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• libavformat/libavcodec providing invalid container header

  14 décembre 2017, par seanr8

  I’m using libavcodec to encode a stream to h264 and libavformat to store it in an mp4. The resulting container has an invalid header that can be played in VLC, but not any other player.

  I’ve found that using the mp4 container and the "mpeg4" codec produces a valid mp4 file, but using libx265 (HEVC) or the libx264 codec produces invalid mp4s.

  I can use ffmpeg -i invalid.mp4 -vcodec copy valid.mp4 and I get a file of almost the exact same size, but in a valid container.

  Examples of these files are here : Broken file and
  Repaied file [use the download links in the upper right to examine]

  I used a hex editor to see the differences in the headers of the two files and the invalid one is 1 byte smaller than the valid one.

  The code I’m using to open the container and codec and to write the header is here :

  AVOutputFormat *container_format;
  
  AVFormatContext *container_format_context;
  
  AVStream *video_stream;
  
  int ret;
  
  
  
  /* allocate the output media context */
  
  avformat_alloc_output_context2(&container_format_context, NULL, NULL, out_file);
  
  if (!container_format_context) {
  
      log(INFO, "Unable to determine container format from filename, exiting\n");
  
      exit(1);
  
  }
  
  else {
  
      log(INFO, "Using container %s\n", container_format_context->oformat->name);
  
  }
  
  
  
  if (!container_format_context) {
  
      log(ERROR, "Could not build container format context. Encoding failed.");
  
      exit(1);
  
  }
  
  
  
  container_format = container_format_context->oformat;
  
  
  
  /* Pull codec based on name */
  
  AVCodec* codec = avcodec_find_encoder_by_name(codec_name);
  
  if (codec == NULL) {
  
      log(ERROR, "Failed to locate codec \"%s\".",
  
              codec_name);
  
      exit(1);
  
  }
  
  
  
  /* create stream */
  
  video_stream = NULL;
  
  video_stream = avformat_new_stream(container_format_context, codec);
  
  if (!video_stream) {
  
      log(ERROR, "Could not allocate encoder stream. Cannot continue.\n");
  
      exit(1);
  
  }
  
  video_stream->id = container_format_context->nb_streams - 1;
  
  
  
  video_stream->time_base = video_stream->codec->time_base = (AVRational) { 1, 25};
  
  
  
  av_dump_format(container_format_context, 0, out_file, 1);
  
  
  
  
  
  
  
  /* Retrieve encoding context */
  
  AVCodecContext* avcodec_context = video_stream->codec;
  
  if (avcodec_context == NULL) {
  
      log(ERROR, "Failed to allocate context for "
  
              "codec \"%s\".", codec_name);
  
      exit(1);
  
  }
  
  
  
  
  
  /* Init context with encoding parameters */
  
  avcodec_context->bit_rate = bitrate;
  
  avcodec_context->width = width;
  
  avcodec_context->height = height;
  
  avcodec_context->gop_size = 10;
  
  avcodec_context->max_b_frames = 1;
  
  avcodec_context->qmax = 31;
  
  avcodec_context->qmin = 2;
  
  avcodec_context->pix_fmt = AV_PIX_FMT_YUV420P;
  
  
  
  av_dump_format(container_format_context, 0, out_file, 1);
  
  
  
  /* Open codec for use */
  
  if (avcodec_open2(avcodec_context, codec, NULL) < 0) {
  
      log(ERROR, "Failed to open codec \"%s\".", codec_name);
  
      exit(1);
  
  }
  
  
  
  /* Allocate corresponding frame */
  
  AVFrame* frame = av_frame_alloc();
  
  if (frame == NULL) {
  
      exit(1);
  
  }
  
  
  
  /* Copy necessary data for frame from avcodec_context */
  
  frame->format = avcodec_context->pix_fmt;
  
  frame->width = avcodec_context->width;
  
  frame->height = avcodec_context->height;
  
  
  
  /* Allocate actual backing data for frame */
  
  if (av_image_alloc(frame->data, frame->linesize, frame->width,
  
              frame->height, frame->format, 32) < 0) {
  
      exit(1);
  
  }
  
  
  
  /* open the output file, if the container needs it */
  
  if (!(container_format->flags & AVFMT_NOFILE)) {
  
      ret = avio_open(&container_format_context->pb, out_file, AVIO_FLAG_WRITE);
  
      if (ret < 0) {
  
          log(ERROR, "Error occurred while opening output file: %s\n",
  
                  av_err2str(ret));
  
          exit(1);
  
      }
  
  }
  
  
  
  /* write the stream header, if needed */
  
  ret = avformat_write_header(container_format_context, NULL);
  
  if (ret < 0) {
  
      log(ERROR, "Error occurred while writing output file header: %s\n",
  
                  av_err2str(ret));
  
