diff options
Diffstat (limited to 'Documentation/sound/alsa')
| -rw-r--r-- | Documentation/sound/alsa/ALSA-Configuration.txt | 60 | ||||
| -rw-r--r-- | Documentation/sound/alsa/DocBook/alsa-driver-api.tmpl | 4 | ||||
| -rw-r--r-- | Documentation/sound/alsa/DocBook/writing-an-alsa-driver.tmpl | 33 | ||||
| -rw-r--r-- | Documentation/sound/alsa/hda_codec.txt | 10 | ||||
| -rw-r--r-- | Documentation/sound/alsa/soc/DAI.txt | 56 | ||||
| -rw-r--r-- | Documentation/sound/alsa/soc/clocking.txt | 51 | ||||
| -rw-r--r-- | Documentation/sound/alsa/soc/codec.txt | 197 | ||||
| -rw-r--r-- | Documentation/sound/alsa/soc/dapm.txt | 297 | ||||
| -rw-r--r-- | Documentation/sound/alsa/soc/machine.txt | 113 | ||||
| -rw-r--r-- | Documentation/sound/alsa/soc/overview.txt | 83 | ||||
| -rw-r--r-- | Documentation/sound/alsa/soc/platform.txt | 58 | ||||
| -rw-r--r-- | Documentation/sound/alsa/soc/pops_clicks.txt | 52 |
12 files changed, 982 insertions, 32 deletions
diff --git a/Documentation/sound/alsa/ALSA-Configuration.txt b/Documentation/sound/alsa/ALSA-Configuration.txt index 9fef210ab50..c30ff1bb2d1 100644 --- a/Documentation/sound/alsa/ALSA-Configuration.txt +++ b/Documentation/sound/alsa/ALSA-Configuration.txt @@ -242,6 +242,12 @@ Prior to version 0.9.0rc4 options had a 'snd_' prefix. This was removed. ac97_clock - AC'97 clock (default = 48000) ac97_quirk - AC'97 workaround for strange hardware See "AC97 Quirk Option" section below. + ac97_codec - Workaround to specify which AC'97 codec + instead of probing. If this works for you + file a bug with your `lspci -vn` output. + -2 -- Force probing. + -1 -- Default behavior. + 0-2 -- Use the specified codec. spdif_aclink - S/PDIF transfer over AC-link (default = 1) This module supports one card and autoprobe. @@ -779,6 +785,7 @@ Prior to version 0.9.0rc4 options had a 'snd_' prefix. This was removed. asus-dig ASUS with SPDIF out asus-dig2 ASUS with SPDIF out (using GPIO2) uniwill 3-jack + fujitsu Fujitsu Laptops (Pi1536) F1734 2-jack lg LG laptop (m1 express dual) lg-lw LG LW20/LW25 laptop @@ -800,14 +807,18 @@ Prior to version 0.9.0rc4 options had a 'snd_' prefix. This was removed. ALC262 fujitsu Fujitsu Laptop hp-bpc HP xw4400/6400/8400/9400 laptops + hp-bpc-d7000 HP BPC D7000 benq Benq ED8 + hippo Hippo (ATI) with jack detection, Sony UX-90s + hippo_1 Hippo (Benq) with jack detection basic fixed pin assignment w/o SPDIF auto auto-config reading BIOS (default) ALC882/885 3stack-dig 3-jack with SPDIF I/O - 6stck-dig 6-jack digital with SPDIF I/O + 6stack-dig 6-jack digital with SPDIF I/O arima Arima W820Di1 + macpro MacPro support auto auto-config reading BIOS (default) ALC883/888 @@ -817,6 +828,10 @@ Prior to version 0.9.0rc4 options had a 'snd_' prefix. This was removed. 3stack-6ch-dig 3-jack 6-channel with SPDIF I/O 6stack-dig-demo 6-jack digital for Intel demo board acer Acer laptops (Travelmate 3012WTMi, Aspire 5600, etc) + medion Medion Laptops + targa-dig Targa/MSI + targa-2ch-dig Targs/MSI with 2-channel + laptop-eapd 3-jack with SPDIF I/O and EAPD (Clevo M540JE, M550JE) auto auto-config reading BIOS (default) ALC861/660 @@ -825,6 +840,16 @@ Prior to version 0.9.0rc4 options had a 'snd_' prefix. This was removed. 6stack-dig 6-jack with SPDIF I/O 3stack-660 3-jack (for ALC660) uniwill-m31 Uniwill M31 laptop + toshiba Toshiba laptop support + asus Asus laptop support + asus-laptop ASUS F2/F3 laptops + auto auto-config reading BIOS (default) + + ALC861VD/660VD + 3stack 3-jack + 3stack-dig 3-jack with SPDIF OUT + 6stack-dig 6-jack with SPDIF OUT + 3stack-660 3-jack (for ALC660VD) auto auto-config reading BIOS (default) CMI9880 @@ -845,6 +870,7 @@ Prior to version 0.9.0rc4 options had a 'snd_' prefix. This was removed. 3stack 3-stack, shared surrounds laptop 2-channel only (FSC V2060, Samsung M50) laptop-eapd 2-channel with EAPD (Samsung R65, ASUS A6J) + ultra 2-channel with EAPD (Samsung Ultra tablet PC) AD1988 6stack 6-jack @@ -854,12 +880,31 @@ Prior to version 0.9.0rc4 options had a 'snd_' prefix. This was removed. laptop 3-jack with hp-jack automute laptop-dig ditto with SPDIF auto auto-config reading BIOS (default) + + Conexant 5045 + laptop Laptop config + test for testing/debugging purpose, almost all controls + can be adjusted. Appearing only when compiled with + $CONFIG_SND_DEBUG=y + + Conexant 5047 + laptop Basic Laptop config + laptop-hp Laptop config for some HP models (subdevice 30A5) + laptop-eapd Laptop config with EAPD support + test for testing/debugging purpose, almost all controls + can be adjusted. Appearing only when compiled with + $CONFIG_SND_DEBUG=y STAC9200/9205/9220/9221/9254 ref Reference board 3stack D945 3stack 5stack D945 5stack + SPDIF + STAC9202/9250/9251 + ref Reference board, base config + m2-2 Some Gateway MX series laptops + m6 Some Gateway NX series laptops + STAC9227/9228/9229/927x ref Reference board 3stack D965 3stack @@ -974,6 +1019,7 @@ Prior to version 0.9.0rc4 options had a 'snd_' prefix. This was removed. Module for Envy24HT (VT/ICE1724), Envy24PT (VT1720) based PCI sound cards. * MidiMan M Audio Revolution 5.1 * MidiMan M Audio Revolution 7.1 + * MidiMan M Audio Audiophile 192 * AMP Ltd AUDIO2000 * TerraTec Aureon 5.1 Sky * TerraTec Aureon 7.1 Space @@ -993,7 +1039,7 @@ Prior to version 0.9.0rc4 options had a 'snd_' prefix. This was removed. model - Use the given board model, one of the following: revo51, revo71, amp2000, prodigy71, prodigy71lt, - prodigy192, aureon51, aureon71, universe, + prodigy192, aureon51, aureon71, universe, ap192, k8x800, phase22, phase28, ms300, av710 This module supports multiple cards and autoprobe. @@ -1049,6 +1095,9 @@ Prior to version 0.9.0rc4 options had a 'snd_' prefix. This was removed. buggy_semaphore - Enable workaround for hardwares with buggy semaphores (e.g. on some ASUS laptops) (default off) + spdif_aclink - Use S/PDIF over AC-link instead of direct connection + from the controller chip + (0 = off, 1 = on, -1 = default) This module supports one chip and autoprobe. @@ -1371,6 +1420,13 @@ Prior to version 0.9.0rc4 options had a 'snd_' prefix. This was removed. This module supports multiple cards. + Module snd-portman2x4 + --------------------- + + Module for Midiman Portman 2x4 parallel port MIDI interface + + This module supports multiple cards. + Module snd-powermac (on ppc only) --------------------------------- diff --git a/Documentation/sound/alsa/DocBook/alsa-driver-api.tmpl b/Documentation/sound/alsa/DocBook/alsa-driver-api.tmpl index 1f3ae3e32d6..c4d2e3507af 100644 --- a/Documentation/sound/alsa/DocBook/alsa-driver-api.tmpl +++ b/Documentation/sound/alsa/DocBook/alsa-driver-api.tmpl @@ -36,7 +36,7 @@ </bookinfo> <chapter><title>Management of Cards and Devices</title> - <sect1><title>Card Managment</title> + <sect1><title>Card Management</title> !Esound/core/init.c </sect1> <sect1><title>Device Components</title> @@ -59,7 +59,7 @@ <sect1><title>PCM Format Helpers</title> !Esound/core/pcm_misc.c </sect1> - <sect1><title>PCM Memory Managment</title> + <sect1><title>PCM Memory Management</title> !Esound/core/pcm_memory.c </sect1> </chapter> diff --git a/Documentation/sound/alsa/DocBook/writing-an-alsa-driver.tmpl b/Documentation/sound/alsa/DocBook/writing-an-alsa-driver.tmpl index ccd0a953953..74d3a35b59b 100644 --- a/Documentation/sound/alsa/DocBook/writing-an-alsa-driver.tmpl +++ b/Documentation/sound/alsa/DocBook/writing-an-alsa-driver.tmpl @@ -1360,8 +1360,7 @@ <informalexample> <programlisting> <![CDATA[ - static irqreturn_t snd_mychip_interrupt(int irq, void *dev_id, - struct pt_regs *regs) + static irqreturn_t snd_mychip_interrupt(int irq, void *dev_id) { struct mychip *chip = dev_id; .... @@ -2127,7 +2126,7 @@ accessible via <constant>substream->runtime</constant>. This runtime pointer holds the various information; it holds the copy of hw_params and sw_params configurations, the buffer - pointers, mmap records, spinlocks, etc. Almost everyhing you + pointers, mmap records, spinlocks, etc. Almost everything you need for controlling the PCM can be found there. </para> @@ -2340,7 +2339,7 @@ struct _snd_pcm_runtime { <para> When the PCM substreams can be synchronized (typically, - synchorinized start/stop of a playback and a capture streams), + synchronized start/stop of a playback and a capture streams), you can give <constant>SNDRV_PCM_INFO_SYNC_START</constant>, too. In this case, you'll need to check the linked-list of PCM substreams in the trigger callback. This will be @@ -3062,8 +3061,7 @@ struct _snd_pcm_runtime { <title>Interrupt Handler Case #1</title> <programlisting> <![CDATA[ - static irqreturn_t snd_mychip_interrupt(int irq, void *dev_id, - struct pt_regs *regs) + static irqreturn_t snd_mychip_interrupt(int irq, void *dev_id) { struct mychip *chip = dev_id; spin_lock(&chip->lock); @@ -3106,8 +3104,7 @@ struct _snd_pcm_runtime { <title>Interrupt Handler Case #2</title> <programlisting> <![CDATA[ - static irqreturn_t snd_mychip_interrupt(int irq, void *dev_id, - struct pt_regs *regs) + static irqreturn_t snd_mychip_interrupt(int irq, void *dev_id) { struct mychip *chip = dev_id; spin_lock(&chip->lock); @@ -3247,7 +3244,7 @@ struct _snd_pcm_runtime { You can even define your own constraint rules. For example, let's suppose my_chip can manage a substream of 1 channel if and only if the format is S16_LE, otherwise it supports any format - specified in the <structname>snd_pcm_hardware</structname> stucture (or in any + specified in the <structname>snd_pcm_hardware</structname> structure (or in any other constraint_list). You can build a rule like this: <example> @@ -3691,16 +3688,6 @@ struct _snd_pcm_runtime { </para> <para> - Here, the chip instance is retrieved via - <function>snd_kcontrol_chip()</function> macro. This macro - just accesses to kcontrol->private_data. The - kcontrol->private_data field is - given as the argument of <function>snd_ctl_new()</function> - (see the later subsection - <link linkend="control-interface-constructor"><citetitle>Constructor</citetitle></link>). - </para> - - <para> The <structfield>value</structfield> field is depending on the type of control as well as on info callback. For example, the sb driver uses this field to store the register offset, @@ -3780,7 +3767,7 @@ struct _snd_pcm_runtime { <para> Like <structfield>get</structfield> callback, when the control has more than one elements, - all elemehts must be evaluated in this callback, too. + all elements must be evaluated in this callback, too. </para> </section> @@ -5541,12 +5528,12 @@ struct _snd_pcm_runtime { #ifdef CONFIG_PM static int snd_my_suspend(struct pci_dev *pci, pm_message_t state) { - .... /* do things for suspsend */ + .... /* do things for suspend */ return 0; } static int snd_my_resume(struct pci_dev *pci) { - .... /* do things for suspsend */ + .... /* do things for suspend */ return 0; } #endif @@ -6111,7 +6098,7 @@ struct _snd_pcm_runtime { <!