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• FFmpeg library : webm (vorbis) audio to aac conversion

  3 janvier 2014, par taansari

  I have written a small program to convert webm (vorbis) audio to aac format, using FFmpeg libraries - C++ (on Windows using 32 bit Zeranoe FFmpeg builds). After writing this program, I find it is sometimes converting files as per expectation, and at other times, results in larger duration files, and audio playback is broken/awkward as well.

  This code appears to be working fine for mp3, which also uses FLTP format (same as vorbis), so technically both look similar.

  Please see below sample code I am using :

  ////////////////////////////////////////////////
  
  #include "stdafx.h"
  
  
  
  #include <iostream>
  
  #include <fstream>
  
  
  
  #include <string>
  
  #include <vector>
  
  #include <map>
  
  
  
  #include <deque>
  
  #include <queue>
  
  
  
  #include 
  
  #include 
  
  #include 
  
  #include 
  
  
  
  extern "C"
  
  {
  
  #include "libavcodec/avcodec.h"
  
  #include "libavformat/avformat.h"
  
  #include "libavdevice/avdevice.h"
  
  #include "libswscale/swscale.h"
  
  #include "libavutil/dict.h"
  
  #include "libavutil/error.h"
  
  #include "libavutil/opt.h"
  
  #include <libavutil></libavutil>fifo.h>
  
  #include <libavutil></libavutil>imgutils.h>
  
  #include <libavutil></libavutil>samplefmt.h>
  
  #include <libswresample></libswresample>swresample.h>
  
  }
  
  
  
  AVFormatContext*    fmt_ctx= NULL;
  
  int                    audio_stream_index = -1;
  
  AVCodecContext *    codec_ctx_audio = NULL;
  
  AVCodec*            codec_audio = NULL;
  
  AVFrame*            decoded_frame = NULL;
  
  uint8_t**            audio_dst_data = NULL;
  
  int                    got_frame = 0;
  
  int                    audiobufsize = 0;
  
  AVPacket            input_packet;
  
  int                    audio_dst_linesize = 0;
  
  int                    audio_dst_bufsize = 0;
  
  SwrContext *        swr = NULL;
  
  
  
  AVOutputFormat *    output_format = NULL ;
  
  AVFormatContext *    output_fmt_ctx= NULL;
  
  AVStream *            audio_st = NULL;
  
  AVCodec *            audio_codec = NULL;
  
  double                audio_pts = 0.0;
  
  AVFrame *            out_frame = avcodec_alloc_frame();
  
  
  
  int                    audio_input_frame_size = 0;
  
  
  
  uint8_t *            audio_data_buf = NULL;
  
  uint8_t *            audio_out = NULL;
  
  int                    audio_bit_rate;
  
  int                    audio_sample_rate;
  
  int                    audio_channels;
  
  
  
  int decode_packet();
  
  int open_audio_input(char* src_filename);
  
  int decode_frame();
  
  
  
  int open_encoder(char* output_filename);
  
  AVStream *add_audio_stream(AVFormatContext *oc, AVCodec **codec,
  
      enum AVCodecID codec_id);
  
  int open_audio(AVFormatContext *oc, AVCodec *codec, AVStream *st);
  
  void close_audio(AVFormatContext *oc, AVStream *st);
  
  void write_audio_frame(uint8_t ** audio_src_data, int audio_src_bufsize);
  
  
  
  int open_audio_input(char* src_filename)
  
  {
  
      int i =0;
  
      /* open input file, and allocate format context */
  
      if (avformat_open_input(&amp;fmt_ctx, src_filename, NULL, NULL) &lt; 0)
  
      {
  
          fprintf(stderr, "Could not open source file %s\n", src_filename);
  
          exit(1);
  
      }
  
  
  
      // Retrieve stream information
  
      if(avformat_find_stream_info(fmt_ctx, NULL)&lt;0)
  
          return -1; // Couldn&#39;t find stream information
  
  
  
