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• FFMpeg slowed down sound sounds too metalic

  8 mars 2015, par user1571299

  this is the command line I use :

  ffmpeg -i VID_20150216_185018.mp4 -filter_complex setpts=4*PTS;atempo=0.5,atempo=0.5 output.mp4

  The video plays just fine and the audio is in sync...but it’s just too metalic compared with most slo-mo videos I’ve seen out there.
  Here’s an example how someone fixes it with Sony Vegas which is exactly what I want to achieve :

  https://www.youtube.com/watch?v=nTNoqdPCUfQ

• MP3 to Youtube video format [on hold]

  17 janvier 2014, par user3027136

  I am trying to convert some audio files and upload them to Youtube ; I am using a .jpg image as video and the .mp3 files as audio.
  I've already managed to do it using ffmpeg, and it works on my computer (video plays the audio and shows the picture), but every time I upload it Youtube tells me the format is wrong, no matter what I instruct ffmpeg to do : avi, mp4, mpeg, mov atc.
  Below is the command I used (ffmpeg) :

  for %A IN (*.mp3) DO ffmpeg -i "Youtube.jpg" -i "%A" -acodec copy "%A.avi"

  Any improvement is appreciated as well as any other suggestion on how to upload mp3s on Youtube

• Transcoding to vorbis using FFmpeg libraries, C++

  6 février 2014, par taansari

  I have made a test application to transcode to vorbis format (webm container).

  So far, based on FFmpeg examples, things are somewhat working, and output file plays properly, but sound in right channel is missing. I tried looking at different possibilities, but so far could not find any answer.

  For reference, this is the code I am using :

  #include "stdafx.h"
  
  #define MAX_AUDIO_PACKET_SIZE (128 * 1024)
  
  
  
  #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>
  
  }
  
  AVCodecID           outputAudioFormat = AV_CODEC_ID_VORBIS;
  
  
  
  
  
  static int sws_flags = SWS_BICUBIC;
  
  #define STREAM_DURATION   50.0
  
  #define STREAM_FRAME_RATE 25 /* 25 images/s */
  
  #define STREAM_NB_FRAMES  ((int)(STREAM_DURATION * STREAM_FRAME_RATE))
  
  #define STREAM_PIX_FMT    AV_PIX_FMT_YUV420P /* default pix_fmt */
  
  
  
  
  
  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 *        swrContext = NULL;
  
  
  
  AVOutputFormat *    output_format = NULL ;
  
  AVFormatContext *    output_fmt_ctx= NULL;
  
  AVStream *            audio_st = NULL;
  
  AVStream*           video_st = NULL;
  
  AVCodec *            audio_codec = NULL;
  
  AVCodec*            video_codec = NULL;
  
  double                audio_pts = 0.0;
  
  AVFrame *            out_frame = avcodec_alloc_frame();
  
  
  
  int                    audio_input_frame_size = 64;
  
  
  
  uint8_t *            audio_data_buf = NULL;
  
  uint8_t *            audio_out = NULL;
  
  int                    audio_bit_rate;
  
  int                    audio_sample_rate;
  
  int                    audio_channels;
  
  int                 sourceSampleRate=0;
  
  int                 destSampleRate = 0;
  
  
  
  int                 dst_nb_samples = 0;
  
  int                 pivotIndex = 0;
  
  int                 max_dst_nb_samples = 0;
  
  int                 samples_count=0;
  
  
  
  
  
  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);
  
  
  
  static AVFrame *frame;
  
  static AVPicture src_picture, dst_picture;
  
  static int frame_count;
  
  /* Add an output stream. */
  
  static AVStream *add_stream(AVFormatContext *oc, AVCodec **codec,
  
                              enum AVCodecID codec_id)
  
  {
  
      AVCodecContext *c;
  
      AVStream *st;
  
  
  
