[alsa-cvslog] alsa-kmirror: ALSA kernel mirror repository branch, master now at v1.0.22.1-93-g88eeb
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[alsa-cvslog] alsa-kmirror: ALSA kernel mirror repository branch, master now at v1.0.22.1-93-g88eeb2a
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From: <nor...@al...> - 2010-01-26 13:10:28
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Hello,
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generated because a ref change was pushed to the repository.
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- Log -----------------------------------------------------------------
commit 88eeb2ae78c2fb7dfe664188d6e20b146269184c
Author: Takashi Iwai <ti...@su...>
AuthorDate: Tue Jan 19 15:46:37 2010 +0100
Commit: Jaroslav Kysela <pe...@pe...>
CommitDate: Tue Jan 26 13:12:22 2010 +0100
ALSA: hda - Turn on EAPD only if available for Realtek codecs
Some codecs disable widgets used for output pins and reserve as vendor-
spec widgets. Thus we need to check the widget type and pin cap before
actually sending SET_EAPD verbs in the auto-configuration mode.
Signed-off-by: Takashi Iwai <ti...@su...>
Signed-off-by: Jaroslav Kysela <pe...@pe...>
commit be7e096b103c67fb18a95c396cac082b20c83156
Author: Wei Ni <wn...@nv...>
AuthorDate: Tue Jan 26 15:59:33 2010 +0800
Commit: Jaroslav Kysela <pe...@pe...>
CommitDate: Tue Jan 26 13:04:22 2010 +0100
ALSA: hda - Add support for more the 8 streams
In azx_stream_start() and azx_stream_stop(),
it use azx_readb/azx_writeb to read/write SIE,
it just enable/disable 8 streams.
But according to the HDA spec, it support 30 streams,
and the new HDA controller will support more then 8
streams. So we should use azx_readl/azx_writel to
read/write SIE.
Signed-off-by: Wei Ni <wn...@nv...>
Signed-off-by: Takashi Iwai <ti...@su...>
Signed-off-by: Jaroslav Kysela <pe...@pe...>
commit 4211d533d2f3d535620a871bf41c866cbdf68f91
Author: Florian Zumbiehl <fl...@fl...>
AuthorDate: Tue Jan 26 09:06:14 2010 +0100
Commit: Jaroslav Kysela <pe...@pe...>
CommitDate: Tue Jan 26 13:04:21 2010 +0100
ALSA: cs46xx: Fix cpu idling with resume
Make sure that capture DMA doesn't stay enabled after system resume
as that potentially prevents the processor from entering deep sleep
states.
Signed-off-by: Florian Zumbiehl <fl...@fl...>
Signed-off-by: Takashi Iwai <ti...@su...>
Signed-off-by: Jaroslav Kysela <pe...@pe...>
commit ef34f4881b797586a0ed6f63324879366e98f69c
Author: Guennadi Liakhovetski <g.l...@gm...>
AuthorDate: Fri Jan 22 19:09:03 2010 +0100
Commit: Jaroslav Kysela <pe...@pe...>
CommitDate: Tue Jan 26 13:04:20 2010 +0100
ASoC: add DAI and platform / DMA drivers for SH SIU
Several SuperH platforms, including sh7722, sh7343, sh7354, sh7367 include
a Sound Interface Unit (SIU). This patch adds DAI and platform / DMA
drivers for this interface.
Signed-off-by: Guennadi Liakhovetski <g.l...@gm...>
Acked-by: Liam Girdwood <lr...@sl...>
Signed-off-by: Mark Brown <br...@op...>
Signed-off-by: Jaroslav Kysela <pe...@pe...>
commit b72824ef757fd24abf7e2ca04dfa2c9b198d751c
Author: Guennadi Liakhovetski <g.l...@gm...>
AuthorDate: Thu Jan 21 22:04:03 2010 +0100
Commit: Jaroslav Kysela <pe...@pe...>
CommitDate: Tue Jan 26 13:04:19 2010 +0100
ASoC: add helper macros to declare struct soc_enum instances
Several shortcuts for popular uses of some of SOC_ENUM_* and
SOC_VALUE_ENUM_* macros.
Signed-off-by: Guennadi Liakhovetski <g.l...@gm...>
Acked-by: Liam Girdwood <lr...@sl...>
Signed-off-by: Mark Brown <br...@op...>
Signed-off-by: Jaroslav Kysela <pe...@pe...>
commit a228feaef05deb8f408018f1518c835a5f5f46c9
Author: Guennadi Liakhovetski <g.l...@gm...>
AuthorDate: Thu Jan 21 21:10:47 2010 +0100
Commit: Jaroslav Kysela <pe...@pe...>
CommitDate: Tue Jan 26 13:04:18 2010 +0100
ASoC: add simplified versions of widget macros
Many macros from include/sound/soc-dapm.h take an array and a number of
elements in it as arguments, whereas most users use static arrays and use
"x, ARRAY_SIZE(x)" as arguments. This patch adds simplified versions of
those macros, calling ARRAY_SIZE() internally.
Signed-off-by: Guennadi Liakhovetski <g.l...@gm...>
Acked-by: Liam Girdwood <lr...@sl...>
Signed-off-by: Mark Brown <br...@op...>
Signed-off-by: Jaroslav Kysela <pe...@pe...>
commit 802ecf423dd122e2aea80d0ff1b444ae9eed389d
Author: Takashi Iwai <ti...@su...>
AuthorDate: Mon Jan 25 15:44:11 2010 +0100
Commit: Jaroslav Kysela <pe...@pe...>
CommitDate: Tue Jan 26 13:04:17 2010 +0100
ALSA: hda - Remove the COEF setup for ALC267/ALC268
The COEF setup for model=auto seems problematic on some laptops,
resulting in the silent speaker output. Better to disable it for now.
Signed-off-by: Takashi Iwai <ti...@su...>
Signed-off-by: Jaroslav Kysela <pe...@pe...>
commit 1045271ed06714d473897021fa5b2826226ee463
Author: Takashi Iwai <ti...@su...>
AuthorDate: Mon Jan 25 15:41:11 2010 +0100
Commit: Jaroslav Kysela <pe...@pe...>
CommitDate: Tue Jan 26 13:04:16 2010 +0100
ALSA: hda - Remove coef output in Realtek proc files
The output of COEF index/value in the proc file for Realtek codecs is
rather useless since the value varies together with the index.
Let's get rid of it again.
Signed-off-by: Takashi Iwai <ti...@su...>
Signed-off-by: Jaroslav Kysela <pe...@pe...>
commit 6b1c80f31c0600f64d282c2486a833cc0a1aac73
Author: Takashi Iwai <ti...@su...>
AuthorDate: Sun Jan 24 11:19:27 2010 +0100
Commit: Jaroslav Kysela <pe...@pe...>
CommitDate: Tue Jan 26 13:04:15 2010 +0100
ALSA: hda - Change headphone pin control with master volume on cx5051
The HP pin (0x16) control has to be changed dynamically depending on
the master volume switch as well as the speaker pin (0x1a). Otherwise
the headphone still sounds with master off.
Signed-off-by: Takashi Iwai <ti...@su...>
Signed-off-by: Jaroslav Kysela <pe...@pe...>
commit 03b515c45e2a93331d2ac9dc1c16440f7fa313ec
Author: Takashi Iwai <ti...@su...>
AuthorDate: Sun Jan 24 11:14:36 2010 +0100
Commit: Jaroslav Kysela <pe...@pe...>
CommitDate: Tue Jan 26 13:04:14 2010 +0100
ALSA: hda - Fix SPDIF output widget for Cxt5051 codec
Fixed the wrongly set up for SPDIF output on Conexant 5051 codec.
It must point to the audio out widget instead of a pin.
Signed-off-by: Takashi Iwai <ti...@su...>
Signed-off-by: Jaroslav Kysela <pe...@pe...>
commit 88a44460759effc9d85df534664a8d15762c479c
Author: Takashi Iwai <ti...@su...>
AuthorDate: Sun Jan 24 11:00:27 2010 +0100
Commit: Jaroslav Kysela <pe...@pe...>
CommitDate: Tue Jan 26 13:04:13 2010 +0100
ALSA: hda - initialize mic port on cxt5051 codec dynamically
Initialize the mic ports B & C on Conexant 5051 codec dynamically
according to the mic jack detection, instead of static init arrays.
Signed-off-by: Takashi Iwai <ti...@su...>
Signed-off-by: Jaroslav Kysela <pe...@pe...>
commit 2ad93669f27024f1bf88a8ebfd990d20c1f8445e
Author: Takashi Iwai <ti...@su...>
AuthorDate: Sun Jan 24 10:47:02 2010 +0100
Commit: Jaroslav Kysela <pe...@pe...>
CommitDate: Tue Jan 26 13:04:12 2010 +0100
ALSA: hda - Merge playback controls for Cx5051 codec models
All cx5051 codec models have the same Master playback mixer definitions.
Merge them together.
Signed-off-by: Takashi Iwai <ti...@su...>
Signed-off-by: Jaroslav Kysela <pe...@pe...>
commit 181ce63a54d41ab0687d694f41fa96dc482e50c3
Author: Takashi Iwai <ti...@su...>
AuthorDate: Sat Jan 23 22:31:36 2010 +0100
Commit: Jaroslav Kysela <pe...@pe...>
CommitDate: Tue Jan 26 13:04:11 2010 +0100
ALSA: hda - Add support for Toshiba Satellite M300
Added the support for Toshiba Satellite M300 with Conexant 5051 codec.
Since the laptop has no port C connection and the pin reports always
the jack sense true, we need to ignore port-C unsol event.
