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• Static Compilation of FFmpeg with x264 support

  17 février 2014, par b1izzard

  Based on the Tutorial, I tried to build the latest FFmpeg binary with x264 support for Command line execution using Android-ndk-r9c.

  I had built the x264 using the below script :

   #!/bin/bash        
  
  
  
  
  
  PREBUILT=/home/blizzard/bin/android-ndk-r9c/toolchains/arm-linux-androideabi-4.6/prebuilt
  
                  PLATFORM=/home/blizzard/bin/android-ndk-r9c/platforms/android-9/arch-arm
  
                  PREFIX=/home/blizzard/bin/android-ndk-r9c/sources/ffmpeg
  
  
  
  
  
              ./configure --prefix=$PREFIX \
  
              --enable-static \
  
              --enable-pic \
  
              --disable-asm \
  
              --disable-cli \             
  
              --host=arm-linux \
  
              --cross-prefix=$PREBUILT/linux-x86_64/bin/arm-linux-androideabi- \
  
              --sysroot=$PLATFORM
  
  
  
              make
  
              sudo make install
  
              sudo ldconfig

  For building FFmpeg I had used the below script :

   #!/bin/bash                    
  
                PLATFORM=/home/blizzard/bin/android-ndk-r9c/platforms/android-9/arch-arm
  
                PREBUILT=/home/blizzard/bin/android-ndk-r9c/toolchains/arm-linux-androideabi-4.6/prebuilt/linux-x86_64
  
                PREFIX=/home/blizzard/bin/android-ndk-r9c/sources/ffmpeg
  
  
  
                  function build_one
  
                  {
  
                  ./configure --target-os=linux --prefix=$PREFIX \
  
                  --arch=arm \
  
                  --cpu=cortex-a8 \
  
                  --enable-cross-compile \
  
                  --enable-runtime-cpudetect \
  
                  --disable-asm \
  
                  --enable-static \
  
                  --disable-shared \
  
                  --arch=arm \
  
                  --cc=$PREBUILT/bin/arm-linux-androideabi-gcc \
  
                  --cross-prefix=$PREBUILT/bin/arm-linux-androideabi- \
  
                  --disable-stripping \
  
                  --nm=$PREBUILT/bin/arm-linux-androideabi-nm \
  
                  --sysroot=$PLATFORM \
  
                  --enable-nonfree \
  
                  --enable-version3 \
  
                  --disable-everything \
  
                  --enable-gpl \
  
                  --disable-doc \
  
                  --enable-avresample \
  
                  --enable-demuxer=rtsp \
  
                  --enable-muxer=rtsp \
  
                  --disable-ffplay \
  
                  --disable-ffserver \
  
                  --enable-ffmpeg \
  
                  --disable-ffprobe \
  
                  --enable-libx264 \
  
                  --enable-encoder=libx264 \
  
                  --enable-decoder=h264 \
  
                  --enable-protocol=rtp \
  
                  --enable-hwaccels \
  
                  --enable-zlib \
  
                  --disable-devices \
  
                  --disable-avdevice \
  
                  --extra-cflags="-I/home/blizzard/bin/android-ndk-r9c/sources/ffmpeg/include -fPIC -DANDROID -D__thumb__ -mthumb -Wfatal-errors -Wno-deprecated -mfloat-abi=softfp -marm -march=armv7-a" \
  
                  --extra-ldflags="-L/home/blizzard/bin/android-ndk-r9c/sources/ffmpeg/lib"
  
                  make -j4 install
  
  
  
                  $PREBUILT/bin/arm-linux-androideabi-ar d libavcodec/libavcodec.a inverse.o
  
                  $PREBUILT/bin/arm-linux-androideabi-ld -rpath-link=$PLATFORM/usr/lib -L$PLATFORM/usr/lib -L$PREFIX/lib  -soname libffmpeg.so -shared -nostdlib  -z noexecstack -Bsymbolic --whole-archive --no-undefined -o $PREFIX/libffmpeg.so libavcodec/libavcodec.a libavfilter/libavfilter.a libavresample/libavresample.a libavformat/libavformat.a libavutil/libavutil.a libswscale/libswscale.a -lc -lm -lz -ldl -llog -lx264  --dynamic-linker=/system/bin/linker $PREBUILT/lib/gcc/arm-linux-androideabi/4.6/libgcc.a
  
                  }
  
  
  
                  build_one

  The output of the script is

  install prefix            /home/blizzard/bin/android-ndk-r9c/sources/ffmpeg
  
  source path               .
  
