[LAD] [PATCH] [SOUND] Fixes recording gain control
Have tested two usb audio mic devices.
At the time of initialization, SNDRV_MIXER_OSS_PRESENT_PVOLUME bit is not
set for MIC (slot 7).
So, the same should not be checked when an application tries to do gain
control
for audio recording devices. Just check
slot-present for SNDRV_MIXER_OSS_PRESENT_CVOLUME independently.
Verified with a simple application which opens /dev/dsp for recording and
/dev/mixer for volume control.
Signed-off-by: Viral Mehta viral.me...@einfochips.com
---
sound/core/oss/mixer_oss.c |2 +-
1 files changed, 1 insertions(+), 1 deletions(-)
diff --git a/sound/core/oss/mixer_oss.c b/sound/core/oss/mixer_oss.c
index 4690b8b..919c7eb 100644
--- a/sound/core/oss/mixer_oss.c
+++ b/sound/core/oss/mixer_oss.c
@@ -690,7 +690,7 @@ static int snd_mixer_oss_put_volume1(struct
snd_mixer_oss_file *fmixer,
if (slot-present SNDRV_MIXER_OSS_PRESENT_PVOLUME) {
snd_mixer_oss_put_volume1_vol(fmixer, pslot,
slot-numid[SNDRV_MIXER_OSS_ITEM_PVOLUME], left, right);
- if (slot-present SNDRV_MIXER_OSS_PRESENT_CVOLUME)
+ } else if (slot-present SNDRV_MIXER_OSS_PRESENT_CVOLUME) {
snd_mixer_oss_put_volume1_vol(fmixer, pslot,
slot-numid[SNDRV_MIXER_OSS_ITEM_CVOLUME], left, right);
} else if (slot-present SNDRV_MIXER_OSS_PRESENT_GVOLUME) {
snd_mixer_oss_put_volume1_vol(fmixer, pslot,
slot-numid[SNDRV_MIXER_OSS_ITEM_GVOLUME], left, right);
--
1.5.3.3
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Re: [LAD] [alsa-devel] [PATCH] [SOUND] recording gain control
Takashi Iwai wrote:
At Tue, 10 Mar 2009 10:00:13 +0100,
Clemens Ladisch wrote:
(CC alsa-devel)
Viral Mehta wrote:
Hi,
I am trying to do gain control while RECORDING for my Audio Mic device.
My application opens /dev/mixer device and calls ioctl(fdmixer,
MIXER_WRITE(SOUND_MIXER_MIC), ...)
But, the call fails. It traces to sound/core/oss/mixer_oss.c file and
snd_mixer_oss_put_volume1() function.
It never falls in if (slot-present SNDRV_MIXER_OSS_PRESENT_CVOLUME)
as condition if (slot-present SNDRV_MIXER_OSS_PRESENT_PVOLUME)
comes as false.
I think there is a bug in kernel and I think it should be like as below,
--- sound/core/oss/mixer_oss.c 2008-12-03 13:24:02.0 +0530
+++ sound/core/oss/mixer_oss.c 2009-03-09 16:22:06.548766896 +0530
@@ -688,7 +688,7 @@ static int snd_mixer_oss_put_volume1(str
if (slot-present SNDRV_MIXER_OSS_PRESENT_PVOLUME) {
snd_mixer_oss_put_volume1_vol(fmixer, pslot,
slot-numid[SNDRV_MIXER_OSS_ITEM_PVOLUME], left, right);
- if (slot-present SNDRV_MIXER_OSS_PRESENT_CVOLUME)
+ } else if (slot-present SNDRV_MIXER_OSS_PRESENT_CVOLUME) {
snd_mixer_oss_put_volume1_vol(fmixer, pslot,
slot-numid[SNDRV_MIXER_OSS_ITEM_CVOLUME], left, right);
} else if (slot-present SNDRV_MIXER_OSS_PRESENT_GVOLUME) {
snd_mixer_oss_put_volume1_vol(fmixer, pslot,
slot-numid[SNDRV_MIXER_OSS_ITEM_GVOLUME], left, right);
Your mailer mangled the patch; see linux/Documentation/email-clients.txt.
