ASoC updates

The following changes since commit 32c8dabc97d436582298ebd0e33af041c69f5a4b:
Mark Brown (1):
ASoC: Remove obsolete declaration of struct snd_soc_clock_info

are available in the git repository at:

git://opensource.wolfsonmicro.com/linux-2.6-asoc for-tiwai

Daniel Mack (1):
ASoC: tlv320aic3x: control additions and cleanups

Grazvydas Ignotas (1):
ASoC: Add support for OMAP3 Pandora

 sound/soc/codecs/tlv320aic3x.c |   18 ++-
 sound/soc/omap/Kconfig         |    8 +
 sound/soc/omap/Makefile        |    2 +
 sound/soc/omap/omap3pandora.c  |  311 ++++++++++++++++++++++++++++++++++++++++
 4 files changed, 334 insertions(+), 5 deletions(-)
 create mode 100644 sound/soc/omap/omap3pandora.c

From: Grazvydas Ignotas <[email protected]>

This patch adds basic support for OMAP3 Pandora.

---
 sound/soc/omap/Kconfig        |    8 +
 sound/soc/omap/Makefile       |    2 +
 sound/soc/omap/omap3pandora.c |  311 +++++++++++++++++++++++++++++++++++++++++
 3 files changed, 321 insertions(+), 0 deletions(-)
 create mode 100644 sound/soc/omap/omap3pandora.c

index da39f27..a7b1d77 100644
@@ -46,3 +46,11 @@ config SND_OMAP_SOC_SDP3430
 	help
 	  Say Y if you want to add support for SoC audio on Texas Instruments
 	  SDP3430.
+
+config SND_OMAP_SOC_OMAP3_PANDORA
+	tristate "SoC Audio support for OMAP3 Pandora"
+	depends on TWL4030_CORE && SND_OMAP_SOC && MACH_OMAP3_PANDORA
+	select SND_OMAP_SOC_MCBSP
+	select SND_SOC_TWL4030
+	help
+	  Say Y if you want to add support for SoC audio on the OMAP3 Pandora.

index 29cf3a8..76fedd9 100644
@@ -11,9 +11,11 @@ snd-soc-osk5912-objs := osk5912.o
 snd-soc-overo-objs := overo.o
 snd-soc-omap2evm-objs := omap2evm.o
 snd-soc-sdp3430-objs := sdp3430.o
+snd-soc-omap3pandora-objs := omap3pandora.o
 
 obj-$(CONFIG_SND_OMAP_SOC_N810) += snd-soc-n810.o
 obj-$(CONFIG_SND_OMAP_SOC_OSK5912) += snd-soc-osk5912.o
 obj-$(CONFIG_SND_OMAP_SOC_OVERO) += snd-soc-overo.o
 obj-$(CONFIG_MACH_OMAP2EVM) += snd-soc-omap2evm.o
 obj-$(CONFIG_SND_OMAP_SOC_SDP3430) += snd-soc-sdp3430.o
+obj-$(CONFIG_SND_OMAP_SOC_OMAP3_PANDORA) += snd-soc-omap3pandora.o

1.5.6.5

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From: Daniel Mack <[email protected]>

- split "Line Playback Switch" into "LineL Playback Switch" and "LineR Playback Switch"
- split "Line PGA Bypass Playback Volume" into "LineL Left PGA Bypass Playback Volume" and "LineR Right PGA Bypass Playback Volume"
- split "Line Line2 Bypass Playback Volume" into "LineL Line2 Bypass Playback Volume" and "LineR Line2 Bypass Playback Volume"
- Added "HP Right PGA Bypass Playback Volume"

---
 sound/soc/codecs/tlv320aic3x.c |   18 +++++++++++++-----
 1 files changed, 13 insertions(+), 5 deletions(-)

index 341e1ad..6a05829 100644
@@ -253,11 +253,17 @@ static const struct snd_kcontrol_new aic3x_snd_controls[] = {
 
 	SOC_DOUBLE_R("Line DAC Playback Volume", DACL1_2_LLOPM_VOL,
 		     DACR1_2_RLOPM_VOL, 0, 0x7f, 1),
-	SOC_DOUBLE_R("Line DAC Playback Switch", LLOPM_CTRL, RLOPM_CTRL, 3,
-		     0x01, 0),
-	SOC_DOUBLE_R("Line PGA Bypass Playback Volume", PGAL_2_LLOPM_VOL,
-		     PGAR_2_RLOPM_VOL, 0, 0x7f, 1),
-	SOC_DOUBLE_R("Line Line2 Bypass Playback Volume", LINE2L_2_LLOPM_VOL,
+	SOC_SINGLE("LineL Playback Switch", LLOPM_CTRL, 3, 0x01, 0),
+	SOC_SINGLE("LineR Playback Switch", RLOPM_CTRL, 3, 0x01, 0),
+	SOC_DOUBLE_R("LineL DAC Playback Volume", DACL1_2_LLOPM_VOL,
+		     DACR1_2_LLOPM_VOL, 0, 0x7f, 1),
+	SOC_SINGLE("LineL Left PGA Bypass Playback Volume", PGAL_2_LLOPM_VOL,
+		     0, 0x7f, 1),
+	SOC_SINGLE("LineR Right PGA Bypass Playback Volume", PGAR_2_RLOPM_VOL,
+		     0, 0x7f, 1),
+	SOC_DOUBLE_R("LineL Line2 Bypass Playback Volume", LINE2L_2_LLOPM_VOL,
+		     LINE2R_2_LLOPM_VOL, 0, 0x7f, 1),
+	SOC_DOUBLE_R("LineR Line2 Bypass Playback Volume", LINE2L_2_RLOPM_VOL,
 		     LINE2R_2_RLOPM_VOL, 0, 0x7f, 1),
 
