Merge branch 'mpi-master' into wip-k-fmlpwip-k-fmlp

Conflicts:
	litmus/sched_cedf.c

10 files changed, 140 insertions, 94 deletions

diff --git a/Documentation/sound/alsa/ALSA-Configuration.txt b/Documentation/sound/alsa/ALSA-Configuration.txt
index 7f4dcebda9c6..89757012c7ff 100644
--- a/Documentation/sound/alsa/ALSA-Configuration.txt
+++ b/Documentation/sound/alsa/ALSA-Configuration.txt
@@ -300,6 +300,74 @@ 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-azt1605
+  ------------------
+
+    Module for Aztech Sound Galaxy soundcards based on the Aztech AZT1605
+    chipset.
+
+    port        - port # for BASE (0x220,0x240,0x260,0x280)
+    wss_port    - port # for WSS (0x530,0x604,0xe80,0xf40)
+    irq         - IRQ # for WSS (7,9,10,11)
+    dma1        - DMA # for WSS playback (0,1,3)
+    dma2        - DMA # for WSS capture (0,1), -1 = disabled (default)
+    mpu_port    - port # for MPU-401 UART (0x300,0x330), -1 = disabled (default)
+    mpu_irq     - IRQ # for MPU-401 UART (3,5,7,9), -1 = disabled (default)
+    fm_port     - port # for OPL3 (0x388), -1 = disabled (default)
+
+    This module supports multiple cards. It does not support autoprobe: port,
+    wss_port, irq and dma1 have to be specified. The other values are
+    optional.
+
+    "port" needs to match the BASE ADDRESS jumper on the card (0x220 or 0x240)
+    or the value stored in the card's EEPROM for cards that have an EEPROM and
+    their "CONFIG MODE" jumper set to "EEPROM SETTING". The other values can
+    be chosen freely from the options enumerated above.
+
+    If dma2 is specified and different from dma1, the card will operate in
+    full-duplex mode. When dma1=3, only dma2=0 is valid and the only way to
+    enable capture since only channels 0 and 1 are available for capture.
+
+    Generic settings are "port=0x220 wss_port=0x530 irq=10 dma1=1 dma2=0
+    mpu_port=0x330 mpu_irq=9 fm_port=0x388".
+
+    Whatever IRQ and DMA channels you pick, be sure to reserve them for
+    legacy ISA in your BIOS.
+
+  Module snd-azt2316
+  ------------------
+
+    Module for Aztech Sound Galaxy soundcards based on the Aztech AZT2316
+    chipset.
+
+    port        - port # for BASE (0x220,0x240,0x260,0x280)
+    wss_port    - port # for WSS (0x530,0x604,0xe80,0xf40)
+    irq         - IRQ # for WSS (7,9,10,11)
+    dma1        - DMA # for WSS playback (0,1,3)
+    dma2        - DMA # for WSS capture (0,1), -1 = disabled (default)
+    mpu_port    - port # for MPU-401 UART (0x300,0x330), -1 = disabled (default)
+    mpu_irq     - IRQ # for MPU-401 UART (5,7,9,10), -1 = disabled (default)
+    fm_port     - port # for OPL3 (0x388), -1 = disabled (default)
+
+    This module supports multiple cards. It does not support autoprobe: port,
+    wss_port, irq and dma1 have to be specified. The other values are
+    optional.
+
+    "port" needs to match the BASE ADDRESS jumper on the card (0x220 or 0x240)
+    or the value stored in the card's EEPROM for cards that have an EEPROM and
+    their "CONFIG MODE" jumper set to "EEPROM SETTING". The other values can
+    be chosen freely from the options enumerated above.
+
+    If dma2 is specified and different from dma1, the card will operate in
+    full-duplex mode. When dma1=3, only dma2=0 is valid and the only way to
+    enable capture since only channels 0 and 1 are available for capture.
+
+    Generic settings are "port=0x220 wss_port=0x530 irq=10 dma1=1 dma2=0
+    mpu_port=0x330 mpu_irq=9 fm_port=0x388".
+
+    Whatever IRQ and DMA channels you pick, be sure to reserve them for
+    legacy ISA in your BIOS.
+
   Module snd-aw2
   --------------
 
@@ -906,13 +974,6 @@ Prior to version 0.9.0rc4 options had a 'snd_' prefix. This was removed.
 
     See hdspm.txt for details.
 
-  Module snd-hifier
-  -----------------
-
-    Module for the MediaTek/TempoTec HiFier Fantasia sound card.
-
-    This module supports autoprobe and multiple cards.
-
   Module snd-ice1712
   ------------------
 
@@ -1169,6 +1230,13 @@ Prior to version 0.9.0rc4 options had a 'snd_' prefix. This was removed.
     This module supports multiple cards.
     The driver requires the firmware loader support on kernel.
 