      // Dump information about file onto standard error
  
      av_dump_format(fmt_ctx, 0, src_filename, 0);
  
  
  
      // Find the first video stream
  
      for(i=0; inb_streams; i++)
  
      {
  
          if(fmt_ctx->streams[i]->codec->codec_type==AVMEDIA_TYPE_AUDIO)
  
          {
  
              audio_stream_index=i;
  
              break;
  
          }
  
      }
  
      if ( audio_stream_index != -1 )
  
      {
  
          // Get a pointer to the codec context for the audio stream
  
          codec_ctx_audio=fmt_ctx->streams[audio_stream_index]->codec;
  
  
  
          // Find the decoder for the video stream
  
          codec_audio=avcodec_find_decoder(codec_ctx_audio->codec_id);
  
          if(codec_audio==NULL) {
  
              fprintf(stderr, "Unsupported audio codec!\n");
  
              return -1; // Codec not found
  
          }
  
  
  
          // Open codec
  
          AVDictionary *codecDictOptions = NULL;
  
          if(avcodec_open2(codec_ctx_audio, codec_audio, &amp;codecDictOptions)&lt;0)
  
              return -1; // Could not open codec
  
  
  
          // Set up SWR context once you&#39;ve got codec information
  
          swr = swr_alloc();
  
          av_opt_set_int(swr, "in_channel_layout",  codec_ctx_audio->channel_layout, 0);
  
          av_opt_set_int(swr, "out_channel_layout", codec_ctx_audio->channel_layout,  0);
  
          av_opt_set_int(swr, "in_sample_rate",     codec_ctx_audio->sample_rate, 0);
  
          av_opt_set_int(swr, "out_sample_rate",    codec_ctx_audio->sample_rate, 0);
  
          av_opt_set_sample_fmt(swr, "in_sample_fmt",  codec_ctx_audio->sample_fmt, 0);
  
          av_opt_set_sample_fmt(swr, "out_sample_fmt", AV_SAMPLE_FMT_S16,  0);
  
          swr_init(swr);
  
  
  
          // Allocate audio frame
  
          if ( decoded_frame == NULL ) decoded_frame = avcodec_alloc_frame();
  
          int nb_planes = 0;
  
          AVStream* audio_stream = fmt_ctx->streams[audio_stream_index];
  
          nb_planes = av_sample_fmt_is_planar(codec_ctx_audio->sample_fmt) ? codec_ctx_audio->channels : 1;
  
          int tempSize =  sizeof(uint8_t *) * nb_planes;
  
          audio_dst_data = (uint8_t**)av_mallocz(tempSize);
  
          if (!audio_dst_data)
  
          {
  
              fprintf(stderr, "Could not allocate audio data buffers\n");
  
          }
  
          else
  
          {
  
              for ( int i = 0 ; i &lt; nb_planes ; i ++ )
  
              {
  
                  audio_dst_data[i] = NULL;
  
              }
  
          }
  
      }
  
  }
  
  
  
  
  
  int decode_frame()
  
  {
  
      int rv = 0;
  
      got_frame = 0;
  
      if ( fmt_ctx == NULL  )
  
      {
  
          return rv;
  
      }
  
      int ret = 0;
  
      audiobufsize = 0;
  
      rv = av_read_frame(fmt_ctx, &amp;input_packet);
  
      if ( rv &lt; 0 )
  
      {
  
          return rv;
  
      }
  
      rv = decode_packet();
  
      // Free the input_packet that was allocated by av_read_frame
  
      av_free_packet(&amp;input_packet);
  
      return rv;
  
  }
  
  
  
  int decode_packet()
  
  {
  
      int rv = 0;
  
      int ret = 0;
  
  
  
      //audio stream?
  
      if(input_packet.stream_index == audio_stream_index)
  
      {
  
          /* decode audio frame */
  
          rv = avcodec_decode_audio4(codec_ctx_audio, decoded_frame, &amp;got_frame, &amp;input_packet);
  
          if (rv &lt; 0)
  
