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• ffmpeg : images to 29.97fps mpeg2, audio not sync [migrated]

  21 novembre 2011, par Andy Le

  I have spent a lot of time on this issue. Hope someone can help.

  I want to convert 3147 images + ac3 audio file into an mpeg2 video at 29.97fps (about 1m45s). My command :

  ~/ffmpeg/ffmpeg/ffmpeg -loop_input -t 105 -i v%4d.tga -i final.ac3 -vcodec mpeg2video -qscale 1 -s 400x400 -r 30000/1001 -acodec copy -y out.mpeg 2> out.txt

  However, the audio file ends before the frame sequence. Which means the video is slower then audio.

  I checked the output file with imageinfo and see :

  General
  
  Complete name                    : out.mpeg
  
  Format                           : MPEG-PS
  
  File size                        : 7.18 MiB
  
  Duration                         : 1mn 44s
  
  Overall bit rate                 : 574 Kbps
  
  
  
  Video
  
  ID                               : 224 (0xE0)
  
  Format                           : MPEG Video
  
  Format version                   : Version 2
  
  Format profile                   : Main@Main
  
  Format settings, BVOP            : No
  
  Format settings, Matrix          : Default
  
  Format_Settings_GOP              : M=1, N=12
  
  Duration                         : 1mn 44s
  
  Bit rate mode                    : Variable
  
  Bit rate                         : 103 Kbps
  
  Width                            : 400 pixels
  
  Height                           : 400 pixels
  
  Display aspect ratio             : 1.000
  
  Frame rate                       : 29.970 fps
  
  Resolution                       : 8 bits
  
  Colorimetry                      : 4:2:0
  
  Scan type                        : Progressive
  
  Bits/(Pixel*Frame)               : 0.021
  
  Stream size                      : 1.29 MiB (18%)
  
  
  
  Audio
  
  ID                               : 128 (0x80)
  
  Format                           : AC-3
  
  Format/Info                      : Audio Coding 3
  
  Duration                         : 1mn 44s
  
  Bit rate mode                    : Constant
  
  Bit rate                         : 448 Kbps
  
  Channel(s)                       : 6 channels
  
  Channel positions                : Front: L C R, Side: L R, LFE
  
  Sampling rate                    : 44.1 KHz
  
  Stream size                      : 5.61 MiB (78%)

  The log from ffmpeg shows many duplicate frames. But I don't know how to get rid of that.

  -loop_input is deprecated, use -loop 1
  
  [image2 @ 0x9c17a80] max_analyze_duration 5000000 reached at 5000000
  
  Input #0, image2, from 'v%4d.tga':
  
    Duration: 00:02:05.88, start: 0.000000, bitrate: N/A
  
      Stream #0:0: Video: targa, bgr24, 400x400, 25 fps, 25 tbr, 25 tbn, 25 tbc
  
  -loop_input is deprecated, use -loop 1
  
  [ac3 @ 0x9ca5420] max_analyze_duration 5000000 reached at 5014400
  
  [ac3 @ 0x9ca5420] Estimating duration from bitrate, this may be inaccurate
  
  Input #1, ac3, from 'Final.ac3':
  
    Duration: 00:20:10.68, start: 0.000000, bitrate: 447 kb/s
  
      Stream #1:0: Audio: ac3, 44100 Hz, 5.1(side), s16, 448 kb/s
  
  Incompatible pixel format 'bgr24' for codec 'mpeg2video', auto-selecting format 'yuv420p'
  
  [buffer @ 0x9c1e060] w:400 h:400 pixfmt:bgr24 tb:1/1000000 sar:0/1 sws_param:
  
  [buffersink @ 0x9dd56c0] auto-inserting filter 'auto-inserted scale 0' between the filter 'src' and the filter 'out'
  
  [scale @ 0x9c178e0] w:400 h:400 fmt:bgr24 -> w:400 h:400 fmt:yuv420p flags:0x4
  
  [mpeg @ 0x9d58060] VBV buffer size not set, muxing may fail
  
  Output #0, mpeg, to 'out.mpeg':
  
    Metadata:
  
      encoder         : Lavf53.21.0
  
      Stream #0:0: Video: mpeg2video, yuv420p, 400x400, q=2-31, 200 kb/s, 90k tbn, 29.97 tbc
  
      Stream #0:1: Audio: ac3, 44100 Hz, 5.1(side), 448 kb/s
  
  Stream mapping:
  