  C compiler                /home/blizzard/bin/android-ndk-r9c/toolchains/arm-linux-androideabi-4.6/prebuilt/linux-x86_64/bin/arm-linux-androideabi-gcc
  
  C library                 bionic
  
  host C compiler           gcc
  
  host C library            glibc
  
  ARCH                      c (cortex-a8)
  
  big-endian                no
  
  runtime cpu detection     yes
  
  debug symbols             yes
  
  strip symbols             no
  
  optimize for size         no
  
  optimizations             yes
  
  static                    yes
  
  shared                    no
  
  postprocessing support    yes
  
  new filter support        yes
  
  network support           yes
  
  threading support         pthreads
  
  safe bitstream reader     yes
  
  SDL support               no
  
  opencl enabled            no
  
  libzvbi enabled           no
  
  texi2html enabled         yes
  
  perl enabled              yes
  
  pod2man enabled           yes
  
  makeinfo enabled          no
  
  
  
  External libraries:
  
  libx264         zlib
  
  
  
  Enabled decoders:
  
  h264
  
  
  
  Enabled encoders:
  
  libx264
  
  
  
  Enabled hwaccels:
  
  
  
  Enabled parsers:
  
  
  
  Enabled demuxers:
  
  asf         mpegts          rtsp
  
  mov         rm
  
  
  
  Enabled muxers:
  
  rtp         rtsp
  
  
  
  Enabled protocols:
  
  http            tcp         udp
  
  rtp
  
  
  
  Enabled filters:
  
  aformat         format          setpts
  
  anull           null            trim
  
  atrim
  
  
  
  Enabled bsfs:
  
  
  
  Enabled indevs:
  
  
  
  Enabled outdevs:
  
  
  
  License: nonfree and unredistributable
  
  Creating config.mak, config.h, and doc/config.texi...
  
  config.h is unchanged
  
  libavutil/avconfig.h is unchanged
  
  
  
  WARNING: /home/blizzard/bin/android-ndk-r9c/toolchains/arm-linux-androideabi-4.6/prebuilt/linux-x86_64/bin/arm-linux-androideabi-pkg-config not found, library detection may fail.
  