  }

  The code to encode a frame is here :

  /* Init video packet */
  
  AVPacket packet;
  
  av_init_packet(&packet);
  
  
  
  /* Request that encoder allocate data for packet */
  
  packet.data = NULL;
  
  packet.size = 0;
  
  
  
  /* Write frame to video */
  
  int got_data;
  
  if (avcodec_encode_video2(avcontext, &packet, frame, &got_data) < 0) {
  
      log(WARNING, "Error encoding frame #%" PRId64,
  
              video_struct->next_pts);
  
      return -1;
  
  }
  
  
  
  /* Write corresponding data to file */
  
  if (got_data) {
  
      if (packet.pts != AV_NOPTS_VALUE) {
  
          packet.pts = av_rescale_q(packet.pts, video_struct->output_stream->codec->time_base, video_struct->output_stream->time_base);
  
      }
  
      if (packet.dts != AV_NOPTS_VALUE) {
  
          packet.dts = av_rescale_q(packet.dts, video_struct->output_stream->codec->time_base, video_struct->output_stream->time_base);
  
      }
  
      write_packet(video_struct, &packet, packet.size);
  
      av_packet_unref(&packet);
  
  }

  And the code to write the packet to the video stream :

  static int write_packet(video_struct* video, void* data, int size) {
  
  
  
  int ret;
  
  
  
  /* use AVStream is not null, otherwise write to output fd */
  
  AVPacket *pkt = (AVPacket*) data;
  
  pkt->stream_index = video->output_stream->index;
  
  ret = av_interleaved_write_frame(video->container_format_context, pkt);
  
  if (ret != 0) {
  
      return -1;
  
  }
  
  
  
  /* Data was written successfully */
  
  return ret;
  
  }

• How to compute the number of extra samples added by LAME or FFMPEG

  24 janvier 2018, par actuallyaswin

  I am attempting to build a MP3 decoder / parser in Python which supports files encoded by LAME or FFMPEG.

  My encoding shell script is shown here :

  #!/bin/bash
  
  for i in wav/*.wav; do
  
      i=${i##*/};
  
      lame --nores --strictly-enforce-ISO -t --cbr -b 64 -h "wav/${i}" "mpeg/lame/${i%.wav}.mp3";
  
      ffmpeg -i "wav/${i}" -codec:a libmp3lame -qscale:a 2 "mpeg/ffmpeg/${i%.wav}.mp3";
  
  done

  This scripts reads WAVE files located in ./wav/ and produces a controlled-bitrate MP3 of 64kbps in my ./mp3/lame/ directory, and a variable-bitrate MP3 of quality 2 in my ./mp3/ffmpeg/.

  I have written a Python script that iterates through both resultant MP3s, counting the number of frames and samples. Both the LAME and FFMPEG results are equivalent (in terms of frames and samples), but their binary files are different.

  The LAME/FFMPEG sample count was done by iterating through the binary MP3 files, locating and parsing the frame header, then using the MP3 spec to determine the number of samples per frame.

  • Number of MP3 data-frames : 112 (ignoring the Xing/Info first frame)
  • Number of output frames : 112*576 = 64512

  Here is a comparison of the sample count for a single 4-second input file :

  • Input WAV # of samples = 62996
  • Output LAME/FFMPEG # of samples = 64512
  • Difference = 1516

  I understand that according to the LAME FAQ file, resultant MP3 files are zero padded in the front and back to make sure the inverse MDCT is performed properly, but also because the windows overlap.

  What I can’t ascertain from the above FAQ, or from any previous StackOverflow post, is how to compute the number of artificially added samples. If I can be sure that all 1516 of these samples are zeros, and I can be sure of their position in the bytestream, I’d like to be able to confidently toss them out. Since there are 1516 "extra" samples and there are 576 samples per frame for a V2LIII encoding, then there must be more than two (but less than three) erroneous MPEG frames here.

  Is anyone here savvy enough with MPEG encoding/decoding to know how many samples are added, and in which frames those samples will be in ? In other words, will the first frame and last frame always contain blank data, or are there more frames ?

• Why does my video have an audio/mp4 mime type [duplicate]

  22 février 2018, par Tucker Peebles

  This question already has an answer here :

  • How do you safely identify whether an mp4 is an audio only or video file
    
    1 answer
    
  • Using ffprobe to check audio-only files
    
    3 answers
    

  I have a video file but it has a mime type of "audio/mp4". Why would that happen. The user who sent it to me said it came from a Flip video camera and was uploaded by usb to her computer.

  In my program, I’m checking for ’video/mp4’ mime type when I’m trying to determine if it’s a video, but I want to eventually support audio uploads too which would need to check ’audio/mp4’.

  Unfortunately, the video is a personal video of the user’s child so I don’t want to upload it here. Let me know if there is any information I can provide about the video that would help.