Signed-off-by: Takashi Iwai <ti...@su...>
Signed-off-by: Jaroslav Kysela <pe...@pe...>
commit a135237bc7bd752e8a40eea693579c0970147792
Author: Takashi Iwai <ti...@su...>
AuthorDate: Sat Jan 23 22:29:54 2010 +0100
Commit: Jaroslav Kysela <pe...@pe...>
CommitDate: Tue Jan 26 13:04:10 2010 +0100
ALSA: hda - Fix HP dv6736 capture mixer name
Use the standard "Capture" mixer name for HP dv6736 with Cxt5051 codec.
Signed-off-by: Takashi Iwai <ti...@su...>
Signed-off-by: Jaroslav Kysela <pe...@pe...>
commit 7c2419c3ece4d72884f98835129e6927788f7151
Author: Takashi Iwai <ti...@su...>
AuthorDate: Sat Jan 23 22:19:29 2010 +0100
Commit: Jaroslav Kysela <pe...@pe...>
CommitDate: Tue Jan 26 13:04:09 2010 +0100
ALSA: hda - Minor fixes for Compaq Presario F700 quirk
Minor fixes for HP Compaq Presario F700 quirks with Cxt5051 codec:
- changed the capture mixer elements to the standard name.
- fixed the quirk name string without a space
- sorted the quirk list
- updated the documentation
Signed-off-by: Takashi Iwai <ti...@su...>
Signed-off-by: Jaroslav Kysela <pe...@pe...>
commit def40544afdf72602f5b069f88876b79f2967d69
Author: Mark Brown <br...@op...>
AuthorDate: Thu Jan 21 11:33:20 2010 +0000
Commit: Jaroslav Kysela <pe...@pe...>
CommitDate: Tue Jan 26 13:04:08 2010 +0100
ASoC: Use BIAS_OFF when idle for wm_hubs devices
This provides a small power saving when audio is inactive.
Signed-off-by: Mark Brown <br...@op...>
Acked-by: Liam Girdwood <lr...@sl...>
Signed-off-by: Jaroslav Kysela <pe...@pe...>
commit dd668803acd611e08d36a24e80d726589286c9ab
Author: Mark Brown <br...@op...>
AuthorDate: Tue Jan 19 22:49:43 2010 +0000
Commit: Jaroslav Kysela <pe...@pe...>
CommitDate: Tue Jan 26 13:04:06 2010 +0100
ASoC: Support turning off bias when the CODEC is idle
Currently ASoC always maintains the bias of the CODEC while the system
is active. With older mobile CODECs this is required since the outputs
are referenced to a non-zero voltage and enabling or disabling this
voltage without audible pops or clicks in the output takes too long to
do when starting or stopping audio.
As a result of features such as ground referenced outputs and class D
speaker drivers current generation devices are able to power on and off
much more quickly without these system level issues so provide a new
flag idle_bias_off in snd_soc_codec which will cause the core to turn
off the CODEC bias. The distinction between STANDBY and OFF is still
maintained. This is partly for consistency but also allows for
potential future extensions such as per-machine overrides or deferring
the bias removal.
Signed-off-by: Mark Brown <br...@op...>
Acked-by: Liam Girdwood <lr...@sl...>
Signed-off-by: Jaroslav Kysela <pe...@pe...>
commit f7ae250639b60e24a7184a33a525486631a7872f
Author: Mark Brown <br...@op...>
AuthorDate: Wed Jan 20 18:18:35 2010 +0000
Commit: Jaroslav Kysela <pe...@pe...>
CommitDate: Tue Jan 26 13:04:05 2010 +0100
ASoC: Remove console DAPM debug code
The same information is now visible via debugfs and with large modern
devices dumping everything to the console can be very resource
intensive, causing more harm than good.
Signed-off-by: Mark Brown <br...@op...>
Acked-by: Liam Girdwood <lr...@sl...>
Signed-off-by: Jaroslav Kysela <pe...@pe...>
commit d20dc6e8da1d75b38a2aab4fddef0b3b5f89ed0b
Author: Peter Ujfalusi <pet...@no...>
AuthorDate: Wed Jan 20 09:39:36 2010 +0200
Commit: Jaroslav Kysela <pe...@pe...>
CommitDate: Tue Jan 26 13:04:05 2010 +0100
ASoC: tlv320dac33: Burst mode BCLK divider configuration
Add possibility to configure the burst mode BCLK divider through platform
data structure.
The BCLK divider changes the actual speed of the serial bus in burst mode,
which is faster than the sampling frequency of the running stream.
In this way platforms can experiment with the optimal burst speed without
the need to modify the codec driver itself.
Signed-off-by: Peter Ujfalusi <pet...@no...>
Acked-by: Liam Girdwood <lr...@sl...>
Signed-off-by: Mark Brown <br...@op...>
Signed-off-by: Jaroslav Kysela <pe...@pe...>
commit 752882e5a1c41a966571c48e8a531cb25d6915af
Author: Peter Ujfalusi <pet...@no...>
AuthorDate: Wed Jan 20 09:39:35 2010 +0200
Commit: Jaroslav Kysela <pe...@pe...>
CommitDate: Tue Jan 26 13:04:04 2010 +0100
ASoC: tlv320dac33: BCLK divider fix
The BCLK divider was not configured in case of mode7.
This leads to unpredictable behavior when switching between FIFO modes.
Configure the BCLK divider depending on the fifo_mode (FIFO is in use,
or FIFO bypass).
Signed-off-by: Peter Ujfalusi <pet...@no...>
Acked-by: Liam Girdwood <lr...@sl...>
Signed-off-by: Mark Brown <br...@op...>
Signed-off-by: Jaroslav Kysela <pe...@pe...>
commit 2b498b5165dbdcd864b50d35fb9a354a802fa78d
Author: Guennadi Liakhovetski <g.l...@gm...>
AuthorDate: Tue Jan 19 08:39:05 2010 +0100
Commit: Jaroslav Kysela <pe...@pe...>
CommitDate: Tue Jan 26 13:04:03 2010 +0100
ASoC: fix compile breakage - add a missing header include
Signed-off-by: Guennadi Liakhovetski <g.l...@gm...>
Acked-by: Liam Girdwood <lr...@sl...>
Signed-off-by: Mark Brown <br...@op...>
Signed-off-by: Jaroslav Kysela <pe...@pe...>
commit 9fb924f1575219265e82d99b0469fbfa31d71a2f
Author: Peter Ujfalusi <pet...@no...>
AuthorDate: Tue Jan 19 11:15:45 2010 +0200
Commit: Jaroslav Kysela <pe...@pe...>
CommitDate: Tue Jan 26 13:04:01 2010 +0100
ASoC: tlv320dac33: Correct the prefill number of samples
Set the prefill number of samples as the same as the lower
threshold in mode7.
In this way the codec will read the same amount of data on
startup and during the running playback.
Signed-off-by: Peter Ujfalusi <pet...@no...>
Acked-by: Liam Girdwood <lr...@sl...>
Signed-off-by: Mark Brown <br...@op...>
Signed-off-by: Jaroslav Kysela <pe...@pe...>
commit f68a59d739efd855a8a31234a0b695ce2628bc75
Author: Takashi Iwai <ti...@su...>
AuthorDate: Mon Jan 18 14:58:57 2010 +0100
Commit: Jaroslav Kysela <pe...@pe...>
CommitDate: Tue Jan 26 13:04:00 2010 +0100
ALSA: pcm - Call pgprot_noncached() for vmalloc'ed buffers
pgprot_noncached() can be set for vmalloc'ed buffers safely, and we'd
need non-cached behavior more or less, even for the intermediate ring-
buffers.
Now snd_pcm_lib_mmap_vmalloc() is added as the common PCM mmap callback
that is coupled with snd_pcm_lib_alloc_vmalloc_buffer() & co.
Signed-off-by: Takashi Iwai <ti...@su...>
Signed-off-by: Jaroslav Kysela <pe...@pe...>
commit 8c35b091a057ff5490335d4ed568ad688d404fc8
Author: Takashi Iwai <ti...@su...>
AuthorDate: Mon Jan 18 14:49:50 2010 +0100
Commit: Jaroslav Kysela <pe...@pe...>
CommitDate: Tue Jan 26 13:03:59 2010 +0100
ALSA: pcm - Remove unneeded ifdef pgprot_noncached
Signed-off-by: Takashi Iwai <ti...@su...>
Signed-off-by: Jaroslav Kysela <pe...@pe...>
commit a936ec287d44237f439537966884206a7bd7992a
Author: Seth Heasley <set...@in...>
AuthorDate: Tue Jan 12 17:03:35 2010 -0800
Commit: Jaroslav Kysela <pe...@pe...>
CommitDate: Tue Jan 26 13:03:58 2010 +0100
ALSA: hda_intel: ALSA HD Audio patch for Intel Cougar Point DeviceIDs
This patch adds the Intel Cougar Point (PCH) HD Audio Controller DeviceIDs.
Signed-off-by: Seth Heasley <set...@in...>
Signed-off-by: Takashi Iwai <ti...@su...>
Signed-off-by: Jaroslav Kysela <pe...@pe...>
commit 4ba6fe71a10100c6ef737a61d4a7381b6566c68e
Author: Guennadi Liakhovetski <g.l...@gm...>
AuthorDate: Fri Jan 22 18:00:03 2010 +0100
Commit: Mark Brown <br...@op...>
CommitDate: Mon Jan 25 14:41:05 2010 +0000
ASoC: fix a memory-leak in wm8903
Remember to free the temporary register-cache.