-- ****************************************************** --> <!-- Acknowledgments --> <!-- ****************************************************** --> - <chapter id="acknowledments"> + <chapter id="acknowledgments"> <title>Acknowledgments</title> <para> I would like to thank Phil Kerr for his help for improvement and diff --git a/Documentation/sound/alsa/hda_codec.txt b/Documentation/sound/alsa/hda_codec.txt index 0be57ed8130..4eaae2a4553 100644 --- a/Documentation/sound/alsa/hda_codec.txt +++ b/Documentation/sound/alsa/hda_codec.txt @@ -277,11 +277,11 @@ Helper Functions snd_hda_get_codec_name() stores the codec name on the given string. snd_hda_check_board_config() can be used to obtain the configuration -information matching with the device. Define the table with struct -hda_board_config entries (zero-terminated), and pass it to the -function. The function checks the modelname given as a module -parameter, and PCI subsystem IDs. If the matching entry is found, it -returns the config field value. +information matching with the device. Define the model string table +and the table with struct snd_pci_quirk entries (zero-terminated), +and pass it to the function. The function checks the modelname given +as a module parameter, and PCI subsystem IDs. If the matching entry +is found, it returns the config field value. snd_hda_add_new_ctls() can be used to create and add control entries. Pass the zero-terminated array of struct snd_kcontrol_new. The same array diff --git a/Documentation/sound/alsa/soc/DAI.txt b/Documentation/sound/alsa/soc/DAI.txt new file mode 100644 index 00000000000..58cbfd01ea8 --- /dev/null +++ b/Documentation/sound/alsa/soc/DAI.txt @@ -0,0 +1,56 @@ +ASoC currently supports the three main Digital Audio Interfaces (DAI) found on +SoC controllers and portable audio CODECS today, namely AC97, I2S and PCM. + + +AC97 +==== + + AC97 is a five wire interface commonly found on many PC sound cards. It is +now also popular in many portable devices. This DAI has a reset line and time +multiplexes its data on its SDATA_OUT (playback) and SDATA_IN (capture) lines. +The bit clock (BCLK) is always driven by the CODEC (usually 12.288MHz) and the +frame (FRAME) (usually 48kHz) is always driven by the controller. Each AC97 +frame is 21uS long and is divided into 13 time slots. + +The AC97 specification can be found at :- +http://www.intel.com/design/chipsets/audio/ac97_r23.pdf + + +I2S +=== + + I2S is a common 4 wire DAI used in HiFi, STB and portable devices. The Tx and +Rx lines are used for audio transmision, whilst the bit clock (BCLK) and +left/right clock (LRC) synchronise the link. I2S is flexible in that either the +controller or CODEC can drive (master) the BCLK and LRC clock lines. Bit clock +usually varies depending on the sample rate and the master system clock +(SYSCLK). LRCLK is the same as the sample rate. A few devices support separate +ADC and DAC LRCLK's, this allows for similtanious capture and playback at +different sample rates. + +I2S has several different operating modes:- + + o I2S - MSB is transmitted on the falling edge of the first BCLK after LRC + transition. + + o Left Justified - MSB is transmitted on transition of LRC. + + o Right Justified - MSB is transmitted sample size BCLK's before LRC + transition. + +PCM +=== + +PCM is another 4 wire interface, very similar to I2S, that can support a more +flexible protocol. It has bit clock (BCLK) and sync (SYNC) lines that are used +to synchronise the link whilst the Tx and Rx lines are used to transmit and +receive the audio data. Bit clock usually varies depending on sample rate +whilst sync runs at the sample rate. PCM also supports Time Division +Multiplexing (TDM) in that several devices can use the bus similtaniuosly (This +is sometimes referred to as network mode). + +Common PCM operating modes:- + + o Mode A - MSB is transmitted on falling edge of first BCLK after FRAME/SYNC. + + o Mode B - MSB is transmitted on rising edge of FRAME/SYNC. diff --git a/Documentation/sound/alsa/soc/clocking.txt b/Documentation/sound/alsa/soc/clocking.txt new file mode 100644 index 00000000000..e93960d53a1 --- /dev/null +++ b/Documentation/sound/alsa/soc/clocking.txt @@ -0,0 +1,51 @@ +Audio Clocking +============== + +This text describes the audio clocking terms in ASoC and digital audio in +general. Note: Audio clocking can be complex ! + + +Master Clock +------------ + +Every audio subsystem is driven by a master clock (sometimes refered to as MCLK +or SYSCLK). This audio master clock can be derived from a number of sources +(e.g. crystal, PLL, CPU clock) and is responsible for producing the correct +audio playback and capture sample rates. + +Some master clocks (e.g. PLL's and CPU based clocks) are configuarble in that +their speed can be altered by software (depending on the system use and to save +power). Other master clocks are fixed at at set frequency (i.e. crystals). + + +DAI Clocks +---------- +The Digital Audio Interface is usually driven by a Bit Clock (often referred to +as BCLK). This clock is used to drive the digital audio data across the link +between the codec and CPU. + +The DAI also has a frame clock to signal the start of each audio frame. This +clock is sometimes referred to as LRC (left right clock) or FRAME. This clock +runs at exactly the sample rate (LRC = Rate). + +Bit Clock can be generated as follows:- + +BCLK = MCLK / x + + or + +BCLK = LRC * x + + or + +BCLK = LRC * Channels * Word Size + +This relationship depends on the codec or SoC CPU in particular. In general +it's best to configure BCLK to the lowest possible speed (depending on your +rate, number of channels and wordsize) to save on power. + +It's also desireable to use the codec (if possible) to drive (or master) the +audio clocks as it's usually gives more accurate sample rates than the CPU. + + + diff --git a/Documentation/sound/alsa/soc/codec.txt b/Documentation/sound/alsa/soc/codec.txt new file mode 100644 index 00000000000..48983c75aad --- /dev/null +++ b/Documentation/sound/alsa/soc/codec.txt @@ -0,0 +1,197 @@ +ASoC Codec Driver +================= + +The codec driver is generic and hardware independent code that configures the +codec to provide audio capture and playback. It should contain no code that is +specific to the target platform or machine. All platform and machine specific +code should be added to the platform and machine drivers respectively. + +Each codec driver *must* provide the following features:- + + 1) Codec DAI and PCM configuration + 2) Codec control IO - using I2C, 3 Wire(SPI) or both API's + 3) Mixers and audio controls + 4) Codec audio operations + +Optionally, codec drivers can also provide:- + + 5) DAPM description. + 6) DAPM event handler. + 7) DAC Digital mute control. + +It's probably best to use this guide in conjuction with the existing codec +driver code in sound/soc/codecs/ + +ASoC Codec driver breakdown +=========================== + +1 - Codec DAI and PCM configuration +----------------------------------- +Each codec driver must have a struct snd_soc_codec_dai to define it's DAI and +PCM's capablities and operations. This struct is exported so that it can be +registered with the core by your machine driver. + +e.g. + +struct snd_soc_codec_dai wm8731_dai = { + .name = "WM8731", + /* playback capabilities */ + .playback = { + .stream_name = "Playback", + .channels_min = 1, + .channels_max = 2, + .rates = WM8731_RATES, + .formats = WM8731_FORMATS,}, + /* capture capabilities */ + .capture = { + .stream_name = "Capture", + .channels_min = 1, + .channels_max = 2, + .rates = WM8731_RATES, + .formats = WM8731_FORMATS,}, + /* pcm operations - see section 4 below */ + .ops = { + .prepare = wm8731_pcm_prepare, + .hw_params = wm8731_hw_params, + .shutdown = wm8731_shutdown, + }, + /* DAI operations - see DAI.txt */ + .dai_ops = { + .digital_mute = wm8731_mute, + .set_sysclk = wm8731_set_dai_sysclk, + .set_fmt = wm8731_set_dai_fmt, + } +}; +EXPORT_SYMBOL_GPL(wm8731_dai); + + +2 - Codec control IO +-------------------- +The codec can ususally be controlled via an I2C or SPI style interface (AC97 +combines control with data in the DAI). The codec drivers will have to provide +functions to read and write the codec registers along with supplying a register +cache:- + + /* IO control data and register cache */ + void *control_data; /* codec control (i2c/3wire) data */ + void *reg_cache; + +Codec read/write should do any data formatting and call the hardware read write +below to perform the IO. These functions are called by the core and alsa when +performing DAPM or changing the mixer:- + + unsigned int (*read)(struct snd_soc_codec *, unsigned int); + int (*write)(struct snd_soc_codec *, unsigned int, unsigned int); + +Codec hardware IO functions - usually points to either the I2C, SPI or AC97 +read/write:- + + hw_write_t hw_write; + hw_read_t hw_read; + + +3 - Mixers and audio controls +----------------------------- +All the codec mixers and audio controls can be defined using the convenience +macros defined in soc.h. + + #define SOC_SINGLE(xname, reg, shift, mask, invert) + +Defines a single control as follows:- + + xname = Control name e.g. "Playback Volume" + reg = codec register + shift = control bit(s) offset in register + mask = control bit size(s) e.g. mask of 7 = 3 bits + invert = the control is inverted + +Other macros include:- + + #define SOC_DOUBLE(xname, reg, shift_left, shift_right, mask, invert) + +A stereo control + + #define SOC_DOUBLE_R(xname, reg_left, reg_right, shift, mask, invert) + +A stereo control spanning 2 registers + + #define SOC_ENUM_SINGLE(xreg, xshift, xmask, xtexts) + +Defines an single enumerated control as follows:- + + xreg = register + xshift = control bit(s) offset in register + xmask = control bit(s) size + xtexts = pointer to array of strings that describe each setting + + #define SOC_ENUM_DOUBLE(xreg, xshift_l, xshift_r, xmask, xtexts) + +Defines a stereo enumerated control + + +4 - Codec Audio Operations +-------------------------- +The codec driver also supports the following alsa operations:- + +/* SoC audio ops */ +struct snd_soc_ops { + int (*startup)(struct snd_pcm_substream *); + void (*shutdown)(struct snd_pcm_substream *); + int (*hw_params)(struct snd_pcm_substream *, struct snd_pcm_hw_params *); + int (*hw_free)(struct snd_pcm_substream *); + int (*prepare)(struct snd_pcm_substream *); +}; + +Please refer to the alsa driver PCM documentation for details. +http://www.alsa-project.org/~iwai/writing-an-alsa-driver/c436.htm + + +5 - DAPM description. +--------------------- +The Dynamic Audio Power Management description describes the codec's power +components, their relationships and registers to the ASoC core. Please read +dapm.txt for details of building the description. + +Please also see the examples in other codec drivers. + + +6 - DAPM event handler +---------------------- +This function is a callback that handles codec domain PM calls and system +domain PM calls (e.g. suspend and resume). It's used to put the codec to sleep +when not in use. + +Power states:- + + SNDRV_CTL_POWER_D0: /* full On */ + /* vref/mid, clk and osc on, active */ + + SNDRV_CTL_POWER_D1: /* partial On */ + SNDRV_CTL_POWER_D2: /* partial On */ + + SNDRV_CTL_POWER_D3hot: /* Off, with power */ + /* everything off except vref/vmid, inactive */ + + SNDRV_CTL_POWER_D3cold: /* Everything Off, without power */ + + +7 - Codec DAC digital mute control. +------------------------------------ +Most codecs have a digital mute before the DAC's that can be used to minimise +any system noise. The mute stops any digital data from entering the DAC. + +A callback can be created that is called by the core for each codec DAI when the +mute is applied or freed. + +i.e. + +static int wm8974_mute(struct snd_soc_codec *codec, + struct snd_soc_codec_dai *dai, int mute) +{ + u16 mute_reg = wm8974_read_reg_cache(codec, WM8974_DAC) & 0xffbf; + if(mute) + wm8974_write(codec, WM8974_DAC, mute_reg | 0x40); + else + wm8974_write(codec, WM8974_DAC, mute_reg); + return 0; +} diff --git a/Documentation/sound/alsa/soc/dapm.txt b/Documentation/sound/alsa/soc/dapm.txt new file mode 100644 index 00000000000..c11877f5b4a --- /dev/null +++ b/Documentation/sound/alsa/soc/dapm.txt @@ -0,0 +1,297 @@ +Dynamic Audio Power Management for Portable Devices +=================================================== + +1. Description +============== + +Dynamic Audio Power Management (DAPM) is designed to allow portable Linux devices +to use the minimum amount of power within the audio subsystem at all times. It +is independent of other kernel PM and as such, can easily co-exist with the +other PM systems. + +DAPM is also completely transparent to all user space applications as all power +switching is done within the ASoC core. No code changes or recompiling are +required for user space applications. DAPM makes power switching descisions based +upon any audio stream (capture/playback) activity and audio mixer settings +within the device. + +DAPM spans the whole machine. It covers power control within the entire audio +subsystem, this includes internal codec power blocks and machine level power +systems. + +There are 4 power domains within DAPM + + 1. Codec domain - VREF, VMID (core codec and audio power) + Usually controlled at codec probe/remove and suspend/resume, although + can be set at stream time if power is not needed for sidetone, etc. + + 2. Platform/Machine domain - physically connected inputs and outputs + Is platform/machine and user action specific, is configured by the + machine driver and responds to asynchronous events e.g when HP + are inserted + + 3. Path domain - audio susbsystem signal paths + Automatically set when mixer and mux settings are changed by the user. + e.g. alsamixer, amixer. + + 4. Stream domain - DAC's and ADC's. + Enabled and disabled when stream playback/capture is started and + stopped respectively. e.g. aplay, arecord. + +All DAPM power switching descisons are made automatically by consulting an audio +routing map of the whole machine. This map is specific to each machine and +consists of the interconnections between every audio component (including +internal codec components). All audio components that effect power are called +widgets hereafter. + + +2. DAPM Widgets +=============== + +Audio DAPM widgets fall into a number of types:- + + o Mixer - Mixes several analog signals into a single analog signal. + o Mux - An analog switch that outputs only 1 of it's inputs. + o PGA - A programmable gain amplifier or attenuation widget. + o ADC - Analog to Digital Converter + o DAC - Digital to Analog Converter + o Switch - An analog switch + o Input - A codec input pin + o Output - A codec output pin + o Headphone - Headphone (and optional Jack) + o Mic - Mic (and optional Jack) + o Line - Line Input/Output (and optional Jack) + o Speaker - Speaker + o Pre - Special PRE widget (exec before all others) + o Post - Special POST widget (exec after all others) + +(Widgets are defined in include/sound/soc-dapm.h) + +Widgets are usually added in the codec driver and the machine driver. There are +convience macros defined in soc-dapm.h that can be used to quickly build a +list of widgets of the codecs and machines DAPM widgets. + +Most widgets have a name, register, shift and invert. Some widgets have extra +parameters for stream name and kcontrols. + + +2.1 Stream Domain Widgets +------------------------- + +Stream Widgets relate to the stream power domain and only consist of ADC's +(analog to digital converters) and DAC's (digital to analog converters). + +Stream widgets have the following format:- + +SND_SOC_DAPM_DAC(name, stream name, reg, shift, invert), + +NOTE: the stream name must match the corresponding stream name in your codecs +snd_soc_codec_dai. + +e.g. stream widgets for HiFi playback and capture + +SND_SOC_DAPM_DAC("HiFi DAC", "HiFi Playback", REG, 3, 1), +SND_SOC_DAPM_ADC("HiFi ADC", "HiFi Capture", REG, 2, 1), + + +2.2 Path Domain Widgets +----------------------- + +Path domain widgets have a ability to control or effect the audio signal or +audio paths within the audio subsystem. They have the following form:- + +SND_SOC_DAPM_PGA(name, reg, shift, invert, controls, num_controls) + +Any widget kcontrols can be set using the controls and num_controls members. + +e.g. Mixer widget (the kcontrols are declared first) + +/* Output Mixer */ +static const snd_kcontrol_new_t wm8731_output_mixer_controls[] = { +SOC_DAPM_SINGLE("Line Bypass Switch", WM8731_APANA, 3, 1, 0), +SOC_DAPM_SINGLE("Mic Sidetone Switch", WM8731_APANA, 5, 1, 0), +SOC_DAPM_SINGLE("HiFi Playback Switch", WM8731_APANA, 4, 1, 0), +}; + +SND_SOC_DAPM_MIXER("Output Mixer", WM8731_PWR, 4, 1, wm8731_output_mixer_controls, + ARRAY_SIZE(wm8731_output_mixer_controls)), + + +2.3 Platform/Machine domain Widgets +----------------------------------- + +Machine widgets are different from codec widgets in that they don't have a +codec register bit associated with them. A machine widget is assigned to each +machine audio component (non codec) that can be independently powered. e.g. + + o Speaker Amp + o Microphone Bias + o Jack connectors + +A machine widget can have an optional call back. + +e.g. Jack connector widget for an external Mic that enables Mic Bias +when the Mic is inserted:- + +static int spitz_mic_bias(struct snd_soc_dapm_widget* w, int event) +{ + if(SND_SOC_DAPM_EVENT_ON(event)) + set_scoop_gpio(&spitzscoop2_device.dev, SPITZ_SCP2_MIC_BIAS); + else + reset_scoop_gpio(&spitzscoop2_device.dev, SPITZ_SCP2_MIC_BIAS); + + return 0; +} + +SND_SOC_DAPM_MIC("Mic Jack", spitz_mic_bias), + + +2.4 Codec Domain +---------------- + +The Codec power domain has no widgets and is handled by the codecs DAPM event +handler. This handler is called when the codec powerstate is changed wrt to any +stream event or by kernel PM events. + + +2.5 Virtual Widgets +------------------- + +Sometimes widgets exist in the codec or machine audio map that don't have any +corresponding register bit for power control. In this case it's necessary to +create a virtual widget - a widget with no control bits e.g. + +SND_SOC_DAPM_MIXER("AC97 Mixer", SND_SOC_DAPM_NOPM, 0, 0, NULL, 0), + +This can be used to merge to signal paths together in software. + +After all the widgets have been defined, they can then be added to the DAPM +subsystem individually with a call to snd_soc_dapm_new_control(). + + +3. Codec Widget Interconnections +================================ + +Widgets are connected to each other within the codec and machine by audio +paths (called interconnections). Each interconnection must be defined in order +to create a map of all audio paths between widgets. +This is easiest with a diagram of the codec (and schematic of the machine audio +system), as it requires joining widgets together via their audio signal paths. + +i.e. from the WM8731 codec's output mixer (wm8731.c) + +The WM8731 output mixer has 3 inputs (sources) + + 1. Line Bypass Input + 2. DAC (HiFi playback) + 3. Mic Sidetone Input + +Each input in this example has a kcontrol associated with it (defined in example +above) and is connected to the output mixer via it's kcontrol name. We can now +connect the destination widget (wrt audio signal) with it's source widgets. + + /* output mixer */ + {"Output Mixer", "Line Bypass Switch", "Line Input"}, + {"Output Mixer", "HiFi Playback Switch", "DAC"}, + {"Output Mixer", "Mic Sidetone Switch", "Mic Bias"}, + +So we have :- + + Destination Widget <=== Path Name <=== Source Widget + +Or:- + + Sink, Path, Source + +Or :- + + "Output Mixer" is connected to the "DAC" via the "HiFi Playback Switch". + +When there is no path name connecting widgets (e.g. a direct connection) we +pass NULL for the path name. + +Interconnections are created with a call to:- + +snd_soc_dapm_connect_input(codec, sink, path, source); + +Finally, snd_soc_dapm_new_widgets(codec) must be called after all widgets and +interconnections have been registered with the core. This causes the core to +scan the codec and machine so that the internal DAPM state matches the +physical state of the machine. + + +3.1 Machine Widget Interconnections +----------------------------------- +Machine widget interconnections are created in the same way as codec ones and +directly connect the codec pins to machine level widgets. + +e.g. connects the speaker out codec pins to the internal speaker. + + /* ext speaker connected to codec pins LOUT2, ROUT2 */ + {"Ext Spk", NULL , "ROUT2"}, + {"Ext Spk", NULL , "LOUT2"}, + +This allows the DAPM to power on and off pins that are connected (and in use) +and pins that are NC respectively. + + +4 Endpoint Widgets +=================== +An endpoint is a start or end point (widget) of an audio signal within the +machine and includes the codec. e.g. + + o Headphone Jack + o Internal Speaker + o Internal Mic + o Mic Jack + o Codec Pins + +When a codec pin is NC it can be marked as not used with a call to + +snd_soc_dapm_set_endpoint(codec, "Widget Name", 