gmane.linux.alsa.devel

Dynamic PCM and Tegra AHUB

Stephen Warren — 2012-05-26T00:23:48

Mark, Liam,

Tegra30's AHUB is structured as follows:


Notes on the diagram:

Each FIFO above is a separate TX and RX FIFO. I merged them in the
drawing for simplicity, but they operate completely independently;
different memory packing formats, data flow rates, ...

The CIF can convert the audio format, e.g. mono<->stereo conversion and
change the # of bits in the data in pretty arbitrary combinations. This
is true for all CIFs; those that join the AHUB core to either the DMA
FIFOs or the I2S/SPDIF/DAM controllers.

The AHUB core is a complete cross-bar; each output selects 1 of the n
inputs.

The I2S and SPDIF controllers take audio from the AHUB, format it to the
appropriate protocol, and send to external IO (or the other way around).
The I2S and SPDIF modules don't perform any additional data
rate/width/channel conversion; the CIF must do whatever conversions are
needed.

The DAMs take 2 input channels from the AHUB, optionally perform some
sample rate conversion and/or bit size conversion beyond what t

Help requested: new HSS1394 MIDI back-end

Sean M. Pappalardo - D.J. Pegasus — 2012-05-25T19:43:17

Hello everyone.

I am a developer for Mixxx, GPL cross-platform digital DJ software: 
http://www.mixxx.org. My focus is the controller subsystem. I am ready 
to finally add Linux support for HSS1394 DJ controllers such as 
Stanton's SCS.1m and SCS.1d.

HSS1394 is just MIDI over IEEE1394 (Firewire) at wire speed. The Windows 
and OSX library source code is LGPL and available on LaunchPad here: 
https://launchpad.net/hss1394

The Stanton SCS.1m is a sound interface integrated with a two-way 
control surface that speaks HSS1394. (The sound card uses an Oxford chip 
set and works fine with FFADO 2.0.1 today up to 96kHz. I use it with 
Mixxx via JACK.)

The Stanton SCS.1d is a two-way control surface (no sound card) that 
speaks HSS1394 and is capable of sending ALOT of latency-sensitive data: 
about 4000 messages per platter rotation. (Now imagine if you had four 
or even two of them on the same network running at +50% speed. This 
would quickly overwhelm USB, at least at the time these devices were 
designed.)

[PATCH] ASoC: fsi: use PIO handler if DMA handler wasinvalid

Kuninori Morimoto — 2012-05-25T06:56:19

PIO handler is not good performance, but works on all platform.
So, switch to PIO handler if DMA handler was invalid case.

Signed-off-by: Kuninori Morimoto <kuninori.morimoto.gx< at >renesas.com>
---
 sound/soc/sh/fsi.c |   29 ++++++++++++++++++++---------
 1 files changed, 20 insertions(+), 9 deletions(-)

diff --git a/sound/soc/sh/fsi.c b/sound/soc/sh/fsi.c
index 05582c1..0da021a 100644
--- a/sound/soc/sh/fsi.c
+++ b/sound/soc/sh/fsi.c
< at >< at > -247,7 +247,7 < at >< at > struct fsi_priv {
 struct fsi_stream_handler {
 int (*init)(struct fsi_priv *fsi, struct fsi_stream *io);
 int (*quit)(struct fsi_priv *fsi, struct fsi_stream *io);
-int (*probe)(struct fsi_priv *fsi, struct fsi_stream *io);
+int (*probe)(struct fsi_priv *fsi, struct fsi_stream *io, struct device *dev);
 int (*transfer)(struct fsi_priv *fsi, struct fsi_stream *io);
 int (*remove)(struct fsi_priv *fsi, struct fsi_stream *io);
 void (*start_stop)(struct fsi_priv *fsi, struct fsi_stream *io,
< at >< at > -571,16 +571,16 < at >< at > static int fsi_stream_transfer(struct fsi_

[PATCH] ASoC: fsi: bugfix: enable master clock controlon DMA stream

Kuninori Morimoto — 2012-05-25T06:55:11

DMA stream handler didn't care about master clock.
This patch fixes it up.