+  Module snd-lola
+  ---------------
+
+    Module for Digigram Lola PCI-e boards
+
+    This module supports multiple cards.
+
   Module snd-lx6464es
   -------------------
 
@@ -1463,15 +1531,20 @@ Prior to version 0.9.0rc4 options had a 'snd_' prefix. This was removed.
   Module snd-oxygen
   -----------------
 
-    Module for sound cards based on the C-Media CMI8788 chip:
+    Module for sound cards based on the C-Media CMI8786/8787/8788 chip:
     * Asound A-8788
+    * Asus Xonar DG
     * AuzenTech X-Meridian
+    * AuzenTech X-Meridian 2G
     * Bgears b-Enspirer
     * Club3D Theatron DTS
     * HT-Omega Claro (plus)
     * HT-Omega Claro halo (XT)
+    * Kuroutoshikou CMI8787-HG2PCI
     * Razer Barracuda AC-1
     * Sondigo Inferno
+    * TempoTec HiFier Fantasia
+    * TempoTec HiFier Serenade
 
     This module supports autoprobe and multiple cards.
 
@@ -1641,20 +1714,6 @@ Prior to version 0.9.0rc4 options had a 'snd_' prefix. This was removed.
 
     This card is also known as Audio Excel DSP 16 or Zoltrix AV302.
 
-  Module snd-sgalaxy
-  ------------------
-
-    Module for Aztech Sound Galaxy sound card.
-
-    sbport      - Port # for SB16 interface (0x220,0x240)
-    wssport     - Port # for WSS interface (0x530,0xe80,0xf40,0x604)
-    irq         - IRQ # (7,9,10,11)
-    dma1        - DMA #
-
-    This module supports multiple cards.
-
-    The power-management is supported.
-
   Module snd-sscape
   -----------------
 
@@ -1952,9 +2011,9 @@ Prior to version 0.9.0rc4 options had a 'snd_' prefix. This was removed.
   Module snd-virtuoso
   -------------------
 
-    Module for sound cards based on the Asus AV100/AV200 chips,
-    i.e., Xonar D1, DX, D2, D2X, DS, HDAV1.3 (Deluxe), Essence ST
-    (Deluxe) and Essence STX.
+    Module for sound cards based on the Asus AV66/AV100/AV200 chips,
+    i.e., Xonar D1, DX, D2, D2X, DS, Essence ST (Deluxe), Essence STX,
+    HDAV1.3 (Deluxe), and HDAV1.3 Slim.
 
     This module supports autoprobe and multiple cards.
 
@@ -2177,7 +2236,7 @@ Proc interfaces (/proc/asound)
 
 /proc/asound/card#/pcm#[cp]/oss
 -------------------------------
-  String "erase" - erase all additional informations about OSS applications
+  String "erase" - erase all additional information about OSS applications
   String "<app_name> <fragments> <fragment_size> [<options>]"
 
    <app_name> - name of application with (higher priority) or without path
diff --git a/Documentation/sound/alsa/HD-Audio-Models.txt b/Documentation/sound/alsa/HD-Audio-Models.txt
index 37c6aad5e590..d70c93bdcadf 100644
--- a/Documentation/sound/alsa/HD-Audio-Models.txt
+++ b/Documentation/sound/alsa/HD-Audio-Models.txt
@@ -94,7 +94,7 @@ ALC662/663/272
   3stack-dig    3-stack (2-channel) with SPDIF
   3stack-6ch     3-stack (6-channel)
   3stack-6ch-dig 3-stack (6-channel) with SPDIF
-  6stack-dig     6-stack with SPDIF
+  5stack-dig     5-stack with SPDIF
   lenovo-101e    Lenovo laptop
   eeepc-p701    ASUS Eeepc P701
   eeepc-ep20    ASUS Eeepc EP20
@@ -149,7 +149,6 @@ ALC882/883/885/888/889
   acer-aspire-7730g Acer Aspire 7730G
   acer-aspire-8930g Acer Aspire 8930G
   medion        Medion Laptops
-  medion-md2    Medion MD2
   targa-dig     Targa/MSI
   targa-2ch-dig Targa/MSI with 2-channel
   targa-8ch-dig Targa/MSI with 8-channel (MSI GX620)
@@ -297,6 +296,7 @@ Conexant 5066
 =============
   laptop        Basic Laptop config (default)
   hp-laptop     HP laptops, e g G60
+  asus          Asus K52JU, Lenovo G560
   dell-laptop   Dell laptops
   dell-vostro   Dell Vostro
   olpc-xo-1_5   OLPC XO 1.5
diff --git a/Documentation/sound/alsa/HD-Audio.txt b/Documentation/sound/alsa/HD-Audio.txt
index 278cc2122ea0..c82beb007634 100644
--- a/Documentation/sound/alsa/HD-Audio.txt
+++ b/Documentation/sound/alsa/HD-Audio.txt
@@ -57,9 +57,11 @@ dead.  However, this detection isn't perfect on some devices.  In such
 a case, you can change the default method via `position_fix` option.
 