    Stream #0:0 -> #0:0 (targa -> mpeg2video)
  
    Stream #1:0 -> #0:1 (copy)
  
  Press [q] to stop, [?] for help
  
  frame=  267 fps=  0 q=1.0 size=     564kB time=00:00:08.87 bitrate= 520.6kbits/s dup=43 drop=0    
  
  frame=  544 fps=542 q=1.0 size=    1186kB time=00:00:18.11 bitrate= 536.2kbits/s dup=89 drop=0    
  
  frame=  821 fps=546 q=1.0 size=    1818kB time=00:00:27.36 bitrate= 544.3kbits/s dup=135 drop=0    
  
  frame= 1098 fps=548 q=1.0 size=    2444kB time=00:00:36.60 bitrate= 547.0kbits/s dup=181 drop=0    
  
  frame= 1376 fps=549 q=1.0 size=    3072kB time=00:00:45.87 bitrate= 548.5kbits/s dup=227 drop=0    
  
  frame= 1653 fps=550 q=1.0 size=    3700kB time=00:00:55.12 bitrate= 549.9kbits/s dup=273 drop=0    
  
  frame= 1930 fps=550 q=1.0 size=    4326kB time=00:01:04.36 bitrate= 550.6kbits/s dup=319 drop=0    
  
  frame= 2208 fps=551 q=1.0 size=    4960kB time=00:01:13.64 bitrate= 551.8kbits/s dup=365 drop=0    
  
  frame= 2462 fps=546 q=1.0 size=    5746kB time=00:01:22.11 bitrate= 573.2kbits/s dup=407 drop=0    
  
  frame= 2728 fps=544 q=1.0 size=    6354kB time=00:01:30.99 bitrate= 572.1kbits/s dup=451 drop=0    
  
  frame= 3007 fps=545 q=1.0 size=    6980kB time=00:01:40.28 bitrate= 570.2kbits/s dup=498 drop=0    
  
  frame= 3146 fps=546 q=1.0 Lsize=    7352kB time=00:01:44.93 bitrate= 573.9kbits/s dup=521 drop=0    
  
  
  
  video:1518kB audio:5745kB global headers:0kB muxing overhead 1.230493%

• convert images to video with differing time ranges

  11 juin 2013, par FaultyJuggler

  I have images coming in at random times labeled with their epoch time they were taken. I want to create a video that shows their real time creation in order. FFMPEG (far as I can tell) only allows you to set the framerate.

  For now I'm looking at creating a video file per image that is as long as the gap between the current image and the next image's timestamp, then concatenating all videos together after.

  Is there a better way to do this ?

• Conversion from mp3 to aac/mp4 container (FFmpeg/c++)

  1er juillet 2013, par taansari

  I have made a small application to extract audio from an mp4 file, or simply convert an existing audio file to AAC/mp4 format (both raw AAC, or inside mp4 container). I have run this application with existing mp4 files as input, and it properly extracts audio, and encodes to mp4 (audio only:AAC), or even directly in AAC format (i.e. test.aac also works). But when I tried running it on mp3 files, output clip plays faster than it should be (a clip of 1:12 seconds plays back till 1:05 seconds only).

  Edit : I have made improvements in code - now, it no longer plays back faster, but is still only converted till 1:05 seconds, remaining clip is missing (this is about 89% conversion done, and remaining 11% remaining).

  Here is the code I have written to achieve this :

  ////////////////////////////////////////////////
  
      #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->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;
  
      char src_filename[90] = "mp3.mp3";
  
      char dst_filename[90] = "test.mp4";
  
      open_audio_input(src_filename);
  
      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;
  
          }
  
          av_free_packet(&amp;input_packet);
  
          printf("\naudio_pts: %.3f", audio_pts);
  
      }
  
      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);
  
      return 0;
  
  }
  
  ///////////////////////////////////////////////
  
  </queue></deque></map></vector></string></fstream></iostream>

  I have been looking at this problem from many angles since about two days now, but cant seem to figure out what I'm doing wrong.

  Note also : the printf() statement I've inserted shows audio_pts up to 64.551 (that's about 1:05 seconds that also proves encoder is not going to full duration of input file : 1:12 secs).

  Can anyone please guide me what I may be doing wrong ?

  Thanks in advance for any guidance !

  p.s. when run through command line like : ffmpeg -i test.mp3 test.mp4, it converts the file just fine.