Signed-off-by: Kuninori Morimoto <kuninori.morimoto.gx< at >renesas.com>
---
 sound/soc/sh/fsi.c |    5 +++++
 1 files changed, 5 insertions(+), 0 deletions(-)

diff --git a/sound/soc/sh/fsi.c b/sound/soc/sh/fsi.c
index 7cee225..05582c1 100644
--- a/sound/soc/sh/fsi.c
+++ b/sound/soc/sh/fsi.c
< at >< at > -1172,9 +1172,14 < at >< at > static int fsi_dma_transfer(struct fsi_priv *fsi, struct fsi_stream *io)
 static void fsi_dma_push_start_stop(struct fsi_priv *fsi, struct fsi_stream *io,
  int start)
 {
+struct fsi_master *master = fsi_get_master(fsi);
+u32 clk  = fsi_is_port_a(fsi) ? CRA  : CRB;
 u32 enable = start ? DMA_ON : 0;
 
 fsi_reg_mask_set(fsi, OUT_DMAC, DMA_ON, enable);
+
+if (fsi_is_clk_master(fsi))
+fsi_master_mask_set(master, CLK_RST, clk, (enable) ? clk : 0);
 }
 
 static int fsi_dma_probe(struct fsi_priv *fsi, struct fsi_stream *io)

emu10k1 (Audigy) FXBus routing query

Dan Swain — 2012-05-25T01:12:05

Dear all,

I am writing to ask how to route certain DSP 'outputs' from my Audigy 2 ZS
back into the equivalent 'ASIO inputs' for recording purposes.

Specifically, I'd like to route outputs from the Wavetable Synths on my
card to inputs that would go straight into a DAW (such as Ardour), so that
I can apply effects, record and mix-down as if they were external synth
signals.

I know that with the kX project drivers in Windows this is possible, by
routing the Synth outputs to FXBus outputs 16-63, leaving the ASIO inputs
as 0-15 for the audio input, and then patching which channels I require
from the Synth to those inputs (e.g. Synth Channel 1 -> FXBus out 16&17 ->
FXBus in 0&1).

After reading a thread that contained responses from Jaroslav, I believe
the answer is in emufx.c, but I'm unsure as to where to start as I need to
assign the outputs from the Synths correctly first.

Thanks in advance! Any help would be greatly appreciated!

Kind Regards,

Dan

runtime change in plugin behaviour

Massimiliano Cialdi — 2012-05-24T14:58:36

Hi,

I have developed a plugin that implements a bi-quad filter. filter's 
parameters are given into asoundrc file.

Now I need to change those parameters during playback.
How can I do?
what help can gives alsalib to achieve this?

best regards

Max

[PATCH 5/5] ASoC: Ux500: Add machine-driver

Ola Lilja — 2012-05-24T13:26:45

Add machine-driver for ST-Ericsson U8500 platform, including
support for the AB8500-codec.

Signed-off-by: Ola Lilja <ola.o.lilja< at >stericsson.com>
---
 sound/soc/ux500/Kconfig         |   11 +
 sound/soc/ux500/Makefile        |    3 +
 sound/soc/ux500/mop500.c        |  117 +++++++++++
 sound/soc/ux500/mop500_ab8500.c |  441 +++++++++++++++++++++++++++++++++++++++
 sound/soc/ux500/mop500_ab8500.h |   22 ++
 5 files changed, 594 insertions(+), 0 deletions(-)
 create mode 100644 sound/soc/ux500/mop500.c
 create mode 100644 sound/soc/ux500/mop500_ab8500.c
 create mode 100644 sound/soc/ux500/mop500_ab8500.h

diff --git a/sound/soc/ux500/Kconfig b/sound/soc/ux500/Kconfig
index 1d38515..069330d 100644
--- a/sound/soc/ux500/Kconfig
+++ b/sound/soc/ux500/Kconfig
< at >< at > -19,3 +19,14 < at >< at > config SND_SOC_UX500_PLAT_DMA
 select SND_SOC_DMAENGINE_PCM
 help
 Say Y if you want to enable the Ux500 platform-driver.
+
++config SND_SOC_UX500_MACH_MOP500
++tristate "Machine - MOP500 (Ux500 + AB8500)"
+depends on AB8500_CORE && A

