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• Correct command to transmit audio to ip camera using ffmpeg ?

  4 novembre 2016, par the_naive

  So I found some hints in this discussion on the correct command to transmit audio to Axis IP camera through using ffmpeg in windows, but still I have not managed to successfully transmit audio to the camera.

  The command I’m using is the following :

  ffmpeg -v debug -y -re -f dshow -i "audio=Microphone (2- High Definition Audio Device)" -c:a pcm_mulaw -ac 1 -ar 16000 -b:a 128k -f flv http://oper
  
  ator:operator@10.10.210.2/axis-cgi/audio/transmit.cgi -multiple_requests 1 -reconnect_at_eof 1 -reconnect_streamed 1 -content_type "audio/basic" -report

  The ouput I get following this command is the following :

  ffmpeg started on 2016-11-04 at 17:32:13
  
  Report written to "ffmpeg-20161104-173213.log"
  
  Command line:
  
  ffmpeg -v debug -y -re -f dshow -i "audio=Microphone (2- High Definition Audio Device)" -c:a pcm_mulaw -ac 1 -ar 16000 -b:a 128k -f flv http://operator:operator@10.10.210.2/axis-cgi/audio/transmit.cgi -content_type audio/basic -multiple_requests 1 -reconnect 1 -reconnect_at_eof 1 -reconnect_streamed 1 -report
  
  ffmpeg version N-82225-gb4e9252 Copyright (c) 2000-2016 the FFmpeg developers
  
    built with gcc 5.4.0 (GCC)
  
    configuration: --enable-gpl --enable-version3 --disable-w32threads --enable-dxva2 --enable-libmfx --enable-nvenc --enable-avisynth --enable-bzlib --enable-libebur128 --enable-fontconfig --enable-frei0r --enable-gnutls --enable-iconv --enable-libass --enable-libbluray --enable-libbs2b --enable-libcaca --enable-libfreetype --enable-libgme --enable-libgsm --enable-libilbc --enable-libmodplug --enable-libmp3lame --enable-libopencore-amrnb --enable-libopencore-amrwb --enable-libopenh264 --enable-libopenjpeg --enable-libopus --enable-librtmp --enable-libschroedinger --enable-libsnappy --enable-libsoxr --enable-libspeex --enable-libtheora --enable-libtwolame --enable-libvidstab --enable-libvo-amrwbenc --enable-libvorbis --enable-libvpx --enable-libwavpack --enable-libwebp --enable-libx264 --enable-libx265 --enable-libxavs --enable-libxvid --enable-libzimg --enable-lzma --enable-decklink --enable-zlib
  
    libavutil      55. 35.100 / 55. 35.100
  
    libavcodec     57. 66.101 / 57. 66.101
  
    libavformat    57. 57.100 / 57. 57.100
  
    libavdevice    57.  2.100 / 57.  2.100
  
    libavfilter     6. 66.100 /  6. 66.100
  
    libswscale      4.  3.100 /  4.  3.100
  
    libswresample   2.  4.100 /  2.  4.100
  
    libpostproc    54.  2.100 / 54.  2.100
  
  Splitting the commandline.
  
  Reading option '-v' ... matched as option 'v' (set logging level) with argument 'debug'.
  
  Reading option '-y' ... matched as option 'y' (overwrite output files) with argument '1'.
  
  Reading option '-re' ... matched as option 're' (read input at native frame rate) with argument '1'.
  
  Reading option '-f' ... matched as option 'f' (force format) with argument 'dshow'.
  
  Reading option '-i' ... matched as input file with argument 'audio=Microphone (2- High Definition Audio Device)'.
  
  Reading option '-c:a' ... matched as option 'c' (codec name) with argument 'pcm_mulaw'.
  
  Reading option '-ac' ... matched as option 'ac' (set number of audio channels) with argument '1'.
  
  Reading option '-ar' ... matched as option 'ar' (set audio sampling rate (in Hz)) with argument '16000'.
  
  Reading option '-b:a' ... matched as option 'b' (video bitrate (please use -b:v)) with argument '128k'.
  
