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• bugs in avcodec_decode_audio4() in ios pad [on hold]

  29 novembre 2016, par LL2012

  I read the mp3 frame and try to decode it to PCM on ios, but the data is totally disordered(from memory comparing with the original pcm)

  it is quite ok with the same code on windows.

  anyone who have met the same thing ?

  seems ffmpeg is not so attractive subject to get any reply, since my last question has been left there for several days. I have to re-create a new one

      while (av_read_frame(format_ctx_org_, &avpkt) >= 0)
  
  {
  
      if (avpkt.stream_index == audio_stream_)
  
      {
  
          pkt_size = avpkt.size;
  
          pkt_data = avpkt.data;
  
  
  
          while (pkt_size > 0)
  
          {
  
              len = avcodec_decode_audio4(codec_ctx_, decoded_frame, &got_frame, &avpkt);
  
              if (len < 0)
  
              {
  
                  //decode has completed
  
                  if ((audio_duration_ - avpkt.dts <= avpkt.duration) ||
  
                      (file_size_ - avpkt.pos == avpkt.size))
  
                  {
  
                      sink_->OnAudioDataDecodeCompleted(true, "decoding loops end!");
  
                      return kThreadProcessBreak;
  
                  }
  
  
  
                  //meet bad frame, just skip it
  
                  break;
  
              }
  
  
  
              if (got_frame)
  
              {
  
                  if (ConverFrame(decoded_frame) < 0)             
  
                      return kThreadProcessBreak;
  
              }
  
  
  
              pkt_size -= len;
  
              pkt_data += len;
  
          }
  
  
  
      }
  
  }
  
  
  
      int FfmpegAudioFileDecoder::ConverFrame(AVFrame * frame)
  
  {
  
      int64_t src_ch_layout, dst_ch_layout ;
  
      int src_rate = codec_ctx_org_->sample_rate, dst_rate = param_->sample_rate;
  
      uint8_t **src_data = 0, **dst_data = 0;
  
      int src_nb_channels = codec_ctx_org_->channels, dst_nb_channels = param_->sample_fmt;
  
      int src_linesize, dst_linesize;
  
      int src_nb_samples = frame->nb_samples, dst_nb_samples, max_dst_nb_samples;
  
      enum AVSampleFormat src_sample_fmt = codec_ctx_org_->sample_fmt, dst_sample_fmt = (enum AVSampleFormat)param_->sample_fmt;
  
      int ret;
  
  
  
      //we already have the frame data, no need to alloc
  
      /*ret = av_samples_alloc_array_and_samples(&src_data, &src_linesize, src_nb_channels,
  
          src_nb_samples, src_sample_fmt, 0);*/
  
  
  
      //for (int i = 0 ; i < src_nb_channels; i++)
  
      //{
  
      //  src_data[i] = frame->data[i];
  
      //}
  
  
  
      //if (ret < 0) {
  
      //  av_log(0, "Could not allocate source samples\n");
  
      //  return -1;
  
      //}
  
  
  
      max_dst_nb_samples = dst_nb_samples =
  
          av_rescale_rnd(src_nb_samples, dst_rate, src_rate, AV_ROUND_UP);
  
  
  
      ret = av_samples_alloc_array_and_samples(&dst_data, &dst_linesize, dst_nb_channels,
  
          dst_nb_samples, dst_sample_fmt, 0);
  
      if (ret < 0) {
  
          sink_->OnAudioDataDecodeCompleted(false, "ConverFrame():Could not allocate destination samples" );
  
          av_log(0,log_level_, "Could not allocate destination samples\n");
  
          return -1;
  
      }
  
  
  
  
  
      dst_nb_samples = av_rescale_rnd(swr_get_delay(swr_ctx_, src_rate) +
  
          src_nb_samples, dst_rate, src_rate, AV_ROUND_UP);
  