[PATCH 3/5] ASoC: Ux500: Add platform-driver

Ola Lilja — 2012-05-24T13:26:32

Add platform-driver handling all DMA-activities.

Signed-off-by: Ola Lilja <ola.o.lilja< at >stericsson.com>
---
 sound/soc/ux500/Kconfig     |    7 +
 sound/soc/ux500/Makefile    |    3 +
 sound/soc/ux500/ux500_pcm.c |  318 +++++++++++++++++++++++++++++++++++++++++++
 sound/soc/ux500/ux500_pcm.h |   35 +++++
 4 files changed, 363 insertions(+), 0 deletions(-)
 create mode 100644 sound/soc/ux500/ux500_pcm.c
 create mode 100644 sound/soc/ux500/ux500_pcm.h

diff --git a/sound/soc/ux500/Kconfig b/sound/soc/ux500/Kconfig
index 44cf434..1d38515 100644
--- a/sound/soc/ux500/Kconfig
+++ b/sound/soc/ux500/Kconfig
< at >< at > -12,3 +12,10 < at >< at > menuconfig SND_SOC_UX500
 config SND_SOC_UX500_PLAT_MSP_I2S
 tristate
 depends on SND_SOC_UX500
+
+config SND_SOC_UX500_PLAT_DMA
+tristate "Platform - DB8500 (DMA)"
+depends on SND_SOC_UX500
+select SND_SOC_DMAENGINE_PCM
+help
+Say Y if you want to enable the Ux500 platform-driver.
diff --git a/sound/soc/ux500/Makefile b/sound/soc/ux500/Makefile
index 19974c5..4634bf0 100644
--- a/soun

[PATCH 2/5] ASoC: core: Add widgetSND_SOC_DAPM_CLOCK_SUPPLY

Ola Lilja — 2012-05-24T13:26:25

Adds a supply-widget variant for connection to the clock-framework.
This widget-type corresponds to the variant for regulators.

Signed-off-by: Ola Lilja <ola.o.lilja< at >stericsson.com>
---
 include/sound/soc-dapm.h |   10 ++++++++++
 sound/soc/soc-dapm.c     |   38 ++++++++++++++++++++++++++++++++++++++
 2 files changed, 48 insertions(+), 0 deletions(-)

diff --git a/include/sound/soc-dapm.h b/include/sound/soc-dapm.h
index e3833d9..05559e5 100644
--- a/include/sound/soc-dapm.h
+++ b/include/sound/soc-dapm.h
< at >< at > -229,6 +229,10 < at >< at > struct device;
 {.id = snd_soc_dapm_adc, .name = wname, .sname = stname, .reg = wreg, \
 .shift = wshift, .invert = winvert, \
 .event = wevent, .event_flags = wflags}
+#define SND_SOC_DAPM_CLOCK_SUPPLY(wname) \
+{.id = snd_soc_dapm_clock_supply, .name = wname, \
+.reg = SND_SOC_NOPM, .event = dapm_clock_event, \
+.event_flags = SND_SOC_DAPM_PRE_PMU | SND_SOC_DAPM_POST_PMD }
 
 /* generic widgets */
 #define SND_SOC_DAPM_REG(wid, wname, wreg, wshift, wmask, won_val, woff_val) \
< at >

[PATCH 1/5] ALSA: pcm: Add debug-print helper function

Ola Lilja — 2012-05-24T13:26:03

Adds a function getting the stream-name as a string for
a specific stream.