      /* find the encoder */
  
      *codec = avcodec_find_encoder(codec_id);
  
      if (!(*codec)) {
  
          fprintf(stderr, "Could not find encoder for &#39;%s&#39;\n",
  
                  avcodec_get_name(codec_id));
  
          exit(1);
  
      }
  
  
  
      st = avformat_new_stream(oc, *codec);
  
      if (!st) {
  
          fprintf(stderr, "Could not allocate stream\n");
  
          exit(1);
  
      }
  
      st->id = oc->nb_streams-1;
  
      c = st->codec;
  
  
  
      switch ((*codec)->type) {
  
      case AVMEDIA_TYPE_AUDIO:
  
          c->sample_fmt  = (*codec)->sample_fmts ?
  
              (*codec)->sample_fmts[0] : AV_SAMPLE_FMT_FLTP;
  
          c->bit_rate    = 64000;
  
          c->sample_rate = 44100;
  
          c->channels    = 2;
  
          break;
  
  
  
      case AVMEDIA_TYPE_VIDEO:
  
          c->codec_id = codec_id;
  
  
  
          c->bit_rate = 400000;
  
          /* Resolution must be a multiple of two. */
  
          c->width    = 352;
  
          c->height   = 288;
  
          /* timebase: This is the fundamental unit of time (in seconds) in terms
  
           * of which frame timestamps are represented. For fixed-fps content,
  
           * timebase should be 1/framerate and timestamp increments should be
  
           * identical to 1. */
  
          c->time_base.den = STREAM_FRAME_RATE;
  
          c->time_base.num = 1;
  
          c->gop_size      = 12; /* emit one intra frame every twelve frames at most */
  
          c->pix_fmt       = STREAM_PIX_FMT;
  
          if (c->codec_id == AV_CODEC_ID_MPEG2VIDEO) {
  
              /* just for testing, we also add B frames */
  
              c->max_b_frames = 2;
  
          }
  
          if (c->codec_id == AV_CODEC_ID_MPEG1VIDEO) {
  
              /* Needed to avoid using macroblocks in which some coeffs overflow.
  
               * This does not happen with normal video, it just happens here as
  
               * the motion of the chroma plane does not match the luma plane. */
  
              c->mb_decision = 2;
  
          }
  
      break;
  
  
  
      default:
  
          break;
  
      }
  
  
  
      /* Some formats want stream headers to be separate. */
  
      if (oc->oformat->flags &amp; AVFMT_GLOBALHEADER)
  
          c->flags |= CODEC_FLAG_GLOBAL_HEADER;
  
  
  
      return st;
  
  }
  
  
  
  
  
  static void open_video(AVFormatContext *oc, AVCodec *codec, AVStream *st)
  
  {
  
      int ret;
  
      AVCodecContext *c = st->codec;
  
  
  
      /* open the codec */
  
      ret = avcodec_open2(c, codec, NULL);
  
      if (ret &lt; 0) {
  
          //fprintf(stderr, "Could not open video codec: %s\n", av_err2str(ret));
  
          exit(1);
  
      }
  
  
  
      /* allocate and init a re-usable frame */
  
      frame = av_frame_alloc();
  
      if (!frame) {
  
          fprintf(stderr, "Could not allocate video frame\n");
  
          exit(1);
  
      }
  
      frame->format = c->pix_fmt;
  
      frame->width = c->width;
  
      frame->height = c->height;
  
  
  
      /* Allocate the encoded raw picture. */
  
      ret = avpicture_alloc(&amp;dst_picture, c->pix_fmt, c->width, c->height);
  
      if (ret &lt; 0) {
  
          //fprintf(stderr, "Could not allocate picture: %s\n", av_err2str(ret));
  
          exit(1);
  
      }
  
  
  
      /* If the output format is not YUV420P, then a temporary YUV420P
  
       * picture is needed too. It is then converted to the required
  
       * output format. */
  
      if (c->pix_fmt != AV_PIX_FMT_YUV420P) {
  
          ret = avpicture_alloc(&amp;src_picture, AV_PIX_FMT_YUV420P, c->width, c->height);
  
          if (ret &lt; 0) {
  
              //fprintf(stderr, "Could not allocate temporary picture: %s\n",
  
              //        av_err2str(ret));
  
              exit(1);
  