          {
  
              fprintf(stderr, "Error decoding audio frame\n");
  
              //return ret;
  
          }
  
          else
  
          {
  
              if (got_frame)
  
              {
  
                  if ( audio_dst_data[0] == NULL )
  
                  {
  
                       ret = av_samples_alloc(audio_dst_data, &amp;audio_dst_linesize, decoded_frame->channels,
  
                          decoded_frame->nb_samples, (AVSampleFormat)decoded_frame->format, 1);
  
                      if (ret &lt; 0)
  
                      {
  
                          fprintf(stderr, "Could not allocate audio buffer\n");
  
                          return AVERROR(ENOMEM);
  
                      }
  
                      /* TODO: extend return code of the av_samples_* functions so that this call is not needed */
  
                      audio_dst_bufsize = av_samples_get_buffer_size(NULL, audio_st->codec->channels,
  
                          decoded_frame->nb_samples, (AVSampleFormat)decoded_frame->format, 1);
  
  
  
                      //int16_t* outputBuffer = ...;
  
                      swr_convert(swr, audio_dst_data, out_frame->nb_samples,
  
                                  (const uint8_t **)(decoded_frame->data), decoded_frame->nb_samples);
  
                      //swr_convert( swr, audio_dst_data, out_frame->nb_samples, (const uint8_t**) decoded_frame->extended_data, decoded_frame->nb_samples );
  
                  }
  
                  /* copy audio data to destination buffer:
  
                  * this is required since rawaudio expects non aligned data */
  
                  //av_samples_copy(audio_dst_data, decoded_frame->data, 0, 0,
  
                  //    decoded_frame->nb_samples, decoded_frame->channels, (AVSampleFormat)decoded_frame->format);
  
              }
  
          }
  
      }
  
      return rv;
  
  }
  
  
  
  
  
  int open_encoder(char* output_filename )
  
  {
  
      int rv = 0;
  
  
  
      /* allocate the output media context */
  
      AVOutputFormat *opfmt = NULL;
  
  
  
      avformat_alloc_output_context2(&amp;output_fmt_ctx, opfmt, NULL, output_filename);
  
      if (!output_fmt_ctx) {
  
          printf("Could not deduce output format from file extension: using MPEG.\n");
  
          avformat_alloc_output_context2(&amp;output_fmt_ctx, NULL, "mpeg", output_filename);
  
      }
  
      if (!output_fmt_ctx) {
  
          rv = -1;
  
      }
  
      else
  
      {
  
          output_format = output_fmt_ctx->oformat;
  
      }
  
  
  
      /* Add the audio stream using the default format codecs
  
      * and initialize the codecs. */
  
      audio_st = NULL;
  
  
  
      if ( output_fmt_ctx )
  
      {
  
          if (output_format->audio_codec != AV_CODEC_ID_NONE)
  
          {
  
              audio_st = add_audio_stream(output_fmt_ctx, &amp;audio_codec, output_format->audio_codec);
  
          }
  
  
  
          /* Now that all the parameters are set, we can open the audio and
  
          * video codecs and allocate the necessary encode buffers. */
  
          if (audio_st)
  
          {
  
              rv = open_audio(output_fmt_ctx, audio_codec, audio_st);
  
              if ( rv &lt; 0 ) return rv;
  
          }
  
  
  
          av_dump_format(output_fmt_ctx, 0, output_filename, 1);
  
          /* open the output file, if needed */
  
          if (!(output_format->flags &amp; AVFMT_NOFILE))
  
          {
  
              if (avio_open(&amp;output_fmt_ctx->pb, output_filename, AVIO_FLAG_WRITE) &lt; 0) {
  
                  fprintf(stderr, "Could not open &#39;%s&#39;\n", output_filename);
  
                  rv = -1;
  
              }
  
              else
  
              {
  
                  /* Write the stream header, if any. */
  
                  if (avformat_write_header(output_fmt_ctx, NULL) &lt; 0)
  