 	SOC_DOUBLE_R("Mono DAC Playback Volume", DACL1_2_MONOLOPM_VOL,

@@ -272,6 +278,8 @@ static const struct snd_kcontrol_new aic3x_snd_controls[] = {
 		     DACR1_2_HPROUT_VOL, 0, 0x7f, 1),
 	SOC_DOUBLE_R("HP DAC Playback Switch", HPLOUT_CTRL, HPROUT_CTRL, 3,
 		     0x01, 0),
+	SOC_DOUBLE_R("HP Right PGA Bypass Playback Volume", PGAR_2_HPLOUT_VOL,
+		     PGAR_2_HPROUT_VOL, 0, 0x7f, 1),
 	SOC_SINGLE("HPL PGA Bypass Playback Volume", PGAL_2_HPLOUT_VOL,
 		     0, 0x7f, 1),
 	SOC_SINGLE("HPR PGA Bypass Playback Volume", PGAL_2_HPROUT_VOL,

1.5.6.5

Hi,

If my sound card supports hardware mixing, how do I need to code my sound card driver such that I can send multiple streams through ALSA to the sound card driver at the same time?

Thanks,
Harsha

It does that by default. Just run multiple applications.

2008/12/5 Harsha, Priya <[email protected]>:

Vedran Miletić
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Thanks.

If I need to send a voice stream and an audio stream to the hardware as two separate streams... I just need to open 2 ALSA device from user space and send it right?

If I want to send some private data for each stream (say I want to differentiate that it's a voice/music...for which I did not find any provision) how do I do that?

-----Original Message-----
From: Vedran Miletić [mailto:[email protected]]
Sent: Friday, December 05, 2008 10:42 PM
To: Harsha, Priya
Cc: [email protected]
Subject: Re: [alsa-devel] question on hardware mixing

It does that by default. Just run multiple applications.

2008/12/5 Harsha, Priya <[email protected]>:

Vedran Miletić

Check out the emu10k1 driver. You need something like the voice allocator in that driver that assigns a hardware mixing channel or channels when ALSA asks the driver for a new PCM stream.

User space would open the same device multiple times (up to the hardware voice limit) and the driver manages the allocations. From userspace POV a hardware mixing device looks just like a single stream device, but it doesn't return -EBUSY when you open it multiple times.

Lee

Hi,

Thanks.

A quick follow up question. In the trigger function - I see that from substream->runtime->private_data, voice specific parameters are extracted. How and can somehow user space fill in some private data to the driver so that driver can process voice in a different way and music in a different way?

Thanks,
Harsha

-----Original Message-----
From: [email protected] [mailto:[email protected]] On Behalf Of Lee Revell
Sent: Saturday, December 06, 2008 10:29 AM
To: Harsha, Priya
Cc: [email protected]
Subject: Re: [alsa-devel] question on hardware mixing

Check out the emu10k1 driver. You need something like the voice allocator in that driver that assigns a hardware mixing channel or channels when ALSA asks the driver for a new PCM stream.

User space would open the same device multiple times (up to the hardware voice limit) and the driver manages the allocations. From userspace POV a hardware mixing device looks just like a single stream device, but it doesn't return -EBUSY when you open it multiple times.

Lee

What exactly do you want to do differently?

Lee

I have a DSP engine with FW. I need to pass this information to the FW which does some processing for voice and music differently.

Thanks,
Harsha

-----Original Message-----
From: [email protected] [mailto:[email protected]] On Behalf Of Lee Revell
Sent: Tuesday, December 09, 2008 8:46 AM
To: Harsha, Priya
Cc: [email protected]
Subject: Re: [alsa-devel] question on hardware mixing

What exactly do you want to do differently?

Lee

I was thinking to pass this information from snd_pcm_t->private_data. This gets copied to substream->private_data internally which would reach the driver. Am I right here?

-----Original Message-----
From: Harsha, Priya
Sent: Tuesday, December 09, 2008 8:54 AM
To: 'Lee Revell'
Cc: [email protected]
Subject: RE: [alsa-devel] question on hardware mixing

I have a DSP engine with FW. I need to pass this information to the FW which does some processing for voice and music differently.

Thanks,
Harsha

-----Original Message-----
From: [email protected] [mailto:[email protected]] On Behalf Of Lee Revell
Sent: Tuesday, December 09, 2008 8:46 AM
To: Harsha, Priya
Cc: [email protected]
Subject: Re: [alsa-devel] question on hardware mixing

What exactly do you want to do differently?