 `position_fix=1` means to use LPIB method explicitly.
-`position_fix=2` means to use the position-buffer.  0 is the default
-value, the automatic check and fallback to LPIB as described in the
-above.  If you get a problem of repeated sounds, this option might
+`position_fix=2` means to use the position-buffer.
+`position_fix=3` means to use a combination of both methods, needed
+for some VIA and ATI controllers.  0 is the default value for all other
+controllers, the automatic check and fallback to LPIB as described in
+the above.  If you get a problem of repeated sounds, this option might
 help.
 
 In addition to that, every controller is known to be broken regarding
diff --git a/Documentation/sound/alsa/SB-Live-mixer.txt b/Documentation/sound/alsa/SB-Live-mixer.txt
index f5639d40521d..f4b5988f450c 100644
--- a/Documentation/sound/alsa/SB-Live-mixer.txt
+++ b/Documentation/sound/alsa/SB-Live-mixer.txt
@@ -87,14 +87,14 @@ accumulator. ALSA uses accumulators 0 and 1 for left and right PCM.
 The result is forwarded to the ADC capture FIFO (thus to the standard capture
 PCM device).
 
-name='Music Playback Volume',index=0
+name='Synth Playback Volume',index=0
 
 This control is used to attenuate samples for left and right MIDI FX-bus
 accumulators. ALSA uses accumulators 4 and 5 for left and right MIDI samples.
 The result samples are forwarded to the front DAC PCM slots of the AC97 codec.
 
-name='Music Capture Volume',index=0
-name='Music Capture Switch',index=0
+name='Synth Capture Volume',index=0
+name='Synth Capture Switch',index=0
 
 These controls are used to attenuate samples for left and right MIDI FX-bus
 accumulator. ALSA uses accumulators 4 and 5 for left and right PCM.
diff --git a/Documentation/sound/alsa/soc/codec.txt b/Documentation/sound/alsa/soc/codec.txt
index 37ba3a72cb76..bce23a4a7875 100644
--- a/Documentation/sound/alsa/soc/codec.txt
+++ b/Documentation/sound/alsa/soc/codec.txt
@@ -27,42 +27,38 @@ ASoC Codec driver breakdown
 
 1 - Codec DAI and PCM configuration
 -----------------------------------
-Each codec driver must have a struct snd_soc_codec_dai to define its DAI and
+Each codec driver must have a struct snd_soc_dai_driver to define its DAI and
 PCM capabilities 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 */
+static struct snd_soc_dai_ops wm8731_dai_ops = {
+        .prepare        = wm8731_pcm_prepare,
+        .hw_params      = wm8731_hw_params,
+        .shutdown       = wm8731_shutdown,
+        .digital_mute   = wm8731_mute,
+        .set_sysclk     = wm8731_set_dai_sysclk,
+        .set_fmt        = wm8731_set_dai_fmt,
+};
+
+struct snd_soc_dai_driver wm8731_dai = {
+        .name = "wm8731-hifi",
         .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,
-        }
+        .ops = &wm8731_dai_ops,
+        .symmetric_rates = 1,
 };
-EXPORT_SYMBOL_GPL(wm8731_dai);
 
 
 2 - Codec control IO
@@ -186,13 +182,14 @@ 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)
+static int wm8974_mute(struct snd_soc_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);
+        struct snd_soc_codec *codec = dai->codec;
+        u16 mute_reg = snd_soc_read(codec, WM8974_DAC) & 0xffbf;
+
+        if (mute)
+                snd_soc_write(codec, WM8974_DAC, mute_reg | 0x40);
         else
-                wm8974_write(codec, WM8974_DAC, mute_reg);
+                snd_soc_write(codec, WM8974_DAC, mute_reg);
         return 0;
 }
diff --git a/Documentation/sound/alsa/soc/machine.txt b/Documentation/sound/alsa/soc/machine.txt
index 2524c75557df..3e2ec9cbf397 100644
--- a/Documentation/sound/alsa/soc/machine.txt
+++ b/Documentation/sound/alsa/soc/machine.txt
@@ -12,6 +12,8 @@ the following struct:-
 struct snd_soc_card {
         char *name;
 
+        ...
+
         int (*probe)(struct platform_device *pdev);
         int (*remove)(struct platform_device *pdev);
 
