Gstreamer dropping samples and voice cracking when using g722

able.eldhose able.eldhose at vvdntech.in
Mon Feb 2 05:05:26 PST 2015


Jan Schmidt-6 wrote
> On 02/02/15 21:05, able.eldhose wrote:
>> I am trying to create a g722 based audio streaming application.
>>
>> For that i tried the below pipeline which samples from mic, encode it ,
>> decode it and output to the headphone
>>
>>      gst-launch-0.10 -v alsasrc ! 'audio/x-raw-int,channels=1,rate=16000'
>> !
>> ffenc_g722 ! ffdec_g722 ! alsasink
>>
>> When i run this in my PC it work perfect, but when i use this in a
>> embedded
>> linux board, voice is cracking heavily and cant hear anything useful.
>>
>> Iam using a zedboard based on zynq-7000.
>>
>> Debug prints shows like below
>>
>>      Additional debug info:
>>      gstbaseaudiosrc.c(840): gst_base_audio_src_create ():
>> /GstPipeline:pipeline0/GstAlsaSrc:alsasrc0:
>>      Dropped 3200 samples. This is most likely because downstream can't
>> keep
>> up and is consuming samples too slowly.
>>      WARNING: from element /GstPipeline:pipeline0/GstAlsaSrc:alsasrc0:
>> Can't
>> record audio fast enough
>>
>> When i removed g722 from the pipeline it works well on the board with
>> good
>> sound clarity.
> 
> Are you still using the capsfilter when testing without g722?
*
> When not using g722 capsfilter is not needed, when using g722 in board
> capsfilter is needed.
> But when i try it on PC pipeline works without capsfilter
*
> 
>  I want to 
> make sure you're not seeing CPU consumption from ALSA resampling to give 
> you 16khz.
*
> Iam sorry i didnt understand what you said... Should i enable any debug
> prints to see that
>  iam a novice in this... Please advice
*
> 
> You may try inserting a queue between ffenc_g722 and ffdec_g722 to 
> decouple the encoding and decoding.
*
> I tried adding queue, the issue of dropping samples was solved now... But
> still the voice is very choppy.
> Should i check anything else.
> I tried using g722 with a wav file as source and still the audio is choppy
> but much better than when using alsasrc. Audio is understandable when
> using wav file as source
*
> 
> Regards,
> Jan.
> 
>> My basic requirement was to stream audio to a multicast IP group using
>> gstreamer and g722 encoding. Any hint for correcting this issue will be
>> very
>> helpfull...
>>
>>
>>
>> --
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