Signed-off-by: Guennadi Liakhovetski <g.l...@gm...>
Acked-by: Liam Girdwood <lr...@sl...>
Signed-off-by: Mark Brown <br...@op...>
Cc: st...@ke...
commit c3fe3bf41955cc58c4de30c9e1527c1bdbcbf4af
Author: Åukasz WojniÅowicz <luk...@gm...>
AuthorDate: Sun Jan 24 14:12:37 2010 +0100
Commit: Takashi Iwai <ti...@su...>
CommitDate: Mon Jan 25 08:00:02 2010 +0100
ALSA: hda - add possibility to choose speakers configuration for 4930g
Now one can choose speaker configuration in e.g. PulseAudio mixer
Signed-off-by: Åukasz WojniÅowicz <luk...@gm...>
Signed-off-by: Takashi Iwai <ti...@su...>
commit 3a91f812e2f7132c325bdebdfda89de1017c2742
Author: Takashi Iwai <ti...@su...>
AuthorDate: Wed Jan 20 08:35:06 2010 +0100
Commit: Takashi Iwai <ti...@su...>
CommitDate: Wed Jan 20 08:35:06 2010 +0100
ALSA: hda - Fix HP T5735 automute
This patch fixes the aut-mute setup on HP T5735 with ALC262 codec.
Instead of wrong amp, use pin control toggling for muting the speaker now.
Tested-by: Lee Trager <lee...@hp...>
Cc: <st...@ke...>
Signed-off-by: Takashi Iwai <ti...@su...>
commit ff00a10e0efd491c53c79e83685c68da0bd598f8
Author: Takashi Iwai <ti...@su...>
AuthorDate: Tue Jan 19 15:38:44 2010 +0100
Commit: Takashi Iwai <ti...@su...>
CommitDate: Tue Jan 19 15:38:44 2010 +0100
ALSA: hda - Fix parsing pin node 0x21 on ALC259
ALC259 has a widget NID 0x21 for the output pin, but it wasn't handled
properly in alc268_new_analog_output().
Signed-off-by: Takashi Iwai <ti...@su...>
commit 3ba8c48f89fc82119eb2e8bcd93c37f64f8a1e44
Author: Takashi Iwai <ti...@su...>
AuthorDate: Mon Jan 18 14:16:24 2010 +0100
Commit: Takashi Iwai <ti...@su...>
CommitDate: Mon Jan 18 14:18:55 2010 +0100
ALSA: Remove warning message for invalid OSS minor ranges
When a card instance with a higher card number is registered, warning
messages are spewed eventually with stack traces due to the invalid minor
number for OSS device registration. For example, thinkpad-acpi registers
the card number 29 as default, and you'll see always these messages.
This is rather confusing (and worries users), thus better to return
simply the error code.
Signed-off-by: Takashi Iwai <ti...@su...>
-----------------------------------------------------------------------
Diffstat:
Documentation/HD-Audio-Models.txt | 2 +
core/pcm_native.c | 11 +-
core/sound_oss.c | 2 +-
drivers/vx/vx_pcm.c | 2 +
include/pcm.h | 4 +
include/soc-dai.h | 2 +
include/soc-dapm.h | 32 ++
include/soc.h | 19 +
include/tlv320dac33-plat.h | 1 +
mips/sgio2audio.c | 3 +
pci/cs46xx/cs46xx_lib.c | 10 +
pci/hda/hda_intel.c | 11 +-
pci/hda/patch_conexant.c | 123 +++---
pci/hda/patch_realtek.c | 64 +--
pcmcia/pdaudiocf/pdaudiocf_pcm.c | 1 +
soc/codecs/tlv320dac33.c | 24 +-
soc/codecs/wm8903.c | 3 +-
soc/codecs/wm_hubs.c | 6 +
soc/sh/Kconfig | 6 +
soc/sh/Makefile | 2 +
soc/sh/siu.h | 193 +++++++++
soc/sh/siu_dai.c | 847 +++++++++++++++++++++++++++++++++++++
soc/sh/siu_pcm.c | 616 +++++++++++++++++++++++++++
soc/soc-dapm.c | 105 ++----
usb/ua101.c | 2 +
usb/usbaudio.c | 2 +
26 files changed, 1903 insertions(+), 190 deletions(-)
Patch -------------- (max 1500 lines)
diff --git a/Documentation/HD-Audio-Models.txt b/Documentation/HD-Audio-Models.txt
index e72cee9..8f06f20 100644
--- a/Documentation/HD-Audio-Models.txt
+++ b/Documentation/HD-Audio-Models.txt
@@ -279,7 +279,9 @@ Conexant 5051
laptop Basic Laptop config (default)
hp HP Spartan laptop
hp-dv6736 HP dv6736
+ hp-f700 HP Compaq Presario F700
lenovo-x200 Lenovo X200 laptop
+ toshiba Toshiba Satellite M300
Conexant 5066
=============
diff --git a/core/pcm_native.c b/core/pcm_native.c
index 43552fd..b53fa84 100644
--- a/core/pcm_native.c
+++ b/core/pcm_native.c
@@ -3165,9 +3165,7 @@ int snd_pcm_lib_mmap_iomem(struct snd_pcm_substream *substream,
long size;
unsigned long offset;
-#ifdef pgprot_noncached
area->vm_page_prot = pgprot_noncached(area->vm_page_prot);
-#endif
area->vm_flags |= VM_IO;
size = area->vm_end - area->vm_start;
offset = area->vm_pgoff << PAGE_SHIFT;
@@ -3181,6 +3179,15 @@ int snd_pcm_lib_mmap_iomem(struct snd_pcm_substream *substream,
EXPORT_SYMBOL(snd_pcm_lib_mmap_iomem);
#endif /* SNDRV_PCM_INFO_MMAP */
+/* mmap callback with pgprot_noncached */
+int snd_pcm_lib_mmap_noncached(struct snd_pcm_substream *substream,
+ struct vm_area_struct *area)
+{
+ area->vm_page_prot = pgprot_noncached(area->vm_page_prot);
+ return snd_pcm_default_mmap(substream, area);
+}
+EXPORT_SYMBOL(snd_pcm_lib_mmap_noncached);
+
/*
* mmap DMA buffer
*/
diff --git a/core/sound_oss.c b/core/sound_oss.c
index 7fe1226..0c164e5 100644
--- a/core/sound_oss.c
+++ b/core/sound_oss.c
@@ -93,7 +93,7 @@ static int snd_oss_kernel_minor(int type, struct snd_card *card, int dev)
default:
return -EINVAL;
}
- if (snd_BUG_ON(minor < 0 || minor >= SNDRV_OSS_MINORS))
+ if (minor < 0 || minor >= SNDRV_OSS_MINORS)
return -EINVAL;
return minor;
}
diff --git a/drivers/vx/vx_pcm.c b/drivers/vx/vx_pcm.c
index c8385d2..35a2f71 100644
--- a/drivers/vx/vx_pcm.c
+++ b/drivers/vx/vx_pcm.c
@@ -905,6 +905,7 @@ static struct snd_pcm_ops vx_pcm_playback_ops = {
.trigger = vx_pcm_trigger,
.pointer = vx_pcm_playback_pointer,
.page = snd_pcm_lib_get_vmalloc_page,
+ .mmap = snd_pcm_lib_mmap_vmalloc,
};
@@ -1125,6 +1126,7 @@ static struct snd_pcm_ops vx_pcm_capture_ops = {
.trigger = vx_pcm_trigger,
.pointer = vx_pcm_capture_pointer,
.page = snd_pcm_lib_get_vmalloc_page,
+ .mmap = snd_pcm_lib_mmap_vmalloc,
};
diff --git a/include/pcm.h b/include/pcm.h
index 768e309..ee16122 100644
--- a/include/pcm.h
+++ b/include/pcm.h
@@ -1022,6 +1022,10 @@ int snd_pcm_lib_mmap_iomem(struct snd_pcm_substream *substream, struct vm_area_s
#define snd_pcm_lib_mmap_iomem NULL
#endif
+int snd_pcm_lib_mmap_noncached(struct snd_pcm_substream *substream,
+ struct vm_area_struct *area);
+#define snd_pcm_lib_mmap_vmalloc snd_pcm_lib_mmap_noncached
+
static inline void snd_pcm_limit_isa_dma_size(int dma, size_t *max)
{
*max = dma < 4 ? 64 * 1024 : 128 * 1024;
diff --git a/include/soc-dai.h b/include/soc-dai.h
index ca24e7f..061f16d 100644
--- a/include/soc-dai.h
+++ b/include/soc-dai.h
@@ -16,6 +16,8 @@
#include <linux/list.h>
+#include <sound/soc.h>
+
struct snd_pcm_substream;
/*
diff --git a/include/soc-dapm.h b/include/soc-dapm.h
index c5c95e1..c0922a0 100644
--- a/include/soc-dapm.h
+++ b/include/soc-dapm.h
@@ -95,6 +95,21 @@
.shift = wshift, .invert = winvert, .kcontrols = wcontrols, \
.num_kcontrols = 1}
+/* Simplified versions of above macros, assuming wncontrols = ARRAY_SIZE(wcontrols) */
