_48000 | SNDRV_PCM_RATE_64000 | \ + SNDRV_PCM_RATE_88200 | SNDRV_PCM_RATE_96000 | \ + SNDRV_PCM_RATE_176400 | SNDRV_PCM_RATE_192000) + +#define GPIO_RATE_16000 (GPIO_FREQ_32KHZ | GPIO_MULTI_HALF | \ + GPIO_INTERNAL_CLOCK) +#define GPIO_RATE_22050 (GPIO_FREQ_44KHZ | GPIO_MULTI_HALF | \ + GPIO_INTERNAL_CLOCK) +#define GPIO_RATE_24000 (GPIO_FREQ_48KHZ | GPIO_MULTI_HALF | \ + GPIO_INTERNAL_CLOCK) +#define GPIO_RATE_32000 (GPIO_FREQ_32KHZ | GPIO_MULTI_1X | \ + GPIO_INTERNAL_CLOCK) +#define GPIO_RATE_44100 (GPIO_FREQ_44KHZ | GPIO_MULTI_1X | \ + GPIO_INTERNAL_CLOCK) +#define GPIO_RATE_48000 (GPIO_FREQ_48KHZ | GPIO_MULTI_1X | \ + GPIO_INTERNAL_CLOCK) +#define GPIO_RATE_64000 (GPIO_FREQ_32KHZ | GPIO_MULTI_2X | \ + GPIO_INTERNAL_CLOCK) +#define GPIO_RATE_88200 (GPIO_FREQ_44KHZ | GPIO_MULTI_2X | \ + GPIO_INTERNAL_CLOCK) +#define GPIO_RATE_96000 (GPIO_FREQ_48KHZ | GPIO_MULTI_2X | \ + GPIO_INTERNAL_CLOCK) +#define GPIO_RATE_176400 (GPIO_FREQ_44KHZ | GPIO_MULTI_4X | \ + GPIO_INTERNAL_CLOCK) +#define GPIO_RATE_192000 (GPIO_FREQ_48KHZ | GPIO_MULTI_4X | \ + GPIO_INTERNAL_CLOCK) + +/* + * Initial setup of the conversion array GPIO <-> rate + */ +static unsigned int juli_rates[] = { + 16000, 22050, 24000, 32000, + 44100, 48000, 64000, 88200, + 96000, 176400, 192000, +}; + +static unsigned int gpio_vals[] = { + GPIO_RATE_16000, GPIO_RATE_22050, GPIO_RATE_24000, GPIO_RATE_32000, + GPIO_RATE_44100, GPIO_RATE_48000, GPIO_RATE_64000, GPIO_RATE_88200, + GPIO_RATE_96000, GPIO_RATE_176400, GPIO_RATE_192000, +}; + +static struct snd_pcm_hw_constraint_list juli_rates_info = { + .count = ARRAY_SIZE(juli_rates), + .list = juli_rates, + .mask = 0, +}; + +static int get_gpio_val(int rate) +{ + int i; + for (i = 0; i < ARRAY_SIZE(juli_rates); i++) + if (juli_rates[i] == rate) + return gpio_vals[i]; + return 0; +} static void juli_ak4114_write(void *private_data, unsigned char reg, unsigned char val) { @@ -78,6 +171,27 @@ static unsigned char juli_ak4114_read(void *private_data, unsigned char reg) } /* + * If SPDIF capture and slaved to SPDIF-IN, setting runtime rate + * to the external rate + */ +static void juli_spdif_in_open(struct snd_ice1712 *ice, + struct snd_pcm_substream *substream) +{ + struct juli_spec *spec = ice->spec; + struct snd_pcm_runtime *runtime = substream->runtime; + int rate; + + if (substream->stream == SNDRV_PCM_STREAM_PLAYBACK || + !ice->is_spdif_master(ice)) + return; + rate = snd_ak4114_external_rate(spec->ak4114); + if (rate >= runtime->hw.rate_min && rate <= runtime->hw.rate_max) { + runtime->hw.rate_min = rate; + runtime->hw.rate_max = rate; + } +} + +/* * AK4358 section */ @@ -99,57 +213,285 @@ static void juli_akm_write(struct snd_akm4xxx *ak, int chip, } /* - * change the rate of envy24HT, AK4358 + * change the rate of envy24HT, AK4358, AK5385 */ static void juli_akm_set_rate_val(struct snd_akm4xxx *ak, unsigned int rate) { - unsigned char old, tmp, dfs; + unsigned char old, tmp, ak4358_dfs; + unsigned int ak5385_pins, old_gpio, new_gpio; + struct snd_ice1712 *ice = ak->private_data[0]; + struct juli_spec *spec = ice->spec; - if (rate == 0) /* no hint - S/PDIF input is master, simply return */ + if (rate == 0) /* no hint - S/PDIF input is master or the new spdif + input rate undetected, simply return */ return; - + /* adjust DFS on codecs */ - if (rate > 96000) - dfs = 2; - else if (rate > 48000) - dfs = 1; - else - dfs = 0; - + if (rate > 96000) { + ak4358_dfs = 2; + ak5385_pins = GPIO_AK5385A_DFS1 | GPIO_AK5385A_CKS0; + } else if (rate > 48000) { + ak4358_dfs = 1; + ak5385_pins = GPIO_AK5385A_DFS0; + } else { + ak4358_dfs = 0; + ak5385_pins = 0; + } + /* AK5385 first, since it requires cold reset affecting both codecs */ + old_gpio = ice->gpio.get_data(ice); + new_gpio = (old_gpio & ~GPIO_AK5385A_MASK) | ak5385_pins; + /* printk(KERN_DEBUG "JULI - ak5385 set_rate_val: new gpio 0x%x\n", + new_gpio); */ + ice->gpio.set_data(ice, new_gpio); + + /* cold reset */ + old = inb(ICEMT1724(ice, AC97_CMD)); + outb(old | VT1724_AC97_COLD, ICEMT1724(ice, AC97_CMD)); + udelay(1); + outb(old & ~VT1724_AC97_COLD, ICEMT1724(ice, AC97_CMD)); + + /* AK4358 */ + /* set new value, reset DFS */ tmp = snd_akm4xxx_get(ak, 0, 2); - old = (tmp >> 4) & 0x03; - if (old == dfs) - return; - /* reset DFS */ snd_akm4xxx_reset(ak, 1); tmp = snd_akm4xxx_get(ak, 0, 2); tmp &= ~(0x03 << 4); - tmp |= dfs << 4; + tmp |= ak4358_dfs << 4; snd_akm4xxx_set(ak, 0, 2, tmp); snd_akm4xxx_reset(ak, 0); + + /* reinit ak4114 */ + snd_ak4114_reinit(spec->ak4114); } +#define AK_DAC(xname, xch) { .name = xname, .num_channels = xch } +#define PCM_VOLUME "PCM Playback Volume" +#define MONITOR_AN_IN_VOLUME "Monitor Analog In Volume" +#define MONITOR_DIG_IN_VOLUME "Monitor Digital In Volume" +#define MONITOR_DIG_OUT_VOLUME "Monitor Digital Out Volume" + +static const struct snd_akm4xxx_dac_channel juli_dac[] = { + AK_DAC(PCM_VOLUME, 2), + AK_DAC(MONITOR_AN_IN_VOLUME, 2), + AK_DAC(MONITOR_DIG_OUT_VOLUME, 2), + AK_DAC(MONITOR_DIG_IN_VOLUME, 2), +}; + + static struct snd_akm4xxx akm_juli_dac __devinitdata = { .type = SND_AK4358, - .num_dacs = 2, + .num_dacs = 8, /* DAC1 - analog out + DAC2 - analog in monitor + DAC3 - digital out monitor + DAC4 - digital in monitor + */ .ops = { .lock = juli_akm_lock, .unlock = juli_akm_unlock, .write = juli_akm_write, .set_rate_val = juli_akm_set_rate_val + }, + .dac_info = juli_dac, +}; + +#define juli_mute_info snd_ctl_boolean_mono_info + +static int juli_mute_get(struct snd_kcontrol *kcontrol, + struct snd_ctl_elem_value *ucontrol) +{ + struct snd_ice1712 *ice = snd_kcontrol_chip(kcontrol); + unsigned int val; + val = ice->gpio.get_data(ice) & (unsigned int) kcontrol->private_value; + if (kcontrol->private_value == GPIO_MUTE_CONTROL) + /* val 0 = signal on */ + ucontrol->value.integer.value[0] = (val) ? 0 : 1; + else + /* val 1 = signal on */ + ucontrol->value.integer.value[0] = (val) ? 