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• parser not found for codec wmav2

  5 décembre 2011, par HashCoder

  I am getting a warning when I run the below command.

  Warning : [asf @ 01C787A0] parser not found for codec wmav2, packets or times may be inval
  id.

  I am using the latest ffmpeg.exe, did I miss any parameters. Any suggestions please.

  ffmpeg -i Assets\Logitech_webcam_on_PC.wmv -sameq -f swf -y -an -s 640x360 MySlide.swf
  
  ffmpeg version N-35295-gb55dd10, Copyright (c) 2000-2011 the FFmpeg developers
  
    built on Nov 30 2011 00:52:52 with gcc 4.6.2
  
    configuration: --enable-gpl --enable-version3 --disable-w32threads --enable-ru
  
  ntime-cpudetect --enable-avisynth --enable-bzlib --enable-frei0r --enable-libope
  
  ncore-amrnb --enable-libopencore-amrwb --enable-libfreetype --enable-libgsm --en
  
  able-libmp3lame --enable-libopenjpeg --enable-librtmp --enable-libschroedinger -
  
  -enable-libspeex --enable-libtheora --enable-libvo-aacenc --enable-libvo-amrwben
  
  c --enable-libvorbis --enable-libvpx --enable-libx264 --enable-libxavs --enable-
  
  libxvid --enable-zlib
  
    libavutil    51. 29. 1 / 51. 29. 1
  
    libavcodec   53. 39. 1 / 53. 39. 1
  
    libavformat  53. 22. 0 / 53. 22. 0
  
    libavdevice  53.  4. 0 / 53.  4. 0
  
    libavfilter   2. 50. 0 /  2. 50. 0
  
    libswscale    2.  1. 0 /  2.  1. 0
  
    libpostproc  51.  2. 0 / 51.  2. 0
  
  [asf @ 01C787A0] parser not found for codec wmav2, packets or times may be inval
  
  id.
  
  
  
  Seems stream 1 codec frame rate differs from container frame rate: 1000.00 (1000
  
  /1) -> 0.08 (1/12)
  
  Input #0, asf, from 'Assets\Logitech_webcam_on_PC.wmv':
  
    Metadata:
  
      WMFSDKVersion   : 11.0.5721.5265
  
      WMFSDKNeeded    : 0.0.0.0000
  
      IsVBR           : 1
  
      VBR Peak        : 50500.0000
  
      Buffer Average  : 66550.0000
  
    Duration: 00:00:36.22, start: 0.000000, bitrate: 497 kb/s
  
      Stream #0:0(eng): Audio: wmav2 (a[1][0][0] / 0x0161), 32000 Hz, 1 channels,
  
  s16, 20 kb/s
  
      Stream #0:1(eng): Video: wmv2 (WMV2 / 0x32564D57), yuv420p, 320x180, 422 kb/
  
  s, 0.08 tbr, 1k tbn, 1k tbc
  
  [buffer @ 02AA9760] w:320 h:180 pixfmt:yuv420p tb:1/1000000 sar:0/1 sws_param:
  
  [scale @ 02AA9A80] w:320 h:180 fmt:yuv420p -> w:640 h:360 fmt:yuv420p flags:0x4
  
  Output #0, swf, to 'MySlide.swf':
  
    Metadata:
  
      WMFSDKVersion   : 11.0.5721.5265
  
      WMFSDKNeeded    : 0.0.0.0000
  
      IsVBR           : 1
  
      VBR Peak        : 50500.0000
  
      Buffer Average  : 66550.0000
  
      encoder         : Lavf53.22.0
  
      Stream #0:0(eng): Video: flv1, yuv420p, 640x360, q=2-31, 200 kb/s, 90k tbn,
  
  0.08 tbc
  
  Stream mapping:
  
    Stream #0:1 -> #0:0 (wmv2 -> flv)
  
  Press [q] to stop, [?] for help
  
  frame=    4 fps=  0 q=0.0 size=      97kB time=00:00:48.00 bitrate=  16.6kbits/s
  
  frame=    5 fps=  0 q=0.0 Lsize=     111kB time=00:01:00.00 bitrate=  15.2kbits/
  
  s dup=0 drop=599
  
  video:111kB audio:0kB global headers:0kB muxing overhead 0.128646%

• How to concat mp4 files using libffmpeg in c program ?