Also, the current code behaves intentionally so (as it's designed for
mic-loopback volume).
A more feasible fix would be to add another if check for *_CVOLUME
instead of moving it.
Oh okie. It makes sense.
I just sent out a patch on LAD. Should I resubmit with what you proposed ?
Takashi
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is not the intended recipient, or an employee or agent responsible for
delivering this message to the intended recipient, you are hereby
notified that any dissemination, distribution, copying, or other use of
this message or its attachments is strictly prohibited. If you have
received this message in error, please notify the sender immediately by
replying to this message and please delete it from your computer. Any
views expressed in this message are those of the individual sender
unless otherwise stated.Company has taken enough precautions to prevent
the spread of viruses. However the company accepts no liability for any
damage caused by any virus transmitted by this email.
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Re: [LAD] [alsa-devel] [PATCH] [SOUND] recording gain control
At Tue, 10 Mar 2009 17:52:46 +0530,
Viral Mehta wrote:
Takashi Iwai wrote:
At Tue, 10 Mar 2009 10:00:13 +0100,
Clemens Ladisch wrote:
(CC alsa-devel)
Viral Mehta wrote:
Hi,
I am trying to do gain control while RECORDING for my Audio Mic device.
My application opens /dev/mixer device and calls ioctl(fdmixer,
MIXER_WRITE(SOUND_MIXER_MIC), ...)
But, the call fails. It traces to sound/core/oss/mixer_oss.c file and
snd_mixer_oss_put_volume1() function.
It never falls in if (slot-present SNDRV_MIXER_OSS_PRESENT_CVOLUME)
as condition if (slot-present SNDRV_MIXER_OSS_PRESENT_PVOLUME)
comes as false.
I think there is a bug in kernel and I think it should be like as below,
--- sound/core/oss/mixer_oss.c 2008-12-03 13:24:02.0 +0530
+++ sound/core/oss/mixer_oss.c 2009-03-09 16:22:06.548766896 +0530
@@ -688,7 +688,7 @@ static int snd_mixer_oss_put_volume1(str
if (slot-present SNDRV_MIXER_OSS_PRESENT_PVOLUME) {
snd_mixer_oss_put_volume1_vol(fmixer, pslot,
slot-numid[SNDRV_MIXER_OSS_ITEM_PVOLUME], left, right);
- if (slot-present SNDRV_MIXER_OSS_PRESENT_CVOLUME)
+ } else if (slot-present SNDRV_MIXER_OSS_PRESENT_CVOLUME) {
snd_mixer_oss_put_volume1_vol(fmixer, pslot,
slot-numid[SNDRV_MIXER_OSS_ITEM_CVOLUME], left, right);
} else if (slot-present SNDRV_MIXER_OSS_PRESENT_GVOLUME) {
snd_mixer_oss_put_volume1_vol(fmixer, pslot,
slot-numid[SNDRV_MIXER_OSS_ITEM_GVOLUME], left, right);
Your mailer mangled the patch; see linux/Documentation/email-clients.txt.
Also, the current code behaves intentionally so (as it's designed for
mic-loopback volume).
A more feasible fix would be to add another if check for *_CVOLUME
instead of moving it.
Oh okie. It makes sense.
I just sent out a patch on LAD. Should I resubmit with what you proposed ?
LAD is no good place for sending a patch of sound driver.
Please post to alsa-devel ML (and add Cc to me).
thanks,
Takashi
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[LAD] Anyone out there familiar with the code base of sonic visualizer?