  INSTALL install-progs-yes
  
  INSTALL ffmpeg
  
  INSTALL presets/libvpx-1080p50_60.ffpreset
  
  INSTALL presets/libvpx-1080p.ffpreset
  
  INSTALL presets/libvpx-360p.ffpreset
  
  INSTALL presets/libvpx-720p50_60.ffpreset
  
  INSTALL presets/libvpx-720p.ffpreset
  
  INSTALL presets/libx264-ipod320.ffpreset
  
  INSTALL presets/libx264-ipod640.ffpreset
  
  INSTALL doc/ffprobe.xsd
  
  INSTALL doc/examples/decoding_encoding.c
  
  INSTALL doc/examples/demuxing_decoding.c
  
  INSTALL doc/examples/filtering_audio.c
  
  INSTALL doc/examples/filtering_video.c
  
  INSTALL doc/examples/metadata.c
  
  INSTALL doc/examples/muxing.c
  
  INSTALL doc/examples/remuxing.c
  
  INSTALL doc/examples/resampling_audio.c
  
  INSTALL doc/examples/scaling_video.c
  
  INSTALL doc/examples/transcode_aac.c
  
  INSTALL doc/examples/Makefile
  
  INSTALL doc/examples/README
  
  INSTALL libavfilter/libavfilter.a
  
  INSTALL libavformat/libavformat.a
  
  INSTALL presets/libvpx-1080p50_60.ffpreset
  
  INSTALL presets/libvpx-1080p.ffpreset
  
  INSTALL presets/libvpx-360p.ffpreset
  
  INSTALL presets/libvpx-720p50_60.ffpreset
  
  INSTALL presets/libvpx-720p.ffpreset
  
  INSTALL presets/libx264-ipod320.ffpreset
  
  INSTALL presets/libx264-ipod640.ffpreset
  
  INSTALL doc/ffprobe.xsd
  
  INSTALL doc/examples/decoding_encoding.c
  
  INSTALL doc/examples/demuxing_decoding.c
  
  INSTALL doc/examples/filtering_audio.c
  
  INSTALL doc/examples/filtering_video.c
  
  INSTALL doc/examples/metadata.c
  
  INSTALL doc/examples/muxing.c
  
  INSTALL doc/examples/remuxing.c
  
  INSTALL doc/examples/resampling_audio.c
  
  INSTALL doc/examples/scaling_video.c
  
  INSTALL doc/examples/transcode_aac.c
  
  INSTALL doc/examples/Makefile
  
  INSTALL doc/examples/README
  
  INSTALL libavresample/libavresample.a
  
  INSTALL libavcodec/libavcodec.a
  
  INSTALL libpostproc/libpostproc.a
  
  INSTALL libswresample/libswresample.a
  
  INSTALL libavutil/libavutil.a
  
  a
  
  INSTALL libavfilter/asrc_abuffer.h
  
  INSTALL libavfilter/avcodec.h
  
  INSTALL libavfilter/avfilter.h
  
  INSTALL libavfilter/avfiltergraph.h
  
  INSTALL libavfilter/buffersink.h
  
  INSTALL libavfilter/buffersrc.h
  
  INSTALL libavfilter/version.h
  
  INSTALL libavformat/avformat.h
  
  INSTALL libavformat/avio.h
  
  INSTALL libavformat/version.h
  
  INSTALL libavfilter/libavfilter.pc
  
  INSTALL libavformat/libavformat.pc
  
  INSTALL libavresample/avresample.h
  
  INSTALL libavresample/version.h
  
  INSTALL libavresample/libavresample.pc
  
  INSTALL libavcodec/avcodec.h
  
  INSTALL libavcodec/avfft.h
  
  INSTALL libavcodec/dxva2.h
  
  INSTALL libavcodec/old_codec_ids.h
  
  INSTALL libavcodec/vaapi.h
  
  INSTALL libavcodec/vda.h
  
  INSTALL libavcodec/vdpau.h
  
  INSTALL libavcodec/version.h
  
  INSTALL libavcodec/xvmc.h
  
  INSTALL libavcodec/libavcodec.pc
  
  INSTALL libpostproc/postprocess.h
  
  INSTALL libpostproc/version.h
  
  INSTALL libpostproc/libpostproc.pc
  
  INSTALL libswresample/swresample.h
  
  INSTALL libswresample/version.h
  
  INSTALL libswscale/swscale.h
  
  INSTALL libswscale/version.h
  
  INSTALL libavutil/adler32.h
  
  INSTALL libavutil/aes.h
  
  INSTALL libavutil/attributes.h
  
  INSTALL libavutil/audio_fifo.h
  
  INSTALL libavutil/audioconvert.h
  
  INSTALL libavutil/avassert.h
  
  INSTALL libavutil/avstring.h
  
  INSTALL libavutil/avutil.h
  
  INSTALL libavutil/base64.h
  
  INSTALL libavutil/blowfish.h
  
  INSTALL libavutil/bprint.h
  
  INSTALL libavutil/bswap.h
  
  INSTALL libavutil/buffer.h
  
  INSTALL libavutil/channel_layout.h
  
  INSTALL libavutil/common.h
  
  INSTALL libavutil/cpu.h
  
  INSTALL libavutil/crc.h
  
  INSTALL libavutil/error.h
  
  INSTALL libavutil/eval.h
  
  INSTALL libavutil/fifo.h
  
  INSTALL libavutil/file.h
  
  INSTALL libavutil/frame.h
  
  INSTALL libavutil/hmac.h
  
  INSTALL libavutil/imgutils.h
  
  INSTALL libavutil/intfloat.h
  
  INSTALL libavutil/intfloat_readwrite.h
  
  INSTALL libavutil/intreadwrite.h
  
  INSTALL libavutil/lfg.h
  
  INSTALL libavutil/log.h
  
  INSTALL libavutil/macros.h
  
  INSTALL libavutil/mathematics.h
  
  INSTALL libavutil/md5.h
  
  INSTALL libavutil/mem.h
  
  INSTALL libavutil/murmur3.h
  
  INSTALL libavutil/dict.h
  
  INSTALL libavutil/old_pix_fmts.h
  
  INSTALL libavutil/opt.h
  
  INSTALL libavutil/parseutils.h
  
  INSTALL libavutil/pixdesc.h
  
  INSTALL libavutil/pixfmt.h
  
  INSTALL libavutil/random_seed.h
  
  INSTALL libavutil/rational.h
  
  INSTALL libavutil/ripemd.h
  
  INSTALL libavutil/samplefmt.h
  
  INSTALL libavutil/sha.h
  
  INSTALL libavutil/sha512.h
  
  INSTALL libavutil/stereo3d.h
  
  INSTALL libavutil/time.h
  
  INSTALL libavutil/timecode.h
  
  INSTALL libavutil/timestamp.h
  
  INSTALL libavutil/version.h
  
  INSTALL libavutil/xtea.h
  
  INSTALL libavutil/avconfig.h
  
  INSTALL libavutil/ffversion.h
  
  INSTALL libswresample/libswresample.pc
  
  INSTALL libswscale/libswscale.pc
  
  INSTALL libavutil/libavutil.pc

  When I tried to execute the following command using the resulting FFmpeg binary in Android Application

  ffmpeg -y -loop 1 -r 30 -i  myImage.jpg -b:v "4096k" -vf "scale=640:480" -t 3 result.mp4