+#define SOC_PGA_ARRAY(wname, wreg, wshift, winvert,\
+ wcontrols) \
+{ .id = snd_soc_dapm_pga, .name = wname, .reg = wreg, .shift = wshift, \
+ .invert = winvert, .kcontrols = wcontrols, .num_kcontrols = ARRAY_SIZE(wcontrols)}
+#define SOC_MIXER_ARRAY(wname, wreg, wshift, winvert, \
+ wcontrols)\
+{ .id = snd_soc_dapm_mixer, .name = wname, .reg = wreg, .shift = wshift, \
+ .invert = winvert, .kcontrols = wcontrols, .num_kcontrols = ARRAY_SIZE(wcontrols)}
+#define SOC_MIXER_NAMED_CTL_ARRAY(wname, wreg, wshift, winvert, \
+ wcontrols)\
+{ .id = snd_soc_dapm_mixer_named_ctl, .name = wname, .reg = wreg, \
+ .shift = wshift, .invert = winvert, .kcontrols = wcontrols, \
+ .num_kcontrols = ARRAY_SIZE(wcontrols)}
+
/* path domain with event - event handler must return 0 for success */
#define SND_SOC_DAPM_PGA_E(wname, wreg, wshift, winvert, wcontrols, \
wncontrols, wevent, wflags) \
@@ -126,6 +141,23 @@
.invert = winvert, .kcontrols = wcontrols, .num_kcontrols = 1, \
.event = wevent, .event_flags = wflags}
+/* Simplified versions of above macros, assuming wncontrols = ARRAY_SIZE(wcontrols) */
+#define SOC_PGA_E_ARRAY(wname, wreg, wshift, winvert, wcontrols, \
+ wevent, wflags) \
+{ .id = snd_soc_dapm_pga, .name = wname, .reg = wreg, .shift = wshift, \
+ .invert = winvert, .kcontrols = wcontrols, .num_kcontrols = ARRAY_SIZE(wcontrols), \
+ .event = wevent, .event_flags = wflags}
+#define SOC_MIXER_E_ARRAY(wname, wreg, wshift, winvert, wcontrols, \
+ wevent, wflags) \
+{ .id = snd_soc_dapm_mixer, .name = wname, .reg = wreg, .shift = wshift, \
+ .invert = winvert, .kcontrols = wcontrols, .num_kcontrols = ARRAY_SIZE(wcontrols), \
+ .event = wevent, .event_flags = wflags}
+#define SOC_MIXER_NAMED_CTL_E_ARRAY(wname, wreg, wshift, winvert, \
+ wcontrols, wevent, wflags) \
+{ .id = snd_soc_dapm_mixer, .name = wname, .reg = wreg, .shift = wshift, \
+ .invert = winvert, .kcontrols = wcontrols, \
+ .num_kcontrols = ARRAY_SIZE(wcontrols), .event = wevent, .event_flags = wflags}
+
/* events that are pre and post DAPM */
#define SND_SOC_DAPM_PRE(wname, wevent) \
{ .id = snd_soc_dapm_pre, .name = wname, .kcontrols = NULL, \
diff --git a/include/soc.h b/include/soc.h
index 08909cc..4bbeb9f 100644
--- a/include/soc.h
+++ b/include/soc.h
@@ -169,6 +169,23 @@
.private_value = (unsigned long)&xenum }
/*
+ * Simplified versions of above macros, declaring a struct and calculating
+ * ARRAY_SIZE internally
+ */
+#define SOC_ENUM_DOUBLE_DECL(name, xreg, xshift_l, xshift_r, xtexts) \
+ struct soc_enum name = SOC_ENUM_DOUBLE(xreg, xshift_l, xshift_r, \
+ ARRAY_SIZE(xtexts), xtexts)
+#define SOC_ENUM_SINGLE_DECL(name, xreg, xshift, xtexts) \
+ SOC_ENUM_DOUBLE_DECL(name, xreg, xshift, xshift, xtexts)
+#define SOC_ENUM_SINGLE_EXT_DECL(name, xtexts) \
+ struct soc_enum name = SOC_ENUM_SINGLE_EXT(ARRAY_SIZE(xtexts), xtexts)
+#define SOC_VALUE_ENUM_DOUBLE_DECL(name, xreg, xshift_l, xshift_r, xmask, xtexts, xvalues) \
+ struct soc_enum name = SOC_VALUE_ENUM_DOUBLE(xreg, xshift_l, xshift_r, xmask, \
+ ARRAY_SIZE(xtexts), xtexts, xvalues)
+#define SOC_VALUE_ENUM_SINGLE_DECL(name, xreg, xshift, xmask, xtexts, xvalues) \
+ SOC_VALUE_ENUM_DOUBLE_DECL(name, xreg, xshift, xshift, xmask, xtexts, xvalues)
+
+/*
* Bias levels
*
* @ON: Bias is fully on for audio playback and capture operations.
@@ -405,6 +422,8 @@ struct snd_soc_codec {
short reg_cache_size;
short reg_cache_step;
+ unsigned int idle_bias_off:1; /* Use BIAS_OFF instead of STANDBY */
+
/* dapm */
u32 pop_time;
struct list_head dapm_widgets;
diff --git a/include/tlv320dac33-plat.h b/include/tlv320dac33-plat.h
index 5858d06..ac06652 100644
--- a/include/tlv320dac33-plat.h
+++ b/include/tlv320dac33-plat.h
@@ -15,6 +15,7 @@
struct tlv320dac33_platform_data {
int power_gpio;
+ u8 burst_bclkdiv;
};
#endif /* __TLV320DAC33_PLAT_H */
diff --git a/mips/sgio2audio.c b/mips/sgio2audio.c
index 9b486be..6aff217 100644
--- a/mips/sgio2audio.c
+++ b/mips/sgio2audio.c
@@ -691,6 +691,7 @@ static struct snd_pcm_ops snd_sgio2audio_playback1_ops = {
.trigger = snd_sgio2audio_pcm_trigger,
.pointer = snd_sgio2audio_pcm_pointer,
.page = snd_pcm_lib_get_vmalloc_page,
+ .mmap = snd_pcm_lib_mmap_vmalloc,
};
static struct snd_pcm_ops snd_sgio2audio_playback2_ops = {
@@ -703,6 +704,7 @@ static struct snd_pcm_ops snd_sgio2audio_playback2_ops = {
.trigger = snd_sgio2audio_pcm_trigger,
.pointer = snd_sgio2audio_pcm_pointer,
.page = snd_pcm_lib_get_vmalloc_page,
+ .mmap = snd_pcm_lib_mmap_vmalloc,
};
static struct snd_pcm_ops snd_sgio2audio_capture_ops = {
@@ -715,6 +717,7 @@ static struct snd_pcm_ops snd_sgio2audio_capture_ops = {
.trigger = snd_sgio2audio_pcm_trigger,
.pointer = snd_sgio2audio_pcm_pointer,
.page = snd_pcm_lib_get_vmalloc_page,
+ .mmap = snd_pcm_lib_mmap_vmalloc,
};
/*
diff --git a/pci/cs46xx/cs46xx_lib.c b/pci/cs46xx/cs46xx_lib.c
index e6b4a87..56fcf00 100644
--- a/pci/cs46xx/cs46xx_lib.c
+++ b/pci/cs46xx/cs46xx_lib.c
@@ -3644,6 +3644,7 @@ int snd_cs46xx_resume(struct pci_dev *pci)
#ifdef CONFIG_SND_CS46XX_NEW_DSP
int i;
#endif
+ unsigned int tmp;
pci_set_power_state(pci, PCI_D0);
pci_restore_state(pci);
@@ -3685,6 +3686,15 @@ int snd_cs46xx_resume(struct pci_dev *pci)
snd_ac97_resume(chip->ac97[CS46XX_PRIMARY_CODEC_INDEX]);
snd_ac97_resume(chip->ac97[CS46XX_SECONDARY_CODEC_INDEX]);
+ /*
+ * Stop capture DMA.
+ */
+ tmp = snd_cs46xx_peek(chip, BA1_CCTL);
+ chip->capt.ctl = tmp & 0x0000ffff;
+ snd_cs46xx_poke(chip, BA1_CCTL, tmp & 0xffff0000);
+
+ mdelay(5);
+
/* reset playback/capture */
snd_cs46xx_set_play_sample_rate(chip, 8000);
snd_cs46xx_set_capture_sample_rate(chip, 8000);
diff --git a/pci/hda/hda_intel.c b/pci/hda/hda_intel.c
index 1f516e6..6eeefda 100644
--- a/pci/hda/hda_intel.c
+++ b/pci/hda/hda_intel.c
@@ -125,6 +125,7 @@ MODULE_SUPPORTED_DEVICE("{{Intel, ICH6},"
"{Intel, ICH9},"
"{Intel, ICH10},"
"{Intel, PCH},"
+ "{Intel, CPT},"
"{Intel, SCH},"
"{ATI, SB450},"
"{ATI, SB600},"
@@ -953,8 +954,8 @@ static void azx_stream_start(struct azx *chip, struct azx_dev *azx_dev)
azx_dev->insufficient = 1;
/* enable SIE */
- azx_writeb(chip, INTCTL,
- azx_readb(chip, INTCTL) | (1 << azx_dev->index));
+ azx_writel(chip, INTCTL,
+ azx_readl(chip, INTCTL) | (1 << azx_dev->index));
/* set DMA start and interrupt mask */
azx_sd_writeb(azx_dev, SD_CTL, azx_sd_readb(azx_dev, SD_CTL) |
SD_CTL_DMA_START | SD_INT_MASK);
@@ -973,8 +974,8 @@ static void azx_stream_stop(struct azx *chip, struct azx_dev *azx_dev)
{
azx_stream_clear(chip, azx_dev);
/* disable SIE */
- azx_writeb(chip, INTCTL,
