Commit 60261795 authored by Linus Torvalds's avatar Linus Torvalds
Browse files

Merge branch 'linus' of master.kernel.org:/pub/scm/linux/kernel/git/perex/alsa

* 'linus' of master.kernel.org:/pub/scm/linux/kernel/git/perex/alsa: (212 commits)
  [PATCH] Fix breakage with CONFIG_SYSFS_DEPRECATED
  [ALSA] version 1.0.14rc2
  [ALSA] ASoC documentation updates
  [ALSA] ca0106 - Add missing sysfs device assignment
  [ALSA] aoa i2sbus: Stop Apple i2s DMA gracefully
  [ALSA] hda-codec - Add support for Fujitsu PI1556 Realtek ALC880
  [ALSA] aoa: remove suspend/resume printks
  [ALSA] Fix possible deadlocks in sequencer at removal of ports
  [ALSA] emu10k1 - Fix STAC9758 front channel
  [ALSA] soc - Clean up with kmemdup()
  [ALSA] snd-ak4114: Fix two array overflows
  [ALSA] ac97_bus power management
  [ALSA] usbaudio - Add support for Edirol UA-101
  [ALSA] hda-codec - Add ALC861VD/ALC660VD support
  [ALSA] soc - ASoC 0.13 Sharp poodle machine
  [ALSA] soc - ASoC 0.13 Sharp tosa machine
  [ALSA] soc - ASoC 0.13 spitz machine
  [ALSA] soc - ASoC Sharp corgi machine
  [ALSA] soc - ASoC 0.13 pxa2xx DMA
  [ALSA] soc - ASoC 0.13 pxa2xx AC97 driver
  ...
parents d27146dd c2902c8a
......@@ -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)
---------------------------------
......
......@@ -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>
......
......@@ -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-&gt;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>
......@@ -3690,16 +3687,6 @@ struct _snd_pcm_runtime {
</example>
</para>
<para>
Here, the chip instance is retrieved via
<function>snd_kcontrol_chip()</function> macro. This macro
just accesses to kcontrol-&gt;private_data. The
kcontrol-&gt;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,
......@@ -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
......
......@@ -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
......
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.
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.
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;
}
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