gstreamer alsasrc with 6 channel

[nagendra sarma]

hi All
i am observing strange behavior with alsasrc plugin (gstreamer-1.0)
if i capture pcm raw data 6 channel i am not able to get proper data, always corrupted data
in fact, if i enable -v for pipe line the caps displayed in the system terminal are part of my pcm data (strange)
but this is not happening for 2 channel (the pcm data is clean and perfect)  

can somebody help me ?

the pipeline used
gst-launch-1.0  -v alsasrc device=hw:0,0 num-buffers=1  ! audio/x-raw,rate=48000,channels=6,depth=16,width=16 ! filesink location=/tmp/test.pcm

regards
nagendra

[Dmitry Valento]

Hello nagendra sarma

1. Have you tried to use another ALSA plugins (i.e. plughw instead of hw)? See more information about ALSA plugins here: http://www.alsa-project.org/alsa-doc/alsa-lib/pcm_plugins.html
2. Have you tried to capture data with alsa utility - arecord?
3. What is mean "corrupted data"? Either some samples are missed, or data is asynced, or noise, or nothing.

[nagendra sarma]

Hi Valento

thanks for reply,

the pipeline posted is working with gstreamer-0.10 but not with gstreamer 1.0
the pcm data captured is having some spikes (looks like clock | | | | | | | | | )
when open with audscity tool
and this data is present in  2 nd forth, fifth and 6th channels.

if i try to capture 2 channel audio everything works well (problem with 6 channel + gstreamer 1.0)

i will check plughw option

regards
nagendra
 
 

[nagendra sarma]

hi All

is it possible to know after snd_pcm_readi()  where this buffer is going in alsasrc ?

could please share gstreamer alsasrc architecture ?

also any idea  why the "caps"  are part of my pcm raw file ? (strange ?)

regards
nagendra

[Nicolas Dufresne-5]

Le mercredi 15 février 2017 à 22:31 -0800, nagendra sarma a écrit :
> the pipeline used
> gst-launch-1.0  -v alsasrc device=hw:0,0 num-buffers=1  !
> audio/x-raw,rate=48000,channels=6,depth=16,width=16 ! filesink
> location=/tmp/test.pcm

Jus a note that you have .10 specifics in your caps. In GStreamer 1.0,
depth and width fields no longer exists and is replaced with format=.
What is likely happening is that alsasrc picks another format, which
then looks like corrupted data. Try the following instead:

gst-launch-1.0 alsasrc device=hw:0,0 num-buffers=1 ! audio/x-raw,rate=48000,channels=6,format=S16LE ! filesink location=/tmp/test.pcm

Note that you could use wavenc before filesink to create a raw file
playable by normal media players.

regards,
Nicolas
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[Nicolas Dufresne-5]

Le jeudi 16 février 2017 à 05:56 -0800, nagendra sarma a écrit :
> could please share gstreamer alsasrc architecture ?

Not much to say in this regards. It's a straight implementation of
GstAudioSource [0] class and a very basic usage of libalsa API. Have a
look at the code [1].

regards,
Nicolas

[0] https://gstreamer.freedesktop.org/data/doc/gstreamer/head/gst-plugi
ns-base-libs/html/gst-plugins-base-libs-gstaudiosrc.html
[1] https://cgit.freedesktop.org/gstreamer/gst-plugins-base/tree/ext/al
sa/gstalsasrc.c

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[nagendra sarma]

hi  Nicolas
thanks,

but i already tried with format option format=SL16LE but same result

my points are:

1. why the caps are part of pcm file ? if i add -v option to pipeline
     the caps which are printing in debug console are part of my pcm data (checked in hexeditor) strange ?

2. why this strange behavior for 6 channel ? two channel working fine

3. i displayed raw data coming from  pcm_snd_readi() which is proper in case of 6 channel
    but the data send/push by alsasrc to next element is not matching with data comes from  pcm_snd_read() which is not the case in 2 channel ?

3. so, i wanted to know , where this pcm data going, and which is responsible to push to next element ?

thanks
nagendra
 

[nagendra sarma]

hi all

i just commented the following code, everything started working, thanks for support

#if 0  
  if (spec->type == GST_AUDIO_RING_BUFFER_FORMAT_TYPE_RAW && alsa->channels < 9)
    gst_audio_ring_buffer_set_channel_positions (GST_AUDIO_BASE_SRC
        (alsa)->ringbuffer, alsa_position[alsa->channels - 1]);
#endif

[Nicolas Dufresne-5]

Le 17 févr. 2017 2:18 AM, "nagendra sarma" <[hidden email]> a écrit :

hi  Nicolas
thanks,

but i already tried with format option format=SL16LE but same result

my points are:

1. why the caps are part of pcm file ? if i add -v option to pipeline
     the caps which are printing in debug console are part of my pcm data
(checked in hexeditor) strange ?

2. why this strange behavior for 6 channel ? two channel working fine

3. i displayed raw data coming from  pcm_snd_readi() which is proper in case
of 6 channel
    but the data send/push by alsasrc to next element is not matching with
data comes from  pcm_snd_read() which is not the case in 2 channel ?

This data is not the same as we need to reorder the channels to GStreamer/SMPTE order. 

3. so, i wanted to know , where this pcm data going, and which is
responsible to push to next element ?

This is done by the ring buffer, base on hardware provided information.

thanks
nagendra






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[Gst-Geek]

Hi,

I am also facing similar problem while capturing 8 channel audio.
With above changes capture is clear.

Is there bug in BugZilla related to this ??





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[tpearson-raptor]

nagendra sarma wrote

> i just commented the following code, everything started working, thanks
> for
> support
>
> #if 0  
>   if (spec->type == GST_AUDIO_RING_BUFFER_FORMAT_TYPE_RAW &&
> alsa->channels
> < 9)
>     gst_audio_ring_buffer_set_channel_positions (GST_AUDIO_BASE_SRC
>         (alsa)->ringbuffer, alsa_position[alsa->channels - 1]);
> #endif

Quoted from:
http://gstreamer-devel.966125.n4.nabble.com/gstreamer-alsasrc-with-6-channel-tp4681900p4681924.html

I also required this patch trying to record from an M-Track USB 8-channel
capture device.  Without it, all I got was silence or very loud digital
buzzing, which seems to line up with the report of an unwanted pulse train
earlier.

What was the original code even trying to do?



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