          }
  
      }
  
  
  
      /* copy data and linesize picture pointers to frame */
  
      *((AVPicture *)frame) = dst_picture;
  
  }
  
  
  
  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
  
          swrContext = swr_alloc();
  
          av_opt_set_int(swrContext, "in_channel_layout",  codec_ctx_audio->channel_layout, 0);
  
          av_opt_set_int(swrContext, "out_channel_layout", codec_ctx_audio->channel_layout,  0);
  
          av_opt_set_int(swrContext, "in_sample_rate",     codec_ctx_audio->sample_rate, 0);
  
          av_opt_set_int(swrContext, "out_sample_rate",    codec_ctx_audio->sample_rate, 0);
  
          av_opt_set_sample_fmt(swrContext, "in_sample_fmt",  codec_ctx_audio->sample_fmt, 0);
  
          if ( outputAudioFormat == AV_CODEC_ID_VORBIS )
  
          {
  
              av_opt_set_sample_fmt(swrContext, "out_sample_fmt", AV_SAMPLE_FMT_FLTP,  0);
  
          }
  
          else
  
          {
  
              av_opt_set_sample_fmt(swrContext, "out_sample_fmt", AV_SAMPLE_FMT_S16,  0);
  
          }
  
          int rv = swr_init(swrContext);
  
  
  
          sourceSampleRate    =   destSampleRate = codec_ctx_audio->sample_rate;
  
  
  
          // 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)
  
      {
  
          avcodec_get_frame_defaults(decoded_frame);
  
          while( input_packet.size > 0 )
  
          {
  
              int result = avcodec_decode_audio4(codec_ctx_audio, decoded_frame, &amp;got_frame, &amp;input_packet);
  
              if ( result &lt; 0)
  
              {
  
                  fprintf(stderr, "Error decoding audio frame\n");
  
                  //return ret;
  
              }
  
              else
  
              {
  
                  if ( got_frame )
  
                  {
  
                      dst_nb_samples = (int)av_rescale_rnd(swr_get_delay(swrContext, sourceSampleRate) + decoded_frame->nb_samples, sourceSampleRate, destSampleRate, AV_ROUND_UP);
  
                      if ( dst_nb_samples > max_dst_nb_samples )
  
                      {
  
                          max_dst_nb_samples = dst_nb_samples;
  
                          if ( audio_dst_data[0] )
  
                          {
  
                              av_freep(&amp;audio_dst_data[0]);
  
                              audio_dst_data[0] = NULL;
  
                          }
  
                      }
  
                      if ( audio_dst_data[0] == NULL )
  
                      {
  
                          if ( outputAudioFormat == AV_CODEC_ID_VORBIS )
  
                          {
  
                              ret = av_samples_alloc(audio_dst_data, &amp;audio_dst_linesize, codec_ctx_audio->channels,
  
                                  decoded_frame->nb_samples, (AVSampleFormat)AV_SAMPLE_FMT_FLTP, 0);
  
                          }
  
                          else
  
                          {
  
                              ret = av_samples_alloc(audio_dst_data, &amp;audio_dst_linesize, codec_ctx_audio->channels,
  
                                  decoded_frame->nb_samples, (AVSampleFormat)AV_SAMPLE_FMT_S16, 0);
  
                          }
  
                      }
  
                      /* TODO: extend return code of the av_samples_* functions so that this call is not needed */
  
                      int resampled  = swr_convert(swrContext, audio_dst_data, out_frame->nb_samples,
  
                          (const uint8_t **)(decoded_frame->extended_data), decoded_frame->nb_samples);
  
                      char str[900]="";
  
                      sprintf(str,"out_frame->nb_samples:\t%d; decoded_frame->nb_samples:\t%d",out_frame->nb_samples,decoded_frame->nb_samples );
  
                      if ( outputAudioFormat == AV_CODEC_ID_VORBIS )
  
