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I'm working on a ffmpeg playout application for Decklink but I'm facing some audio issues. I've seen other questions about this topic but none of them are currently helping.

I've tried Reubens code (https://stackoverflow.com/a/15372417/12610231) with the swr_convert for playing out ffmpeg/libav frames to a Decklink board (this needs to be 16 bits PCM interleaved) but the audio sounds wrong. It sounds like it's missing samples/ only getting half of the required samples).

When I record the samples in a raw audio file and play it out with Audacity the timeline is half the length of the actual recording and playing the samples on double speed.

I also tried the 'manual' conversion (https://stackoverflow.com/a/15372417/12610231) but unfortunately, not the result I was hoping for.

Here are some snippets of my code

swr_ctx = swr_alloc();
av_opt_set_int(swr_ctx, "in_channel_count", pAudioCodecCtx->channels, 0);
av_opt_set_int(swr_ctx, "in_sample_rate", pAudioCodecCtx->sample_rate, 0);
av_opt_set_int(swr_ctx, "in_channel_layout", pAudioCodecCtx->channel_layout, 0);
av_opt_set_sample_fmt(swr_ctx, "in_sample_fmt", pAudioCodecCtx->sample_fmt, 0);
av_opt_set_int(swr_ctx, "out_channel_count", 2, 0);
av_opt_set_int(swr_ctx, "out_sample_rate", 48000, 0);
av_opt_set_int(swr_ctx, "out_channel_layout", AV_CH_LAYOUT_STEREO, 0);
av_opt_set_sample_fmt(swr_ctx, "out_sample_fmt", AV_SAMPLE_FMT_S16, 0);

if (swr_init(swr_ctx))
{
    printf("Error SWR");
}

///

ret = avcodec_decode_audio4(pAudioCodecCtx, pFrame, &frameFinished, &packet);

if (ret < 0) {
    printf("Error in decoding audio frame.\n");
}

swr_convert(swr_ctx, (uint8_t**)&m_audioBuffer, pFrame->nb_samples, (const uint8_t *)pFrame->extended_data, pFrame->nb_samples);

It also looks like that the FFmpeg packet contains out of 1 video packet en 2 audio packets, not sure what to do with the second audio packet, I already tried to combine the first and second audio package without any good result on the audio side.

Any help is appreciated.

Rick Jansen
  • 1
  • prob the source to get accurate list of sample rate, bits per sample and channelCount.. when u know that for the input review your settings for your playback api – Robert Rowntree Dec 28 '19 at 01:32
  • My output device is set to sample rate 48khz, sample depth 16 bit and 2 channels. The video output format is 1080i50. If I'm correct the samples per channel should be 48000hz/25 frames = 1920. The internal audio buffer is defined by: m_audioBuffer = (int16_t*)malloc(sizeof(int16_t) * (m_sampleRate / m_framesPerSecond) * m_channels); memset(m_audioBuffer, 0, sizeof(int16_t) * (m_sampleRate / m_framesPerSecond) * m_channels); I also have a self generated 1khz tone buffer and this is working fine on the output. – Rick Jansen Dec 28 '19 at 15:24

0 Answers0