1 : 0; + return 0; +} + +static int juli_mute_put(struct snd_kcontrol *kcontrol, + struct snd_ctl_elem_value *ucontrol) +{ + struct snd_ice1712 *ice = snd_kcontrol_chip(kcontrol); + unsigned int old_gpio, new_gpio; + old_gpio = ice->gpio.get_data(ice); + if (ucontrol->value.integer.value[0]) { + /* unmute */ + if (kcontrol->private_value == GPIO_MUTE_CONTROL) { + /* 0 = signal on */ + new_gpio = old_gpio & ~GPIO_MUTE_CONTROL; + /* un-smuting DAC */ + snd_akm4xxx_write(ice->akm, 0, 0x01, 0x01); + } else + /* 1 = signal on */ + new_gpio = old_gpio | + (unsigned int) kcontrol->private_value; + } else { + /* mute */ + if (kcontrol->private_value == GPIO_MUTE_CONTROL) { + /* 1 = signal off */ + new_gpio = old_gpio | GPIO_MUTE_CONTROL; + /* smuting DAC */ + snd_akm4xxx_write(ice->akm, 0, 0x01, 0x03); + } else + /* 0 = signal off */ + new_gpio = old_gpio & + ~((unsigned int) kcontrol->private_value); + } + /* printk("JULI - mute/unmute: control_value: 0x%x, old_gpio: 0x%x, \ + new_gpio 0x%x\n", + (unsigned int)ucontrol->value.integer.value[0], old_gpio, + new_gpio); */ + if (old_gpio != new_gpio) { + ice->gpio.set_data(ice, new_gpio); + return 1; + } + /* no change */ + return 0; +} + +static struct snd_kcontrol_new juli_mute_controls[] __devinitdata = { + { + .iface = SNDRV_CTL_ELEM_IFACE_MIXER, + .name = "Master Playback Switch", + .info = juli_mute_info, + .get = juli_mute_get, + .put = juli_mute_put, + .private_value = GPIO_MUTE_CONTROL, + }, + /* Although the following functionality respects the succint NDA'd + * documentation from the card manufacturer, and the same way of + * operation is coded in OSS Juli driver, only Digital Out monitor + * seems to work. Surprisingly, Analog input monitor outputs Digital + * output data. The two are independent, as enabling both doubles + * volume of the monitor sound. + * + * Checking traces on the board suggests the functionality described + * by the manufacturer is correct - I2S from ADC and AK4114 + * go to ICE as well as to Xilinx, I2S inputs of DAC2,3,4 (the monitor + * inputs) are fed from Xilinx. + * + * I even checked traces on board and coded a support in driver for + * an alternative possiblity - the unused I2S ICE output channels + * switched to HW-IN/SPDIF-IN and providing the monitoring signal to + * the DAC - to no avail. The I2S outputs seem to be unconnected. + * + * The windows driver supports the monitoring correctly. + */ + { + .iface = SNDRV_CTL_ELEM_IFACE_MIXER, + .name = "Monitor Analog In Switch", + .info = juli_mute_info, + .get = juli_mute_get, + .put = juli_mute_put, + .private_value = GPIO_ANAIN_MONITOR, + }, + { + .iface = SNDRV_CTL_ELEM_IFACE_MIXER, + .name = "Monitor Digital Out Switch", + .info = juli_mute_info, + .get = juli_mute_get, + .put = juli_mute_put, + .private_value = GPIO_DIGOUT_MONITOR, + }, + { + .iface = SNDRV_CTL_ELEM_IFACE_MIXER, + .name = "Monitor Digital In Switch", + .info = juli_mute_info, + .get = juli_mute_get, + .put = juli_mute_put, + .private_value = GPIO_DIGIN_MONITOR, + }, +}; + + +static void ak4358_proc_regs_read(struct snd_info_entry *entry, + struct snd_info_buffer *buffer) +{ + struct snd_ice1712 *ice = (struct snd_ice1712 *)entry->private_data; + int reg, val; + for (reg = 0; reg <= 0xf; reg++) { + val = snd_akm4xxx_get(ice->akm, 0, reg); + snd_iprintf(buffer, "0x%02x = 0x%02x\n", reg, val); } +} + +static void ak4358_proc_init(struct snd_ice1712 *ice) +{ + struct snd_info_entry *entry; + if (!snd_card_proc_new(ice->card, "ak4358_codec", &entry)) + snd_info_set_text_ops(entry, ice, ak4358_proc_regs_read); +} + +static char *slave_vols[] __devinitdata = { + PCM_VOLUME, + MONITOR_AN_IN_VOLUME, + MONITOR_DIG_IN_VOLUME, + MONITOR_DIG_OUT_VOLUME, + NULL }; +static __devinitdata +DECLARE_TLV_DB_SCALE(juli_master_db_scale, -6350, 50, 1); + +static struct snd_kcontrol __devinit *ctl_find(struct snd_card *card, + const char *name) +{ + struct snd_ctl_elem_id sid; + memset(&sid, 0, sizeof(sid)); + /* FIXME: strcpy is bad. */ + strcpy(sid.name, name); + sid.iface = SNDRV_CTL_ELEM_IFACE_MIXER; + return snd_ctl_find_id(card, &sid); +} + +static void __devinit add_slaves(struct snd_card *card, + struct snd_kcontrol *master, char **list) +{ + for (; *list; list++) { + struct snd_kcontrol *slave = ctl_find(card, *list); + /* printk(KERN_DEBUG "add_slaves - %s\n", *list); */ + if (slave) { + /* printk(KERN_DEBUG "slave %s found\n", *list); */ + snd_ctl_add_slave(master, slave); + } + } +} + static int __devinit juli_add_controls(struct snd_ice1712 *ice) { struct juli_spec *spec = ice->spec; int err; + unsigned int i; + struct snd_kcontrol *vmaster; + err = snd_ice1712_akm4xxx_build_controls(ice); if (err < 0) return err; + + for (i = 0; i < ARRAY_SIZE(juli_mute_controls); i++) { + err = snd_ctl_add(ice->card, + snd_ctl_new1(&juli_mute_controls[i], ice)); + if (err < 0) + return err; + } + /* Create virtual master control */ + vmaster = snd_ctl_make_virtual_master("Master Playback Volume", + juli_master_db_scale); + if (!vmaster) + return -ENOMEM; + add_slaves(ice->card, vmaster, slave_vols); + err = snd_ctl_add(ice->card, vmaster); + if (err < 0) + return err; + /* only capture SPDIF over AK4114 */ err = snd_ak4114_build(spec->ak4114, NULL, - ice->pcm_pro->streams[SNDRV_PCM_STREAM_CAPTURE].substream); + ice->pcm->streams[SNDRV_PCM_STREAM_CAPTURE].substream); + + ak4358_proc_init(ice); if (err < 0) return err; return 0; @@ -158,6 +500,74 @@ static int __devinit juli_add_controls(struct snd_ice1712 *ice) /* * initialize the chip */ + +static inline int juli_is_spdif_master(struct snd_ice1712 *ice) +{ + return (ice->gpio.get_data(ice) & GPIO_INTERNAL_CLOCK) ? 0 : 1; +} + +static unsigned int juli_get_rate(struct snd_ice1712 *ice) +{ + int i; + unsigned char result; + + result = ice->gpio.get_data(ice) & GPIO_RATE_MASK; + for (i = 0; i < ARRAY_SIZE(gpio_vals); i++) + if (gpio_vals[i] == result) + return juli_rates[i]; + return 0; +} + +/* setting new rate */ +static void juli_set_rate(struct snd_ice1712 *ice, unsigned int rate) +{ + unsigned int old, new; + unsigned char val; + + old = ice->gpio.get_data(ice); + new = (old & ~GPIO_RATE_MASK) | get_gpio_val(rate); + /* printk(KERN_DEBUG "JULI - set_rate: old %x, new %x\n", + old & GPIO_RATE_MASK, + new & GPIO_RATE_MASK); */ + + ice->gpio.set_data(ice, new); + /* switching to external clock - supplied by external circuits */ + val = inb(ICEMT1724(ice, RATE)); + outb(val | VT1724_SPDIF_MASTER, ICEMT1724(ice, RATE)); +} + +static inline unsigned char juli_set_mclk(struct snd_ice1712 *ice, + unsigned int rate) +{ + /* no change in master clock */ + return 0; +} + +/* setting clock to external - SPDIF */ +static void juli_set_spdif_clock(struct snd_ice1712 *ice) +{ + unsigned int old; + old = ice->gpio.get_data(ice); + /* external clock (= 0), multiply 1x, 48kHz */ + ice->gpio.set_data(ice, (old & ~GPIO_RATE_MASK) | GPIO_MULTI_1X | + GPIO_FREQ_48KHZ); +} + +/* Called when ak4114 detects change in the input SPDIF stream */ +static void juli_ak4114_change(struct ak4114 *ak4114, unsigned char c0, + unsigned char c1) +{ + struct snd_ice1712 *ice = ak4114->change_callback_private; + int rate; + if (ice->is_spdif_master(ice) && c1) { + /* only for SPDIF master mode, rate was changed */ + rate = snd_ak4114_external_rate(ak4114); + /* printk(KERN_DEBUG "ak4114 - input rate changed to %d\n", + rate); */ + juli_akm_set_rate_val(ice->akm, rate); + } +} + static int __devinit juli_init(struct snd_ice1712 *ice) { static const unsigned char ak4114_init_vals[] = { @@ -187,6 +597,11 @@ static int __devinit juli_init(struct snd_ice1712 *ice) ice, &spec->ak4114); if (err < 0) return err; + /* callback for codecs rate setting */ + spec->ak4114->change_callback = juli_ak4114_change; + spec->ak4114->change_callback_private = ice; + /* AK4114 in Juli can detect external rate correctly */ + spec->ak4114->check_flags = 0; #if 0 /* it seems that the analog doughter board detection does not work @@ -210,6 +625,15 @@ static int __devinit juli_init(struct snd_ice1712 *ice) return err; } + /* juli is clocked by Xilinx array */ + ice->hw_rates = &juli_rates_info; + ice->is_spdif_master = juli_is_spdif_master; + ice->get_rate = juli_get_rate; + ice->set_rate = juli_set_rate; + ice->set_mclk = juli_set_mclk; + ice->set_spdif_clock = juli_set_spdif_clock; + + ice->spdif.ops.open = juli_spdif_in_open; return 0; } @@ -220,18 +644,20 @@ static int __devinit juli_init(struct snd_ice1712 *ice) */ static unsigned char juli_eeprom[] __devinitdata = { - [ICE_EEP2_SYSCONF] = 0x20, /* clock 512, mpu401, 1xADC, 1xDACs */ + [ICE_EEP2_SYSCONF] = 0x2b, /* clock 512, mpu401, 1xADC, 1xDACs, + SPDIF in */ [ICE_EEP2_ACLINK] = 0x80, /* I2S */ [ICE_EEP2_I2S] = 0xf8, /* vol, 96k, 24bit, 192k */ [ICE_EEP2_SPDIF] = 0xc3, /* out-en, out-int, spdif-in */ - [ICE_EEP2_GPIO_DIR] = 0x9f, + [ICE_EEP2_GPIO_DIR] = 0x9f, /* 5, 6:inputs; 7, 4-0 outputs*/ [ICE_EEP2_GPIO_DIR1] = 0xff, [ICE_EEP2_GPIO_DIR2] = 0x7f, - [ICE_EEP2_GPIO_MASK] = 0x9f, - [ICE_EEP2_GPIO_MASK1] = 0xff, + [ICE_EEP2_GPIO_MASK] = 0x60, /* 5, 6: locked; 7, 4-0 writable */ + [ICE_EEP2_GPIO_MASK1] = 0x00, /* 0-7 writable */ [ICE_EEP2_GPIO_MASK2] = 0x7f, - [ICE_EEP2_GPIO_STATE] = 0x16, /* internal clock, multiple 1x, 48kHz */ - [ICE_EEP2_GPIO_STATE1] = 0x80, /* mute */ + [ICE_EEP2_GPIO_STATE] = GPIO_FREQ_48KHZ | GPIO_MULTI_1X | + GPIO_INTERNAL_CLOCK, /* internal clock, multiple 1x, 48kHz*/ + [ICE_EEP2_GPIO_STATE1] = 0x00, /* unmuted */ [ICE_EEP2_GPIO_STATE2] = 0x00, }; diff --git a/sound/pci/ice1712/pontis.c b/sound/pci/ice1712/pontis.c index 4945c81..203cdc1 100644 --- a/sound/pci/ice1712/pontis.c +++ b/sound/pci/ice1712/pontis.c @@ -246,7 +246,7 @@ static int wm_adc_mux_put(struct snd_kcontrol *kcontrol, struct snd_ctl_elem_val wm_put(ice, WM_ADC_MUX, nval); } mutex_unlock(&ice->gpio_mutex); - return 0; + return change; } /* @@ -450,7 +450,7 @@ static int cs_source_put(struct snd_kcontrol *kcontrol, struct snd_ctl_elem_valu change = 1; } mutex_unlock(&ice->gpio_mutex); - return 0; + return change; } diff --git a/sound/pci/ice1712/prodigy192.c b/sound/pci/ice1712/prodigy192.c index 48cf40a..25ceb67 100644 --- a/sound/pci/ice1712/prodigy192.c +++ b/sound/pci/ice1712/prodigy192.c @@ -319,12 +319,11 @@ static int stac9460_mic_sw_put(struct snd_kcontrol *kcontrol, /* * Handler for setting correct codec rate - called when rate change is detected */ -static void stac9460_set_rate_val(struct snd_akm4xxx *ak, unsigned int rate) +static void stac9460_set_rate_val(struct snd_ice1712 *ice, unsigned int rate) { unsigned char old, new; int idx; unsigned char changed[7]; - struct snd_ice1712 *ice = ak->private_data[0]; struct prodigy192_spec *spec = ice->spec; if (rate == 0) /* no hint - S/PDIF input is master, simply return */ @@ -357,16 +356,6 @@ static void stac9460_set_rate_val(struct snd_akm4xxx *ak, unsigned int rate) mutex_unlock(&spec->mute_mutex); } -/* using akm infrastructure for setting rate of the codec */ -static struct snd_akm4xxx akmlike_stac9460 __devinitdata = { - .type = NON_AKM, /* special value */ - .num_adcs = 6, /* not used in any way, just for completeness */ - .num_dacs = 2, - .ops = { - .set_rate_val = stac9460_set_rate_val - } -}; - static const DECLARE_TLV_DB_SCALE(db_scale_dac, -19125, 75, 0); static const DECLARE_TLV_DB_SCALE(db_scale_adc, 0, 150, 0); @@ -642,12 +631,19 @@ static int prodigy192_ak4114_init(struct snd_ice1712 *ice) 0x41, 0x02, 0x2c, 0x00, 0x00 }; struct prodigy192_spec *spec = ice->spec; + int err; - return snd_ak4114_create(ice->card, + err = snd_ak4114_create(ice->card, prodigy192_ak4114_read, prodigy192_ak4114_write, ak4114_init_vals, ak4114_init_txcsb, ice, &spec->ak4114); + if (err < 0) + return err; + /* AK4114 in Prodigy192 cannot detect external rate correctly. + * No reason to stop capture stream due to incorrect checks */ + spec->ak4114->check_flags = AK4114_CHECK_NO_RATE; + return 0; } static void stac9460_proc_regs_read(struct snd_info_entry *entry, @@ -743,7 +739,6 @@ static int __devinit prodigy192_init(struct snd_ice1712 *ice) }; const unsigned short *p; int err = 0; - struct snd_akm4xxx *ak; struct prodigy192_spec *spec; /* prodigy 192 */ @@ -761,15 +756,7 @@ static int __devinit prodigy192_init(struct snd_ice1712 *ice) p = stac_inits_prodigy; for (; *p != (unsigned short)-1; p += 2) stac9460_put(ice, p[0], p[1]); - /* reusing the akm codecs infrastructure, - * for setting rate on stac9460 */ - ak = ice->akm = kmalloc(sizeof(struct snd_akm4xxx), GFP_KERNEL); - if (!ak) - return -ENOMEM; - ice->akm_codecs = 1; - err = snd_ice1712_akm4xxx_init(ak, &akmlike_stac9460, NULL, ice); - if (err < 0) - return err; + ice->gpio.set_pro_rate = stac9460_set_rate_val; /* MI/ODI/O add on card with AK4114 */ if (prodigy192_miodio_exists(ice)) { diff --git a/sound/pci/ice1712/revo.c b/sound/pci/ice1712/revo.c index 301bf92..dba93d8 100644 --- a/sound/pci/ice1712/revo.c +++ b/sound/pci/ice1712/revo.c @@ -322,17 +322,23 @@ static struct snd_pt2258 ptc_revo51_volume; static void ap192_set_rate_val(struct snd_akm4xxx *ak, unsigned int rate) { struct snd_ice1712 *ice = ak->private_data[0]; + int dfs; revo_set_rate_val(ak, rate); -#if 1 /* FIXME: do we need this procedure? */ - /* reset DFS pin of AK5385A for ADC, too */ - /* DFS0 (pin 18) -- GPIO10 pin 77 */ - snd_ice1712_save_gpio_status(ice); - snd_ice1712_gpio_write_bits(ice, 1 << 10, - rate > 48000 ? (1 << 10) : 0); - snd_ice1712_restore_gpio_status(ice); -#endif + /* reset CKS */ + snd_ice1712_gpio_write_bits(ice, 1 << 8, rate > 96000 ? 