Audio lags in aSPICE client while using gstreamer framework

[Ritesh Prajapati]

Hi Nicolas or others,

     does anyone has any idea to solve below issue?

Regards,
Ritesh Prajapati,
System Level Solutions (India) Pvt.Ltd.

On Friday 08 July 2016 09:25 AM, Ritesh Prajapati wrote:
> Hi Nicolas,
>
>     Thanks for Reply.
>
>     When I am getting too much audio discontinuity timestamps messages 
> at that time Audio data is lagged compare to video data and goes out 
> of sync for that duration. So, if I set threshold timestamp to 40ms 
> then will my sync problem be solved or reduced?
>
>     Please let me know if any other configuration require in appsrc 
> pipeline to solve this problem.
> Regards,
> Ritesh Prajapati,
> System Level Solutions (India) Pvt.Ltd.
> On Thursday 07 July 2016 06:45 PM, Nicolas Dufresne wrote:
>> See reply inline.
>>
>> Le jeudi 07 juillet 2016 à 14:12 +0530, Ritesh Prajapati a écrit :
>>> Hi All,
>>>
>>>      I am working on one of our product in which android 4.4 Kitkat is
>>> running. We have one application called it as SPICE client which runs
>>> on our product and is used to capture Audio+Video data streamed over
>>> network from SPICE server which is installed into Ubuntu 16.04
>>> Server.
>>>
>>>      We are using SPICE client gtk based topology in which Gstreamer
>>> Framework 1.0 is used.
>>>
>>>       We are facing one audio sync issue like when Streaming process
>>> of Audio+Video are started from SPICE server side at that time we are
>>> getting video data perfectly but not getting audio data in sync
>>> compared to video frames and are dropped for some initial durations.
>>> so, it seems like Audio data is not synced or is lagged compare to
>>> video data at that time.
>>>
>>>       Please find following code snippet of Gstreamer Pipeline we have
>>> configured and used in our SPICE client android code.
>>>
>>>> #ifdef WITH_GST1AUDIO
>>>>              g_strdup_printf("audio/x-
>>>> raw,format=\"S16LE\",channels=%d,rate=%d,"
>>>>                              "layout=interleaved", channels,
>>>> frequency);
>>>>          #else
>>>>              g_strdup_printf("audio/x-raw-
>>>> int,channels=%d,rate=%d,signed=(boolean)true,"
>>>>                              "width=16,depth=16,endianness=1234",
>>>> channels, frequency);
>>>>          #endif
>>>>          gchar *pipeline = g_strdup
>>>> (g_getenv("SPICE_GST_AUDIOSINK"));
>>>>          if (pipeline == NULL)
>>>>              pipeline = g_strdup_printf("appsrc is-live=1 do-
>>>> timestamp=1 format=time min-latency=0 caps=\"%s\" name=\"appsrc\" !
>>>> "
>>>>                                         "audioconvert !
>>>> audioresample ! autoaudiosink name=\"audiosink\"", audio_caps);
>>>   
>>>      Also, we are getting below warning message sometimes from Android
>>> Gstreamer Studio.
>>>          
>>>         
>>> gstaudiobasesink.c:1807:gst_audio_base_sink_get_alignment:<audiosink-
>>> actual-sink-opensles> Unexpected discontinuity in audio timestamps of
>>> -0:00:00.076145124, resyncing
>> As audio may arrive in small burst, what will happen is that the
>> timestamp distance between buffer becomes too small. The timestamp +
>> duration of the current buffer may endup after the next timestamp. In
>> this case the audio sink will resync. To prevent that, you should
>> timestamp the buffer yourself, so you pick an initial timestamp, and
>> add the duration. You then monitor the calculated timestamp and the
>> time now, if it drift over a certain threshold (generally 40ms), you
>> resync (and set the discontinuity flag).
>>
>> In an ideal world, the streaming protocol should provide you with
>> timing information that let you correlate in time the video and the
>> audio frames. This should serve in creating timestamps and ensuring
>> perfect A/V sync. I don't know Spice too much, but it might not be part
>> of the protocol.
>>
>> Nicolas
>>
>>
>> _______________________________________________
>> gstreamer-devel mailing list
>> gstreamer-devel at lists.freedesktop.org
>> https://lists.freedesktop.org/mailman/listinfo/gstreamer-devel
>

-------------- next part --------------
An HTML attachment was scrubbed...
URL: <https://lists.freedesktop.org/archives/gstreamer-devel/attachments/20160709/303d8e0d/attachment.html>