  Reading option '-f' ... matched as option 'f' (force format) with argument 'flv'.
  
  Reading option 'http://operator:operator@10.10.210.2/axis-cgi/audio/transmit.cgi' ... matched as output file.
  
  Reading option '-content_type' ... matched as AVOption 'content_type' with argument 'audio/basic'.
  
  Reading option '-multiple_requests' ... matched as AVOption 'multiple_requests' with argument '1'.
  
  Reading option '-reconnect' ... matched as AVOption 'reconnect' with argument '1'.
  
  Reading option '-reconnect_at_eof' ... matched as AVOption 'reconnect_at_eof' with argument '1'.
  
  Reading option '-reconnect_streamed' ... matched as AVOption 'reconnect_streamed' with argument '1'.
  
  Reading option '-report' ... matched as option 'report' (generate a report) with argument '1'.
  
  Trailing options were found on the commandline.
  
  Finished splitting the commandline.
  
  Parsing a group of options: global .
  
  Applying option v (set logging level) with argument debug.
  
  Applying option y (overwrite output files) with argument 1.
  
  Applying option report (generate a report) with argument 1.
  
  Successfully parsed a group of options.
  
  Parsing a group of options: input file audio=Microphone (2- High Definition Audio Device).
  
  Applying option re (read input at native frame rate) with argument 1.
  
  Applying option f (force format) with argument dshow.
  
  Successfully parsed a group of options.
  
  Opening an input file: audio=Microphone (2- High Definition Audio Device).
  
  [dshow @ 00000000000279e0] Selecting pin Capture on audio only
  
  dshow passing through packet of type audio size    88200 timestamp 310221040000 orig timestamp 310221040000 graph timestamp 310226130000 diff 5090000 Microphone (2- High Definition Audio Device)
  
  [dshow @ 00000000000279e0] All info found
  
  Guessed Channel Layout for Input Stream #0.0 : stereo
  
  Input #0, dshow, from 'audio=Microphone (2- High Definition Audio Device)':
  
    Duration: N/A, start: 31022.104000, bitrate: 1411 kb/s
  
      Stream #0:0, 1, 1/10000000: Audio: pcm_s16le, 44100 Hz, stereo, s16, 1411 kb/s
  
  Successfully opened the file.
  
  Parsing a group of options: output file http://operator:operator@10.10.210.2/axis-cgi/audio/transmit.cgi.
  
  Applying option c:a (codec name) with argument pcm_mulaw.
  
  Applying option ac (set number of audio channels) with argument 1.
  
  Applying option ar (set audio sampling rate (in Hz)) with argument 16000.
  
  Applying option b:a (video bitrate (please use -b:v)) with argument 128k.
  
  Applying option f (force format) with argument flv.
  
  Successfully parsed a group of options.
  
  Opening an output file: http://operator:operator@10.10.210.2/axis-cgi/audio/transmit.cgi.
  
  [http @ 0000000001c94040] Setting default whitelist 'http,https,tls,rtp,tcp,udp,crypto,httpproxy'
  
  [http @ 0000000001c94040] request: POST /axis-cgi/audio/transmit.cgi HTTP/1.1
  
  
  
  Transfer-Encoding: chunked
  
  
  
  User-Agent: Lavf/57.57.100
  
  
  
  Accept: */*
  
  
  
  Expect: 100-continue
  
  
  
  Connection: close
  
  
  
  Host: 10.10.210.2
  
  
  
  Icy-MetaData: 1
  
  
  
  
  
  
  
  
  
  [http @ 0000000001c94040] request: POST /axis-cgi/audio/transmit.cgi HTTP/1.1
  
  
  
  Transfer-Encoding: chunked
  
  
  
  User-Agent: Lavf/57.57.100
  
  
  
  Accept: */*
  
  
  
  Connection: close
  
  
  
  Host: 10.10.210.2
  
  
  
  Icy-MetaData: 1
  
  
  
