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• How to change metadata with ffmpeg/avconv without creating a new file ?

  15 avril 2014, par tampis

  I am writing a python script for producing audio and video podcasts. There are a bunch of recorded media files (audio and video) and text files containing the meta information.

  Now I want to program a function which shall add the information from the meta data text files to all media files (the original and the converted ones). Because I have to handle many different file formats (wav, flac, mp3, mp4, ogg, ogv...) it would be great to have a tool which add meta data to arbitrary formats.

  My Question :

  How can I change the metadata of a file with ffmpeg/avconv without changing the audio or video of it and without creating a new file ? Is there another commandline/python tool which would do the job for me ?

  What I tried so far :

  I thought ffmpeg/avconv could be such a tool, because it can handle nearly all media formats. I hoped, that if I set -i input_file and the output_file to the same file, ffmpeg/avconv will be smart enough to leave the file unchanged. Then I could set -metadata key=value and just the metadata will be changed.

  But I noticed, that if I type avconv -i test.mp3 -metadata title='Test title' test.mp3 the audio test.mp3 will be reconverted in another bitrate.

  So I thought to use -c copy to copy all video and audio information. Unfortunately also this does not work :

  :~$ du -h test.wav # test.wav is 303 MB big
  
  303M    test.wav
  
  
  
  :~$ avconv -i test.wav -c copy -metadata title='Test title' test.wav
  
  avconv version 0.8.3-4:0.8.3-0ubuntu0.12.04.1, Copyright (c) 2000-2012 the
  
  Libav    developers
  
  built on Jun 12 2012 16:37:58 with gcc 4.6.3
  
  [wav @ 0x846b260] max_analyze_duration reached
  
  Input #0, wav, from 'test.wav':
  
  Duration: 00:29:58.74, bitrate: 1411 kb/s
  
      Stream #0.0: Audio: pcm_s16le, 44100 Hz, 2 channels, s16, 1411 kb/s
  
  File 'test.wav' already exists. Overwrite ? [y/N] y
  
  Output #0, wav, to 'test.wav':
  
  Metadata:
  
      title           : Test title
  
      encoder         : Lavf53.21.0
  
      Stream #0.0: Audio: pcm_s16le, 44100 Hz, 2 channels, 1411 kb/s
  
  Stream mapping:
  
  Stream #0:0 -> #0:0 (copy)
  
  Press ctrl-c to stop encoding
  
  size=     896kB time=5.20 bitrate=1411.3kbits/s    
  
  video:0kB audio:896kB global headers:0kB muxing overhead 0.005014%
  
  
  
  :~$ du -h test.wav # file size of test.wav changed dramatically
  
  900K    test.wav

  You see, that I cannot use -c copy if input_file and output_file are the same. Of course I could produce a temporarily file :

  :-$ avconv -i test.wav -c copy -metadata title='Test title' test_temp.mp3
  
  :-$ mv test_tmp.mp3 test.mp3

  But this solution would create (temporarily) a new file on the filesystem and is therefore not preferable.

• How to encode resampled PCM-audio to AAC using ffmpeg-API when input pcm samples count not equal 1024

  26 septembre 2016, par Aleksei2414904

  I am working on capturing and streaming audio to RTMP server at a moment. I work under MacOS (in Xcode), so for capturing audio sample-buffer I use AVFoundation-framework. But for encoding and streaming I need to use ffmpeg-API and libfaac encoder. So output format must be AAC (for supporting stream playback on iOS-devices).

  And I faced with such problem : audio-capturing device (in my case logitech camera) gives me sample-buffer with 512 LPCM samples, and I can select input sample-rate from 16000, 24000, 36000 or 48000 Hz. When I give these 512 samples to AAC-encoder (configured for appropriate sample-rate), I hear a slow and jerking audio (seems as like pice of silence after each frame).

  I figured out (maybe I am wrong), that libfaac encoder accepts audio frames only with 1024 samples. When I set input samplerate to 24000 and resample input sample-buffer to 48000 before encoding, I obtain 1024 resampled samples. After encoding these 1024 sampels to AAC, I hear proper sound on output. But my web-cam produce 512 samples in buffer for any input samplerate, when output sample-rate must be 48000 Hz. So I need to do resampling in any case, and I will not obtain exactly 1024 samples in buffer after resampling.

  Is there a way to solve this problem within ffmpeg-API functionality ?

  I would be grateful for any help.

  PS :
  I guess that I can accumulate resampled buffers until count of samples become 1024, and then encode it, but this is stream so there will be troubles with resulting timestamps and with other input devices, and such solution is not suitable.

  The current issue came out of the problem described in [question] : How to fill audio AVFrame (ffmpeg) with the data obtained from CMSampleBufferRef (AVFoundation) ?