- azx_readb(chip, INTCTL) & ~(1 << azx_dev->index));
+ azx_writel(chip, INTCTL,
+ azx_readl(chip, INTCTL) & ~(1 << azx_dev->index));
}
@@ -2677,6 +2678,8 @@ static struct pci_device_id azx_ids[] = {
{ PCI_DEVICE(0x8086, 0x3a6e), .driver_data = AZX_DRIVER_ICH },
/* PCH */
{ PCI_DEVICE(0x8086, 0x3b56), .driver_data = AZX_DRIVER_ICH },
+ /* CPT */
+ { PCI_DEVICE(0x8086, 0x1c20), .driver_data = AZX_DRIVER_ICH },
/* SCH */
{ PCI_DEVICE(0x8086, 0x811b), .driver_data = AZX_DRIVER_SCH },
/* ATI SB 450/600 */
diff --git a/pci/hda/patch_conexant.c b/pci/hda/patch_conexant.c
index 685015a..9077e41 100644
--- a/pci/hda/patch_conexant.c
+++ b/pci/hda/patch_conexant.c
@@ -42,10 +42,12 @@
/* Conexant 5051 specific */
-#define CXT5051_SPDIF_OUT 0x1C
+#define CXT5051_SPDIF_OUT 0x12
#define CXT5051_PORTB_EVENT 0x38
#define CXT5051_PORTC_EVENT 0x39
+#define AUTO_MIC_PORTB (1 << 1)
+#define AUTO_MIC_PORTC (1 << 2)
struct conexant_jack {
@@ -74,7 +76,7 @@ struct conexant_spec {
*/
unsigned int cur_eapd;
unsigned int hp_present;
- unsigned int no_auto_mic;
+ unsigned int auto_mic;
unsigned int need_dac_fix;
/* capture */
@@ -1603,6 +1605,11 @@ static void cxt5051_update_speaker(struct hda_codec *codec)
{
struct conexant_spec *spec = codec->spec;
unsigned int pinctl;
+ /* headphone pin */
+ pinctl = (spec->hp_present && spec->cur_eapd) ? PIN_HP : 0;
+ snd_hda_codec_write(codec, 0x16, 0, AC_VERB_SET_PIN_WIDGET_CONTROL,
+ pinctl);
+ /* speaker pin */
pinctl = (!spec->hp_present && spec->cur_eapd) ? PIN_OUT : 0;
snd_hda_codec_write(codec, 0x1a, 0, AC_VERB_SET_PIN_WIDGET_CONTROL,
pinctl);
@@ -1626,7 +1633,7 @@ static void cxt5051_portb_automic(struct hda_codec *codec)
struct conexant_spec *spec = codec->spec;
unsigned int present;
- if (spec->no_auto_mic)
+ if (!(spec->auto_mic & AUTO_MIC_PORTB))
return;
present = snd_hda_jack_detect(codec, 0x17);
snd_hda_codec_write(codec, 0x14, 0,
@@ -1641,7 +1648,7 @@ static void cxt5051_portc_automic(struct hda_codec *codec)
unsigned int present;
hda_nid_t new_adc;
- if (spec->no_auto_mic)
+ if (!(spec->auto_mic & AUTO_MIC_PORTC))
return;
present = snd_hda_jack_detect(codec, 0x18);
if (present)
@@ -1687,13 +1694,7 @@ static void cxt5051_hp_unsol_event(struct hda_codec *codec,
conexant_report_jack(codec, nid);
}
-static struct snd_kcontrol_new cxt5051_mixers[] = {
- HDA_CODEC_VOLUME("Internal Mic Volume", 0x14, 0x00, HDA_INPUT),
- HDA_CODEC_MUTE("Internal Mic Switch", 0x14, 0x00, HDA_INPUT),
- HDA_CODEC_VOLUME("External Mic Volume", 0x14, 0x01, HDA_INPUT),
- HDA_CODEC_MUTE("External Mic Switch", 0x14, 0x01, HDA_INPUT),
- HDA_CODEC_VOLUME("Docking Mic Volume", 0x15, 0x00, HDA_INPUT),
- HDA_CODEC_MUTE("Docking Mic Switch", 0x15, 0x00, HDA_INPUT),
+static struct snd_kcontrol_new cxt5051_playback_mixers[] = {
HDA_CODEC_VOLUME("Master Playback Volume", 0x10, 0x00, HDA_OUTPUT),
{
.iface = SNDRV_CTL_ELEM_IFACE_MIXER,
@@ -1703,7 +1704,16 @@ static struct snd_kcontrol_new cxt5051_mixers[] = {
.put = cxt5051_hp_master_sw_put,
.private_value = 0x1a,
},
+ {}
+};
+static struct snd_kcontrol_new cxt5051_capture_mixers[] = {
+ HDA_CODEC_VOLUME("Internal Mic Volume", 0x14, 0x00, HDA_INPUT),
+ HDA_CODEC_MUTE("Internal Mic Switch", 0x14, 0x00, HDA_INPUT),
+ HDA_CODEC_VOLUME("External Mic Volume", 0x14, 0x01, HDA_INPUT),
+ HDA_CODEC_MUTE("External Mic Switch", 0x14, 0x01, HDA_INPUT),
+ HDA_CODEC_VOLUME("Docking Mic Volume", 0x15, 0x00, HDA_INPUT),
+ HDA_CODEC_MUTE("Docking Mic Switch", 0x15, 0x00, HDA_INPUT),
{}
};
@@ -1712,48 +1722,26 @@ static struct snd_kcontrol_new cxt5051_hp_mixers[] = {
HDA_CODEC_MUTE("Internal Mic Switch", 0x14, 0x00, HDA_INPUT),
HDA_CODEC_VOLUME("External Mic Volume", 0x15, 0x00, HDA_INPUT),
HDA_CODEC_MUTE("External Mic Switch", 0x15, 0x00, HDA_INPUT),
- HDA_CODEC_VOLUME("Master Playback Volume", 0x10, 0x00, HDA_OUTPUT),
- {
- .iface = SNDRV_CTL_ELEM_IFACE_MIXER,
- .name = "Master Playback Switch",
- .info = cxt_eapd_info,
- .get = cxt_eapd_get,
- .put = cxt5051_hp_master_sw_put,
- .private_value = 0x1a,
- },
-
{}
};
static struct snd_kcontrol_new cxt5051_hp_dv6736_mixers[] = {
- HDA_CODEC_VOLUME("Mic Volume", 0x14, 0x00, HDA_INPUT),
- HDA_CODEC_MUTE("Mic Switch", 0x14, 0x00, HDA_INPUT),
- HDA_CODEC_VOLUME("Master Playback Volume", 0x10, 0x00, HDA_OUTPUT),
- {
- .iface = SNDRV_CTL_ELEM_IFACE_MIXER,
- .name = "Master Playback Switch",
- .info = cxt_eapd_info,
- .get = cxt_eapd_get,
- .put = cxt5051_hp_master_sw_put,
- .private_value = 0x1a,
- },
-
+ HDA_CODEC_VOLUME("Capture Volume", 0x14, 0x00, HDA_INPUT),
+ HDA_CODEC_MUTE("Capture Switch", 0x14, 0x00, HDA_INPUT),
{}
};
static struct snd_kcontrol_new cxt5051_f700_mixers[] = {
- HDA_CODEC_VOLUME("Mic Volume", 0x14, 0x01, HDA_INPUT),
- HDA_CODEC_MUTE("Mic Switch", 0x14, 0x01, HDA_INPUT),
- HDA_CODEC_VOLUME("Master Playback Volume", 0x10, 0x00, HDA_OUTPUT),
- {
- .iface = SNDRV_CTL_ELEM_IFACE_MIXER,
- .name = "Master Playback Switch",
- .info = cxt_eapd_info,
- .get = cxt_eapd_get,
- .put = cxt5051_hp_master_sw_put,
- .private_value = 0x1a,
- },
+ HDA_CODEC_VOLUME("Capture Volume", 0x14, 0x01, HDA_INPUT),
+ HDA_CODEC_MUTE("Capture Switch", 0x14, 0x01, HDA_INPUT),
+ {}
+};
+static struct snd_kcontrol_new cxt5051_toshiba_mixers[] = {
+ HDA_CODEC_VOLUME("Internal Mic Volume", 0x14, 0x00, HDA_INPUT),
+ HDA_CODEC_MUTE("Internal Mic Switch", 0x14, 0x00, HDA_INPUT),
+ HDA_CODEC_VOLUME("External Mic Volume", 0x14, 0x01, HDA_INPUT),
+ HDA_CODEC_MUTE("External Mic Switch", 0x14, 0x01, HDA_INPUT),
{}
};
@@ -1782,8 +1770,6 @@ static struct hda_verb cxt5051_init_verbs[] = {
/* EAPD */
{0x1a, AC_VERB_SET_EAPD_BTLENABLE, 0x2}, /* default on */
{0x16, AC_VERB_SET_UNSOLICITED_ENABLE, AC_USRSP_EN|CONEXANT_HP_EVENT},
- {0x17, AC_VERB_SET_UNSOLICITED_ENABLE, AC_USRSP_EN|CXT5051_PORTB_EVENT},
- {0x18, AC_VERB_SET_UNSOLICITED_ENABLE, AC_USRSP_EN|CXT5051_PORTC_EVENT},
{ } /* end */
};
@@ -1809,7 +1795,6 @@ static struct hda_verb cxt5051_hp_dv6736_init_verbs[] = {
/* EAPD */
{0x1a, AC_VERB_SET_EAPD_BTLENABLE, 0x2}, /* default on */
{0x16, AC_VERB_SET_UNSOLICITED_ENABLE, AC_USRSP_EN|CONEXANT_HP_EVENT},
- {0x17, AC_VERB_SET_UNSOLICITED_ENABLE, AC_USRSP_EN|CXT5051_PORTB_EVENT},
{ } /* end */
};
@@ -1841,8 +1826,6 @@ static struct hda_verb cxt5051_lenovo_x200_init_verbs[] = {
/* EAPD */
{0x1a, AC_VERB_SET_EAPD_BTLENABLE, 0x2}, /* default on */
{0x16, AC_VERB_SET_UNSOLICITED_ENABLE, AC_USRSP_EN|CONEXANT_HP_EVENT},
- {0x17, AC_VERB_SET_UNSOLICITED_ENABLE, AC_USRSP_EN|CXT5051_PORTB_EVENT},
- {0x18, AC_VERB_SET_UNSOLICITED_ENABLE, AC_USRSP_EN|CXT5051_PORTC_EVENT},