  Authorization: Digest username="operator", realm="AXIS_ACCC8E027F47", nonce="0EcsO3xABQA=ab5efc4740a6c625ecf6a6729d0d67d2b62b615a", uri="/axis-cgi/audio/transmit.cgi", response="4bd3a627b20d6bcaba9e2f595ef6cd2a", algorithm="MD5", qop="auth", cnonce="6a579dd6664b57eb", nc=00000001
  
  
  
  
  
  
  
  
  
  Successfully opened the file.
  
  detected 8 logical cores
  
  [graph 0 input from stream 0:0 @ 0000000001c9f6e0] Setting 'time_base' to value '1/44100'
  
  [graph 0 input from stream 0:0 @ 0000000001c9f6e0] Setting 'sample_rate' to value '44100'
  
  [graph 0 input from stream 0:0 @ 0000000001c9f6e0] Setting 'sample_fmt' to value 's16'
  
  [graph 0 input from stream 0:0 @ 0000000001c9f6e0] Setting 'channel_layout' to value '0x3'
  
  [graph 0 input from stream 0:0 @ 0000000001c9f6e0] tb:1/44100 samplefmt:s16 samplerate:44100 chlayout:0x3
  
  [audio format for output stream 0:0 @ 0000000001c9fa20] Setting 'sample_fmts' to value 's16'
  
  [audio format for output stream 0:0 @ 0000000001c9fa20] Setting 'sample_rates' to value '16000'
  
  [audio format for output stream 0:0 @ 0000000001c9fa20] Setting 'channel_layouts' to value '0x4'
  
  [audio format for output stream 0:0 @ 0000000001c9fa20] auto-inserting filter 'auto-inserted resampler 0' between the filter 'Parsed_anull_0' and the filter 'audio format for output stream 0:0'
  
  [AVFilterGraph @ 000000000002ab20] query_formats: 4 queried, 6 merged, 3 already done, 0 delayed
  
  [auto-inserted resampler 0 @ 0000000001ca4060] [SWR @ 0000000001ca4a80] Using s16p internally between filters
  
  [auto-inserted resampler 0 @ 0000000001ca4060] [SWR @ 0000000001ca4a80] Matrix coefficients:
  
  [auto-inserted resampler 0 @ 0000000001ca4060] [SWR @ 0000000001ca4a80] FC: FL:0.500000 FR:0.500000 
  
  [auto-inserted resampler 0 @ 0000000001ca4060] ch:2 chl:stereo fmt:s16 r:44100Hz -> ch:1 chl:mono fmt:s16 r:16000Hz
  
  Output #0, flv, to 'http://operator:operator@10.10.210.2/axis-cgi/audio/transmit.cgi':
  
    Metadata:
  
      encoder         : Lavf57.57.100
  
      Stream #0:0, 0, 1/1000: Audio: pcm_mulaw ([8][0][0][0] / 0x0008), 16000 Hz, mono, s16, 128 kb/s
  
      Metadata:
  
        encoder         : Lavc57.66.101 pcm_mulaw
  
  Stream mapping:
  
    Stream #0:0 -> #0:0 (pcm_s16le (native) -> pcm_mulaw (native))
  
  Press [q] to stop, [?] for help
  
  cur_dts is invalid (this is harmless if it occurs once at the start per stream)
  
  av_interleaved_write_frame(): Unknown error
  
  No more output streams to write to, finishing.
  
  Error writing trailer of http://operator:operator@10.10.210.2/axis-cgi/audio/transmit.cgi: Error number -10053 occurredsize=       8kB time=00:00:00.49 bitrate= 131.2kbits/s speed=79.6x    
  
  video:0kB audio:8kB subtitle:0kB other streams:0kB global headers:0kB muxing overhead: 2.492485%
  
  Input file #0 (audio=Microphone (2- High Definition Audio Device)):
  
    Input stream #0:0 (audio): 1 packets read (88200 bytes); 1 frames decoded (22050 samples); 
  
    Total: 1 packets (88200 bytes) demuxed
  
  Output file #0 (http://operator:operator@10.10.210.2/axis-cgi/audio/transmit.cgi):
  