  Here is a code with audio-codec configs (there also was video stream but video work fine) :

      /*global variables*/
  
      static AVFrame *aframe;
  
      static AVFrame *frame;
  
      AVOutputFormat *fmt; 
  
      AVFormatContext *oc; 
  
      AVStream *audio_st, *video_st;
  
  Init ()
  
  {
  
      AVCodec *audio_codec, *video_codec;
  
      int ret;
  
  
  
      avcodec_register_all();  
  
      av_register_all();
  
      avformat_network_init();
  
      avformat_alloc_output_context2(&oc, NULL, "flv", filename);
  
      fmt = oc->oformat;
  
      oc->oformat->video_codec = AV_CODEC_ID_H264;
  
      oc->oformat->audio_codec = AV_CODEC_ID_AAC;
  
      video_st = NULL;
  
      audio_st = NULL;
  
      if (fmt->video_codec != AV_CODEC_ID_NONE) 
  
        { //…  /*init video codec*/}
  
      if (fmt->audio_codec != AV_CODEC_ID_NONE) {
  
      audio_codec= avcodec_find_encoder(fmt->audio_codec);
  
  
  
      if (!(audio_codec)) {
  
          fprintf(stderr, "Could not find encoder for '%s'\n",
  
                  avcodec_get_name(fmt->audio_codec));
  
          exit(1);
  
      }
  
      audio_st= avformat_new_stream(oc, audio_codec);
  
      if (!audio_st) {
  
          fprintf(stderr, "Could not allocate stream\n");
  
          exit(1);
  
      }
  
      audio_st->id = oc->nb_streams-1;
  
  
  
      //AAC:
  
      audio_st->codec->sample_fmt  = AV_SAMPLE_FMT_S16;
  
      audio_st->codec->bit_rate    = 32000;
  
      audio_st->codec->sample_rate = 48000;
  
      audio_st->codec->profile=FF_PROFILE_AAC_LOW;
  
      audio_st->time_base = (AVRational){1, audio_st->codec->sample_rate };
  
      audio_st->codec->channels    = 1;
  
      audio_st->codec->channel_layout = AV_CH_LAYOUT_MONO;      
  
  
  
  
  
      if (oc->oformat->flags & AVFMT_GLOBALHEADER)
  
          audio_st->codec->flags |= CODEC_FLAG_GLOBAL_HEADER;
  
      }
  
  
  
      if (video_st)
  
      {
  
      //   …
  
      /*prepare video*/
  
      }
  
      if (audio_st)
  
      {
  
      aframe = avcodec_alloc_frame();
  
      if (!aframe) {
  
          fprintf(stderr, "Could not allocate audio frame\n");
  
          exit(1);
  
      }
  
      AVCodecContext *c;
  
      int ret;
  
  
  
      c = audio_st->codec;
  
  
  
  
  
      ret = avcodec_open2(c, audio_codec, 0);
  
      if (ret < 0) {
  
          fprintf(stderr, "Could not open audio codec: %s\n", av_err2str(ret));
  
          exit(1);
  
      }
  
  
  
      //…
  
  }

  And resampling and encoding audio :

  if (mType == kCMMediaType_Audio)
  
  {
  
      CMSampleTimingInfo timing_info;
  
      CMSampleBufferGetSampleTimingInfo(sampleBuffer, 0, &timing_info);
  
      double  pts=0;
  
      double  dts=0;
  
      AVCodecContext *c;
  
      AVPacket pkt = { 0 }; // data and size must be 0;
  
      int got_packet, ret;
  
       av_init_packet(&pkt);
  
      c = audio_st->codec;
  
        CMItemCount numSamples = CMSampleBufferGetNumSamples(sampleBuffer);
  
  
  
      NSUInteger channelIndex = 0;
  
  
  
      CMBlockBufferRef audioBlockBuffer = CMSampleBufferGetDataBuffer(sampleBuffer);
  
      size_t audioBlockBufferOffset = (channelIndex * numSamples * sizeof(SInt16));
  
      size_t lengthAtOffset = 0;
  
      size_t totalLength = 0;
  
      SInt16 *samples = NULL;
  
      CMBlockBufferGetDataPointer(audioBlockBuffer, audioBlockBufferOffset, &lengthAtOffset, &totalLength, (char **)(&samples));
  
  
  
      const AudioStreamBasicDescription *audioDescription = CMAudioFormatDescriptionGetStreamBasicDescription(CMSampleBufferGetFormatDescription(sampleBuffer));
  
  
  
      SwrContext *swr = swr_alloc();
  
  
  
      int in_smprt = (int)audioDescription->mSampleRate;
  
      av_opt_set_int(swr, "in_channel_layout",  AV_CH_LAYOUT_MONO, 0);
  
  
  
      av_opt_set_int(swr, "out_channel_layout", audio_st->codec->channel_layout,  0);
  
  
  
      av_opt_set_int(swr, "in_channel_count", audioDescription->mChannelsPerFrame,  0);
  
      av_opt_set_int(swr, "out_channel_count", audio_st->codec->channels,  0);
  