                  {
  
                      fprintf(stderr, "Error occurred when opening output file\n");
  
                      rv = -1;
  
                  }
  
              }
  
          }
  
      }
  
  
  
      return rv;
  
  }
  
  
  
  AVStream *add_audio_stream(AVFormatContext *oc, AVCodec **codec,
  
      enum AVCodecID codec_id)
  
  {
  
      AVCodecContext *c;
  
      AVStream *st;
  
  
  
      /* find the audio encoder */
  
      *codec = avcodec_find_encoder(codec_id);
  
      if (!(*codec)) {
  
          fprintf(stderr, "Could not find codec\n");
  
          exit(1);
  
      }
  
  
  
      st = avformat_new_stream(oc, *codec);
  
      if (!st) {
  
          fprintf(stderr, "Could not allocate stream\n");
  
          exit(1);
  
      }
  
      st->id = 1;
  
  
  
      c = st->codec;
  
  
  
      /* put sample parameters */
  
      c->sample_fmt  = AV_SAMPLE_FMT_S16;
  
      c->bit_rate    = audio_bit_rate;
  
      c->sample_rate = audio_sample_rate;
  
      c->channels    = audio_channels;
  
  
  
      // some formats want stream headers to be separate
  
      if (oc->oformat->flags &amp; AVFMT_GLOBALHEADER)
  
          c->flags |= CODEC_FLAG_GLOBAL_HEADER;
  
  
  
      return st;
  
  }
  
  
  
  int open_audio(AVFormatContext *oc, AVCodec *codec, AVStream *st)
  
  {
  
      int ret=0;
  
      AVCodecContext *c;
  
  
  
      st->duration = fmt_ctx->duration;
  
      c = st->codec;
  
  
  
      /* open it */
  
      ret = avcodec_open2(c, codec, NULL) ;
  
      if ( ret &lt; 0)
  
      {
  
          fprintf(stderr, "could not open codec\n");
  
          return -1;
  
          //exit(1);
  
      }
  
  
  
      if (c->codec->capabilities &amp; CODEC_CAP_VARIABLE_FRAME_SIZE)
  
          audio_input_frame_size = 10000;
  
      else
  
          audio_input_frame_size = c->frame_size;
  
      int tempSize = audio_input_frame_size *
  
          av_get_bytes_per_sample(c->sample_fmt) *
  
          c->channels;
  
      return ret;
  
  }
  
  
  
  void close_audio(AVFormatContext *oc, AVStream *st)
  
  {
  
      avcodec_close(st->codec);
  
  }
  
  
  
  void write_audio_frame(uint8_t ** audio_src_data, int audio_src_bufsize)
  
  {
  
      AVFormatContext *oc = output_fmt_ctx;
  
      AVStream *st = audio_st;
  
      if ( oc == NULL || st == NULL ) return;
  
      AVCodecContext *c;
  
      AVPacket pkt = { 0 }; // data and size must be 0;
  
      int got_packet;
  
  
  
      av_init_packet(&amp;pkt);
  
      c = st->codec;
  
  
  
      out_frame->nb_samples = audio_input_frame_size;
  
      int buf_size =         audio_src_bufsize *
  
          av_get_bytes_per_sample(c->sample_fmt) *
  
          c->channels;
  
      avcodec_fill_audio_frame(out_frame, c->channels, c->sample_fmt,
  
          (uint8_t *) *audio_src_data,
  
          buf_size, 1);
  
      avcodec_encode_audio2(c, &amp;pkt, out_frame, &amp;got_packet);
  
      if (!got_packet)
  
      {
  
      }
  
      else
  
      {
  
          if (pkt.pts != AV_NOPTS_VALUE)
  
              pkt.pts =  av_rescale_q(pkt.pts, st->codec->time_base, st->time_base);
  
          if (pkt.dts != AV_NOPTS_VALUE)
  
              pkt.dts = av_rescale_q(pkt.dts, st->codec->time_base, st->time_base);
  
          if ( c &amp;&amp; c->coded_frame &amp;&amp; c->coded_frame->key_frame)
  
              pkt.flags |= AV_PKT_FLAG_KEY;
  