{0x19, AC_VERB_SET_UNSOLICITED_ENABLE, AC_USRSP_EN|CONEXANT_HP_EVENT},
{ } /* end */
};
@@ -1869,15 +1852,34 @@ static struct hda_verb cxt5051_f700_init_verbs[] = {
/* EAPD */
{0x1a, AC_VERB_SET_EAPD_BTLENABLE, 0x2}, /* default on */
{0x16, AC_VERB_SET_UNSOLICITED_ENABLE, AC_USRSP_EN|CONEXANT_HP_EVENT},
- {0x17, AC_VERB_SET_UNSOLICITED_ENABLE, AC_USRSP_EN|CXT5051_PORTB_EVENT},
{ } /* end */
};
+static void cxt5051_init_mic_port(struct hda_codec *codec, hda_nid_t nid,
+ unsigned int event)
+{
+ snd_hda_codec_write(codec, nid, 0,
+ AC_VERB_SET_UNSOLICITED_ENABLE,
+ AC_USRSP_EN | event);
+#ifdef CONFIG_SND_HDA_INPUT_JACK
+ conexant_add_jack(codec, nid, SND_JACK_MICROPHONE);
+ conexant_report_jack(codec, nid);
+#endif
+}
+
/* initialize jack-sensing, too */
static int cxt5051_init(struct hda_codec *codec)
{
+ struct conexant_spec *spec = codec->spec;
+
conexant_init(codec);
conexant_init_jacks(codec);
+
+ if (spec->auto_mic & AUTO_MIC_PORTB)
+ cxt5051_init_mic_port(codec, 0x17, CXT5051_PORTB_EVENT);
+ if (spec->auto_mic & AUTO_MIC_PORTC)
+ cxt5051_init_mic_port(codec, 0x18, CXT5051_PORTC_EVENT);
+
if (codec->patch_ops.unsol_event) {
cxt5051_hp_automute(codec);
cxt5051_portb_automic(codec);
@@ -1893,6 +1895,7 @@ enum {
CXT5051_HP_DV6736, /* HP without mic switch */
CXT5051_LENOVO_X200, /* Lenovo X200 laptop */
CXT5051_F700, /* HP Compaq Presario F700 */
+ CXT5051_TOSHIBA, /* Toshiba M300 & co */
CXT5051_MODELS
};
@@ -1901,17 +1904,19 @@ static const char *cxt5051_models[CXT5051_MODELS] = {
[CXT5051_HP] = "hp",
[CXT5051_HP_DV6736] = "hp-dv6736",
[CXT5051_LENOVO_X200] = "lenovo-x200",
- [CXT5051_F700] = "hp 700"
+ [CXT5051_F700] = "hp-700",
+ [CXT5051_TOSHIBA] = "toshiba",
};
static struct snd_pci_quirk cxt5051_cfg_tbl[] = {
SND_PCI_QUIRK(0x103c, 0x30cf, "HP DV6736", CXT5051_HP_DV6736),
SND_PCI_QUIRK(0x103c, 0x360b, "Compaq Presario CQ60", CXT5051_HP),
+ SND_PCI_QUIRK(0x103c, 0x30ea, "Compaq Presario F700", CXT5051_F700),
+ SND_PCI_QUIRK(0x1179, 0xff50, "Toshiba M30x", CXT5051_TOSHIBA),
SND_PCI_QUIRK(0x14f1, 0x0101, "Conexant Reference board",
CXT5051_LAPTOP),
SND_PCI_QUIRK(0x14f1, 0x5051, "HP Spartan 1.1", CXT5051_HP),
SND_PCI_QUIRK(0x17aa, 0x20f2, "Lenovo X200", CXT5051_LENOVO_X200),
- SND_PCI_QUIRK(0x103c, 0x30ea, "Compaq Presario F700", CXT5051_F700),
{}
};
@@ -1935,8 +1940,9 @@ static int patch_cxt5051(struct hda_codec *codec)
spec->multiout.dig_out_nid = CXT5051_SPDIF_OUT;
spec->num_adc_nids = 1; /* not 2; via auto-mic switch */
spec->adc_nids = cxt5051_adc_nids;
- spec->num_mixers = 1;
- spec->mixers[0] = cxt5051_mixers;
+ spec->num_mixers = 2;
+ spec->mixers[0] = cxt5051_capture_mixers;
+ spec->mixers[1] = cxt5051_playback_mixers;
spec->num_init_verbs = 1;
spec->init_verbs[0] = cxt5051_init_verbs;
spec->spdif_route = 0;
@@ -1950,6 +1956,7 @@ static int patch_cxt5051(struct hda_codec *codec)
board_config = snd_hda_check_board_config(codec, CXT5051_MODELS,
cxt5051_models,
cxt5051_cfg_tbl);
+ spec->auto_mic = AUTO_MIC_PORTB | AUTO_MIC_PORTC;
switch (board_config) {
case CXT5051_HP:
spec->mixers[0] = cxt5051_hp_mixers;
@@ -1957,7 +1964,7 @@ static int patch_cxt5051(struct hda_codec *codec)
case CXT5051_HP_DV6736:
spec->init_verbs[0] = cxt5051_hp_dv6736_init_verbs;
spec->mixers[0] = cxt5051_hp_dv6736_mixers;
- spec->no_auto_mic = 1;
+ spec->auto_mic = 0;
break;
case CXT5051_LENOVO_X200:
spec->init_verbs[0] = cxt5051_lenovo_x200_init_verbs;
@@ -1965,7 +1972,11 @@ static int patch_cxt5051(struct hda_codec *codec)
case CXT5051_F700:
spec->init_verbs[0] = cxt5051_f700_init_verbs;
spec->mixers[0] = cxt5051_f700_mixers;
- spec->no_auto_mic = 1;
+ spec->auto_mic = 0;
+ break;
+ case CXT5051_TOSHIBA:
+ spec->mixers[0] = cxt5051_toshiba_mixers;
+ spec->auto_mic = AUTO_MIC_PORTB;
break;
}
diff --git a/pci/hda/patch_realtek.c b/pci/hda/patch_realtek.c
index 7e0ea37..d47eee8 100644
--- a/pci/hda/patch_realtek.c
+++ b/pci/hda/patch_realtek.c
@@ -841,27 +841,6 @@ static void add_verb(struct alc_spec *spec, const struct hda_verb *verb)
spec->init_verbs[spec->num_init_verbs++] = verb;
}
-#ifdef CONFIG_PROC_FS
-/*
- * hook for proc
- */
-static void print_realtek_coef(struct snd_info_buffer *buffer,
- struct hda_codec *codec, hda_nid_t nid)
-{
- int coeff;
-
- if (nid != 0x20)
- return;
- coeff = snd_hda_codec_read(codec, nid, 0, AC_VERB_GET_PROC_COEF, 0);
- snd_iprintf(buffer, " Processing Coefficient: 0x%02x\n", coeff);
- coeff = snd_hda_codec_read(codec, nid, 0,
- AC_VERB_GET_COEF_INDEX, 0);
- snd_iprintf(buffer, " Coefficient Index: 0x%02x\n", coeff);
-}
-#else
-#define print_realtek_coef NULL
-#endif
-
/*
* set up from the preset table
*/
@@ -1097,6 +1076,16 @@ static void alc889_coef_init(struct hda_codec *codec)
snd_hda_codec_write(codec, 0x20, 0, AC_VERB_SET_PROC_COEF, tmp|0x2010);
}
+/* turn on/off EAPD control (only if available) */
+static void set_eapd(struct hda_codec *codec, hda_nid_t nid, int on)
+{
+ if (get_wcaps_type(get_wcaps(codec, nid)) != AC_WID_PIN)
+ return;
+ if (snd_hda_query_pin_caps(codec, nid) & AC_PINCAP_EAPD)
+ snd_hda_codec_write(codec, nid, 0, AC_VERB_SET_EAPD_BTLENABLE,
+ on ? 2 : 0);
+}
+
static void alc_auto_init_amp(struct hda_codec *codec, int type)
{
unsigned int tmp;
@@ -1114,25 +1103,22 @@ static void alc_auto_init_amp(struct hda_codec *codec, int type)
case ALC_INIT_DEFAULT:
switch (codec->vendor_id) {
case 0x10ec0260:
- snd_hda_codec_write(codec, 0x0f, 0,
- AC_VERB_SET_EAPD_BTLENABLE, 2);
- snd_hda_codec_write(codec, 0x10, 0,
- AC_VERB_SET_EAPD_BTLENABLE, 2);
+ set_eapd(codec, 0x0f, 1);
+ set_eapd(codec, 0x10, 1);
break;
case 0x10ec0262:
case 0x10ec0267:
case 0x10ec0268:
case 0x10ec0269:
+ case 0x10ec0270:
case 0x10ec0272:
case 0x10ec0660:
case 0x10ec0662:
case 0x10ec0663:
case 0x10ec0862:
case 0x10ec0889:
- snd_hda_codec_write(codec, 0x14, 0,
- AC_VERB_SET_EAPD_BTLENABLE, 2);
- snd_hda_codec_write(codec, 0x15, 0,
- AC_VERB_SET_EAPD_BTLENABLE, 2);
+ set_eapd(codec, 0x14, 1);
+ set_eapd(codec, 0x15, 1);
break;
}
switch (codec->vendor_id) {
@@ -1159,6 +1145,7 @@ static void alc_auto_init_amp(struct hda_codec *codec, int type)
case 0x10ec0888:
alc888_coef_init(codec);
break;
+#if 0 /* XXX: This may cause the silent output on speaker on some machines */
case 0x10ec0267:
case 0x10ec0268:
snd_hda_codec_write(codec, 0x20, 0,
@@ -1171,6 +1158,7 @@ static void alc_auto_init_amp(struct hda_codec *codec, int type)
AC_VERB_SET_PROC_COEF,
tmp | 0x3000);
break;
+#endif /* XXX */
}
break;
}
@@ -5074,7 +5062,6 @@ static int patch_alc880(struct hda_codec *codec)
if (!spec->loopback.amplist)
