[Libav-user] Choppy dshow audio playback with ffplay

[Roger Pack]

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
>