1 : 0); + /* reset DFS pins of AK5385A for ADC, too */ + if (rate > 96000) + dfs = 2; + else if (rate > 48000) + dfs = 1; + else + dfs = 0; + snd_ice1712_gpio_write_bits(ice, 3 << 9, dfs << 9); + /* reset ADC */ + snd_ice1712_gpio_write_bits(ice, 1 << 11, 0); + snd_ice1712_gpio_write_bits(ice, 1 << 11, 1); } static const struct snd_akm4xxx_dac_channel ap192_dac[] = { @@ -353,28 +359,20 @@ static struct snd_ak4xxx_private akm_ap192_priv __devinitdata = { .cif = 0, .data_mask = VT1724_REVO_CDOUT, .clk_mask = VT1724_REVO_CCLK, - .cs_mask = VT1724_REVO_CS0 | VT1724_REVO_CS3, - .cs_addr = VT1724_REVO_CS3, - .cs_none = VT1724_REVO_CS0 | VT1724_REVO_CS3, + .cs_mask = VT1724_REVO_CS0 | VT1724_REVO_CS1, + .cs_addr = VT1724_REVO_CS1, + .cs_none = VT1724_REVO_CS0 | VT1724_REVO_CS1, .add_flags = VT1724_REVO_CCLK, /* high at init */ .mask_flags = 0, }; -#if 0 -/* FIXME: ak4114 makes the sound much lower due to some confliction, - * so let's disable it right now... - */ -#define BUILD_AK4114_AP192 -#endif - -#ifdef BUILD_AK4114_AP192 /* AK4114 support on Audiophile 192 */ /* CDTO (pin 32) -- GPIO2 pin 52 * CDTI (pin 33) -- GPIO3 pin 53 (shared with AK4358) * CCLK (pin 34) -- GPIO1 pin 51 (shared with AK4358) * CSN (pin 35) -- GPIO7 pin 59 */ -#define AK4114_ADDR 0x00 +#define AK4114_ADDR 0x02 static void write_data(struct snd_ice1712 *ice, unsigned int gpio, unsigned int data, int idx) @@ -428,7 +426,7 @@ static unsigned int ap192_4wire_start(struct snd_ice1712 *ice) tmp = snd_ice1712_gpio_read(ice); tmp |= VT1724_REVO_CCLK; /* high at init */ tmp |= VT1724_REVO_CS0; - tmp &= ~VT1724_REVO_CS3; + tmp &= ~VT1724_REVO_CS1; snd_ice1712_gpio_write(ice, tmp); udelay(1); return tmp; @@ -436,7 +434,7 @@ static unsigned int ap192_4wire_start(struct snd_ice1712 *ice) static void ap192_4wire_finish(struct snd_ice1712 *ice, unsigned int tmp) { - tmp |= VT1724_REVO_CS3; + tmp |= VT1724_REVO_CS1; tmp |= VT1724_REVO_CS0; snd_ice1712_gpio_write(ice, tmp); udelay(1); @@ -485,13 +483,17 @@ static int __devinit ap192_ak4114_init(struct snd_ice1712 *ice) struct ak4114 *ak; int err; - return snd_ak4114_create(ice->card, + err = snd_ak4114_create(ice->card, ap192_ak4114_read, ap192_ak4114_write, ak4114_init_vals, ak4114_init_txcsb, ice, &ak); + /* AK4114 in Revo cannot detect external rate correctly. + * No reason to stop capture stream due to incorrect checks */ + ak->check_flags = AK4114_CHECK_NO_RATE; + + return 0; /* error ignored; it's no fatal error */ } -#endif /* BUILD_AK4114_AP192 */ static int __devinit revo_init(struct snd_ice1712 *ice) { @@ -557,6 +559,9 @@ static int __devinit revo_init(struct snd_ice1712 *ice) if (err < 0) return err; + /* unmute all codecs */ + snd_ice1712_gpio_writ