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Axel Lin | 1 Feb 02:56
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Re: [PATCH 1/2] sound: soc: add alias for tlv320aic3x

2012/2/1 Felipe Contreras <felipe.contreras <at> gmail.com>:
> According to the documentation, should help hotplugging.
>
> Signed-off-by: Felipe Contreras <felipe.contreras <at> gmail.com>
> ---
>  sound/soc/codecs/tlv320aic3x.c |    1 +
>  1 files changed, 1 insertions(+), 0 deletions(-)
>
> diff --git a/sound/soc/codecs/tlv320aic3x.c b/sound/soc/codecs/tlv320aic3x.c
> index 492f22f..00eb29d 100644
> --- a/sound/soc/codecs/tlv320aic3x.c
> +++ b/sound/soc/codecs/tlv320aic3x.c
> @@ -1565,3 +1565,4 @@ module_exit(aic3x_exit);
>  MODULE_DESCRIPTION("ASoC TLV320AIC3X codec driver");
>  MODULE_AUTHOR("Vladimir Barinov");
>  MODULE_LICENSE("GPL");
> +MODULE_ALIAS("i2c:tlv320aic3x");
This is not required because "tlv320aic3x" is already in the aic3x_i2c_id table.
MODULE_DEVICE_TABLE will automatically setup this modalias.

> --
> 1.7.9
>
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David Henningsson | 1 Feb 09:19
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Re: [PATCH] ALSA: HDA: Fix jack creation for codecs with front and rear Line In

On 01/31/2012 04:40 PM, Takashi Iwai wrote:
> At Tue, 31 Jan 2012 16:28:14 +0100,
> David Henningsson wrote:
>>
>> On 01/31/2012 03:17 PM, Takashi Iwai wrote:
>>> At Tue, 31 Jan 2012 09:04:15 +0100,
>>> David Henningsson wrote:
>>>>
>>>> If a codec has both a front and a rear Line In, two controls both
>>>> named "Line Jack" will be created, which causes parsing to fail.
>>>> While a long term solution might be to name the jacks differently,
>>>> this extra check is consistent with what is already being done in many
>>>> auto-parsers, and will also protect against other cases when two
>>>> inputs have the same label.
>>>>
>>>> BugLink: https://bugs.launchpad.net/bugs/923409
>>>> Signed-off-by: David Henningsson<david.henningsson <at> canonical.com>
>>>
>>> Thanks, applied now.
>>>
>>> BTW, any chance to change apport to gather alsa-info.sh output?

This has now been provided: 
http://www.alsa-project.org/db/?f=f4a8312f34573a24f60011109e18f0c7371129e0

>>> It'd be a great help, so that we can check it via hda-emu.
>>
>> I have asked for this particular bug reporter to supply alsa-info as
>> well, but may I ask you...do you have some kind of regression test
>> script, and if so, are they public? Or could it a part of hda-emu?
(Continue reading)

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Takashi Iwai | 1 Feb 09:39
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Re: [PATCH] ALSA: HDA: Fix jack creation for codecs with front and rear Line In

At Wed, 01 Feb 2012 09:19:25 +0100,
David Henningsson wrote:
> 
> On 01/31/2012 04:40 PM, Takashi Iwai wrote:
> > At Tue, 31 Jan 2012 16:28:14 +0100,
> > David Henningsson wrote:
> >>
> >> On 01/31/2012 03:17 PM, Takashi Iwai wrote:
> >>> At Tue, 31 Jan 2012 09:04:15 +0100,
> >>> David Henningsson wrote:
> >>>>
> >>>> If a codec has both a front and a rear Line In, two controls both
> >>>> named "Line Jack" will be created, which causes parsing to fail.
> >>>> While a long term solution might be to name the jacks differently,
> >>>> this extra check is consistent with what is already being done in many
> >>>> auto-parsers, and will also protect against other cases when two
> >>>> inputs have the same label.
> >>>>
> >>>> BugLink: https://bugs.launchpad.net/bugs/923409
> >>>> Signed-off-by: David Henningsson<david.henningsson <at> canonical.com>
> >>>
> >>> Thanks, applied now.
> >>>
> >>> BTW, any chance to change apport to gather alsa-info.sh output?
> 
> This has now been provided: 
> http://www.alsa-project.org/db/?f=f4a8312f34573a24f60011109e18f0c7371129e0
> 
> >>> It'd be a great help, so that we can check it via hda-emu.
> >>
(Continue reading)

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Takashi Iwai | 1 Feb 09:45
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Re: [PATCH] ALSA: hda - Fix calling cs_automic twice for Cirrus codecs.

At Tue, 31 Jan 2012 13:04:41 -0800,
Dylan Reid wrote:
> 
> If cs_automic is called twice (like it is during init) while the mic
> is present, it will over-write the last_input with the new one,
> causing it to switch back to the automic input when the mic is
> unplugged. This leaves the driver in a state (cur_input, last_input,
> and automix_idx the same) where the internal mic can not be selected
> until it is rebooted without the mic attached.
> 
> Check that the mic hasn't already been switched to before setting
> last_input.
> 
> Signed-off-by: Dylan Reid <dgreid <at> chromium.org>

Thanks, applied now.

Takashi

> ---
>  sound/pci/hda/patch_cirrus.c |    6 ++++--
>  1 files changed, 4 insertions(+), 2 deletions(-)
> 
> diff --git a/sound/pci/hda/patch_cirrus.c b/sound/pci/hda/patch_cirrus.c
> index 0e99357..bc5a993 100644
> --- a/sound/pci/hda/patch_cirrus.c
> +++ b/sound/pci/hda/patch_cirrus.c
> @@ -988,8 +988,10 @@ static void cs_automic(struct hda_codec *codec)
>  			change_cur_input(codec, !spec->automic_idx, 0);
>  	} else {
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Vinod Koul | 1 Feb 10:07
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Re: soc-dsp programming model for loopbacks

On Wed, 2012-01-25 at 17:07 +0000, Liam Girdwood wrote:
> We could eventually remove steps 2 and 4 for the FE DAI link, and look
> at hard coding the hw_params() in the mach driver for the loopback
> link.
> That way the only user space driven actions would be to configure the
> mixers in the CODEC and DSP for the correct route. DAPM would then
> detect the path and Dynamic PCM would use the hard coded configuration
> or bespoke mach driver logic to configure the loopback DAI link based
> on
> use case. This would have to be done after the basic Dynamic PCM
> infrastructure was upstream though (unless you have a patch atm). 
Hi,
Sorry to join the party late :-)

Liam, if we configure the hw_params in the machine driver statically,
represent the DSP using a map along with a codec kind of modeling.

Would the BEs be triggered on from soc_dsp_runtime_update() when the
loopback is established thru the DSP.
This way we avoid all the "virtual" FEs. Use loopback to turn on codec
and DSP (thru BEs)

Would this make sense, or I need more coffee :)

--

-- 
~Vinod
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Mark Brown | 1 Feb 11:05
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[PATCH] ASoC: wm8962: Fix word length configuration

From: Susan Gao <sgao <at> opensource.wolfsonmicro.com>

Signed-off-by: Susan Gao <sgao <at> opensource.wolfsonmicro.com>
Signed-off-by: Mark Brown <broonie <at> opensource.wolfsonmicro.com>
---
 sound/soc/codecs/wm8962.c |    6 +++---
 1 files changed, 3 insertions(+), 3 deletions(-)

diff --git a/sound/soc/codecs/wm8962.c b/sound/soc/codecs/wm8962.c
index de74af7..c713d55 100644
--- a/sound/soc/codecs/wm8962.c
+++ b/sound/soc/codecs/wm8962.c
@@ -2594,13 +2594,13 @@ static int wm8962_hw_params(struct snd_pcm_substream *substream,
 	case SNDRV_PCM_FORMAT_S16_LE:
 		break;
 	case SNDRV_PCM_FORMAT_S20_3LE:
-		aif0 |= 0x40;
+		aif0 |= 0x4;
 		break;
 	case SNDRV_PCM_FORMAT_S24_LE:
-		aif0 |= 0x80;
+		aif0 |= 0x8;
 		break;
 	case SNDRV_PCM_FORMAT_S32_LE:
-		aif0 |= 0xc0;
+		aif0 |= 0xc;
 		break;
 	default:
 		return -EINVAL;
--

--
(Continue reading)

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Mark Brown | 1 Feb 11:49
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Re: [RFC][PATCH] ASoC: cs42l73: Fix Output [X|A|V]SP_SCLK Sourcing Mode setting for master mode

On Tue, Jan 31, 2012 at 09:49:04AM +0800, Axel Lin wrote:
> For master mode, set Output [X|A|V]SP_SCLK Sourcing Mode to MCLK Mode.

Applied, thanks.
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David Henningsson | 1 Feb 12:05
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[PATCH] ALSA: HDA: Fix duplicated output to more than one codec

This typo caused the wrong codec's nid to be checked for wcaps type.
As a result, sometimes speakers would duplicate the output sent to
HDMI output.

Cc: stable <at> kernel.org
BugLink: https://bugs.launchpad.net/bugs/924320
Signed-off-by: David Henningsson <david.henningsson <at> canonical.com>
---
 sound/pci/hda/hda_codec.c |    2 +-
 1 files changed, 1 insertions(+), 1 deletions(-)

diff --git a/sound/pci/hda/hda_codec.c b/sound/pci/hda/hda_codec.c
index 4df72c0..c2c65f6 100644
--- a/sound/pci/hda/hda_codec.c
+++ b/sound/pci/hda/hda_codec.c
@@ -1447,7 +1447,7 @@ void snd_hda_codec_setup_stream(struct hda_codec *codec, hda_nid_t nid,
 		for (i = 0; i < c->cvt_setups.used; i++) {
 			p = snd_array_elem(&c->cvt_setups, i);
 			if (!p->active && p->stream_tag == stream_tag &&
-			    get_wcaps_type(get_wcaps(codec, p->nid)) == type)
+			    get_wcaps_type(get_wcaps(c, p->nid)) == type)
 				p->dirty = 1;
 		}
 	}
--

-- 
1.7.8.3
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Ashish Chavan | 1 Feb 12:25
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[PATCH v3] ASoC: da7210: Add support for PLL and SRM

Current DA7210 driver does support PLL mode fully. It uses fixed
value of input master clock and PLL mode is enabled and disabled based
on the sampling frequency being used for playback or recording. It also
doesn't support Sample Rate Measurement feature of DA7210 hardware.

This patch adds full support for PLL and SRM. Basically following three
modes of operation are possible for DA7210 hardware,

(1) I2S SLAVE mode with PLL bypassed
(2) I2S SLAVE mode with PLL enabled
(3) I2S Master mode with PLL enabled

This patch adds support for all three modes. Also, in case of SLAVE mode
with PLL, it supports SRM (Sample Rate Measurement) feature of the chip.

This patch has been tested on DA7210 EVB with Samsung SMDK6410 board.

Signed-off-by: Ashish Chavan <ashish.chavan <at> kpitcummins.com>
Signed-off-by: David Dajun Chen <dchen <at> diasemi.com>
---
Changes since v2:
- Combined three tables of pll dividers in to one to minimize duplicate
code
- Revamped logic to fetch appropriate pll dividers
- Replaced magic numbers with proper defines

Changes since v1:
- Removed unnecessary function to configure clock dividers
- Changed data layout and code related to PLL dividers. Tried making it
more readable and robust
(Continue reading)

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Takashi Iwai | 1 Feb 12:14
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Re: [PATCH] ALSA: HDA: Fix duplicated output to more than one codec

At Wed,  1 Feb 2012 12:05:41 +0100,
David Henningsson wrote:
> 
> This typo caused the wrong codec's nid to be checked for wcaps type.
> As a result, sometimes speakers would duplicate the output sent to
> HDMI output.
> 
> Cc: stable <at> kernel.org
> BugLink: https://bugs.launchpad.net/bugs/924320
> Signed-off-by: David Henningsson <david.henningsson <at> canonical.com>

Oh, thanks for spotting out!
Applied it now.

Takashi

> ---
>  sound/pci/hda/hda_codec.c |    2 +-
>  1 files changed, 1 insertions(+), 1 deletions(-)
> 
> diff --git a/sound/pci/hda/hda_codec.c b/sound/pci/hda/hda_codec.c
> index 4df72c0..c2c65f6 100644
> --- a/sound/pci/hda/hda_codec.c
> +++ b/sound/pci/hda/hda_codec.c
> @@ -1447,7 +1447,7 @@ void snd_hda_codec_setup_stream(struct hda_codec *codec, hda_nid_t nid,
>  		for (i = 0; i < c->cvt_setups.used; i++) {
>  			p = snd_array_elem(&c->cvt_setups, i);
>  			if (!p->active && p->stream_tag == stream_tag &&
> -			    get_wcaps_type(get_wcaps(codec, p->nid)) == type)
> +			    get_wcaps_type(get_wcaps(c, p->nid)) == type)
(Continue reading)

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