€•ÏCŒsphinx.addnodes”Œdocument”“”)”}”(Œ rawsource”Œ”Œchildren”]”(Œ translations”Œ LanguagesNode”“”)”}”(hhh]”(hŒ pending_xref”“”)”}”(hhh]”Œdocutils.nodes”ŒText”“”ŒChinese (Simplified)”…””}”Œparent”hsbaŒ attributes”}”(Œids”]”Œclasses”]”Œnames”]”Œdupnames”]”Œbackrefs”]”Œ refdomain”Œstd”Œreftype”Œdoc”Œ reftarget”Œ&/translations/zh_CN/sound/soc/overview”Œmodname”NŒ classname”NŒ refexplicit”ˆuŒtagname”hhh ubh)”}”(hhh]”hŒChinese (Traditional)”…””}”hh2sbah}”(h]”h ]”h"]”h$]”h&]”Œ refdomain”h)Œreftype”h+Œ reftarget”Œ&/translations/zh_TW/sound/soc/overview”Œmodname”NŒ classname”NŒ refexplicit”ˆuh1hhh ubh)”}”(hhh]”hŒItalian”…””}”hhFsbah}”(h]”h ]”h"]”h$]”h&]”Œ refdomain”h)Œreftype”h+Œ reftarget”Œ&/translations/it_IT/sound/soc/overview”Œmodname”NŒ classname”NŒ refexplicit”ˆuh1hhh ubh)”}”(hhh]”hŒJapanese”…””}”hhZsbah}”(h]”h ]”h"]”h$]”h&]”Œ refdomain”h)Œreftype”h+Œ reftarget”Œ&/translations/ja_JP/sound/soc/overview”Œmodname”NŒ classname”NŒ refexplicit”ˆuh1hhh ubh)”}”(hhh]”hŒKorean”…””}”hhnsbah}”(h]”h ]”h"]”h$]”h&]”Œ refdomain”h)Œreftype”h+Œ reftarget”Œ&/translations/ko_KR/sound/soc/overview”Œmodname”NŒ classname”NŒ refexplicit”ˆuh1hhh ubh)”}”(hhh]”hŒSpanish”…””}”hh‚sbah}”(h]”h ]”h"]”h$]”h&]”Œ refdomain”h)Œreftype”h+Œ reftarget”Œ&/translations/sp_SP/sound/soc/overview”Œmodname”NŒ classname”NŒ refexplicit”ˆuh1hhh ubeh}”(h]”h ]”h"]”h$]”h&]”Œcurrent_language”ŒEnglish”uh1h hhŒ _document”hŒsource”NŒline”NubhŒsection”“”)”}”(hhh]”(hŒtitle”“”)”}”(hŒALSA SoC Layer Overview”h]”hŒALSA SoC Layer Overview”…””}”(hh¨hžhhŸNh Nubah}”(h]”h ]”h"]”h$]”h&]”uh1h¦hh£hžhhŸŒ@/var/lib/git/docbuild/linux/Documentation/sound/soc/overview.rst”h KubhŒ paragraph”“”)”}”(hX7The overall project goal of the ALSA System on Chip (ASoC) layer is to provide better ALSA support for embedded system-on-chip processors (e.g. pxa2xx, au1x00, iMX, etc) and portable audio codecs. Prior to the ASoC subsystem there was some support in the kernel for SoC audio, however it had some limitations:-”h]”hX7The overall project goal of the ALSA System on Chip (ASoC) layer is to provide better ALSA support for embedded system-on-chip processors (e.g. pxa2xx, au1x00, iMX, etc) and portable audio codecs. Prior to the ASoC subsystem there was some support in the kernel for SoC audio, however it had some limitations:-”…””}”(hh¹hžhhŸNh Nubah}”(h]”h ]”h"]”h$]”h&]”uh1h·hŸh¶h Khh£hžhubhŒ block_quote”“”)”}”(hX * Codec drivers were often tightly coupled to the underlying SoC CPU. This is not ideal and leads to code duplication - for example, Linux had different wm8731 drivers for 4 different SoC platforms. * There was no standard method to signal user initiated audio events (e.g. Headphone/Mic insertion, Headphone/Mic detection after an insertion event). These are quite common events on portable devices and often require machine specific code to re-route audio, enable amps, etc., after such an event. * Drivers tended to power up the entire codec when playing (or recording) audio. This is fine for a PC, but tends to waste a lot of power on portable devices. There was also no support for saving power via changing codec oversampling rates, bias currents, etc. ”h]”hŒ bullet_list”“”)”}”(hhh]”(hŒ list_item”“”)”}”(hŒÅCodec drivers were often tightly coupled to the underlying SoC CPU. This is not ideal and leads to code