@@ -22,12 +24,13 @@ struct snd_soc_card {
         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()
@@ -42,11 +45,6 @@ of any machine audio tasks that have to be done before or after the codec, DAIs
 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 DAIs together. It can
@@ -61,8 +59,10 @@ struct snd_soc_dai_link is used to set up each DAI in your machine. e.g.
 static struct snd_soc_dai_link corgi_dai = {
         .name = "WM8731",
         .stream_name = "WM8731",
-        .cpu_dai = &pxa_i2s_dai,
-        .codec_dai = &wm8731_dai,
+        .cpu_dai_name = "pxa-is2-dai",
+        .codec_dai_name = "wm8731-hifi",
+        .platform_name = "pxa-pcm-audio",
+        .codec_name = "wm8713-codec.0-001a",
         .init = corgi_wm8731_init,
         .ops = &corgi_ops,
 };
@@ -77,26 +77,6 @@ static struct snd_soc_card snd_soc_corgi = {
 };
 
 
-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_corgi,
-        .platform = &pxa2xx_soc_platform,
-        .codec_dev = &soc_codec_dev_wm8731,
-        .codec_data = &corgi_wm8731_setup,
-};
-
-
 Machine Power Map
 -----------------
 
diff --git a/Documentation/sound/alsa/soc/platform.txt b/Documentation/sound/alsa/soc/platform.txt
index 06d835987c6a..d57efad37e0a 100644
--- a/Documentation/sound/alsa/soc/platform.txt
+++ b/Documentation/sound/alsa/soc/platform.txt
@@ -20,9 +20,10 @@ struct snd_soc_ops {
         int (*trigger)(struct snd_pcm_substream *, int);
 };
 
-The platform driver exports its DMA functionality via struct snd_soc_platform:-
+The platform driver exports its DMA functionality via struct
+snd_soc_platform_driver:-
 
-struct snd_soc_platform {
+struct snd_soc_platform_driver {
         char *name;
 
         int (*probe)(struct platform_device *pdev);
@@ -34,6 +35,13 @@ struct snd_soc_platform {
         int (*pcm_new)(struct snd_card *, struct snd_soc_codec_dai *, struct snd_pcm *);
         void (*pcm_free)(struct snd_pcm *);
 
+        /*
+         * For platform caused delay reporting.
+         * Optional.
+         */
+        snd_pcm_sframes_t (*delay)(struct snd_pcm_substream *,
+                struct snd_soc_dai *);
+
         /* platform stream ops */
         struct snd_pcm_ops *pcm_ops;
 };
diff --git a/Documentation/sound/oss/AudioExcelDSP16 b/Documentation/sound/oss/AudioExcelDSP16
index c0f08922993b..e0dc0641b480 100644
--- a/Documentation/sound/oss/AudioExcelDSP16
+++ b/Documentation/sound/oss/AudioExcelDSP16
@@ -1,10 +1,10 @@
 Driver
 ------
 
-Informations about Audio Excel DSP 16 driver can be found in the source
+Information about Audio Excel DSP 16 driver can be found in the source
 file aedsp16.c
 Please, read the head of the source before using it. It contain useful
-informations.
+information.
 
 Configuration
 -------------
@@ -68,7 +68,7 @@ Sound cards supported
 This driver supports the SC-6000 and SC-6600 based Gallant's sound card.
 It don't support the Audio Excel DSP 16 III (try the SC-6600 code).
 I'm working on the III version of the card: if someone have useful
-informations about it, please let me know.
+information about it, please let me know.
 For all the non-supported audio cards, you have to boot MS-DOS (or WIN95)
 activating the audio card with the MS-DOS device driver, then you have to
 <ctrl>-<alt>-<del> and boot Linux.
diff --git a/Documentation/sound/oss/README.OSS b/Documentation/sound/oss/README.OSS
index c615debbf08d..4be259428a1c 100644
--- a/Documentation/sound/oss/README.OSS
+++ b/Documentation/sound/oss/README.OSS
@@ -1352,7 +1352,7 @@ OSS-mixer.
 The PCM20 contains a radio tuner, which is also controlled by
 ACI. This radio tuner is supported by the ACI driver together with the
 miropcm20.o module. Also the 7-band equalizer is integrated
-(limited by the OSS-design). Developement has started and maybe
+(limited by the OSS-design). Development has started and maybe
 finished for the RDS decoder on this card, too. You will be able to
 read RadioText, the Programme Service name, Programme TYpe and
 others. Even the v4l radio module benefits from it with a refined
diff --git a/Documentation/sound/oss/README.ymfsb b/Documentation/sound/oss/README.ymfsb
index af8a7d3a4e8e..b6b77906b58d 100644
--- a/Documentation/sound/oss/README.ymfsb
+++ b/Documentation/sound/oss/README.ymfsb
@@ -5,7 +5,7 @@ FIRST OF ALL
 ============
 
   This code references YAMAHA's sample codes and data sheets.
-  I respect and thank for all people they made open the informations
+  I respect and thank for all people they made open the information
   about YMF7xx cards.
 
   And this codes heavily based on Jeff Garzik <jgarzik@pobox.com>'s