Signed-off-by: Ola Lilja <ola.o.lilja< at >stericsson.com>
---
 include/sound/pcm.h |   11 +++++++++++
 1 files changed, 11 insertions(+), 0 deletions(-)

diff --git a/include/sound/pcm.h b/include/sound/pcm.h
index 0d11128..a55d5db 100644
--- a/include/sound/pcm.h
+++ b/include/sound/pcm.h
< at >< at > -1073,4 +1073,15 < at >< at > static inline void snd_pcm_limit_isa_dma_size(int dma, size_t *max)
 
 const char *snd_pcm_format_name(snd_pcm_format_t format);
 
+/**
+ * Get a string naming the direction of a stream
+ */
+static inline const char *snd_pcm_stream_str(struct snd_pcm_substream *substream)
+{
+if (substream->stream == SNDRV_PCM_STREAM_PLAYBACK)
+return "Playback";
+else
+return "Capture";
+}
+
 #endif /* __SOUND_PCM_H */

[PATCH 1/5] ALSA: pcm: Add debug-print helper function

Ola Lilja — 2012-05-24T13:24:57

Adds a function getting the stream-name as a string for
a specific stream.

Signed-off-by: Ola Lilja <ola.o.lilja< at >stericsson.com>
---
 include/sound/pcm.h |   11 +++++++++++
 1 files changed, 11 insertions(+), 0 deletions(-)

diff --git a/include/sound/pcm.h b/include/sound/pcm.h
index 0d11128..a55d5db 100644
--- a/include/sound/pcm.h
+++ b/include/sound/pcm.h
< at >< at > -1073,4 +1073,15 < at >< at > static inline void snd_pcm_limit_isa_dma_size(int dma, size_t *max)
 
 const char *snd_pcm_format_name(snd_pcm_format_t format);
 
+/**
+ * Get a string naming the direction of a stream
+ */
+static inline const char *snd_pcm_stream_str(struct snd_pcm_substream *substream)
+{
+if (substream->stream == SNDRV_PCM_STREAM_PLAYBACK)
+return "Playback";
+else
+return "Capture";
+}
+
 #endif /* __SOUND_PCM_H */

[PATCH 0/5] Version 5

Ola Lilja — 2012-05-24T13:24:45

*) ALSA-core helper function made static inline
*) Removed debug-functions from the core-patch
*) Bug-fixes in codec-file
*) Controversial controls (clock-stuff) moved to machine-file
*) Removed custom functions for second codec-interface
*) Streams moved to separete DAPM-widgets

Ola Lilja (5):
  ALSA: pcm: Add debug-print helper function
  ASoC: core: Add widget SND_SOC_DAPM_CLOCK_SUPPLY
  ASoC: Ux500: Add platform-driver
  ASoC: codecs: Add AB8500 codec-driver
  ASoC: Ux500: Add machine-driver

 include/sound/pcm.h             |   11 +
 include/sound/soc-dapm.h        |   10 +
 sound/soc/codecs/Kconfig        |    4 +
 sound/soc/codecs/Makefile       |    2 +
 sound/soc/codecs/ab8500-codec.c | 2569 +++++++++++++++++++++++++++++++++++++++
 sound/soc/codecs/ab8500-codec.h |  606 +++++++++
 sound/soc/soc-dapm.c            |   38 +
 sound/soc/ux500/Kconfig         |   18 +
 sound/soc/ux500/Makefile        |    6 +
 sound/soc/ux500/mop500.c        |  117 ++
 sound/soc/ux500/mop500_ab8500.c |  441 +++++++
 soun

UAC2 defaults to max sampling rate

adelias — 2012-05-23T07:59:56

Hi,

I'm running ALSA 1.0.23 and kernel 2.6.37 on ARM. The UAC2 audio device 
is the XMOS reference board.