spec->loopback.amplist = alc880_loopbacks;
#endif
- codec->proc_widget_hook = print_realtek_coef;
return 0;
}
@@ -6684,7 +6671,6 @@ static int patch_alc260(struct hda_codec *codec)
if (!spec->loopback.amplist)
spec->loopback.amplist = alc260_loopbacks;
#endif
- codec->proc_widget_hook = print_realtek_coef;
return 0;
}
@@ -9580,6 +9566,7 @@ static struct alc_config_preset alc882_presets[] = {
.num_channel_mode = ARRAY_SIZE(alc883_3ST_6ch_modes),
.channel_mode = alc883_3ST_6ch_modes,
.need_dac_fix = 1,
+ .const_channel_count = 6,
.num_mux_defs =
ARRAY_SIZE(alc888_2_capture_sources),
.input_mux = alc888_2_capture_sources,
@@ -10302,7 +10289,6 @@ static int patch_alc882(struct hda_codec *codec)
if (!spec->loopback.amplist)
spec->loopback.amplist = alc882_loopbacks;
#endif
- codec->proc_widget_hook = print_realtek_coef;
return 0;
}
@@ -10490,7 +10476,7 @@ static void alc262_hp_t5735_setup(struct hda_codec *codec)
struct alc_spec *spec = codec->spec;
spec->autocfg.hp_pins[0] = 0x15;
- spec->autocfg.speaker_pins[0] = 0x0c; /* HACK: not actually a pin */
+ spec->autocfg.speaker_pins[0] = 0x14;
}
static struct snd_kcontrol_new alc262_hp_t5735_mixer[] = {
@@ -11919,9 +11905,9 @@ static struct alc_config_preset alc262_presets[] = {
.num_channel_mode = ARRAY_SIZE(alc262_modes),
.channel_mode = alc262_modes,
.input_mux = &alc262_capture_source,
- .unsol_event = alc_automute_amp_unsol_event,
+ .unsol_event = alc_sku_unsol_event,
.setup = alc262_hp_t5735_setup,
- .init_hook = alc_automute_amp,
+ .init_hook = alc_inithook,
},
[ALC262_HP_RP5700] = {
.mixers = { alc262_hp_rp5700_mixer },
@@ -12166,7 +12152,6 @@ static int patch_alc262(struct hda_codec *codec)
if (!spec->loopback.amplist)
spec->loopback.amplist = alc262_loopbacks;
#endif
- codec->proc_widget_hook = print_realtek_coef;
return 0;
}
@@ -12675,6 +12660,7 @@ static int alc268_new_analog_output(struct alc_spec *spec, hda_nid_t nid,
dac = 0x02;
break;
case 0x15:
+ case 0x21:
dac = 0x03;
break;
default:
@@ -13232,8 +13218,6 @@ static int patch_alc268(struct hda_codec *codec)
if (board_config == ALC268_AUTO)
spec->init_hook = alc268_auto_init;
- codec->proc_widget_hook = print_realtek_coef;
-
return 0;
}
@@ -13950,7 +13934,6 @@ static int patch_alc269(struct hda_codec *codec)
if (!spec->loopback.amplist)
spec->loopback.amplist = alc269_loopbacks;
#endif
- codec->proc_widget_hook = print_realtek_coef;
return 0;
}
@@ -15078,7 +15061,6 @@ static int patch_alc861(struct hda_codec *codec)
if (!spec->loopback.amplist)
spec->loopback.amplist = alc861_loopbacks;
#endif
- codec->proc_widget_hook = print_realtek_coef;
return 0;
}
@@ -16058,7 +16040,6 @@ static int patch_alc861vd(struct hda_codec *codec)
if (!spec->loopback.amplist)
spec->loopback.amplist = alc861vd_loopbacks;
#endif
- codec->proc_widget_hook = print_realtek_coef;
return 0;
}
@@ -18193,7 +18174,6 @@ static int patch_alc662(struct hda_codec *codec)
if (!spec->loopback.amplist)
spec->loopback.amplist = alc662_loopbacks;
#endif
- codec->proc_widget_hook = print_realtek_coef;
return 0;
}
diff --git a/pcmcia/pdaudiocf/pdaudiocf_pcm.c b/pcmcia/pdaudiocf/pdaudiocf_pcm.c
index 0afa683..0d668f4 100644
--- a/pcmcia/pdaudiocf/pdaudiocf_pcm.c
+++ b/pcmcia/pdaudiocf/pdaudiocf_pcm.c
@@ -277,6 +277,7 @@ static struct snd_pcm_ops pdacf_pcm_capture_ops = {
.trigger = pdacf_pcm_trigger,
.pointer = pdacf_pcm_capture_pointer,
.page = snd_pcm_lib_get_vmalloc_page,
+ .mmap = snd_pcm_lib_mmap_vmalloc,
};
diff --git a/soc/codecs/tlv320dac33.c b/soc/codecs/tlv320dac33.c
index 2df9c20..1b35d0c 100644
--- a/soc/codecs/tlv320dac33.c
+++ b/soc/codecs/tlv320dac33.c
@@ -91,6 +91,7 @@ struct tlv320dac33_priv {
* this */
enum dac33_fifo_modes fifo_mode;/* FIFO mode selection */
unsigned int nsample; /* burst read amount from host */
+ u8 burst_bclkdiv; /* BCLK divider value in burst mode */
enum dac33_state state;
};
@@ -559,7 +560,7 @@ static inline void dac33_prefill_handler(struct tlv320dac33_priv *dac33)
break;
case DAC33_FIFO_MODE7:
dac33_write16(codec, DAC33_PREFILL_MSB,
- DAC33_THRREG(20));
+ DAC33_THRREG(10));
break;
default:
dev_warn(codec->dev, "Unhandled FIFO mode: %d\n",
@@ -845,11 +846,23 @@ static int dac33_prepare_chip(struct snd_pcm_substream *substream)
dac33_write(codec, DAC33_SER_AUDIOIF_CTRL_A, aictrl_a);
dac33_write(codec, DAC33_SER_AUDIOIF_CTRL_B, aictrl_b);
+ /*
+ * BCLK divide ratio
+ * 0: 1.5
+ * 1: 1
+ * 2: 2
+ * ...
+ * 254: 254
+ * 255: 255
+ */
+ if (dac33->fifo_mode)
+ dac33_write(codec, DAC33_SER_AUDIOIF_CTRL_C,
+ dac33->burst_bclkdiv);
+ else
+ dac33_write(codec, DAC33_SER_AUDIOIF_CTRL_C, 32);
+
switch (dac33->fifo_mode) {
case DAC33_FIFO_MODE1:
- /* 20: BCLK divide ratio */
- dac33_write(codec, DAC33_SER_AUDIOIF_CTRL_C, 3);
-
dac33_write16(codec, DAC33_ATHR_MSB,
DAC33_THRREG(dac33->alarm_threshold));
break;
@@ -864,8 +877,6 @@ static int dac33_prepare_chip(struct snd_pcm_substream *substream)
DAC33_THRREG(10));
break;
default:
- /* BYPASS mode */
- dac33_write(codec, DAC33_SER_AUDIOIF_CTRL_C, 32);
break;
}
@@ -1238,6 +1249,7 @@ static int __devinit dac33_i2c_probe(struct i2c_client *client,
i2c_set_clientdata(client, dac33);
dac33->power_gpio = pdata->power_gpio;
+ dac33->burst_bclkdiv = pdata->burst_bclkdiv;
dac33->irq = client->irq;
dac33->nsample = NSAMPLE_MAX;
/* Disable FIFO use by default */
diff --git a/soc/codecs/wm8903.c b/soc/codecs/wm8903.c
index ce5515e..3595bd5 100644
--- a/soc/codecs/wm8903.c
+++ b/soc/codecs/wm8903.c
@@ -1504,7 +1504,7 @@ static int wm8903_resume(struct platform_device *pdev)
struct i2c_client *i2c = codec->control_data;
int i;
u16 *reg_cache = codec->reg_cache;
- u16 *tmp_cache = kmemdup(codec->reg_cache, sizeof(wm8903_reg_defaults),
+ u16 *tmp_cache = kmemdup(reg_cache, sizeof(wm8903_reg_defaults),
GFP_KERNEL);
/* Bring the codec back up to standby first to minimise pop/clicks */
@@ -1516,6 +1516,7 @@ static int wm8903_resume(struct platform_device *pdev)
for (i = 2; i < ARRAY_SIZE(wm8903_reg_defaults); i++)
if (tmp_cache[i] != reg_cache[i])
snd_soc_write(codec, i, tmp_cache[i]);
+ kfree(tmp_cache);
} else {
dev_err(&i2c->dev, "Failed to allocate temporary cache\n");
}
diff --git a/soc/codecs/wm_hubs.c b/soc/codecs/wm_hubs.c
index d73c305..a67319d 100644
--- a/soc/codecs/wm_hubs.c
+++ b/soc/codecs/wm_hubs.c
@@ -753,6 +753,12 @@ int wm_hubs_handle_analogue_pdata(struct snd_soc_codec *codec,
WM8993_LINEOUT2_MODE,
WM8993_LINEOUT2_MODE);
+ /* If the line outputs are differential then we aren't presenting
+ * VMID as an output and can disable it.