duplication - for example, Linux had different wm8731 drivers for 4 different SoC platforms. ”h]”h¸)”}”(hŒÄCodec drivers were often tightly coupled to the underlying SoC CPU. This is not ideal and leads to code duplication - for example, Linux had different wm8731 drivers for 4 different SoC platforms.”h]”hŒÄCodec drivers were often tightly coupled to the underlying SoC CPU. This is not ideal and leads to code duplication - for example, Linux had different wm8731 drivers for 4 different SoC platforms.”…””}”(hhØhžhhŸNh Nubah}”(h]”h ]”h"]”h$]”h&]”uh1h·hŸh¶h K hhÔubah}”(h]”h ]”h"]”h$]”h&]”uh1hÒhhÏubhÓ)”}”(hX*There was no standard method to signal user initiated audio events (e.g. Headphone/Mic insertion, Headphone/Mic detection after an insertion event). These are quite common events on portable devices and often require machine specific code to re-route audio, enable amps, etc., after such an event. ”h]”h¸)”}”(hX)There was no standard method to signal user initiated audio events (e.g. Headphone/Mic insertion, Headphone/Mic detection after an insertion event). These are quite common events on portable devices and often require machine specific code to re-route audio, enable amps, etc., after such an event.”h]”hX)There was no standard method to signal user initiated audio events (e.g. Headphone/Mic insertion, Headphone/Mic detection after an insertion event). These are quite common events on portable devices and often require machine specific code to re-route audio, enable amps, etc., after such an event.”…””}”(hhðhžhhŸNh Nubah}”(h]”h ]”h"]”h$]”h&]”uh1h·hŸh¶h Khhìubah}”(h]”h ]”h"]”h$]”h&]”uh1hÒhhÏubhÓ)”}”(hXDrivers tended to power up the entire codec when playing (or recording) audio. This is fine for a PC, but tends to waste a lot of power on portable devices. There was also no support for saving power via changing codec oversampling rates, bias currents, etc. ”h]”h¸)”}”(hXDrivers tended to power up the entire codec when playing (or recording) audio. This is fine for a PC, but tends to waste a lot of power on portable devices. There was also no support for saving power via changing codec oversampling rates, bias currents, etc.”h]”hXDrivers tended to power up the entire codec when playing (or recording) audio. This is fine for a PC, but tends to waste a lot of power on portable devices. There was also no support for saving power via changing codec oversampling rates, bias currents, etc.”…””}”(hjhžhhŸNh Nubah}”(h]”h ]”h"]”h$]”h&]”uh1h·hŸh¶h Khjubah}”(h]”h ]”h"]”h$]”h&]”uh1hÒhhÏubeh}”(h]”h ]”h"]”h$]”h&]”Œbullet”Œ*”uh1hÍhŸh¶h K hhÉubah}”(h]”h ]”h"]”h$]”h&]”uh1hÇhŸh¶h K hh£hžhubh¢)”}”(hhh]”(h§)”}”(hŒ ASoC Design”h]”hŒ ASoC Design”…””}”(hj-hžhhŸNh Nubah}”(h]”h ]”h"]”h$]”h&]”uh1h¦hj*hžhhŸh¶h Kubh¸)”}”(hŒXThe ASoC layer is designed to address these issues and provide the following features :-”h]”hŒXThe ASoC layer is designed to address these issues and provide the following features :-”…””}”(hj;hžhhŸNh Nubah}”(h]”h ]”h"]”h$]”h&]”uh1h·hŸh¶h Khj*hžhubhÈ)”}”(hX* Codec independence. Allows reuse of codec drivers on other platforms and machines. * Easy I2S/PCM audio interface setup between codec and SoC. Each SoC interface and codec registers its audio interface capabilities with the core and are subsequently matched and configured when the