[Libav-user] Choppy dshow audio playback with ffplay
Taha Ansari
mtaha.ansari at gmail.com
Mon Dec 24 11:55:17 CET 2012
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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