  
  
      av_opt_set_int(swr, "out_channel_layout", audio_st->codec->channel_layout,  0);
  
      av_opt_set_int(swr, "in_sample_rate",     audioDescription->mSampleRate,0);
  
  
  
      av_opt_set_int(swr, "out_sample_rate",    audio_st->codec->sample_rate,0);
  
  
  
      av_opt_set_sample_fmt(swr, "in_sample_fmt",  AV_SAMPLE_FMT_S16, 0);
  
  
  
      av_opt_set_sample_fmt(swr, "out_sample_fmt", audio_st->codec->sample_fmt,  0);
  
  
  
      swr_init(swr);
  
      uint8_t **input = NULL;
  
      int src_linesize;
  
      int in_samples = (int)numSamples;
  
      ret = av_samples_alloc_array_and_samples(&input, &src_linesize, audioDescription->mChannelsPerFrame,
  
                                               in_samples, AV_SAMPLE_FMT_S16P, 0);
  
  
  
  
  
      *input=(uint8_t*)samples;
  
      uint8_t *output=NULL;
  
  
  
  
  
      int out_samples = av_rescale_rnd(swr_get_delay(swr, in_smprt) +in_samples, (int)audio_st->codec->sample_rate, in_smprt, AV_ROUND_UP);
  
  
  
      av_samples_alloc(&output, NULL, audio_st->codec->channels, out_samples, audio_st->codec->sample_fmt, 0);
  
      in_samples = (int)numSamples;
  
      out_samples = swr_convert(swr, &output, out_samples, (const uint8_t **)input, in_samples);
  
  
  
  
  
      aframe->nb_samples =(int) out_samples;
  
  
  
  
  
      ret = avcodec_fill_audio_frame(aframe, audio_st->codec->channels, audio_st->codec->sample_fmt,
  
                               (uint8_t *)output,
  
                               (int) out_samples *
  
                               av_get_bytes_per_sample(audio_st->codec->sample_fmt) *
  
                               audio_st->codec->channels, 1);
  
  
  
      aframe->channel_layout = audio_st->codec->channel_layout;
  
      aframe->channels=audio_st->codec->channels;
  
      aframe->sample_rate= audio_st->codec->sample_rate;
  
  
  
      if (timing_info.presentationTimeStamp.timescale!=0)
  
          pts=(double) timing_info.presentationTimeStamp.value/timing_info.presentationTimeStamp.timescale;
  
  
  
      aframe->pts=pts*audio_st->time_base.den;
  
      aframe->pts = av_rescale_q(aframe->pts, audio_st->time_base, audio_st->codec->time_base);
  
  
  
      ret = avcodec_encode_audio2(c, &pkt, aframe, &got_packet);
  
  
  
      if (ret < 0) {
  
          fprintf(stderr, "Error encoding audio frame: %s\n", av_err2str(ret));
  
          exit(1);
  
      }
  
      swr_free(&swr);
  
      if (got_packet)
  
      {
  
          pkt.stream_index = audio_st->index;
  
  
  
          pkt.pts = av_rescale_q(pkt.pts, audio_st->codec->time_base, audio_st->time_base);
  
          pkt.dts = av_rescale_q(pkt.dts, audio_st->codec->time_base, audio_st->time_base);
  
  
  
          // Write the compressed frame to the media file.
  
         ret = av_interleaved_write_frame(oc, &pkt);
  
         if (ret != 0) {
  
              fprintf(stderr, "Error while writing audio frame: %s\n",
  
                      av_err2str(ret));
  
              exit(1);
  
          }
  
  
  
  }

• Video Processing via Bluetooth

  16 juillet 2012, par kerim yucel

  The application I am currently developing processes each frames using a native code and it should record the video as well. I tried SDK for this purpose but certain restrictions didn't allow me to do so, so I switched to NDK for a video recording code piece.

  Apparently, my algorithm seriously uses CPU, upto %70 percent in the worst case. Before I actually start working on a video recorder, I wanted to try the following approach.

  I will process the preview frames using an android phone and send it to another phone (which uses same application and same model) for recording. My questions are :

  1.Should I try WiFi instead of Bluetooth ? I am developing the application for API 8 so I don't have WiFi-Direct, therefore I should do some socket programming, which would possibly complicate things a bit for me since Bluetooth can easily be set up using SDK.

  2- Will I be available to record the frames as a video at the receiving end ? I will receive each frame with certain metadata embedded to it and should record them using the other phone. I doubt I will be able to do it using SDK, so NDK along with ffmpeg seems to be the best choice ? Any suggestions related to this question will be more than welcome.

  3-Here comes the best part. I am recording the video with the lowest resolution that,after compressed, takes no more than 14mb space for a 10 minute long video. I have to reach the raw frames to send it to other end, encode and compress it. Any ideas related to possible flooding of Bluetooth/Wi-Fi because of big-sized raw frames ?

  Any other approaches and answers will be much appreciated. Thanks.