  1er août 2013, par chichien

  I know ffmpeg command line is easy, but how to programmatically implement？ I'm not good at this，here is some code from internet, it is used to convert .mp4 to .ts，and i made some changes，but the audio stream problem persists：

  #include 
  
  #include 
  
  #include 
  
  #include 
  
  
  
  #include "libavformat/avformat.h"
  
  #include "libavcodec/avcodec.h"
  
  #include "libavutil/avutil.h"
  
  #include "libavutil/rational.h"
  
  #include "libavdevice/avdevice.h"
  
  #include "libavutil/mathematics.h"
  
  #include "libswscale/swscale.h"
  
  
  
  static AVStream* add_output_stream(AVFormatContext* output_format_context, AVStream* input_stream)
  
  {
  
  
  
      AVCodecContext* input_codec_context = NULL;
  
      AVCodecContext* output_codec_context = NULL;
  
  
  
      AVStream* output_stream = NULL;
  
      output_stream = av_new_stream(output_format_context, 0);
  
      if (!output_stream)
  
      {
  
          printf("Call av_new_stream function failed\n");
  
          return NULL;
  
      }
  
  
  
      input_codec_context = input_stream->codec;
  
      output_codec_context = output_stream->codec;
  
  
  
      output_codec_context->codec_id = input_codec_context->codec_id;
  
      output_codec_context->codec_type = input_codec_context->codec_type;
  
      output_codec_context->codec_tag = input_codec_context->codec_tag;
  
      output_codec_context->bit_rate = input_codec_context->bit_rate;
  
      output_codec_context->extradata = input_codec_context->extradata;
  
      output_codec_context->extradata_size = input_codec_context->extradata_size;
  
  
  
      if (av_q2d(input_codec_context->time_base) * input_codec_context->ticks_per_frame > av_q2d(input_stream->time_base) && av_q2d(input_stream->time_base) < 1.0 / 1000)
  
      {
  
          output_codec_context->time_base = input_codec_context->time_base;
  
          output_codec_context->time_base.num *= input_codec_context->ticks_per_frame;
  
      }
  
      else
  
      {
  
          output_codec_context->time_base = input_stream->time_base;
  
      }
  
      switch (input_codec_context->codec_type)
  
      {
  
      case AVMEDIA_TYPE_AUDIO:
  
          output_codec_context->channel_layout = input_codec_context->channel_layout;
  
          output_codec_context->sample_rate = input_codec_context->sample_rate;
  
          output_codec_context->channels = input_codec_context->channels;
  
          output_codec_context->frame_size = input_codec_context->frame_size;
  
          if ((input_codec_context->block_align == 1 && input_codec_context->codec_id == CODEC_ID_MP3) || input_codec_context->codec_id == CODEC_ID_AC3)
  
          {
  
              output_codec_context->block_align = 0;
  
          }
  
          else
  
          {
  
              output_codec_context->block_align = input_codec_context->block_align;
  
          }
  
          break;
  
      case AVMEDIA_TYPE_VIDEO:
  
          output_codec_context->pix_fmt = input_codec_context->pix_fmt;
  
          output_codec_context->width = input_codec_context->width;
  
          output_codec_context->height = input_codec_context->height;
  
          output_codec_context->has_b_frames = input_codec_context->has_b_frames;
  
          if (output_format_context->oformat->flags & AVFMT_GLOBALHEADER)
  
          {
  
              output_codec_context->flags |= CODEC_FLAG_GLOBAL_HEADER;
  
          }
  
          break;
  
      default:
  
          break;
  
      }
  
  
  
      return output_stream;
  
  }
  
  
  
  //[[** from ffmpeg.c
  
  static void write_frame(AVFormatContext *s, AVPacket *pkt, AVCodecContext *avctx, AVBitStreamFilterContext *bsfc){
  
      int ret;
  
  
  
      while(bsfc){
  
          AVPacket new_pkt= *pkt;
  
          int a= av_bitstream_filter_filter(bsfc, avctx, NULL,
  
                                            &new_pkt.data, &new_pkt.size,
  
                                            pkt->data, pkt->size,
  
                                            pkt->flags & AV_PKT_FLAG_KEY);
  
          if(a>0){
  
              av_free_packet(pkt);
  
              new_pkt.destruct= av_destruct_packet;
  
          } else if(a<0){
  
              fprintf(stderr, "%s failed for stream %d, codec %s\n",
  
                      bsfc->filter->name, pkt->stream_index,
  
                      avctx->codec ? avctx->codec->name : "copy");
  
              //print_error("", a);
  
              //if (exit_on_error)
  
              //    ffmpeg_exit(1);
  
          }
  
          *pkt= new_pkt;
  
  
  
          bsfc= bsfc->next;
  
      }
  
  
  
      ret= av_interleaved_write_frame(s, pkt);
  
      if(ret < 0){
  
          //print_error("av_interleaved_write_frame()", ret);
  
          fprintf(stderr, "av_interleaved_write_frame(%d)\n", ret);
  
          exit(1);
  