0); + +The last argument is 0 for inactive and 1 for active. This way the pin and its +input widget will never be powered up and consume power. + +This also applies to machine widgets. e.g. if a headphone is connected to a +jack then the jack can be marked active. If the headphone is removed, then +the headphone jack can be marked inactive. + + +5 DAPM Widget Events +==================== + +Some widgets can register their interest with the DAPM core in PM events. +e.g. A Speaker with an amplifier registers a widget so the amplifier can be +powered only when the spk is in use. + +/* turn speaker amplifier on/off depending on use */ +static int corgi_amp_event(struct snd_soc_dapm_widget *w, int event) +{ + if (SND_SOC_DAPM_EVENT_ON(event)) + set_scoop_gpio(&corgiscoop_device.dev, CORGI_SCP_APM_ON); + else + reset_scoop_gpio(&corgiscoop_device.dev, CORGI_SCP_APM_ON); + + return 0; +} + +/* corgi machine dapm widgets */ +static const struct snd_soc_dapm_widget wm8731_dapm_widgets = + SND_SOC_DAPM_SPK("Ext Spk", corgi_amp_event); + +Please see soc-dapm.h for all other widgets that support events. + + +5.1 Event types +--------------- + +The following event types are supported by event widgets. + +/* dapm event types */ +#define SND_SOC_DAPM_PRE_PMU 0x1 /* before widget power up */ +#define SND_SOC_DAPM_POST_PMU 0x2 /* after widget power up */ +#define SND_SOC_DAPM_PRE_PMD 0x4 /* before widget power down */ +#define SND_SOC_DAPM_POST_PMD 0x8 /* after widget power down */ +#define SND_SOC_DAPM_PRE_REG 0x10 /* before audio path setup */ +#define SND_SOC_DAPM_POST_REG 0x20 /* after audio path setup */ diff --git a/Documentation/sound/alsa/soc/machine.txt b/Documentation/sound/alsa/soc/machine.txt new file mode 100644 index 00000000000..72bd222f2a2 --- /dev/null +++ b/Documentation/sound/alsa/soc/machine.txt @@ -0,0 +1,113 @@ +ASoC Machine Driver +=================== + +The ASoC machine (or board) driver is the code that glues together the platform +and codec drivers. + +The machine driver can contain codec and platform specific code. It registers +the audio subsystem with the kernel as a platform device and is represented by +the following struct:- + +/* SoC machine */ +struct snd_soc_machine { + char *name; + + int (*probe)(struct platform_device *pdev); + int (*remove)(struct platform_device *pdev); + + /* the pre and post PM functions are used to do any PM work before and + * after the codec and DAI's do any PM work. */ + int (*suspend_pre)(struct platform_device *pdev, pm_message_t state); + int (*suspend_post)(struct platform_device *pdev, pm_message_t state); + int (*resume_pre)(struct platform_device *pdev); + int (*resume_post)(struct platform_device *pdev); + + /* machine stream operations */ + struct snd_soc_ops *ops; + + /* CPU <--> Codec DAI links */ + struct snd_soc_dai_link *dai_link; + int num_links; +}; + +probe()/remove() +---------------- +probe/remove are optional. Do any machine specific probe here. + + +suspend()/resume() +------------------ +The machine driver has pre and post versions of suspend and resume to take care +of any machine audio tasks that have to be done before or after the codec, DAI's +and DMA is suspended and resumed. Optional. + + +Machine operations +------------------ +The machine specific audio operations can be set here. Again this is optional. + + +Machine DAI Configuration +------------------------- +The machine DAI configuration glues all the codec and CPU DAI's together. It can +also be used to set up the DAI system clock and for any machine related DAI +initialisation e.g. the machine audio map can be connected to the codec audio +map, unconnnected codec pins can be set as such. Please see corgi.c, spitz.c +for examples. + +struct snd_soc_dai_link is used to set up each DAI in your machine. e.g. + +/* corgi digital audio interface glue - connects codec <--> CPU */ +static struct snd_soc_dai_link corgi_dai = { + .name = "WM8731", + .stream_name = "WM8731", + .cpu_dai = &pxa_i2s_dai, + .codec_dai = &wm8731_dai, + .init = corgi_wm8731_init, + .ops = &corgi_ops, +}; + +struct snd_soc_machine then sets up the machine with it's DAI's. e.g. + +/* corgi audio machine driver */ +static struct snd_soc_machine snd_soc_machine_corgi = { + .name = "Corgi", + .dai_link = &corgi_dai, + .num_links = 1, +}; + + +Machine Audio Subsystem +----------------------- + +The machine soc device glues the platform, machine and codec driver together. +Private data can also be set here. e.g. + +/* corgi audio private data */ +static struct wm8731_setup_data corgi_wm8731_setup = { + .i2c_address = 0x1b, +}; + +/* corgi audio subsystem */ +static struct snd_soc_device corgi_snd_devdata = { + .machine = &snd_soc_machine_corgi, + .platform = &pxa2xx_soc_platform, + .codec_dev = &soc_codec_dev_wm8731, + .codec_data = &corgi_wm8731_setup, +}; + + +Machine Power Map +----------------- + +The machine driver can optionally extend the codec power map and to become an +audio power map of the audio subsystem. This allows for automatic power up/down +of speaker/HP amplifiers, etc. Codec pins can be connected to the machines jack +sockets in the machine init function. See soc/pxa/spitz.c and dapm.txt for +details. + + +Machine Controls +---------------- + +Machine specific audio mixer controls can be added in the dai init function.