Lee

On the Emu10k1 chip, the hardware voices are used for both PCM playback and wavetable (MIDI) playback. Voice-specific parameters are set for MIDI data; the parameters come from the soundfont that was loaded to specify how the instruments sound. (Here, the term "voice" has no relation to the sound of a human voice.)

As far as PCM playback is concerned, ALSA does not have any predefined mechanism to differentiate between voice and music audio streams. It would be possible to define mixer controls to switch between these settings for each substream.

HTH
Clemens

If I define mixer controls to switch between each substream, I would not be able to handle multiple streams at time.... I had asked this in my earlier follow up email as well - can I use snd_pcm_t->private_data to communicate if it's a voice or audio I understand this private_data gets copied to substream->private data which will reach the driver...
Am I right here?

-----Original Message-----
From: Clemens Ladisch [mailto:[email protected]]
Sent: Tuesday, December 09, 2008 2:23 PM
To: Harsha, Priya
Cc: Lee Revell; [email protected]
Subject: Re: [alsa-devel] question on hardware mixing

On the Emu10k1 chip, the hardware voices are used for both PCM playback and wavetable (MIDI) playback. Voice-specific parameters are set for MIDI data; the parameters come from the soundfont that was loaded to specify how the instruments sound. (Here, the term "voice" has no relation to the sound of a human voice.)

As far as PCM playback is concerned, ALSA does not have any predefined mechanism to differentiate between voice and music audio streams. It would be possible to define mixer controls to switch between these settings for each substream.

HTH
Clemens

Check into the ALSA configuration file syntax -- there are number of things you can do there, like define a 'voice' device that uses hooks to set a control setting (snd_kcontrol) when you open a PCM. You can use that control setting to let your driver know if the PCM channel is being used for voice or audio traffic.

Look at how the EMU10K1 handles surround sound audio in /etc/alsa/cards/EMU10K1.conf

I started to do some similar things for sis7019 surround sound, but they've never made it out of my office due to lack of time/testing HW.

I can't answer this specifically, but I believe the answer is in the name -- 'private'. This is for the ALSA system to use, you can hang your own data off a substream using snd_pcm_substream_chip(). Besides, how do you plan on setting it in the snd_pcm_t type?

I think that if you set up some controls, use the config file language, and have your audio apps open a 'voice' device, you can get where you want.

Dave

Hi ,

I need more help here. I tried looking at the .conf file but I have another requirement here.

Let me put my requirements here. Please suggest me ways to get this done

1. When I get a .prepare call in my driver, I need to know the stream type
2. When I return from that call, I need to send to the user the locally allocated stream id

This is specific to the sound card/ DSP I am developing the driver for. The application is also proprietary. Is this possible at all? Please suggest me ways to do the same.

Thanks,
Harsha

The issue that I am facing with creating a snd_kcontrol is that I am not able to bind that value to a stream. Say I start the application and set the type of that control to voice and open another instance of that application and set the type to audio. There might be a clash and I might get the wrong stream type to the driver. Also, the stream ids passed cannot also be guaranteed to reach the right stream.

At Thu, 9 Apr 2009 13:26:50 +0530,

Why?

Another instance must have a different stream, right? Then it must have a different substream number. In the scenario a la emu10k1, you'll have a control per each substream. So this won't conflict.

The "stream id" you are referring to might be different from what I'm thinking of. Please elaborate your terminology before further questions. It's a bit hard to illustrate your use-case from your descriptions, so far...

thanks,

Takashi

I wrote "mixer controls to switch between these settings for each substream", i.e., each substream would have its own flag.

ALSA does not have a predefined mechanism to set it from userspace. Your driver can initialize it when it is probed, or it can change it when opening the substream.

HTH
Clemens

-----Original Message-----
From: [email protected] [mailto:[email protected]] On Behalf Of Clemens Ladisch
Sent: Tuesday, December 09, 2008 7:36 PM
To: Harsha, Priya
Cc: [email protected]; Lee Revell
Subject: Re: [alsa-devel] question on hardware mixing

I wrote "mixer controls to switch between these settings for each
substream", i.e., each substream would have its own flag.
[Harsha] I am still not comfortable how to do this... I guess EMU10K1 handle this kind of mixer controls for switching... I shall look into this driver to learn more

ALSA does not have a predefined mechanism to set it from userspace.
Your driver can initialize it when it is probed, or it can change it
when opening the substream.
[Harsha] Thanks.... so can I assume that when I open a substream, I can set a private_data to communicate to the driver if its voice/music?

HTH
Clemens
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If I create a mixer control using snd_kcontrol. When two applications in parallel try to create a stream and change the snd_kcontrol, can you help me understand how I would get the stream type stored in each substream as a flag?

Also, when I get a snd_kcontrol's .set function call, I am not sure how to get the substream specific data in that call?

Thanks,
Harsha

At Fri, 5 Dec 2008 16:53:42 +0000,

Thanks, pulled now.

Takashi