+ */
+ if (lineout1_diff && lineout2_diff)
+ codec->idle_bias_off = 1;
+
if (lineout1fb)
snd_soc_update_bits(codec, WM8993_ADDITIONAL_CONTROL,
WM8993_LINEOUT1_FB, WM8993_LINEOUT1_FB);
diff --git a/soc/sh/Kconfig b/soc/sh/Kconfig
index 8072a6d..3f1cd55 100644
--- a/soc/sh/Kconfig
+++ b/soc/sh/Kconfig
@@ -26,6 +26,12 @@ config SND_SOC_SH4_FSI
help
This option enables FSI sound support
+config SND_SOC_SH4_SIU
+ tristate
+ depends on (SUPERH || ARCH_SHMOBILE) && HAVE_CLK
+ select DMADEVICES
+ select SH_DMAE
+
##
## Boards
##
diff --git a/soc/sh/Makefile b/soc/sh/Makefile
index 1d0ec0a..5a97d25 100644
--- a/soc/sh/Makefile
+++ b/soc/sh/Makefile
@@ -6,9 +6,11 @@ obj-$(CONFIG_SND_SOC_PCM_SH7760) += snd-soc-dma-sh7760.o
snd-soc-hac-objs := hac.o
snd-soc-ssi-objs := ssi.o
snd-soc-fsi-objs := fsi.o
+snd-soc-siu-objs := siu_pcm.o siu_dai.o
obj-$(CONFIG_SND_SOC_SH4_HAC) += snd-soc-hac.o
obj-$(CONFIG_SND_SOC_SH4_SSI) += snd-soc-ssi.o
obj-$(CONFIG_SND_SOC_SH4_FSI) += snd-soc-fsi.o
+obj-$(CONFIG_SND_SOC_SH4_SIU) += snd-soc-siu.o
## boards
snd-soc-sh7760-ac97-objs := sh7760-ac97.o
diff --git a/soc/sh/siu.h b/soc/sh/siu.h
new file mode 100644
index 0000000..9cc04ab
--- /dev/null
+++ b/soc/sh/siu.h
@@ -0,0 +1,193 @@
+/*
+ * siu.h - ALSA SoC driver for Renesas SH7343, SH7722 SIU peripheral.
+ *
+ * Copyright (C) 2009-2010 Guennadi Liakhovetski <g.l...@gm...>
+ * Copyright (C) 2006 Carlos Munoz <ca...@ke...>
+ *
+ * This program is free software; you can redistribute it and/or modify
+ * it under the terms of the GNU General Public License as published by
+ * the Free Software Foundation; either version 2 of the License, or
+ * (at your option) any later version.
+ *
+ * This program is distributed in the hope that it will be useful,
+ * but WITHOUT ANY WARRANTY; without even the implied warranty of
+ * MERCHANTABILITY or FITNESS FOR A PARTICULAR PURPOSE. See the
+ * GNU General Public License for more details.
+ *
+ * You should have received a copy of the GNU General Public License
+ * along with this program; if not, write to the Free Software
+ * Foundation, Inc., 675 Mass Ave, Cambridge, MA 02139, USA.
+ */
+
+#ifndef SIU_H
+#define SIU_H
+
+/* Common kernel and user-space firmware-building defines and types */
+
+#define YRAM0_SIZE (0x0040 / 4) /* 16 */
+#define YRAM1_SIZE (0x0080 / 4) /* 32 */
+#define YRAM2_SIZE (0x0040 / 4) /* 16 */
+#define YRAM3_SIZE (0x0080 / 4) /* 32 */
+#define YRAM4_SIZE (0x0080 / 4) /* 32 */
+#define YRAM_DEF_SIZE (YRAM0_SIZE + YRAM1_SIZE + YRAM2_SIZE + \
+ YRAM3_SIZE + YRAM4_SIZE)
+#define YRAM_FIR_SIZE (0x0400 / 4) /* 256 */
+#define YRAM_IIR_SIZE (0x0200 / 4) /* 128 */
+
+#define XRAM0_SIZE (0x0400 / 4) /* 256 */
+#define XRAM1_SIZE (0x0200 / 4) /* 128 */
+#define XRAM2_SIZE (0x0200 / 4) /* 128 */
+
+/* PRAM program array size */
+#define PRAM0_SIZE (0x0100 / 4) /* 64 */
+#define PRAM1_SIZE ((0x2000 - 0x0100) / 4) /* 1984 */
+
+#include <linux/types.h>
+
+struct siu_spb_param {
+ __u32 ab1a; /* input FIFO address */
+ __u32 ab0a; /* output FIFO address */
+ __u32 dir; /* 0=the ather except CPUOUTPUT, 1=CPUINPUT */
+ __u32 event; /* SPB program starting conditions */
+ __u32 stfifo; /* STFIFO register setting value */
+ __u32 trdat; /* TRDAT register setting value */
+};
+
+struct siu_firmware {
+ __u32 yram_fir_coeff[YRAM_FIR_SIZE];
+ __u32 pram0[PRAM0_SIZE];
+ __u32 pram1[PRAM1_SIZE];
+ __u32 yram0[YRAM0_SIZE];
+ __u32 yram1[YRAM1_SIZE];
+ __u32 yram2[YRAM2_SIZE];
+ __u32 yram3[YRAM3_SIZE];
+ __u32 yram4[YRAM4_SIZE];
+ __u32 spbpar_num;
+ struct siu_spb_param spbpar[32];
+};
+
+#ifdef __KERNEL__
+
+#include <linux/dmaengine.h>
+#include <linux/interrupt.h>
+#include <linux/io.h>
+
+#include <asm/dma-sh.h>
+
+#include <sound/core.h>
+#include <sound/pcm.h>
+#include <sound/soc-dai.h>
+
+#define SIU_PERIOD_BYTES_MAX 8192 /* DMA transfer/period size */
+#define SIU_PERIOD_BYTES_MIN 256 /* DMA transfer/period size */
+#define SIU_PERIODS_MAX 64 /* Max periods in buffer */
+#define SIU_PERIODS_MIN 4 /* Min periods in buffer */
+#define SIU_BUFFER_BYTES_MAX (SIU_PERIOD_BYTES_MAX * SIU_PERIODS_MAX)
+
+/* SIU ports: only one can be used at a time */
+enum {
+ SIU_PORT_A,
+ SIU_PORT_B,
+ SIU_PORT_NUM,
+};
+
+/* SIU clock configuration */
+enum {
+ SIU_CLKA_PLL,
+ SIU_CLKA_EXT,
+ SIU_CLKB_PLL,
+ SIU_CLKB_EXT
+};
+
+struct siu_info {
+ int port_id;
+ u32 __iomem *pram;
+ u32 __iomem *xram;
+ u32 __iomem *yram;
+ u32 __iomem *reg;
+ struct siu_firmware fw;
+};
+
+struct siu_stream {
+ struct tasklet_struct tasklet;
+ struct snd_pcm_substream *substream;
+ snd_pcm_format_t format;
+ size_t buf_bytes;
+ size_t period_bytes;
+ int cur_period; /* Period currently in dma */
+ u32 volume;
+ snd_pcm_sframes_t xfer_cnt; /* Number of frames */
+ u8 rw_flg; /* transfer status */
+ /* DMA status */
+ struct dma_chan *chan; /* DMA channel */
+ struct dma_async_tx_descriptor *tx_desc;
+ dma_cookie_t cookie;
+ struct sh_dmae_slave param;
+};
+
+struct siu_port {
+ unsigned long play_cap; /* Used to track full duplex */
+ struct snd_pcm *pcm;
+ struct siu_stream playback;
+ struct siu_stream capture;
+ u32 stfifo; /* STFIFO value from firmware */
+ u32 trdat; /* TRDAT value from firmware */
+};
+
+extern struct siu_port *siu_ports[SIU_PORT_NUM];
+
+static inline struct siu_port *siu_port_info(struct snd_pcm_substream *substream)
+{
+ struct platform_device *pdev =
+ to_platform_device(substream->pcm->card->dev);
+ return siu_ports[pdev->id];
+}
+
+/* Register access */
+static inline void siu_write32(u32 __iomem *addr, u32 val)
+{
+ __raw_writel(val, addr);
+}
+
+static inline u32 siu_read32(u32 __iomem *addr)
+{
+ return __raw_readl(addr);
+}
+
+/* SIU registers */
+#define SIU_IFCTL (0x000 / sizeof(u32))
+#define SIU_SRCTL (0x004 / sizeof(u32))
+#define SIU_SFORM (0x008 / sizeof(u32))
+#define SIU_CKCTL (0x00c / sizeof(u32))
+#define SIU_TRDAT (0x010 / sizeof(u32))
+#define SIU_STFIFO (0x014 / sizeof(u32))
+#define SIU_DPAK (0x01c / sizeof(u32))
+#define SIU_CKREV (0x020 / sizeof(u32))
+#define SIU_EVNTC (0x028 / sizeof(u32))
+#define SIU_SBCTL (0x040 / sizeof(u32))
+#define SIU_SBPSET (0x044 / sizeof(u32))
+#define SIU_SBFSTS (0x068 / sizeof(u32))
+#define SIU_SBDVCA (0x06c / sizeof(u32))
+#define SIU_SBDVCB (0x070 / sizeof(u32))
+#define SIU_SBACTIV (0x074 / sizeof(u32))
+#define SIU_DMAIA (0x090 / sizeof(u32))
+#define SIU_DMAIB (0x094 / sizeof(u32))
+#define SIU_DMAOA (0x098 / sizeof(u32))
+#define SIU_DMAOB (0x09c / sizeof(u32))
+#define SIU_DMAML (0x0a0 / sizeof(u32))
+#define SIU_SPSTS (0x0cc / sizeof(u32))
+#define SIU_SPCTL (0x0d0 / sizeof(u32))
+#define SIU_BRGASEL (0x100 / sizeof(u32))
+#define SIU_BRRA (0x104 / sizeof(u32))
+#define SIU_BRGBSEL (0x108 / sizeof(u32))
+#define SIU_BRRB (0x10c / sizeof(u32))
+
+extern struct snd_soc_platform siu_platform;
+extern struct snd_soc_dai siu_i2s_dai;
+
+int siu_init_port(int port, struct siu_port **port_info, struct snd_card *card);
+void siu_free_port(struct siu_port *port_info);
+
+#endif
+
+#endif /* SIU_H */
diff --git a/soc/sh/siu_dai.c b/soc/sh/siu_dai.c
new file mode 100644
index 0000000..5452d19
--- /dev/null
+++ b/soc/sh/siu_dai.c
@@ -0,0 +1,847 @@
+/*
+ * siu_dai.c - ALSA SoC driver for Renesas SH7343, SH7722 SIU peripheral.
+ *
+ * Copyright (C) 2009-2010 G...
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