    Output stream #0:0 (audio): 1 frames encoded (7984 samples); 1 packets muxed (7984 bytes); 
  
    Total: 1 packets (7984 bytes) muxed
  
  1 frames successfully decoded, 0 decoding errors
  
  [AVIOContext @ 0000000001c9e4c0] Statistics: 0 seeks, 2 writeouts
  
  dshow passing through packet of type audio size    12152 timestamp 310226130000 orig timestamp 310226130000 graph timestamp 310226820000 diff 690000 Microphone (2- High Definition Audio Device)
  
  Conversion failed!

  For some reason, despite setting multiple_requests, reconnect_eof, reconnect_streamed all to true, connection becomes closed.

  Could you please tell me what I’m doing wrong ?

• Qt Video Recorder

  11 mai 2014, par Davlog

  I am trying to create a video recorder with Qt. What I did so far was taking a screenshot of a rectangle on the screen and save it. At the end I use ffmpeg to get a video file out of the images.

  I connected a timer’s signal timeout() to my custom slot which takes the snapshot and saves it to my tmp folder. The timer has an intervall of 1000 / 30. That should be 30 times per second. But 1000 / 30 is a little bit more than 33 milliseconds so I cannot really get 30 fps. It’s a bit more.

  I recorded a youtube video with my recorder and everything was smooth but a little bit faster / slower depending on the intervall.

  So my question basically is how do I get really 30 / 40 / 50 / ... fps ?

• how to convert an MPEGTS file into an FLV file by programming with libavcodec

  9 mai 2014, par Hexing B

  I wanna convert an MPEGTS file into an FLV file with libavcodec APIs(just transformat, video/audio codecs are not changed). Following is the code I found on web. After some hack, it can generate an FLV file, which could not be played. Any clue to fix this ?

  #include 
  
  #include 
  
  #include 
  
  #include 
  
  
  
  #include <libavformat></libavformat>avformat.h>
  
  #include <libavcodec></libavcodec>avcodec.h>
  
  #include <libavutil></libavutil>avutil.h>
  
  #include <libavutil></libavutil>rational.h>
  
  #include <libavdevice></libavdevice>avdevice.h>
  
  #include <libavutil></libavutil>mathematics.h>
  
  #include <libswscale></libswscale>swscale.h>
  
  
  
  static AVStream* add_output_stream(AVFormatContext* output_format_context, AVStream* input_stream) {
  
      AVCodecContext* input_codec_context = NULL;
  
      AVCodecContext* output_codec_context = NULL;
  
  
  
      AVStream* output_stream = NULL;
  
      output_stream = avformat_new_stream(output_format_context, 0);
  
      if (!output_stream) {
  
          printf("Call av_new_stream function failed\n");
  
          return NULL;
  
      }
  
  
  
      input_codec_context = input_stream->codec;
  
      output_codec_context = output_stream->codec;
  
  
  
      output_codec_context->codec_id = input_codec_context->codec_id;
  
      output_codec_context->codec_type = input_codec_context->codec_type;
  
      // output_codec_context->codec_tag = input_codec_context->codec_tag;
  
      output_codec_context->codec_tag = av_codec_get_tag(output_format_context->oformat->codec_tag, input_codec_context->codec_id);
  
      output_codec_context->bit_rate = input_codec_context->bit_rate;
  
      output_codec_context->extradata = input_codec_context->extradata;
  
      output_codec_context->extradata_size = input_codec_context->extradata_size;
  
  
  
      if (av_q2d(input_codec_context->time_base) * input_codec_context->ticks_per_frame > av_q2d(input_stream->time_base) &amp;&amp; av_q2d(input_stream->time_base) &lt; 1.0 / 1000) {
  
          output_codec_context->time_base = input_codec_context->time_base;
  
          output_codec_context->time_base.num *= input_codec_context->ticks_per_frame;
  
      } else {
  
          output_codec_context->time_base = input_stream->time_base;
  