                      {
  
                          audio_dst_bufsize  = av_samples_get_buffer_size(&amp;audio_dst_linesize, decoded_frame->channels, resampled, (AVSampleFormat)AV_SAMPLE_FMT_FLTP, 1);
  
                      }
  
                      else
  
                      {
  
                          audio_dst_bufsize  = av_samples_get_buffer_size(&amp;audio_dst_linesize, decoded_frame->channels, resampled, (AVSampleFormat)AV_SAMPLE_FMT_S16, 1);
  
                      }
  
  
  
                      input_packet.size -= result;
  
                      input_packet.data += result;
  
                  }
  
                  else
  
                  {
  
                      input_packet.size   =   0;
  
                      input_packet.data   =   NULL;
  
                  }
  
              }
  
          }
  
      }
  
      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 */
  
      if ( outputAudioFormat == AV_CODEC_ID_VORBIS )
  
      {
  
          c->sample_fmt  = AV_SAMPLE_FMT_FLTP;
  
      }
  
      else
  
      {
  
          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;
  
      out_frame->nb_samples = audio_input_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_dst_data, int audio_dst_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=0, ret=0;
  
  
  
      av_init_packet(&amp;pkt);
  
      c = st->codec;
  
  
  
      out_frame->nb_samples = audio_input_frame_size;
  
  
  
      AVRational r;
  
      r.num = 1;
  
      r.den = c->sample_rate;
  
      out_frame->pts = av_rescale_q(samples_count, (AVRational)r, c->time_base);
  
      avcodec_fill_audio_frame(out_frame, c->channels, c->sample_fmt,
  
                               audio_dst_data[0], audio_dst_bufsize, 0);
  
      samples_count += out_frame->nb_samples;
  
  
  
      ret = avcodec_encode_audio2(c, &amp;pkt, out_frame, &amp;got_packet);
  
      if (ret &lt; 0) 
  
      {
  
          return;
  
      }
  
  
  
      if (!got_packet)
  
          return;
  
  
  
      /* rescale output packet timestamp values from codec to stream timebase */
  
      pkt.pts = av_rescale_q_rnd(pkt.pts, c->time_base, st->time_base, (AVRounding )(AV_ROUND_NEAR_INF|AV_ROUND_PASS_MINMAX));
  
      pkt.dts = av_rescale_q_rnd(pkt.dts, c->time_base, st->time_base, (AVRounding )(AV_ROUND_NEAR_INF|AV_ROUND_PASS_MINMAX));
  
      pkt.duration = av_rescale_q(pkt.duration, c->time_base, st->time_base);
  
      pkt.stream_index = st->index;
  
  
  
      char str[999]="";
  
      sprintf(str,"out_frame->nb_samples:\t%d",out_frame->nb_samples);
  
      /* Write the compressed frame to the media file. */
  
      ret = av_interleaved_write_frame(oc, &amp;pkt);
  
      if (ret != 0) 
  
      {
  
          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)
  
          {
  
              pkt.pts = av_rescale_q_rnd(pkt.pts, c->time_base, st->time_base, (AVRounding )(AV_ROUND_NEAR_INF|AV_ROUND_PASS_MINMAX));
  
              pkt.dts = av_rescale_q_rnd(pkt.dts, c->time_base, st->time_base, (AVRounding )(AV_ROUND_NEAR_INF|AV_ROUND_PASS_MINMAX));
  
              pkt.duration = av_rescale_q(pkt.duration, c->time_base, st->time_base);
  
              pkt.stream_index = st->index;
  
              if ( c &amp;&amp; c->coded_frame &amp;&amp; c->coded_frame->key_frame)
  
                  pkt.flags |= AV_PKT_FLAG_KEY;
  