  
  
           pkt.stream_index = st->index;
  
          pkt.flags |= AV_PKT_FLAG_KEY;
  
          /* Write the compressed frame to the media file. */
  
          if (av_interleaved_write_frame(oc, &amp;pkt) != 0)
  
          {
  
              fprintf(stderr, "Error while writing audio frame\n");
  
              exit(1);
  
          }
  
      }
  
      av_free_packet(&amp;pkt);
  
  }
  
  
  
  
  
  void write_delayed_frames(AVFormatContext *oc, AVStream *st)
  
  {
  
      AVCodecContext *c = st->codec;
  
      int got_output = 0;
  
      int ret = 0;
  
      AVPacket pkt;
  
      pkt.data = NULL;
  
      pkt.size = 0;
  
      av_init_packet(&amp;pkt);
  
      int i = 0;
  
      for (got_output = 1; got_output; i++)
  
      {
  
          ret = avcodec_encode_audio2(c, &amp;pkt, NULL, &amp;got_output);
  
          if (ret &lt; 0)
  
          {
  
              fprintf(stderr, "error encoding frame\n");
  
              exit(1);
  
          }
  
          static int64_t tempPts = 0;
  
          static int64_t tempDts = 0;
  
          /* If size is zero, it means the image was buffered. */
  
          if (got_output)
  
          {
  
              if (pkt.pts != AV_NOPTS_VALUE)
  
                  pkt.pts =  av_rescale_q(pkt.pts, st->codec->time_base, st->time_base);
  
              if (pkt.dts != AV_NOPTS_VALUE)
  
                  pkt.dts = av_rescale_q(pkt.dts, st->codec->time_base, st->time_base);
  
              if ( c &amp;&amp; c->coded_frame &amp;&amp; c->coded_frame->key_frame)
  
                  pkt.flags |= AV_PKT_FLAG_KEY;
  
  
  
              pkt.stream_index = st->index;
  
              /* Write the compressed frame to the media file. */
  
              ret = av_interleaved_write_frame(oc, &amp;pkt);
  
          }
  
          else
  
          {
  
              ret = 0;
  
          }
  
          av_free_packet(&amp;pkt);
  
      }
  
  }
  
  
  
  int main(int argc, char **argv)
  
  {
  
      /* register all formats and codecs */
  
      av_register_all();
  
      avcodec_register_all();
  
      avformat_network_init();
  
      avdevice_register_all();
  
      int i =0;
  
      int ret=0;
  
      char src_filename[90] = "test_a.webm";
  
      char dst_filename[90] = "output.aac";
  
      open_audio_input(src_filename);
  
      if ( codec_ctx_audio->bit_rate == 0 ) codec_ctx_audio->bit_rate = 112000;
  
      audio_bit_rate        = codec_ctx_audio->bit_rate;
  
      audio_sample_rate    = codec_ctx_audio->sample_rate;
  
      audio_channels        = codec_ctx_audio->channels;
  
      open_encoder( dst_filename );
  
      while(1)
  
      {
  
          int rv = decode_frame();
  
          if ( rv &lt; 0 )
  
          {
  
              break;
  
          }
  
  
  
          if (audio_st)
  
          {
  
              audio_pts = (double)audio_st->pts.val * audio_st->time_base.num /
  
                  audio_st->time_base.den;
  
          }
  
          else
  
          {
  
              audio_pts = 0.0;
  
          }
  
          if ( codec_ctx_audio )
  
          {
  
              if ( got_frame)
  
              {
  
                  write_audio_frame( audio_dst_data, audio_dst_bufsize );
  
              }
  
          }
  
          if ( audio_dst_data[0] )
  
          {
  
              av_freep(&amp;audio_dst_data[0]);
  
              audio_dst_data[0] = NULL;
  