  LogCat displays the following message

  02-17 15:42:45.683: D/(14511): *******Starting FFMPEG
  
  02-17 15:42:45.683: D/(14511): ***ffmpeg version N-60108-gda25a65 Copyright (c) 2000-2014 the FFmpeg developers***
  
  02-17 15:42:45.693: D/(14511): ***  built on Feb 17 2014 15:35:20 with gcc 4.6 (GCC) 20120106 (prerelease)***
  
  02-17 15:42:45.693: D/(14511): ***  configuration: --target-os=linux --prefix=/home/blizzard/bin/android-ndk-r9c/sources/ffmpeg --arch=arm --cpu=cortex-a8 --enable-cross-compile --enable-runtime-cpudetect --disable-asm --enable-static --disable-shared --arch=arm --cc=/home/blizzard/bin/android-ndk-r9c/toolchains/arm-linux-androideabi-4.6/prebuilt/linux-x86_64/bin/arm-linux-androideabi-gcc --cross-prefix=/home/blizzard/bin/android-ndk-r9c/toolchains/arm-linux-androideabi-4.6/prebuilt/linux-x86_64/bin/arm-linux-androideabi- --disable-stripping --nm=/home/blizzard/bin/android-ndk-r9c/toolchains/arm-linux-androideabi-4.6/prebuilt/linux-x86_64/bin/arm-linux-androideabi-nm --sysroot=/home/blizzard/bin/android-ndk-r9c/platforms/android-9/arch-arm --enable-nonfree --enable-version3 --disable-everything --enable-gpl --disable-doc --enable-avresample --enable-demuxer=rtsp --enable-muxer=rtsp --disable-ffplay --disable-ffserver --enable-ffmpeg --disable-ffprobe --enable-libx264 --enable-encoder=libx264 --enable-decoder=h264 --enable-protocol=rtp --enable-hwaccels --enable-zlib --disable-devices --disable-avdevice --extra-cflags='-I/home/blizzard/bin/android-ndk-r9c/sources/ffmpeg/include -fPIC -DANDROID -D__thumb__ -mthumb -Wfatal-errors -Wno-deprecated -mfloat-abi=softfp -mfpu=vfpv3-d16 -marm -march=armv7-a' --extra-ldflags=-L/home/blizzard/bin/android-ndk-r9c/sources/ffmpeg/lib***
  
  02-17 15:42:45.693: D/(14511): ***  libavutil      52. 63.100 / 52. 63.100***
  
  02-17 15:42:45.693: D/(14511): ***  libavcodec     55. 49.100 / 55. 49.100***
  
  02-17 15:42:45.693: D/(14511): ***  libavformat    55. 25.101 / 55. 25.101***
  
  02-17 15:42:45.693: D/(14511): ***  libavfilter     4.  1.100 /  4.  1.100***
  
  02-17 15:42:45.693: D/(14511): ***  libavresample   1.  1.  0 /  1.  1.  0***
  
  02-17 15:42:45.693: D/(14511): ***  libswscale      2.  5.101 /  2.  5.101***
  
  02-17 15:42:45.693: D/(14511): ***  libswresample   0. 17.104 /  0. 17.104***
  
  02-17 15:42:45.693: D/(14511): ***  libpostproc    52.  3.100 / 52.  3.100***
  
  02-17 15:42:45.693: D/(14511): ***Unrecognized option 'loop'.***
  
  02-17 15:42:45.693: D/(14511): ***Error splitting the argument list: Option not found***
  
  02-17 15:42:45.693: D/(14511): ****ending FFMPEG****

  Can anybody help me in building the latest version of FFmpeg or Can you share the latest FFMpeg binary for Android ?

• Evolution #3130 (Fermé) : Langue d’un article syndiqué

  28 janvier 2014, par Fil Up

  je ferme car ça me paraît trop difficile à faire en standard, parce que les sources ne sont pas fiables (inutile d’indiquer hreflang="en" pour une ressource en français sur un site mal configuré ! ce serait pire que de ne rien indiquer) ; pour une application spécifique, et en délimitant précisément la liste des langues qu’il est possible d’avoir dans ses flux, on pourra utiliser lang_detect.

• 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 ?