\ No newline at end of file diff --git a/Documentation/sound/alsa/soc/overview.txt b/Documentation/sound/alsa/soc/overview.txt new file mode 100644 index 00000000000..753c5cc5984 --- /dev/null +++ b/Documentation/sound/alsa/soc/overview.txt @@ -0,0 +1,83 @@ +ALSA SoC Layer +============== + +The overall project goal of the ALSA System on Chip (ASoC) layer is to provide +better ALSA support for embedded system on chip procesors (e.g. pxa2xx, au1x00, +iMX, etc) and portable audio codecs. Currently there is some support in the +kernel for SoC audio, however it has some limitations:- + + * Currently, codec drivers are often tightly coupled to the underlying SoC + cpu. This is not ideal and leads to code duplication i.e. Linux now has 4 + different wm8731 drivers for 4 different SoC platforms. + + * There is 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 ofter require + machine specific code to re route audio, enable amps etc after such an event. + + * Current drivers tend 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 is also no support for saving power via + changing codec oversampling rates, bias currents, etc. + + +ASoC Design +=========== + +The ASoC layer is designed to address these issues and provide the following +features :- + + * 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 it's audio interface capabilities with the core and are + subsequently matched and configured when the application hw params are known. + + * Dynamic Audio Power Management (DAPM). DAPM automatically sets the codec to + it's 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 amp. + +To achieve all this, ASoC basically splits an embedded audio system into 3 +components :- + + * Codec driver: The codec driver is platform independent and contains audio + controls, audio interface capabilities, codec dapm definition and codec IO + functions. + + * Platform driver: The platform driver contains the audio dma engine and audio + interface drivers (e.g. I2S, AC97, PCM) for that platform. + + * Machine driver: The machine driver handles any machine specific controls and + audio events. i.e. turing on an amp at start of playback. + + +Documentation +============= + +The documentation is spilt into the following sections:- + +overview.txt: This file. + +codec.txt: Codec driver internals. + +DAI.txt: Description of Digital Audio Interface standards and how to configure +a DAI within your codec and CPU DAI drivers. + +dapm.txt: Dynamic Audio Power Management + +platform.txt: Platform audio DMA and DAI. + +machine.txt: Machine driver internals. + +pop_clicks.txt: How to minimise audio artifacts. + +clocking.txt: ASoC clocking for best power performance.
\ No newline at end of file diff --git a/Documentation/sound/alsa/soc/platform.txt b/Documentation/sound/alsa/soc/platform.txt new file mode 100644 index 00000000000..e95b16d5a53 --- /dev/null +++ b/Documentation/sound/alsa/soc/platform.txt @@ -0,0 +1,58 @@ +ASoC Platform Driver +==================== + +An ASoC platform driver can be divided into audio DMA and SoC DAI configuration +and control. The platform drivers only target the SoC CPU and must have no board +specific code. + +Audio DMA +========= + +The platform DMA driver optionally supports the following alsa operations:- + +/* SoC audio ops */ +struct snd_soc_ops { + int (*startup)(struct snd_pcm_substream *); + void (*shutdown)(struct snd_pcm_substream *); + int (*hw_params)(struct snd_pcm_substream *, struct snd_pcm_hw_params *); + int (*hw_free)(struct snd_pcm_substream *); + int (*prepare)(struct snd_pcm_substream *); + int (*trigger)(struct snd_pcm_substream *, int); +}; + +The platform driver exports it's DMA functionailty via struct snd_soc_platform:- + +struct snd_soc_platform { + char *name; + + int (*probe)(struct platform_device *pdev); + int (*remove)(struct platform_device *pdev); + int (*suspend)(struct platform_device *pdev, struct snd_soc_cpu_dai *cpu_dai); + int (*resume)(struct platform_device *pdev, struct snd_soc_cpu_dai *cpu_dai); + + /* pcm creation and destruction */ + int (*pcm_new)(struct snd_card *, struct snd_soc_codec_dai *, struct snd_pcm *); + void (*pcm_free)(struct snd_pcm *); + + /* platform stream ops */ + struct snd_pcm_ops *pcm_ops; +}; + +Please refer to the alsa driver documentation for details of audio DMA. +http://www.alsa-project.org/~iwai/writing-an-alsa-driver/c436.htm + +An example DMA driver is soc/pxa/pxa2xx-pcm.c + + +SoC DAI Drivers +=============== + +Each SoC DAI driver must provide the following features:- + + 1) Digital audio interface (DAI) description + 2) Digital audio interface configuration + 3) PCM's description + 4) Sysclk configuration + 5) Suspend and resume (optional) + +Please see codec.txt for a description of items 1 - 4. diff --git a/Documentation/sound/alsa/soc/pops_clicks.txt b/Documentation/sound/alsa/soc/pops_clicks.txt new file mode 100644 index 00000000000..2cf7ee5b3d7 --- /dev/null +++ b/Documentation/sound/alsa/soc/pops_clicks.txt @@ -0,0 +1,52 @@ +Audio Pops and Clicks +===================== + +Pops and clicks are unwanted audio artifacts caused by the powering up and down +of components within the audio subsystem. This is noticable on PC's when an +audio module is either loaded or unloaded (at module load time the sound card is +powered up and causes a popping noise on the speakers). + +Pops and clicks can be more frequent on portable systems with DAPM. This is +because the components within the subsystem are being dynamically powered +depending on the audio usage and this can subsequently cause a small pop or +click every time a component power state is changed. + + +Minimising Playback Pops and Clicks +=================================== + +Playback pops in portable audio subsystems cannot be completely eliminated atm, +however future audio codec hardware will have better pop and click supression. +Pops can be reduced within playback by powering the audio components in a +specific order. This order is different for startup and shutdown and follows +some basic rules:- + + Startup Order :- DAC --> Mixers --> Output PGA --> Digital Unmute + + Shutdown Order :- Digital Mute --> Output PGA --> Mixers --> DAC + +This assumes that the codec PCM output path from the DAC is via a mixer and then +a PGA (programmable gain amplifier) before being output to the speakers. + + +Minimising Capture Pops and Clicks +================================== + +Capture artifacts are somewhat easier to get rid as we can delay activating the +ADC until all the pops have occured. This follows similar power rules to +playback in that components are powered in a sequence depending upon stream +startup or shutdown. + + Startup Order - Input PGA --> Mixers --> ADC + + Shutdown Order - ADC --> Mixers --> Input PGA + + +Zipper Noise +============ +An unwanted zipper noise can occur within the audio playback or capture stream +when a volume control is changed near its maximum gain value. The zipper noise +is heard when the gain increase or decrease changes the mean audio signal +amplitude too quickly. It can be minimised by enabling the zero cross setting +for each volume control. The ZC forces the gain change to occur when the signal +crosses the zero amplitude line. |