          }
  
          printf("\naudio_pts: %.3f", audio_pts);
  
      }
  
      while(1)
  
      {
  
          if ( audio_dst_data &amp;&amp; audio_dst_data[0] )
  
          {
  
              av_freep(&amp;audio_dst_data[0]);
  
              audio_dst_data[0] = NULL;
  
          }
  
          ret = av_samples_alloc(audio_dst_data, NULL, codec_ctx_audio->channels,
  
              decoded_frame->nb_samples, AV_SAMPLE_FMT_S16, 0);
  
          ret = swr_convert(swr, audio_dst_data, out_frame->nb_samples,NULL, 0);
  
          if ( ret &lt;= 0 ) break;
  
          write_audio_frame( audio_dst_data, audio_dst_bufsize );
  
      }
  
      write_delayed_frames( output_fmt_ctx, audio_st );
  
      av_write_trailer(output_fmt_ctx);
  
      close_audio( output_fmt_ctx, audio_st);
  
      swr_free(&amp;swr);
  
      avcodec_free_frame(&amp;out_frame);
  
      getch();
  
      return 0;
  
  }
  
  </queue></deque></map></vector></string></fstream></iostream>

  "test_a.webm" input file results in longer duration (40 second output), and if I change it to "jet.webm", it is converted fine.

  Both input files are approximately 18 second duration.

  For reference, these files can be downloaded from links below :

  http://www.filedropper.com/testa ,
  http://www.filedropper.com/jet

  Alternatively, they are zipped and uploaded elsewhere as well :

  http://www.files.com/shared/52c3eefe990ea/test_audio_files.zip

  Could someone kindly guide on what I am doing wrong here ?

  Thanks in advance...

  p.s. These files are taken/extracted from different online sources/demos.

  Edit 2-1-14 : After debugging, I can see audio_pts variable is being populated incorrectly. It relies on audio_st->pts.val, which is automatically calculated upon calling av_interleaved_write_frame() function. I cannot step inside av_interleaved_write_frame() function since I am on Windows, using libav dlls/libs.

  So,

  For jet.webm file (whose conversion is fine), audio_st->pts.val goes till maximum : 1665567, and audio_pts becomes :

  1665567*1/90000 = 18.5063

  For test_a.webm file (whose conversion is bad), audio_st->pts.val goes till maximum : 3606988, and audio_pts becomes :

  3606988*1/90000 = 40.0776

  • reference line : audio_pts = (double)audio_st->pts.val * audio_st->time_base.num /
    audio_st->time_base.den ;

  Since PTS is very off, it shouldn't be playing fine logically as well. But I cannot say av_interleaved_write_frame() function is doing it wrong ; surely something cleaner can be managed on my end.

  Edit 3-1-14 : Discovered one more thing : while reading jet.webm file, decoded frame's nb_sample are always 1024 (except for 1st frame : 576), but in case of test_a.webm file, nb_samples are either 1024, or 128, with exceptions of 576 (less frequent). If I ignore writing of frame when nb_sample is 128, I get approximately same file length in the end, but you can hear bits and pieces are missing here and there.

  How can I deal with this ?

• ffmpeg commad with colorchannelmixer

  12 décembre 2013, par bindal

  I am using following command :
  command = "ffmpeg -y -i "+strFinalPath+" -vf colorchannelmixer=.3 :.4 :.3:0 :.3 :.4 :.3:0 :.3 :.4 :.3 "+strRotatePath ;

  But this command exit with error code 1

  is there any mistake in this command

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• ffmprg commad with colorchannelmixer

  12 décembre 2013, par bindal

  I am using following command :
  command = "ffmpeg -y -i "+strFinalPath+" -vf colorchannelmixer=.3 :.4 :.3:0 :.3 :.4 :.3:0 :.3 :.4 :.3 "+strRotatePath ;

  But this command exit with error code 1

  is there any mistake in this command

  Please suggest me

  Thanks