[Libav-user] Choppy dshow audio playback with ffplay

[Taha Ansari]

Hi!

Thanks for your reply - I just tried your suggestion, and recorded
approximately 1.5 minutes of audio, and saved it to a file on hard-disk (no
UDP involved). There are no cut-offs or choppy playback; all voice is
apparently recorded fine.

So this means above mentioned routines are working properly. So that leaves
UDP channel and decoding/playback on other end. Since managing changing of
destination file (local vs. UDP) is something managed by FFmpeg itself (and
for testing purpose I am always on same network), there are more chances
there is something not optimized on the decoding end.


On Tue, Apr 2, 2013 at 2:33 AM, Roger Pack <rogerdpack2 at gmail.com> wrote:

> what if you just record locally (no UDP) does it die?
>
> On 12/24/12, Taha Ansari <mtaha.ansari at gmail.com> wrote:
> > Hi!
> >
> > I have a small test application that sends microphone audio over network.
> > But the audio playback is sometimes very choppy/lossy, and also I
> initially
> > need to 'seek' ffplay back to hear audio with minimum latency. I do this
> in
> > Windows, using dshow, zeranoe ffmpeg builds, MSVS; and here is custom
> code
> > of relevance (output file is in extension .mp2, and packets are sent on
> > udp. I tried AAC extension as well, but results are somewhat the same):
> >
> > ***********  +  Decoding part:   +  *************
> > if(this->packet.stream_index == this->audioStream)
> >     {
> >         unsigned int samples_size= 0;
> >         AVCodecContext *c = outputCodecCtxAudio;
> >         int finalPTS = 0;
> >         samples = (short *) av_fast_realloc(samples, &samples_size,
> > FFMAX(packet.size, AVCODEC_MAX_AUDIO_FRAME_SIZE));
> >         finalPTS =  packet.pts;
> >         audiobufsize = AVCODEC_MAX_AUDIO_FRAME_SIZE*2;
> >         avcodec_decode_audio3(pCodecCtxAudio, samples, &audiobufsize,
> > &packet);
> >
> >
> >         if(pCodecCtxAudio->sample_rate != c->sample_rate ||
> > pCodecCtxAudio->channels != c->channels )
> >         {
> >             if ( rs == NULL)
> >             {
> >                 rs = av_audio_resample_init(c->channels,
> > pCodecCtxAudio->channels, c->sample_rate, pCodecCtxAudio->sample_rate,
> > AV_SAMPLE_FMT_S16, AV_SAMPLE_FMT_S16, 0,0,0,0);
> >             }
> >         }
> >         if(pCodecCtxAudio->sample_rate != c->sample_rate ||
> > pCodecCtxAudio->channels != c->channels)
> >         {
> >             int size_out = audio_resample(rs, (short *)buffer_resample,
> > samples, audiobufsize/ (pCodecCtxAudio->channels * 2) );
> >             av_fifo_generic_write(fifo, (uint8_t *)buffer_resample,
> > size_out * c->channels * 2, NULL );
> >         }
> >         else
> >         {
> >             av_fifo_generic_write(fifo, (uint8_t *)samples, audiobufsize,
> > NULL );
> >         }
> >     }
> > ***********  -  Decoding part:   -  *************
> >
> > ***********  + Encoding part:   + ***************
> > if ( decoderData->audiobufsize )
> >     {
> >         AVPacket pkt;
> >         av_init_packet(&pkt);
> >
> >         AVCodecContext* c = encoderData->audio_st->codec;
> >
> >         int frame_bytes = c->frame_size * 2 * c->channels;
> >
> >         while( av_fifo_size(decoderData->fifo) >= frame_bytes )
> >         {
> >             int ret = av_fifo_generic_read( decoderData->fifo, data_buf,
> > frame_bytes, NULL );
> >             /* encode the samples */
> >             pkt.size= avcodec_encode_audio(c, audio_out, frame_bytes
> > /*packet.size*/, (short *)data_buf);
> >
> >             pkt.stream_index= encoderData->audio_st->index;
> >             pkt.data= audio_out;
> >             pkt.flags |= AV_PKT_FLAG_KEY;
> >
> >             pkt.pts = pkt.dts = 0;
> >             /* write the compressed frame in the media file */
> >             if (av_interleaved_write_frame(encoderData->ocAud, &pkt) !=
> 0)
> >             {
> >                 fprintf(stderr, "Error while writing audio frame\n");
> >                 exit(1);
> >             }
> >         }
> >     }
> > ***********  - Encoding part:   - ***************
> >
> > Other code is similar to the muxing.c example that comes with the
> builds. I
> > know the functions used above are kind of outdated, but that is the best
> > working source I could find from the internet.
> >
> > Can anyone kindly highlight how I could improve my code, or do I need to
> > tweak ffplay somehow for better results?
> >
> > Thanks for your time,
> >
> > Best regards
> >
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>
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