This is what I'm observing. As soon as the XMOS board is plugged in to 
the USB port, the display on my DAC indicates that is has locked at 
192kHz which is the maximum sampling rate supported by the XMOS.

Is this the expected behavior or can I specify the defualt sampling rate 
at start up?

Thanks

[PATCH] alsa-lib conf: Add more USB devices to S/PDIFblacklist

David Henningsson — 2012-05-23T07:59:50

These devices don't have digital in/out, so prevent them from being
opened.

Signed-off-by: David Henningsson <david.henningsson< at >canonical.com>
---
 src/conf/cards/USB-Audio.conf |    4 ++++
 1 file changed, 4 insertions(+)

diff --git a/src/conf/cards/USB-Audio.conf b/src/conf/cards/USB-Audio.conf
index 0a0e374..177a7af 100644
--- a/src/conf/cards/USB-Audio.conf
+++ b/src/conf/cards/USB-Audio.conf
< at >< at > -42,6 +42,10 < at >< at > USB-Audio.pcm.iec958_device {
 "Logitech Speaker Lapdesk N700" 999
 "Logitech USB Headset" 999
 "Logitech Wireless Headset" 999
+"Plantronics GameCom 780" 999
+"Plantronics USB Headset" 999
+"Plantronics Wireless Audio" 999
+"SB WoW Headset" 999
 "Sennheiser USB headset" 999
 "USB Device 0x46d:0x992" 999
 }

[PATCH] ASoC: tegra+wm8903: remove non-DT support forSeaboard

Stephen Warren — 2012-05-22T22:11:19

From: Stephen Warren <swarren< at >nvidia.com>

In kernel 3.6, Seaboard will only be supported when booting using device
tree; the board files are being removed. Hence, remove the non-DT support
for Seaboard and derivatives Kaen and Aebl from the audio driver.

Harmony is the only remaining board supported by this driver when not
using DT. This support is currently scheduled for removal in 3.7.

Signed-off-by: Stephen Warren <swarren< at >nvidia.com>
---
This is pretty independent from anything else, but is based on the previous
5-long series I posted in terms of context.

 sound/soc/tegra/tegra_wm8903.c |   48 +--------------------------------------
 1 files changed, 2 insertions(+), 46 deletions(-)

diff --git a/sound/soc/tegra/tegra_wm8903.c b/sound/soc/tegra/tegra_wm8903.c
index 1fd6a41..b75e0e8 100644
--- a/sound/soc/tegra/tegra_wm8903.c
+++ b/sound/soc/tegra/tegra_wm8903.c
< at >< at > -28,8 +28,6 < at >< at >
  *
  */
 
-#include <asm/mach-types.h>
-
 #include <linux/module.h>
 #include <linux/platform_device.h>
 #include <linux/s

[PATCH 1/2] ALSA: update sync header when streams arelinked/unlinked

Pierre-Louis Bossart — 2012-05-22T19:54:01

Previous code only reported card number and was not updated
when devices were linked/unlinked. This makes alsa-lib
snd_pcm_info_get_sync totally useless.
Add hooks to force update of sync header when devices are
linked/unlinked, and provide more information such as
number of devices and indices of capture/playback devices
linked to

Signed-off-by: Pierre-Louis Bossart <pierre-louis.bossart< at >linux.intel.com>
---
 sound/core/pcm_lib.c    |   37 ++++++++++++++++++++++++++++++++-----
 sound/core/pcm_native.c |    6 ++++++
 2 files changed, 38 insertions(+), 5 deletions(-)

diff --git a/sound/core/pcm_lib.c b/sound/core/pcm_lib.c
index faedb14..ae46d75 100644
--- a/sound/core/pcm_lib.c
+++ b/sound/core/pcm_lib.c
< at >< at > -533,11 +533,38 < at >< at > EXPORT_SYMBOL(snd_pcm_set_ops);
 void snd_pcm_set_sync(struct snd_pcm_substream *substream)
 {
 struct snd_pcm_runtime *runtime = substream->runtime;
-
-runtime->sync.id32[0] = substream->pcm->card->number;
-runtime->sync.id32[1] = -1;
-runtime->sync.id32[2] = -1;
-runtime->sync

[PATCH 3/3] sb_mixer: Add extended SB16 mixer controlsfor newer chips

Ondrej Zary — 2012-05-22T19:45:04

Newer SB16 chips (Vibra 16C and later) have extended mixer controls.
This detects them and enables automatically if present.