              /* 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.mp2";
  
      char dst_filename[90] = "output.webm";
  
      outputAudioFormat = AV_CODEC_ID_VORBIS;
  
      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 );
  
      int frames= 0;
  
      while(1)
  
      {
  
          int rv = decode_frame();
  
          if ( rv &lt; 0 )
  
          {
  
              break;
  
          }
  
  
  
          if (audio_st)
  
          {
  
              audio_pts = audio_st->pts.val * av_q2d(audio_st->time_base);
  
          }
  
          else
  
          {
  
              audio_pts = 0.0;
  
          }
  
          if ( codec_ctx_audio )
  
          {
  
              if ( got_frame )
  
              {
  
                  write_audio_frame( audio_dst_data, audio_dst_bufsize );
  
                  frames++;
  
              }
  
          }
  
          printf("\naudio_pts: %f", audio_pts);
  
      }
  
      while(1)
  
      {
  
          dst_nb_samples = (int)av_rescale_rnd(swr_get_delay(swrContext, sourceSampleRate) + decoded_frame->nb_samples, sourceSampleRate, destSampleRate, AV_ROUND_UP);
  
          if ( dst_nb_samples > max_dst_nb_samples )
  
          {
  
              max_dst_nb_samples = dst_nb_samples;
  
              if ( audio_dst_data[0] )
  
              {
  
                  av_freep(&amp;audio_dst_data[0]);
  
                  audio_dst_data[0] = NULL;
  
              }
  
          }
  
          if ( audio_dst_data[0] == NULL )
  
          {
  
              if ( outputAudioFormat == AV_CODEC_ID_VORBIS )
  
              {
  
                  ret = av_samples_alloc(audio_dst_data, NULL, codec_ctx_audio->channels,
  
                      decoded_frame->nb_samples, (AVSampleFormat)AV_SAMPLE_FMT_FLTP, 0);
  
              }
  
              else
  
              {
  
                  ret = av_samples_alloc(audio_dst_data, NULL, codec_ctx_audio->channels,
  
                      decoded_frame->nb_samples, (AVSampleFormat)AV_SAMPLE_FMT_S16, 0);
  
              }
  
          }
  
          int resampled = swr_convert(swrContext, audio_dst_data, out_frame->nb_samples,NULL, 0);
  
          if ( outputAudioFormat == AV_CODEC_ID_VORBIS )
  
          {
  
              audio_dst_bufsize  = av_samples_get_buffer_size(&amp;audio_dst_linesize, decoded_frame->channels, resampled, (AVSampleFormat)AV_SAMPLE_FMT_FLTP, 1);
  
          }
  
          else
  
          {
  
              audio_dst_bufsize  = av_samples_get_buffer_size(&amp;audio_dst_linesize, decoded_frame->channels, resampled, (AVSampleFormat)AV_SAMPLE_FMT_S16, 1);
  
          }
  
          if ( audio_dst_bufsize &lt;= 0 ) break;
  
          audio_pts = audio_st->pts.val * av_q2d(audio_st->time_base);
  
          printf("\naudio_pts: %f", audio_pts);
  
          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;swrContext);
  
      avcodec_free_frame(&amp;out_frame);
  
      getch();
  
      return 0;
  
  }
  
  </queue></deque></map></vector></string></fstream></iostream>

  Working under Windows 7, Zeranoe FFmpeg 32 bit build :

  libavutil      52. 62.100 / 52. 62.100
  
  libavcodec     55. 47.101 / 55. 47.101
  
  libavformat    55. 22.103 / 55. 22.103
  
  libavdevice    55.  5.102 / 55.  5.102
  
  libavfilter     4.  1.100 /  4.  1.100
  
  libswscale      2.  5.101 /  2.  5.101
  
  libswresample   0. 17.104 /  0. 17.104
  
  libpostproc    52.  3.100 / 52.  3.100

  Could anyone point to the place where I might be misunderstanding things ?

  Thanks for any guidance in advance !