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[LAD] New release of jconv
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Re: [LAD] New release of jconv
On Wed, Mar 11, 2009 at 1:28 AM, Fons Adriaensen f...@kokkinizita.netwrote: Hello all, Jconv-0.8.0 is now available at the usual place http://www.kokkinizita.net/linuxaudio/downloads New in this release: - New command line option -L nsamples, tries to compensate for 'nsamples' of processing latency. - Jack ports can now be given meaningful names and optionally be connected, by commands in the config files. - Some new reverb IRs available in a separate package 'jconv-reverbs'. Includes ambisonic and stereo IRs of the concert hall of La Casa della Musica, and the former Santa Elisabetta church, now La Casa del Suono, in Parma. Enjoy ! -- FA Laboratorio di Acustica ed Elettroacustica Parma, Italia Be quiet, Master Land; and you, Professor, will you be so good as to listen to me ? ___ Linux-audio-dev mailing list Linux-audio-dev@lists.linuxaudio.org http://lists.linuxaudio.org/mailman/listinfo/linux-audio-dev Fons, downloaded, but get the following make error: make g++ -O3 -Wall -MMD -MP -DVERSION=\0.8.0\ -c -o jconv.o jconv.cc In file included from jconv.cc:29: jclient.h: In member function ‘void Jclient::start()’: jclient.h:48: error: no matching function for call to ‘Convproc::start_process(int, int)’ /usr/include/zita-convolver.h:250: note: candidates are: int Convproc::start_process(int) make: *** [jconv.o] Error 1 What am i doing wrong here? Alex. -- Parchment Studios (It started as a joke...) ___ Linux-audio-dev mailing list Linux-audio-dev@lists.linuxaudio.org http://lists.linuxaudio.org/mailman/listinfo/linux-audio-dev
Re: [LAD] New release of jconv
On Wed, Mar 11, 2009 at 01:44:45AM +0300, alex stone wrote: Fons, downloaded, but get the following make error: make g++ -O3 -Wall -MMD -MP -DVERSION=\0.8.0\ -c -o jconv.o jconv.cc In file included from jconv.cc:29: jclient.h: In member function ‘void Jclient::start()’: jclient.h:48: error: no matching function for call to ‘Convproc::start_process(int, int)’ /usr/include/zita-convolver.h:250: note: candidates are: int Convproc::start_process(int) make: *** [jconv.o] Error 1 What am i doing wrong here? Nothing, I forgot you need an update of zita-convolver. It's uploaded now. Make sure to check the makefile for zita-convolver and add the best optimisation flags for your CPU. Ciao, -- FA Laboratorio di Acustica ed Elettroacustica Parma, Italia Be quiet, Master Land; and you, Professor, will you be so good as to listen to me ? ___ Linux-audio-dev mailing list Linux-audio-dev@lists.linuxaudio.org http://lists.linuxaudio.org/mailman/listinfo/linux-audio-dev
Re: [LAD] New release of jconv
Fons Adriaensen wrote: On Wed, Mar 11, 2009 at 01:44:45AM +0300, alex stone wrote: What am i doing wrong here? Nothing, I forgot you need an update of zita-convolver. It's uploaded now. Make sure to check the makefile for zita-convolver and add the best optimisation flags for your CPU. thanks, that one fixed the problem for me. there's a spurios makeald in the Makefile that prevents the install target from working. i deleted it without apparent ill effects. best, jörn ___ Linux-audio-dev mailing list Linux-audio-dev@lists.linuxaudio.org http://lists.linuxaudio.org/mailman/listinfo/linux-audio-dev
Re: [LAD] New release of jconv
wow. thanks for the new example configurations and the extensive comments! very helpful. the convolver itself is a bit of a marketing disaster, though. i mean, uh, it just convolves. No fancy new features, such as UltraLowJitter, or at least TrueMultiplyAndAdd (for purists, as opposed to FFT-based, for that extra dash of depth and clarity). And TubeMode would be cool as well. but one cannot have it all, i assume. ;) in the reverb configs, you mention that you cut off the direct sound by setting an offset, which will then make the convolutions suitable to use in a classic aux send setup, so that the reverb return will be 100% wet. but why do you remove the first 5 ms of the room response? i seem to recall it is about control of low frequency energy, but i forgot the details. are those 5ms a natural constant, a matter of taste, or maybe a function of the room's rt60? pointers to TFM would be R with great interest. best, jörn ___ Linux-audio-dev mailing list Linux-audio-dev@lists.linuxaudio.org http://lists.linuxaudio.org/mailman/listinfo/linux-audio-dev
