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• How to use libx264 to encode video in ffmpeg ?

  2 juin 2017, par wzjing

       I want to compress a video to smaller size, and if the video not encode by h264, then I transcode to h264. I have compiled ffmpeg 3.3.2 with libx264, When I using avcodec_send_frame(), it always return an error code, I use av_err2str(ret), and it said "Generic error in an external library" . Is this becouse my code is wrong or my lib is wrong? any way to crrect it or get more details?
       so here is my code of the transcode part, I do encode in the while loop, and the method to encode frame is encodeFrame(), the error happend in the last 3 line in method encodeFrame(), that means avcodec_send_frame(outCxt, outFrame) return whith an error:

  int transcode(){
      ......
  
      /******** Section 1 arguments configure ********/
      AVCodec outCodec = NULL;
      outCodec = avcodec_find_encoder_by_name("libx264");
      //tried avcodec_find_encoder(AV_CODEC_ID_H264) got same error
      if (outCodec == NULL) {
          LOGE("Unable to find encoder libx264");
          exit(1);
      }
      AVCodecContext *outCtx = NULL;
      outCtx = avcodec_alloc_context3(outCodec);
      outCtx->bit_rate = 40000;
      outCtx->width = inCtx->width;
      outCtx->height = inCtx->height;
      outCtx->time_base = (AVRational) {1, 25};
      outCtx->framerate = (AVRational) {25, 1};
      outCtx->gop_size = inCtx->gop_size;
      outCtx->max_b_frames = inCtx->max_b_frames;
      outCtx->pix_fmt = AV_PIX_FMT_YUV420P;
      outCtx->sample_fmt = inCtx->sample_fmt;
      outCtx->sample_rate = inCtx->sample_rate;
      outCtx->frame_size = inCtx->frame_size;
      outCtx->channel_layout = inCtx->channel_layout;
      outCtx->channels = inCtx->channels;
      av_opt_set(outCtx->priv_data, "preset", "slow", 0);
      ret = avcodec_open2(outCtx, outCodec, NULL);
      if (ret < 0)
          error(ret, "Error while inilizing outCtx instanceof AVCodecContext");
  
      /******** Section 2 init AVPacket and AVFrame ********/
      AVPacket *inPacket = NULL;
      inPacket = av_packet_alloc();
      av_init_packet(inPacket);
  
      AVFrame *vFrame;
      vFrame = av_frame_alloc();
      if (!vFrame)
          LOGE("Unable to allocate vFrame");
      vFrame->width = outCtx->width;
      vFrame->height = outCtx->height;
      vFrame->format = outCtx->pix_fmt;
  
      ret = av_frame_get_buffer(vFrame, 32);
      if (ret < 0)
          error(ret, "Unable to execute av_frame_get_buffer");
  
      /******** Section 3 start transcode ********/
      int k = 0;
      while (av_read_frame(inFormatContext, inPacket) == 0) {
          LOGI("--------------------- index:%d -------------------", k);
          resend:
          ret = avcodec_send_packet(inCtx, inPacket);
  
          if (ret == AVERROR(EOF))
              LOGW("> avcodec_send_packet: reache EOF");
          else if (ret == AVERROR(EAGAIN)) {
              LOGW("> avcodec_send_packet: last frame not been read");
              goto resend;
          }
          else if (ret == 0) {
              if (inPacket->stream_index == video_index) {
                  LOGD(">> receiving video frame");
                  avcodec_receive_frame(inCtx, vFrame);
                  encodeFrame(outFormatContext, outCtx, vFrame);
              } else {
                  //Currently, I igonred audio stream for now
                  LOGD(">> receiving audio frame");
                  avcodec_receive_frame(inCtx, aFrame);
              }
              av_packet_unref(inPacket);
              k++;
          } else {
              LOGE("> Error: avcodec_send_packet(%s)", av_err2str(ret));
              exit(1);
          }
  
      }
  
      ......
  }
  
  void encodeFrame(AVFormatContext *outFCxt, AVCodecContext *outCxt, AVFrame *outFrame) {
  
      LOGD(">> Encode Start");
      int ret;
  
      AVPacket *packet;
      packet = av_packet_alloc();
      ret = avcodec_send_frame(outCxt, outFrame);
      switch (ret) {
          case AVERROR(EAGAIN):
              LOGW(">>> avcodec_send_frame(last packet not received yet)");
              av_packet_unref(packet);
              encodeFrame(outFCxt, outCxt, outFrame);
              break;
          case AVERROR(EOF):
              LOGW(">>> avocdec_send_frame(reach end)");
              av_packet_unref(packet);
              break;
          case 0:
              av_frame_unref(outFrame);
              LOGD(">>> avcodec_send_frame(packet send success)");
              while (1){
                  ret = avcodec_receive_packet(outCxt, packet);
                  switch (ret) {
                      case AVERROR(EAGAIN):
                          LOGW(">>>> avcodec_receive_packet(no input)");
                          goto end;
                      case AVERROR(EOF):
                          LOGW(">>>> avcodec_receive_packet(reach end)");
                          goto end;
                      case 0:
                          LOGD(">>>> Writting frame");
                          ret = av_write_frame(outFCxt, packet);
                          if (ret == 0){
                              LOGD(">>>>> Write finished");
                              av_packet_unref(packet);
                          }
                          else {
                              LOGE(">>>>> Error:av_write_frame(%s)", av_err2str(ret));
                              exit(1);
                          }
                          goto end;
                      default:
                          LOGE(">>>> Error:avcode_receive_packet(%s)", av_err2str(ret));
                          exit(1);
  
                  }
                  end:
                  break;
              }
              break;
          default:
              /** I got a "Generic error in an external library" here **/
              LOGE(">>> Error: avcodec_send_frame(%s)", av_err2str(ret));
              exit(1);
      }
  

  }

• FFMPEG concat causing audio glitches in segments boundaries

  28 mai 2017, par Idan

  I am segmenting a video file to 10sec segments, transcoding each of them and then run a concat to merge them all back together. While the whole process works without errors and the final file is valid and playable, I've noticed some sort of glitches/interruptions in the audio stream when passing over the original segments boundaries.

  For example, in a 60sec video that was segmented to 10sec segments and stitched back again, the final output will show audio interruptions in 10/20/30/40/50 sec timestamps.

  For segmenting the files I use:

  ffmpeg -y -i $INPUT_FILE -c copy -flags -global_header -segment_time 10 -break_non_keyframes 0 -reset_timestamps 1 -segment_list file_segs.list -segment_list_type ffconcat -write_empty_segments 0 -segment_format mp4  -f segment file_seg-%d.mp4
  

  And for concat I use:

  ffmpeg -y -f concat -i file_segs.list -c copy -movflags +faststart file_video.mp4
  

  Transcoding is to libx264 / aac

  What might be the cause? Is there any way to work around this problem?

• ffmpeg -movflags +faststart in clipbucket video script

  22 mai 2017, par Spirog

  i am a little new at ffmpeg.

  Have a video script ( ClipBucket ) and would like to know how to add -movflags +faststart to the script to take advantage of this for .mp4 videos being converted.

  here is the code ffmpeg.class.php. Not sure where I can add this to work properly

  https://github.com/arslancb/clipbucket/tree/master/upload/includes/classes/conversion

  ffmpeg.class.php and conversion.class.php

  Seems to be the correct file(s), just not sure where to add it

  Thanks so much in advance.

  Spiro

• FFMPEG (libx264) "height not divisible by 2"

  16 mai 2017, par Andy Hin

  I am trying to encode a .mp4 video from a set of frames using FFMPEG using the libx264 codec.

  This is the command I am running:

  /usr/local/bin/ffmpeg -r 24 -i frame_%05d.jpg -vcodec libx264 -y -an video.mp4
  

  I sometimes get the following error:

  [libx264 @ 0xa3b85a0] height not divisible by 2 (520x369)
  

  After searching around a bit it seems that the issue has something to do with the scaling algorithm and can be fixed by adding a -vf argument.

  However, in my case I don't want to do any scaling. Ideally, I want to keep the dimensions exactly the same as the frames. Any advice? Is there some sort of aspect ratio that h264 enforces?

• ffmpeg x264 CPU core scaling

  12 mai 2017, par alex416

  Does anyone have any experence with this type of thing? Currently using a Ryzen 7 8c/16T CPU, and using a program called BES. Using 10% CPU, is almost no slower at transcoding from mkv to mp4 (both x264, i know i can just copy them) than using 100% cpu on all 16 threads. So, Since this is the case, What is the ideal core count for libx264? I mean, Does the same type of thing happen with 32 threads? Or 8 threads, etc.