      }
  
  }
  
  //]]**
  
  
  
  int main(int argc, char* argv[])
  
  {
  
  
  
      const char* input;
  
      const char* output;
  
      const char* output_prefix = NULL;
  
      char* segment_duration_check = 0;
  
      const char* index = NULL;
  
      char* tmp_index = NULL;
  
      const char* http_prefix = NULL;
  
      long max_tsfiles = NULL;
  
      double prev_segment_time = 0;
  
      double segment_duration = 0;
  
  
  
      AVInputFormat* ifmt = NULL;
  
      AVOutputFormat* ofmt = NULL;
  
      AVFormatContext* ic = NULL;
  
      AVFormatContext* oc = NULL;
  
      AVStream* video_st = NULL;
  
      AVStream* audio_st = NULL;
  
      AVCodec* codec = NULL;
  
      AVDictionary* pAVDictionary = NULL;
  
  
  
      long frame_count = 0;
  
  
  
      if (argc != 3) {
  
          fprintf(stderr, "Usage: %s inputfile outputfile\n", argv[0]);
  
          exit(1);
  
      }
  
  
  
      input = argv[1];
  
      output = argv[2];
  
  
  
      av_register_all();
  
  
  
      char szError[256] = {0};
  
      int nRet = avformat_open_input(&ic, input, ifmt, &pAVDictionary);
  
      if (nRet != 0)
  
      {
  
          av_strerror(nRet, szError, 256);
  
          printf(szError);
  
          printf("\n");
  
          printf("Call avformat_open_input function failed!\n");
  
          return 0;
  
      }
  
  
  
      if (av_find_stream_info(ic) < 0)
  
      {
  
          printf("Call av_find_stream_info function failed!\n");
  
          return 0;
  
      }
  
  
  
      ofmt = av_guess_format("mpegts", NULL, NULL);
  
      if (!ofmt)
  
      {
  
          printf("Call av_guess_format function failed!\n");
  
          return 0;
  
      }
  
  
  
      oc = avformat_alloc_context();
  
      if (!oc)
  
      {
  
          printf("Call av_guess_format function failed!\n");
  
          return 0;
  
      }
  
      oc->oformat = ofmt;
  
  
  
      int video_index = -1, audio_index = -1;
  
      for (unsigned int i = 0; i < ic->nb_streams && (video_index < 0 || audio_index < 0); i++)
  
      {
  
          switch (ic->streams[i]->codec->codec_type)
  
          {
  
          case AVMEDIA_TYPE_VIDEO:
  
              video_index = i;
  
              ic->streams[i]->discard = AVDISCARD_NONE;
  
              video_st = add_output_stream(oc, ic->streams[i]);
  
              break;
  
          case AVMEDIA_TYPE_AUDIO:
  
              audio_index = i;
  
              ic->streams[i]->discard = AVDISCARD_NONE;
  
              audio_st = add_output_stream(oc, ic->streams[i]);
  
              break;
  
          default:
  
              ic->streams[i]->discard = AVDISCARD_ALL;
  
              break;
  
          }
  
      }
  
      codec = avcodec_find_decoder(video_st->codec->codec_id);
  
      if (codec == NULL)
  
      {
  
          printf("Call avcodec_find_decoder function failed!\n");
  
          return 0;
  
      }
  
  
  
      if (avcodec_open(video_st->codec, codec) < 0)
  
      {
  
          printf("Call avcodec_open function failed !\n");
  
          return 0;
  
      }
  
  
  
      if (avio_open(&oc->pb, output, AVIO_FLAG_WRITE) < 0)
  
      {
  
          return 0;
  
      }
  
  
  
      if (avformat_write_header(oc, &pAVDictionary))
  
      {
  
          printf("Call avformat_write_header function failed.\n");
  
          return 0;
  
      }
  
  
  
      //[[++
  
      AVBitStreamFilterContext *bsfc = av_bitstream_filter_init("h264_mp4toannexb");
  
      //AVBitStreamFilterContext *absfc = av_bitstream_filter_init("aac_adtstoasc");
  
      if (!bsfc) {
  
          fprintf(stderr, "bsf init error!\n");
  
          return -1;
  
      }
  
      //]]++
  
  
  
      int decode_done = 0;
  
      do
  
      {
  
          double segment_time = 0;
  
          AVPacket packet;
  
  
  
          decode_done = av_read_frame(ic, &packet);
  
          if (decode_done < 0)
  
              break;
  
  
  
          if (av_dup_packet(&packet) < 0)
  
          {
  
              printf("Call av_dup_packet function failed\n");
  
              av_free_packet(&packet);
  
              break;
  