      }
  
      switch (input_codec_context->codec_type) {
  
      case AVMEDIA_TYPE_AUDIO:
  
          output_codec_context->channel_layout = input_codec_context->channel_layout;
  
          output_codec_context->sample_rate = input_codec_context->sample_rate;
  
          output_codec_context->channels = input_codec_context->channels;
  
          output_codec_context->frame_size = input_codec_context->frame_size;
  
          if ((input_codec_context->block_align == 1 &amp;&amp; input_codec_context->codec_id == CODEC_ID_MP3) || input_codec_context->codec_id == CODEC_ID_AC3) {
  
              output_codec_context->block_align = 0;
  
          } else {
  
              output_codec_context->block_align = input_codec_context->block_align;
  
          }
  
          break;
  
      case AVMEDIA_TYPE_VIDEO:
  
          output_codec_context->pix_fmt = input_codec_context->pix_fmt;
  
          output_codec_context->width = input_codec_context->width;
  
          output_codec_context->height = input_codec_context->height;
  
          output_codec_context->has_b_frames = input_codec_context->has_b_frames;
  
          if (output_format_context->oformat->flags &amp; AVFMT_GLOBALHEADER) {
  
              output_codec_context->flags |= CODEC_FLAG_GLOBAL_HEADER;
  
          }
  
          // output_codec_context->strict_std_compliance = FF_COMPLIANCE_EXPERIMENTAL;
  
          break;
  
      default:
  
          break;
  
      }
  
  
  
      return output_stream;
  
  }
  
  
  
  int main(int argc, char* argv[]) {
  
      const char* input = argv[1];
  
      const char* output_prefix = NULL;
  
      char* segment_duration_check = 0;
  
      const char* index = NULL;
  
      char* tmp_index = NULL;
  
      const char* http_prefix = NULL;
  
      long max_tsfiles = 0;
  
      double prev_segment_time = 0;
  
      double segment_duration = 0;
  
  
  
      AVInputFormat* ifmt = NULL;
  
      AVOutputFormat* ofmt = NULL;
  
      AVFormatContext* ic = NULL;
  
      AVFormatContext* oc = NULL;
  
      AVStream* video_st = NULL;
  
      AVStream* audio_st = NULL;
  
      AVCodec* codec = NULL;
  
      AVDictionary* pAVDictionary = NULL;
  
  
  
      av_register_all();
  
      av_log_set_level(AV_LOG_DEBUG);
  
  
  
      char szError[256] = {0};
  
      int nRet = avformat_open_input(&amp;ic, input, ifmt, &amp;pAVDictionary);
  
      if (nRet != 0) {
  
          av_strerror(nRet, szError, 256);
  
          printf(szError);
  
          printf("\n");
  
          printf("Call avformat_open_input function failed!\n");
  
          return 0;
  
      }
  
  
  
      if (avformat_find_stream_info(ic, NULL) &lt; 0) {
  
          printf("Call av_find_stream_info function failed!\n");
  
          return 0;
  
      }
  
  
  
      ofmt = av_guess_format(NULL, argv[2], NULL);
  
      if (!ofmt) {
  
          printf("Call av_guess_format function failed!\n");
  
          return 0;
  
      }
  
  
  
      oc = avformat_alloc_context();
  
      if (!oc) {
  
          printf("Call av_guess_format function failed!\n");
  
          return 0;
  
      }
  
      oc->oformat = ofmt;
  
  
  
      int video_index = -1, audio_index = -1;
  
      unsigned int i;
  
      for (i = 0; i &lt; ic->nb_streams &amp;&amp; (video_index &lt; 0 || audio_index &lt; 0); i++) {
  
          switch (ic->streams[i]->codec->codec_type) {
  
          case AVMEDIA_TYPE_VIDEO:
  
              video_index = i;
  
              ic->streams[i]->discard = AVDISCARD_NONE;
  
              video_st = add_output_stream(oc, ic->streams[i]);
  
              if (video_st->codec->codec_id == CODEC_ID_H264) {
  
                  video_st->codec->opaque = av_bitstream_filter_init("h264_mp4toannexb");
  