Supported extended controls:
Synth Playback Switch
PCM Playback Switch
Beep Playback Switch
Master Playback Switch
PCM Capture Switch
Master Capture Volume
Master Capture Switch

Vibra 16XV has an additional Aux input which is used on
SF16-FMD and SF16-FMD2 cards to connect the onboard FM tuner:
Aux Playback Switch
Aux Volume
Aux Capture Switch

Signed-off-by: Ondrej Zary <linux< at >rainbow-software.org>

--- a/include/sound/sb.h
+++ b/include/sound/sb.h
< at >< at > -214,6 +214,12 < at >< at > struct snd_sb {
 #define SB_DSP4_IGAIN_DEV0x3f
 #define SB_DSP4_OGAIN_DEV0x41
 #define SB_DSP4_MIC_AGC0x43
+/* extended registers for newer SB16 */
+#define SB_DSP4_OUTPUT2_SW0x48
+#define SB_DSP4_MASTER_SW0x49
+#define SB_DSP4_INPUT2_SW0x4a
+#define SB_DSP4_MASTER_REC_DEV0x50
+#define SB_DSP4_AUX_DEV0x52
 
 /* additional registers for SB 16 mixer */
 #define SB_DSP4_IRQSETUP0x80
--- a/sound/isa/sb/sb_mixer

[PATCH 2/3] sb_mixer: Autodetect optional SB16 mixercontrols

Ondrej Zary — 2012-05-22T19:44:55

Tone (bass/treble), gain and 3D mixer controls are not present on all SB16 cards.

This attempts to detect chip type and enable them only if they're really present.

Signed-off-by: Ondrej Zary <linux< at >rainbow-software.org>

--- a/sound/isa/sb/sb_mixer.c
+++ b/sound/isa/sb/sb_mixer.c
< at >< at > -592,10 +592,6 < at >< at > static struct sbmix_elem snd_sb16_controls[] = {
 SB_SINGLE("Mic Playback Switch", SB_DSP4_OUTPUT_SW, 0, 1),
 SB_SINGLE("Mic Volume", SB_DSP4_MIC_DEV, 3, 31),
 SB_SINGLE("Beep Volume", SB_DSP4_SPEAKER_DEV, 6, 3),
-SB_DOUBLE("Gain Capture Volume",
-  SB_DSP4_IGAIN_DEV, (SB_DSP4_IGAIN_DEV + 1), 6, 6, 3),
-SB_DOUBLE("Gain Playback Volume",
-  SB_DSP4_OGAIN_DEV, (SB_DSP4_OGAIN_DEV + 1), 6, 6, 3),
 SB16_INPUT_SW("Line Capture Route",
       SB_DSP4_INPUT_LEFT, SB_DSP4_INPUT_RIGHT, 4, 3),
 SB_DOUBLE("Line Playback Switch",
< at >< at > -603,13 +599,25 < at >< at > static struct sbmix_elem snd_sb16_controls[] = {
 SB_DOUBLE("Line Volume",
   SB_DSP4_LINE_DEV, (SB_DSP4_LINE_DEV + 1), 3, 3, 31),
 SB_SINGLE("Mic Auto Gain"

[PATCH 1/3] sb_mixer: Correct SB16 mixer control names

Ondrej Zary — 2012-05-22T19:44:47

SB16 volume controls affect both playback and recording. Correct the volume controls to match that.
(Beep volume was already correct).

