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• libFLAC : Fix an out-of-bounds head read

  26 septembre 2015, par Erik de Castro Lopo

  libFLAC : Fix an out-of-bounds head read
  When doing a flac to flac conversion, bad data read from the input file
  
  was making it all the way through the encoder to cause a buffer read
  
  past the end of the buffer in the CRC calculation.
  Fix had two parts :
  * bitwriter.c : Make a debug only assert (assert bits < 32) into a proper
  
    all-build failure.
  
  * stream_encoder.c : Catch the error condition of wasted bits being greater
  
    that bits_pers_sample and limit it to the bits_per_sample value.
  Found using the American Fuzzy Lop fuzzer.
  

  • [DH] src/libFLAC/bitwriter.c
  • [DH] src/libFLAC/stream_encoder.c

• FFmpeg distorted sound when converting audio (C++)

  19 août 2015, par David Norgren

  I’m using the FFmpeg library to generate MP4 files containing audio from various files, such as MP3, WAV, OGG, but I’m having some troubles (I’m also putting video in there, but for simplicity’s sake I’m omitting that for this question, since I’ve got that working). My current code opens an audio file, decodes the content and converts it into the MP4 container and finally writes it into the destination file as interleaved frames.

  It works perfectly for most MP3 files, but when inputting WAV or OGG, the audio in the resulting MP4 is slightly distorted and often plays at the wrong speed (up to many times faster or slower).

  I’ve looked at countless of examples of using the converting functions (swr_convert), but I can’t seem to get rid of the noise in the exported audio.

  Here’s how I add an audio stream to the MP4 (outContext is the AVFormatContext for the output file) :

  audioCodec = avcodec_find_encoder(outContext->oformat->audio_codec);
  
  if (!audioCodec)
  
      die("Could not find audio encoder!");
  
  
  
  
  
  // Start stream
  
  audioStream = avformat_new_stream(outContext, audioCodec);
  
  if (!audioStream)
  
      die("Could not allocate audio stream!");
  
  
  
  audioCodecContext = audioStream->codec;
  
  audioStream->id = 1;
  
  
  
  
  
  // Setup
  
  audioCodecContext->sample_fmt = AV_SAMPLE_FMT_S16;
  
  audioCodecContext->bit_rate = 128000;
  
  audioCodecContext->sample_rate = 44100;
  
  audioCodecContext->channels = 2;
  
  audioCodecContext->channel_layout = AV_CH_LAYOUT_STEREO;
  
  
  
  
  
  // Open the codec
  
  if (avcodec_open2(audioCodecContext, audioCodec, NULL) < 0)
  
      die("Could not open audio codec");

  And to open a sound file from MP3/WAV/OGG (from the filename variable)...

  // Create contex
  
  formatContext = avformat_alloc_context();
  
  if (avformat_open_input(&formatContext, filename, NULL, NULL)<0)
  
      die("Could not open file");
  
  
  
  
  
  // Find info
  
  if (avformat_find_stream_info(formatContext, 0)<0)
  
      die("Could not find file info");
  
  
  
  av_dump_format(formatContext, 0, filename, false);
  
  
  
  
  
  // Find audio stream
  
  streamId = av_find_best_stream(formatContext, AVMEDIA_TYPE_AUDIO, -1, -1, NULL, 0);
  
  if (streamId < 0)
  
      die("Could not find Audio Stream");
  
  
  
  codecContext = formatContext->streams[streamId]->codec;
  
  
  
  
  
  // Find decoder
  
  codec = avcodec_find_decoder(codecContext->codec_id);
  
  if (codec == NULL)
  
      die("cannot find codec!");
  
  
  
  
  
  // Open codec
  
  if (avcodec_open2(codecContext, codec, 0)<0)
  
      die("Codec cannot be found");
  
  
  
  
  
  // Set up resample context
  
  swrContext = swr_alloc();
  
  if (!swrContext)
  
      die("Failed to alloc swr context");
  
  
  
  av_opt_set_int(swrContext, "in_channel_count", codecContext->channels, 0);
  
  av_opt_set_int(swrContext, "in_channel_layout", codecContext->channel_layout, 0);
  
  av_opt_set_int(swrContext, "in_sample_rate", codecContext->sample_rate, 0);
  
  av_opt_set_sample_fmt(swrContext, "in_sample_fmt", codecContext->sample_fmt, 0);
  
  
  
  av_opt_set_int(swrContext, "out_channel_count", audioCodecContext->channels, 0);
  
  av_opt_set_int(swrContext, "out_channel_layout", audioCodecContext->channel_layout, 0);
  
  av_opt_set_int(swrContext, "out_sample_rate", audioCodecContext->sample_rate, 0);
  
  av_opt_set_sample_fmt(swrContext, "out_sample_fmt", audioCodecContext->sample_fmt, 0);
  
  
  
  if (swr_init(swrContext))
  
      die("Failed to init swr context");

  Finally, to decode+convert+encode...

  // Allocate and init re-usable frames
  
  audioFrameDecoded = av_frame_alloc();
  
  if (!audioFrameDecoded)
  
          die("Could not allocate audio frame");
  
  
  
  audioFrameDecoded->format = fileCodecContext->sample_fmt;
  
  audioFrameDecoded->channel_layout = fileCodecContext->channel_layout;
  
  audioFrameDecoded->channels = fileCodecContext->channels;
  
  audioFrameDecoded->sample_rate = fileCodecContext->sample_rate;
  
  
  
  audioFrameConverted = av_frame_alloc();
  
  if (!audioFrameConverted)
  
          die("Could not allocate audio frame");
  
  
  
  audioFrameConverted->nb_samples = audioCodecContext->frame_size;
  
  audioFrameConverted->format = audioCodecContext->sample_fmt;
  
  audioFrameConverted->channel_layout = audioCodecContext->channel_layout;
  
  audioFrameConverted->channels = audioCodecContext->channels;
  
  audioFrameConverted->sample_rate = audioCodecContext->sample_rate;
  
  
  
  AVPacket inPacket;
  
  av_init_packet(&inPacket);
  
  inPacket.data = NULL;
  
  inPacket.size = 0;
  
  
  
  int frameFinished = 0;
  
  
  
  while (av_read_frame(formatContext, &inPacket) >= 0) {
  
  
  
          if (inPacket.stream_index == streamId) {
  
  
  
                  int len = avcodec_decode_audio4(fileCodecContext, audioFrameDecoded, &frameFinished, &inPacket);
  
  
  
                  if (frameFinished) {
  
  
  
                          // Convert
  
  
  
                          uint8_t *convertedData=NULL;
  
  
  
                          if (av_samples_alloc(&convertedData,
  
                                               NULL,
  
                                               audioCodecContext->channels,
  
                                               audioFrameConverted->nb_samples,
  
                                               audioCodecContext->sample_fmt, 0) < 0)
  
                                  die("Could not allocate samples");
  
  
  
                          int outSamples = swr_convert(swrContext,
  
                                                       &convertedData,
  
                                                       audioFrameConverted->nb_samples,
  
                                                       (const uint8_t **)audioFrameDecoded->data,
  
                                                       audioFrameDecoded->nb_samples);
  
                          if (outSamples < 0)
  
                                  die("Could not convert");
  
  
  
                          size_t buffer_size = av_samples_get_buffer_size(NULL,
  
                                                                          audioCodecContext->channels,
  
                                                                          audioFrameConverted->nb_samples,
  
                                                                          audioCodecContext->sample_fmt,
  
                                                                          0);
  
                          if (buffer_size < 0)
  
                                  die("Invalid buffer size");
  
  
  
                          if (avcodec_fill_audio_frame(audioFrameConverted,
  
                                                       audioCodecContext->channels,
  
                                                       audioCodecContext->sample_fmt,
  
                                                       convertedData,
  
                                                       buffer_size,
  
                                                       0) < 0)
  
                                  die("Could not fill frame");
  
  
  
                          AVPacket outPacket;
  
                          av_init_packet(&outPacket);
  
                          outPacket.data = NULL;
  
                          outPacket.size = 0;
  
  
  
                          if (avcodec_encode_audio2(audioCodecContext, &outPacket, audioFrameConverted, &frameFinished) < 0)
  
                                  die("Error encoding audio frame");
  
  
  
                          if (frameFinished) {
  
                                  outPacket.stream_index = audioStream->index;
  
  
  
                                  if (av_interleaved_write_frame(outContext, &outPacket) != 0)
  
                                          die("Error while writing audio frame");
  
  
  
                                  av_free_packet(&outPacket);
  
                          }
  
                  }
  
          }
  
  }
  
  
  
  av_frame_free(&audioFrameConverted);
  
  av_frame_free(&audioFrameDecoded);
  
  av_free_packet(&inPacket);

  I have also tried setting appropriate pts values for outgoing frames, but that doesn’t seem to affect the sound quality at all.

  I’m also unsure how/if I should be allocating the converted data, can av_samples_alloc be used for this ? What about avcodec_fill_audio_frame ? Am I on the right track ?

  Any input is appreciated (I can also send the exported MP4s if necessary, if you want to hear the distortion).

• ffmpeg - loop MP4 for X Hours - sound missing

  9 octobre 2022, par CodeMaker748

  Hey can someone help me with this script :

  


  I want to loop the out.mp4 file up to 1 hous.
It is working it loop the out.mp4 up to 1 hour and save it as finalVid.mp4 but it has no sound. I checked the out.mp4 here the sound is still fine but after executing this script the sound is gone and i just have a 1 hour video.

  


  ffmpeg -y -f lavfi -i \"movie=filename=out.mp4:loop=0, setpts=N/(FRAME_RATE*TB)\" -t 3600 finalVid.mp4


  


  what i need to add here to get sound or someone can tell me other way to loop it up to 1h with ffmpeg ?