              }
  
              break;
  
          case AVMEDIA_TYPE_AUDIO:
  
              audio_index = i;
  
              ic->streams[i]->discard = AVDISCARD_NONE;
  
              audio_st = add_output_stream(oc, ic->streams[i]);
  
              if (audio_st->codec->codec_id == CODEC_ID_AAC &amp;&amp; !audio_st->codec->extradata_size) {
  
                  audio_st->codec->opaque = av_bitstream_filter_init("aac_adtstoasc");
  
              }
  
              break;
  
          default:
  
              ic->streams[i]->discard = AVDISCARD_ALL;
  
              break;
  
          }
  
      }
  
      codec = avcodec_find_decoder(video_st->codec->codec_id);
  
      if (codec == NULL) {
  
          printf("Call avcodec_find_decoder function failed!\n");
  
          return 0;
  
      }
  
  
  
      if (avcodec_open2(video_st->codec, codec, NULL) &lt; 0) {
  
          printf("Call avcodec_open function failed !\n");
  
          return 0;
  
      }
  
  
  
      if (avio_open(&amp;oc->pb, argv[2], AVIO_FLAG_WRITE) &lt; 0) {
  
          return 0;
  
      }
  
  
  
      if (avformat_write_header(oc, &amp;pAVDictionary)) {
  
          printf("Call avformat_write_header function failed.\n");
  
          return 0;
  
      }
  
  
  
      int decode_done = 0;
  
      do {
  
          AVPacket packet;
  
          decode_done = av_read_frame(ic, &amp;packet);
  
          if (decode_done &lt; 0) {
  
              break;
  
          }
  
  
  
          if (packet.stream_index == video_index &amp;&amp; (packet.flags &amp; AV_PKT_FLAG_KEY) &amp;&amp; video_st->codec->opaque != NULL) {
  
              AVPacket pkt = packet;
  
              int a = av_bitstream_filter_filter(video_st->codec->opaque, video_st->codec, NULL, &amp;pkt.data, &amp;pkt.size,
  
                      packet.data, packet.size, packet.flags &amp; AV_PKT_FLAG_KEY);
  
              if (a == 0) {
  
                  memmove(packet.data, pkt.data, pkt.size);
  
                  packet.size = pkt.size;
  
              } else if (a > 0) {
  
                  packet = pkt;
  
              }
  
          }
  
          else if (packet.stream_index == audio_index &amp;&amp; audio_st->codec->opaque != NULL) {
  
              AVPacket pkt = packet;
  
              int a = av_bitstream_filter_filter(audio_st->codec->opaque, audio_st->codec, NULL, &amp;pkt.data, &amp;pkt.size,
  
                      packet.data, packet.size, packet.flags &amp; AV_PKT_FLAG_KEY);
  
              if (a == 0) {
  
                  memmove(packet.data, pkt.data, pkt.size);
  
                  packet.size = pkt.size;
  
              } else if (a > 0) {
  
                  packet = pkt;
  
              }
  
          }
  
          nRet = av_interleaved_write_frame(oc, &amp;packet);
  
          if (nRet &lt; 0) {
  
              printf("Call av_interleaved_write_frame function failed\n");
  
          } else if (nRet > 0) {
  
              printf("End of stream requested\n");
  
              av_free_packet(&amp;packet);
  
                  break;
  
          }
  
          av_free_packet(&amp;packet);
  
      } while(!decode_done);
  
  
  
      av_write_trailer(oc);
  
  
  
      av_bitstream_filter_close(video_st->codec->opaque);  
  
      av_bitstream_filter_close(audio_st->codec->opaque);  
  
      avcodec_close(video_st->codec);
  
      unsigned int k;
  
      for(k = 0; k &lt; oc->nb_streams; k++) {
  
          av_freep(&amp;oc->streams[k]->codec);
  
          av_freep(&amp;oc->streams[k]);
  
      }
  
      av_free(oc);
  
      getchar();
  
      return 0;
  
  }