application hardware parameters are known. * Dynamic Audio Power Management (DAPM). DAPM automatically sets the codec to its minimum power state at all times. This includes powering up/down internal power blocks depending on the internal codec audio routing and any active streams. * Pop and click reduction. Pops and clicks can be reduced by powering the codec up/down in the correct sequence (including using digital mute). ASoC signals the codec when to change power states. * Machine specific controls: Allow machines to add controls to the sound card (e.g. volume control for speaker amplifier). ”h]”hÎ)”}”(hhh]”(hÓ)”}”(hŒSCodec independence. Allows reuse of codec drivers on other platforms and machines. ”h]”h¸)”}”(hŒRCodec independence. Allows reuse of codec drivers on other platforms and machines.”h]”hŒRCodec independence. Allows reuse of codec drivers on other platforms and machines.”…””}”(hjThžhhŸNh Nubah}”(h]”h ]”h"]”h$]”h&]”uh1h·hŸh¶h K!hjPubah}”(h]”h ]”h"]”h$]”h&]”uh1hÒhjMubhÓ)”}”(hŒðEasy I2S/PCM audio interface setup between codec and SoC. Each SoC interface and codec registers its audio interface capabilities with the core and are subsequently matched and configured when the application hardware parameters are known. ”h]”h¸)”}”(hŒïEasy I2S/PCM audio interface setup between codec and SoC. Each SoC interface and codec registers its audio interface capabilities with the core and are subsequently matched and configured when the application hardware parameters are known.”h]”hŒïEasy I2S/PCM audio interface setup between codec and SoC. Each SoC interface and codec registers its audio interface capabilities with the core and are subsequently matched and configured when the application hardware parameters are known.”…””}”(hjlhžhhŸNh Nubah}”(h]”h ]”h"]”h$]”h&]”uh1h·hŸh¶h K$hjhubah}”(h]”h ]”h"]”h$]”h&]”uh1hÒhjMubhÓ)”}”(hŒíDynamic Audio Power Management (DAPM). DAPM automatically sets the codec to its minimum power state at all times. This includes powering up/down internal power blocks depending on the internal codec audio routing and any active streams. ”h]”h¸)”}”(hŒìDynamic Audio Power Management (DAPM). DAPM automatically sets the codec to its minimum power state at all times. This includes powering up/down internal power blocks depending on the internal codec audio routing and any active streams.”h]”hŒìDynamic Audio Power Management (DAPM). DAPM automatically sets the codec to its minimum power state at all times. This includes powering up/down internal power blocks depending on the internal codec audio routing and any active streams.”…””}”(hj„hžhhŸNh Nubah}”(h]”h ]”h"]”h$]”h&]”uh1h·hŸh¶h K)hj€ubah}”(h]”h ]”h"]”h$]”h&]”uh1hÒhjMubhÓ)”}”(hŒÂPop and click reduction. Pops and clicks can be reduced by powering the codec up/down in the correct sequence (including using digital mute). ASoC signals the codec when to change power states. ”h]”h¸)”}”(hŒÁPop and click reduction. Pops and clicks can be reduced by powering the codec up/down in the correct sequence (including using digital mute). ASoC signals the codec when to change power states.”h]”hŒÁPop and click reduction. Pops and clicks can be reduced by powering the codec up/down in the correct sequence (including using digital mute). 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