Correct misleading "Capture Volume" and "Playback Volume" names - they're in fact gain controls.

Also change "3D Enhancement Switch" to standard name "3D Control - Switch".

Signed-off-by: Ondrej Zary <linux< at >rainbow-software.org>

--- a/sound/isa/sb/sb_mixer.c
+++ b/sound/isa/sb/sb_mixer.c
< at >< at > -579,31 +579,31 < at >< at > static struct sbmix_elem snd_sb16_controls[] = {
   SB_DSP4_PCM_DEV, (SB_DSP4_PCM_DEV + 1), 3, 3, 31),
 SB16_INPUT_SW("Synth Capture Route",
       SB_DSP4_INPUT_LEFT, SB_DSP4_INPUT_RIGHT, 6, 5),
-SB_DOUBLE("Synth Playback Volume",
+SB_DOUBLE("Synth Volume",
   SB_DSP4_SYNTH_DEV, (SB_DSP4_SYNTH_DEV + 1), 3, 3, 31),
 SB16_INPUT_SW("CD Capture Route",
       SB_DSP4_INPUT_LEFT, SB_DSP4_INPUT_RIGHT, 2, 1),
 SB_DOUBLE("CD Playback Switch",
   SB_DSP4_OUTPUT_SW, SB_DSP4_OUTPUT_SW, 2, 1, 1),
-SB_DOUBLE("CD Playback Volume",
+SB_DOUBLE("CD Volume",
   SB_

[GIT PULL] ASoC updates for 3.5

Liam Girdwood — 2012-05-22T17:19:25

Hi Takashi,

It's mainly OMAP4 HDMI updates from Ricardo using the new DSS audio
interface and Device Tree updates from Peter for OMAP.

Thanks

Liam 

---

The following changes since commit 766812e6d5e2e23be1e212cf84902d5e834dd865:

  ASoC: sh: fsi: enable chip specific data transfer mode (2012-05-19 19:41:45 +0100)

are available in the git repository at:

  git://git.kernel.org/pub/scm/linux/kernel/git/lrg/asoc.git for-3.5

for you to fetch changes up to ced47a3117ba084b404f3fc28de3ef6eaba1b911:

  ASoC: OMAP: HDMI: Rename sound card source file (2012-05-22 17:33:24 +0100)

----------------------------------------------------------------
Peter Ujfalusi (9):
      ASoC: omap-mcbsp: Use DMA packet mode for non mono streams on OMAP3+
      ASoC: omap-mcbsp: Remove unused FRAME dma_op_mode
      ASoC: omap-mcbsp: Use the common interrupt line if supported by the SoC
      ASoC: omap-mcbsp: buffer size constraint only applies to playback stream
      ASoC: omap-mcpdm: Add device tree bindings
      ASoC: omap

ASoC updates for 3.5

Mark Brown — 2012-05-21T14:57:41

The following changes since commit 5fb86e5d4a951ddb0474cdfd809380c8e2a8d101:

  ARM: mx31_3ds: Add sound support (2012-05-18 16:42:44 +0100)

are available in the git repository at:

  git://git.kernel.org/pub/scm/linux/kernel/git/broonie/sound.git tags/asoc-3.5

for you to fetch changes up to 766812e6d5e2e23be1e212cf84902d5e834dd865:

  ASoC: sh: fsi: enable chip specific data transfer mode (2012-05-19 19:41:45 +0100)

----------------------------------------------------------------
ASoC: Last minute updates

These are all new code, they've been in -next already so should be OK
for merge this time round.  I'd been planning to send a pull request
today after they'd had a bit of exposure there to make sure breakage
didn't propagate into your tree.

----------------------------------------------------------------
Kuninori Morimoto (5):
      ASoC: sh: fsi: use register field macro name on IN/OUT_DMAC
      ASoC: sh: fsi: add fsi_version() and removed meaningless version check
      ASoC: sh: fsi: use same form