Re: [LAD] New release of jconv
On Wed, Mar 11, 2009 at 02:26:06AM +0300, alex stone wrote: Hehe Jorn, there's an echo in here. I'll go hunting for the makeald, then. Just delete the makeald line in the install target. I'll update the release... (Seems like I'm trying to do too much in too little time) Ciao, -- FA Laboratorio di Acustica ed Elettroacustica Parma, Italia Be quiet, Master Land; and you, Professor, will you be so good as to listen to me ? ___ Linux-audio-dev mailing list Linux-audio-dev@lists.linuxaudio.org http://lists.linuxaudio.org/mailman/listinfo/linux-audio-dev
[LAD] Strange network problem ?
Hello all, Today I had a strange problem when using the WFS system here. The 'master' PC sends a multicast message every 1024 samples (21.333 ms) to all 'render' PCs. This has to arrive on time, and when it's late the renderers will mute their output and report the error in their status messages. Today I used some ssh -X logins from my laptop in the WFS room to the WFS master to run Ardour and some other apps for a demo. All this worked well all the time, as it has done before. I left everything running when going for lunch with our visitors, and when I returned restarted Ardour to listen again. I got a lots 'of 'message too late' errors from the rendering machines, and interrupted sound. Strange enough this seemed to be related to the _volume_ of the sounds... In other words to Ardour's level meters. Restarting Ardour in a new ssh login did not help, but running it directly on the WFS master solved the problem. So apparently the network traffic required to update Ardour's meters was causing the delays. And clearly the whole remote X session was slower than normal. CPU loads looked normal. Now all this should be peanuts for a Gbit network that has no other traffic at all, and it worked perfectly before. I just never left it running for such a long time. Anyone an idea as to what is happening here, and how it could be cured ? TIA, -- FA Laboratorio di Acustica ed Elettroacustica Parma, Italia Be quiet, Master Land; and you, Professor, will you be so good as to listen to me ? ___ Linux-audio-dev mailing list Linux-audio-dev@lists.linuxaudio.org http://lists.linuxaudio.org/mailman/listinfo/linux-audio-dev
Re: [LAD] New release of jconv
On Wed, Mar 11, 2009 at 12:45:17AM +0100, Jörn Nettingsmeier wrote: wow. thanks for the new example configurations and the extensive comments! very helpful. I hope you like the AMB ones... the convolver itself is a bit of a marketing disaster, though. i mean, uh, it just convolves. No fancy new features, such as UltraLowJitter, or at least TrueMultiplyAndAdd (for purists, as opposed to FFT-based, for that extra dash of depth and clarity). And TubeMode would be cool as well. but one cannot have it all, i assume. ;) :-) I'll take the IR of a long tube one if these days. in the reverb configs, you mention that you cut off the direct sound by setting an offset, which will then make the convolutions suitable to use in a classic aux send setup, so that the reverb return will be 100% wet. but why do you remove the first 5 ms of the room response? Because it's not room response but speaker response (mainly LF, if the source was not well equalised), or reflections from very close objects which are not perceived as reflections but just add a nasty colouration. Ciao, -- FA Laboratorio di Acustica ed Elettroacustica Parma, Italia Be quiet, Master Land; and you, Professor, will you be so good as to listen to me ? ___ Linux-audio-dev mailing list Linux-audio-dev@lists.linuxaudio.org http://lists.linuxaudio.org/mailman/listinfo/linux-audio-dev