          }
  
  
  
          //[[**
  
          if (packet.stream_index == audio_index) {
  
              segment_time = (double)audio_st->pts.val * audio_st->time_base.num / audio_st->time_base.den;
  
              nRet = av_interleaved_write_frame(oc, &packet);   
  
          } else if (packet.stream_index == video_index) {
  
              if (packet.flags & AV_PKT_FLAG_KEY) {
  
                  segment_time = (double)video_st->pts.val * video_st->time_base.num / video_st->time_base.den;
  
              } else {
  
                  segment_time = prev_segment_time;
  
              }
  
              //nRet = av_interleaved_write_frame(oc, &packet);  
  
              write_frame(oc, &packet, video_st->codec, bsfc);
  
          }
  
          //]]**
  
  
  
          if (nRet < 0)
  
          {
  
              printf("Call av_interleaved_write_frame function failed: %d\n", nRet);
  
          }
  
          else if (nRet > 0)
  
          {
  
              printf("End of stream requested\n");
  
              av_free_packet(&packet);
  
              break;
  
          }
  
          av_free_packet(&packet);
  
          frame_count++;
  
      }while(!decode_done);
  
  
  
      av_write_trailer(oc);
  
      printf("frame_count = %d\n", frame_count);
  
  
  
      av_bitstream_filter_close(bsfc);  
  
      avcodec_close(video_st->codec);
  
      for(unsigned int k = 0; k < oc->nb_streams; k++)
  
      {
  
          av_freep(&oc->streams[k]->codec);
  
          av_freep(&oc->streams[k]);
  
      }
  
      av_free(oc);
  
      //getchar();
  
      return 0;
  
  }

  Compile this code, to got an executable file named muxts, and then :

  $ ./muxts vid1.mp4 vid1.ts

  No error message printed，but the audio stream was unsynchronized and noise。Check the .ts file using ffmpeg :

  $ ffmpeg -i vid1.ts
  
  ffmpeg version 0.8.14-tessus, Copyright (c) 2000-2013 the FFmpeg developers
  
    built on Jul 29 2013 17:05:18 with llvm_gcc 4.2.1 (Based on Apple Inc. build 5658) (LLVM build 2336.1.00)
  
    configuration: --prefix=/usr/local --arch=x86_64 --as=yasm --extra-version=tessus --enable-gpl --enable-nonfree --enable-version3 --disable-ffplay --enable-libvorbis --enable-libmp3lame --enable-libx264 --enable-libxvid --enable-bzlib --enable-zlib --enable-postproc --enable-filters --enable-runtime-cpudetect --enable-debug=3 --disable-optimizations
  
    libavutil    51.  9. 1 / 51.  9. 1
  
    libavcodec   53.  8. 0 / 53.  8. 0
  
    libavformat  53.  5. 0 / 53.  5. 0
  
    libavdevice  53.  1. 1 / 53.  1. 1
  
    libavfilter   2. 23. 0 /  2. 23. 0
  
    libswscale    2.  0. 0 /  2.  0. 0
  
    libpostproc  51.  2. 0 / 51.  2. 0
  
  
  
  Seems stream 0 codec frame rate differs from container frame rate: 180000.00 (180000/1) -> 90000.00 (180000/2)
  
  Input #0, mpegts, from 'vid1.ts':
  
    Duration: 00:00:03.75, start: 0.000000, bitrate: 3656 kb/s
  
    Program 1
  
      Metadata:
  
        service_name    : Service01
  
        service_provider: FFmpeg
  
      Stream #0.0[0x100]: Video: h264 (Baseline), yuv420p, 640x480, 90k tbr, 90k tbn, 180k tbc
  
      Stream #0.1[0x101]: Audio: aac, 48000 Hz, mono, s16, 190 kb/s
  
  At least one output file must be specified

  What should i do ?

  If this issue fixed , how can i concat multi .ts files into single .mp4 file ?

• Revision 38fa487164 : Shortcut 8x8/16x16 inverse 2D-DCT This commit brought back the shortcut impleme

  27 juillet 2013, par Jingning Han

  Changed Paths :
   Modify /vp9/decoder/vp9_idct_blk.c

  
   Modify /vp9/encoder/vp9_encodemb.c

  
  
  Shortcut 8x8/16x16 inverse 2D-DCT

  This commit brought back the shortcut implementation of 8x8/16x16
  inverse 2D-DCT. When the eob <= 10, it skips the inverse transform
  operations on row 4:7/4:15 in the first round. For bus_cif at 1000
  kbps, this provides about 2% speed-up at speed 0.

  Change-Id : I453e2d72956467d75be4ad8c04b4482ab889d572