  
  
      if (dst_nb_samples > max_dst_nb_samples) {
  
          av_freep(&dst_data[0]);
  
          ret = av_samples_alloc(dst_data, &dst_linesize, dst_nb_channels,
  
              dst_nb_samples, dst_sample_fmt, 1);
  
          if (ret < 0)
  
          {
  
              sink_->OnAudioDataDecodeCompleted(false, "ConverFrame():av_samples_alloc failed");
  
              return -1;
  
          }
  
  
  
          max_dst_nb_samples = dst_nb_samples;
  
      }
  
  
  
      ret = swr_convert(swr_ctx_, dst_data, dst_nb_samples, (const uint8_t **)frame->data, src_nb_samples);
  
      if (ret < 0) {
  
          sink_->OnAudioDataDecodeCompleted(false, "ConverFrame():Error while converting");
  
          av_log(0, log_level_,"Error while converting\n");
  
          return -1;
  
      }
  
      int dst_bufsize = av_samples_get_buffer_size(&dst_linesize, dst_nb_channels,
  
          ret, dst_sample_fmt, 1);
  
      if (dst_bufsize < 0) {
  
          sink_->OnAudioDataDecodeCompleted(false, "ConverFrame():Could not get sample buffer size");
  
          av_log(0,log_level_, "Could not get sample buffer size\n");
  
          return -1;
  
      }
  
  
  
  
  
      decoded_duration_ += frame->pkt_duration;
  
      float decode_percentage = (float)decoded_duration_ /audio_duration_;
  
  
  
      av_log(0, log_level_, "decode_percentage %.3f\n" , decode_percentage);
  
  
  
      int nRet = sink_->OnAudioDataDecoded(dst_data[0], dst_bufsize, decode_percentage);
  
      if(nRet < 0)
  
          sink_->OnAudioDataDecodeCompleted(false, "ConverFrame(): user cancelled");
  
  
  
      if (dst_data)
  
          av_freep(&dst_data[0]);
  
      av_freep(&dst_data);
  
  
  
      return nRet;
  
  }

• Can't use ffmpeg 3.2 in Android project

  21 novembre 2016, par world2ashish

  Tried almost everything, spent almost 4-5 days trying and still counting.

  The reason I want to compile recent version of ffmpeg is because https://github.com/WritingMinds/ffmpeg-android-java doesn’t work for video rotation. For this also I tried multiple answers, different versions of answer (transpose=1, transpose=dir=cclock, etc)

  Somewhere it was mentioned that rotate command wont work with previous versions of ffmpeg, so after a lot of trials, I started working on compilation of new ffmpeg. But all the tutorials for ffmpeg belong to earlier versions (of Android NDk, ffmpeg) which wont work with ffmpeg 3.2 and NDK r13b (I also tried with NDK downloaded by studio itself but no luck). In case it worked for you, please help.

  After a lot of frustrating experience I am writing this. Please help with a workflow or a clear direction. Thanks

  Wasnt able to tag ffmpeg3.2 because of reputation.

• Detect audio silence in AVFrame using AutoGen in C# for FFmpeg

  16 novembre 2016, par williamtroup

  I’m currently reading audio frames as follows :

  AVFrame* frame = ffmpeg.av_frame_alloc();
  
  
  
  while (ffmpeg.av_read_frame(formatContext, &packet) >= 0)
  
  {
  
      if (packet.stream_index == streamIndex)
  
      {
  
          while (packet.size > 0)
  
          {
  
              int frameDecoded;
  
              int frameDecodedResult = ffmpeg.avcodec_decode_audio4(codecContext, frame, &frameDecoded, packet);
  
  
  
              if (frameDecoded > 0 && frameDecodedResult >= 0)
  
              {
  
                  packet.data += totalBytesDecoded;
  
                  packet.size -= totalBytesDecoded;
  
              }
  
          }
  
  
  
          frameIndex++;
  
      }
  
  
  
      Avcodec.av_free_packet(&packet);
  
  }

  In this loop, I want to be able to detect if "frame" contains silent audio before doing anything with it. Is there a filter to do this ? Been struggling with this for a few days.

  Many thanks in advance.