gmane.linux.alsa.devel

Re: [PATCH] ASoC: codecs: Add DA732x codec driver

Mark Brown — 2012-05-23T11:14:34


You only ever enable this and it seems like it should be a widget - why
is it not a widget?


This should be a switch not an enum.


Why is this not done using a normal enum?


Similarly here, and this looks like a switch statement.


This should be generic code, there's nothing device specific about this
control type and it'd be useful for other devices.


Just MIC1 Switch.


All these should be Volumes, and really they should be stereo controls.


Volume needs to be the last thing in the name.


This looks like you perhaps need some supply widgets.  Similarly for
most of your events.


Don't use MICBIAS widgets for new code, use supply widgets.



The MICBIASes should be connected on the board.


Use dai->base.


Use snd_soc_update_bits, this will fix the issue where you don't
currently clear the register.  You're also missing error handling in the
default: case.



Don't make up private defines for generic things like this.  Though
really a define like this is a bit silly...


This should either stop an

Re: [PATCH] ASoC: tegra+wm8903: remove non-DT supportfor Seaboard

Mark Brown — 2012-05-23T11:20:58

Applied, thanks.

Re: [PATCH 1/5] ASoC: WM8903: init GPIOs during I2C probe not codec probe

Mark Brown — 2012-05-23T11:20:23


Applied all, thanks.

Re: [PATCH v5] sound/soc/lapis: add platform driverfor ML7213

Mark Brown — 2012-05-23T09:41:00



Nobody seems to be working on it as far as I can tell, I've certainly
not seen any patches.

Re: UAC2 defaults to max sampling rate

Clemens Ladisch — 2012-05-23T09:18:10

This is an as-yet unfixed bug in the driver.
(The usb_set_interface call in snd_usb_add_endpoint is completely
superfluous.)


Regards,
Clemens

Re: [PATCH] alsa-lib conf: Add more USB devices toS/PDIF blacklist

Takashi Iwai — 2012-05-23T09:04:03

At Wed, 23 May 2012 09:59:50 +0200,
David Henningsson wrote:

Applied.  Thanks.


Takashi

[PATCH 2/3 v2] sb_mixer: Autodetect optional SB16mixer controls

Ondrej Zary — 2012-05-23T08:07:15

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>
---
Removed a debug printk.

--- 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),

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
 }

Re: [PATCH 10/11] ASoC: codecs: Add AB8500 codec-driver

Ola Lilja — 2012-05-23T07:04:49



These one we need as ALSA-controls. They are parameters that we can tune during
runtime and are not hard-wired in any way.

Re: [PATCH] alsa-lib/tlv: fix handling of rawvalueranges

Takashi Iwai — 2012-05-22T23:54:30

At Tue, 22 May 2012 18:06:34 +0200 (CEST),
Benoît Thébaudeau wrote:

Thanks, applied now.


Takashi

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Re: [PATCH 2/2] ALSA: core: group read of pointer,tstamp and jiffies

Takashi Iwai — 2012-05-22T23:52:13

At Tue, 22 May 2012 14:54:02 -0500,
Pierre-Louis Bossart wrote:

Applied this one.  Thanks.


Takashi

Re: [PATCH 1/2] ALSA: update sync header when streamsare linked/unlinked

Takashi Iwai — 2012-05-22T23:47:31

At Tue, 22 May 2012 14:54:01 -0500,
Pierre-Louis Bossart wrote:

This looks like a right "fix", but ...


This is a completely different issue, so please don't mix up in a
single patch.

And...


... this isn's safe.  There are more than 32 substreams.  And there
are multiple streams with the same substream index.

Also, the point of the sync id is that it's shared with all linked
streams.  Your patch breaks it.  It updates only the last added sync
id.

The fact that the driver currently sets only the card number is
actually problematic.  It's not unique enough.  This should be fixed.
But, exposing the substream bitmask doesn't help much because it can't
be fully implemented in the sync id size.  If you need to know which
streams are linked, loop over all streams and check the sync id. 


thanks,

Takashi

Re: [GIT PULL] ASoC updates for 3.5

Takashi Iwai — 2012-05-22T23:39:51

At Tue, 22 May 2012 18:19:25 +0100,
Liam Girdwood wrote:

Thanks.  Pulled now, but this won't be included in the pull request
I'm going to send today.  It'll be in the next pull request.


Takashi

[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 2/2] ALSA: core: group read of pointer,tstamp and jiffies

Pierre-Louis Bossart — 2012-05-22T19:54:02

Group read of hw_ptr, tstamp and jiffies in a sequence
for better correlation. Previous code took timestamp at the
end, which could introduce delays between audio time and
system time.

Signed-off-by: Pierre-Louis Bossart <pierre-louis.bossart< at >linux.intel.com>
---
 sound/core/pcm_lib.c |   23 ++++++++++++++++++-----
 1 files changed, 18 insertions(+), 5 deletions(-)

diff --git a/sound/core/pcm_lib.c b/sound/core/pcm_lib.c
index ae46d75..3f3097d 100644
--- a/sound/core/pcm_lib.c
+++ b/sound/core/pcm_lib.c
< at >< at > -313,9 +313,22 < at >< at > static int snd_pcm_update_hw_ptr0(struct snd_pcm_substream *substream,
 snd_pcm_uframes_t old_hw_ptr, new_hw_ptr, hw_base;
 snd_pcm_sframes_t hdelta, delta;
 unsigned long jdelta;
+unsigned long curr_jiffies;
+struct timespec curr_tstamp;
 
 old_hw_ptr = runtime->status->hw_ptr;
+
+/*
+ * group pointer, time and jiffies reads to allow for more
+ * accurate correlations/corrections.
+ * The values are stored at the end of this routine after
+ * corrections for hw_ptr position

[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_

Re: snd-usb endpoint rework

Felix Homann — 2012-05-22T19:36:59

Sorry, I can't help testing at the moment. I'm extremely busy preparing a
couple of shows I'll be playing in the next few weeks.

But Aurélien's problems remind me of an issue I had some weeks ago. The
issue was caused by a jackd process that was unintentionally launched via
D-Bus on start up.

Regards,
Felix
Am 22.05.2012 20:22 schrieb "Aurélien Leblond" <blablack< at >gmail.com>: