Question

我正在尝试使用ffmpeg将音频数据包写入文件。源设备在一段时间后发送数据包。例如

First packet has a time stamp 00:00:00 
Second packet has a time stamp 00:00:00.5000000
Third packet has a time stamp 00:00:01
And so on...

每秒意味着两个数据包。

我想对这些数据包进行编码并写入文件。

我指的是链接Muxing.c

中的Ffmpeg示例

编码和写入时没有错误。但输出文件的音频持续时间仅为2秒，速度也非常快。

视频帧根据设置正确。

我认为这个问题与pts，dts和数据包持续时间的计算有关。

我应该如何计算pts，dts和持续时间的正确值。或者这个问题与其他事情有关吗？

代码：

void AudioWriter::WriteAudioChunk(IntPtr chunk, int lenght, TimeSpan timestamp)
{
    int buffer_size = av_samples_get_buffer_size(NULL, outputStream->tmp_frame->channels, outputStream->tmp_frame->nb_samples,  outputStream->AudioStream->codec->sample_fmt, 0);

    uint8_t *audioData = reinterpret_cast<uint8_t*>(static_cast<void*>(chunk));
    int ret = avcodec_fill_audio_frame(outputStream->tmp_frame,outputStream->Channels, outputStream->AudioStream->codec->sample_fmt, audioData, buffer_size, 1);

    if (!ret)
       throw gcnew System::IO::IOException("A audio file was not opened yet.");

    write_audio_frame(outputStream->FormatContext, outputStream, audioData);
}


static int write_audio_frame(AVFormatContext *oc, AudioWriterData^ ost, uint8_t *audioData)
{
       AVCodecContext *c;
       AVPacket pkt = { 0 };
       int ret;
       int got_packet;
       int dst_nb_samples;

       av_init_packet(&pkt);
       c = ost->AudioStream->codec;

       AVFrame *frame = ost->tmp_frame;

      if (frame)
      {
          dst_nb_samples = av_rescale_rnd(swr_get_delay(ost->swr_ctx, c->sample_rate) + frame->nb_samples, c->sample_rate, c->sample_rate, AV_ROUND_UP);
          if (dst_nb_samples != frame->nb_samples)
            throw gcnew Exception("dst_nb_samples != frame->nb_samples");

          ret = av_frame_make_writable(ost->AudioFrame);
          if (ret < 0)
             throw gcnew Exception("Unable to make writable.");

          ret = swr_convert(ost->swr_ctx, ost->AudioFrame->data, dst_nb_samples, (const uint8_t **)frame->data, frame->nb_samples);
          if (ret < 0) 
            throw gcnew Exception("Unable to convert to destination format.");

          frame = ost->AudioFrame;

          AVRational timebase = { 1, c->sample_rate };
          frame->pts = av_rescale_q(ost->samples_count, timebase, c->time_base);
          ost->samples_count += dst_nb_samples;
      }

      ret = avcodec_encode_audio2(c, &pkt, frame, &got_packet);
      if (ret < 0) 
        throw gcnew Exception("Error encoding audio frame.");

      if (got_packet)
      {
        ret = write_frame(oc, &c->time_base, ost->AudioStream, &pkt);
        if (ret < 0)
            throw gcnew Exception("Audio is not written.");
      }
      else
         throw gcnew Exception("Audio packet encode failed.");

      return (ost->AudioFrame || got_packet) ? 0 : 1;
}

static int write_frame(AVFormatContext *fmt_ctx, const AVRational *time_base, AVStream *st, AVPacket *pkt)
{
    av_packet_rescale_ts(pkt, *time_base, st->time_base);
    pkt->stream_index = st->index;
    return av_interleaved_write_frame(fmt_ctx, pkt);
}

使用ffmpeg将音频数据包写入文件

0 个答案: