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• ffplay cannot play more than one song

  5 février 2020, par Bernie gach

  i have taken ffplay.c file from http://ffmpeg.org/doxygen/trunk/ffplay_8c-source.html and re edited it to a cpp file to embed in my win32 gui application . i have made the following changes to it.

  1. made the int main function into a local function as follows, i can pass the HWND to embedd the player

  void Ffplay::play_song(string file, HWND parent, bool* successfull)
  
  {
  
      int flags;
  
      VideoState* is;
  
      input_filename = file; 
  
      /* register all codecs, demux and protocols */
  
  #if CONFIG_AVDEVICE
  
      avdevice_register_all();
  
  #endif
  
      //avformat_network_init();
  
      //check whether the filename is valid
  
      if (input_filename.empty())
  
      {
  
          logger.log(logger.LEVEL_ERROR, "filename %s is not valid\n", file);
  
          return;
  
      }
  
      if (display_disable)
  
      {
  
          video_disable = 1;
  
      }
  
      flags = SDL_INIT_VIDEO | SDL_INIT_AUDIO | SDL_INIT_TIMER;
  
      if (audio_disable)
  
          flags &= ~SDL_INIT_AUDIO;
  
      else
  
      {
  
          /* Try to work around an occasional ALSA buffer underflow issue when the
  
           * period size is NPOT due to ALSA resampling by forcing the buffer size. */
  
          if (!SDL_getenv("SDL_AUDIO_ALSA_SET_BUFFER_SIZE"))
  
              SDL_setenv("SDL_AUDIO_ALSA_SET_BUFFER_SIZE", "1", 1);
  
      }
  
      if (display_disable)
  
          flags &= ~SDL_INIT_VIDEO;
  
      SDL_SetMainReady();
  
      if (SDL_Init(flags))
  
      {
  
          logger.log(logger.LEVEL_ERROR, "Could not initialize SDL - %s\n", SDL_GetError());
  
          logger.log(logger.LEVEL_ERROR, "(Did you set the DISPLAY variable?)\n");
  
          return;
  
      }
  
      //Initialize optional fields of a packet with default values.
  
      //Note, this does not touch the data and size members, which have to be initialized separately.
  
      av_init_packet(&flush_pkt);
  
      flush_pkt.data = (uint8_t*)&flush_pkt;
  
  
  
      if (!display_disable)
  
      {
  
          int flags = SDL_WINDOW_HIDDEN;
  
          if (alwaysontop)
  
  #if SDL_VERSION_ATLEAST(2,0,5)
  
              flags |= SDL_WINDOW_ALWAYS_ON_TOP;
  
  #else
  
              logger.log(logger.LEVEL_INFO, "SDL version doesn't support SDL_WINDOW_ALWAYS_ON_TOP. Feature will be inactive.\n");
  
  #endif
  
          if (borderless)
  
              flags |= SDL_WINDOW_BORDERLESS;
  
          else
  
              flags |= SDL_WINDOW_RESIZABLE;
  
          SDL_InitSubSystem(flags);
  
          ShowWindow(parent, true);
  
          //window = SDL_CreateWindow(program_name, SDL_WINDOWPOS_UNDEFINED, SDL_WINDOWPOS_UNDEFINED, default_width, default_height, flags);
  
          window = SDL_CreateWindowFrom(parent);
  
          SDL_SetHint(SDL_HINT_RENDER_SCALE_QUALITY, "linear");
  
          if (window) {
  
              renderer = SDL_CreateRenderer(window, -1, SDL_RENDERER_ACCELERATED | SDL_RENDERER_PRESENTVSYNC);
  
              if (!renderer)
  
              {
  
                  logger.log(logger.LEVEL_ERROR, "Failed to initialize a hardware accelerated renderer: %s\n", SDL_GetError());
  
                  renderer = SDL_CreateRenderer(window, -1, 0);
  
              }
  
              if (renderer)
  
              {
  
                  if (!SDL_GetRendererInfo(renderer, &renderer_info))
  
                  {
  
                      logger.log(logger.LEVEL_INFO, "Initialized %s renderer.\n", renderer_info.name);
  
                  }
  
              }
  
          }
  
          if (!window || !renderer || !renderer_info.num_texture_formats)
  
          {
  
              logger.log(logger.LEVEL_ERROR, "Failed to create window or renderer: %s\n", SDL_GetError());
  
              return;
  
          }
  
      }
  
  
  
      is = stream_open(input_filename.c_str(), file_iformat);
  
      if (!is)
  
      {
  
          logger.log(logger.LEVEL_ERROR, "Failed to initialize VideoState!\n");
  
          return;
  
      }
  
      //the song is playing now
  
      *successfull = true;
  
      event_loop(is);
  
      //the song has quit;
  
      *successfull = false;
  
  }

  2. changed the callback functions as the static ones couldn’t be used by c++ eg,

  void Ffplay::static_sdl_audio_callback(void* opaque, Uint8* stream, int len)
  
  {
  
      static_cast(opaque)->sdl_audio_callback(opaque, stream, len);
  
  }

  closing doesn’t change from the main file to close the audio and sdl framework

  void Ffplay::do_exit(VideoState* is)
  
  {
  
      abort = true;
  
      if(is)
  
      {
  
          stream_close(is);
  
      }
  
      if (renderer)
  
          SDL_DestroyRenderer(renderer);
  
      if (window)
  
           SDL_DestroyWindow(window);
  
  #if CONFIG_AVFILTER
  
      av_freep(&vfilters_list);
  
  #endif
  
      avformat_network_deinit();
  
      SDL_Quit();
  
  
  
  }

  i call the functions as follows from main gui

  ft=std::async(launch::async, &Menu::play_song, this, songs_to_play.at(0));

  the menu::play_song function is :

  void Menu::play_song(wstring song_path)
  
  {
  
      ready_to_play_song = false;
  
      OutputDebugString(L"\nbefore song\n");
  
      using std::future;
  
      using std::async;
  
      using std::launch;
  
  
  
      string input{ song_path.begin(),song_path.end() };
  
      Ffplay ffplay;
  
      ffplay.play_song(input, h_sdl_window, &song_opened);
  
  
  
      OutputDebugString(L"\nafter song\n");
  
      ready_to_play_song = true;
  
  }

  THE PROBLEM is i can only play one song . if i call the menu::play_song function again the sound is missing and the video/art cover is occasionally missing also. it seems some resources are not been released or something like that.

  i have localised the proble to this function

  int Ffplay::packet_queue_get(PacketQueue* q, AVPacket* pkt, int block, int* serial)
  
  {
  
  
  
      MyAVPacketList* pkt1;
  
      int ret;
  
      int count=0;
  
      SDL_LockMutex(q->mutex);
  
  
  
      for (;;) 
  
      {
  
  
  
  
  
          if (q->abort_request)
  
          {
  
              ret = -1;
  
              break;
  
          }
  
  
  
          pkt1 = q->first_pkt;
  
          if (pkt1) {
  
              q->first_pkt = pkt1->next;
  
              if (!q->first_pkt)
  
                  q->last_pkt = NULL;
  
              q->nb_packets--;
  
              q->size -= pkt1->pkt.size + sizeof(*pkt1);
  
              q->duration -= pkt1->pkt.duration;
  
              *pkt = pkt1->pkt;
  
              if (serial)
  
                  *serial = pkt1->serial;
  
              av_free(pkt1);
  
              ret = 1;
  
              break;
  
          }
  
          else if (!block) {
  
              ret = 0;
  
              break;
  
          }
  
          else 
  
          {
  
              logger.log(logger.LEVEL_INFO, "packet_queue before");
  
              SDL_CondWait(q->cond, q->mutex);
  
              logger.log(logger.LEVEL_INFO, "packet_queue after");
  
  
  
          }
  
      }
  
      SDL_UnlockMutex(q->mutex);
  
      return ret;
  
  }

  the call to SDL_CondWait(q->cond, q->mutex); never returns

• ffmpeg to Youtube Live stops working. ffmpeg continues to run

  10 février 2020, par Robert Pringle

  I’m streaming a mobotix camera to Youtube Live.

  This works correctly for a period of time. Youtube then reports the stream as "noData".
  ffmpeg continues to run, but no video is being received by Youtube.

  The problem is intermittant.

  Restarting the ffmpeg process makes Youtube start receiving the stream again.

  ffmpeg is being run as a systemd service.

  ffmpeg is run with the following command :

  ffmpeg -nostdin  -f lavfi  -i anullsrc  -rtsp_transport tcp -thread_queue_size 512 -i "rtsp://@192.168.1.1:554/mobotix.h264"   -tune zerolatency  -vcodec libx264  -pix_fmt + -c:v copy  -c:a aac  -strict experimental  -f flv rtmp://a.rtmp.youtube.com/live2/

  Logs only show : ffmpeg[47357]: [48.0K blob data]

  Any ideas on how to solve this ? Or improve the logging.

  Edit :
  Logs from starting ffmpeg :

      Feb 07 17:57:00 localhost.localdomain ffmpeg[49525]: [48.0K blob data]
  
      Feb 07 17:52:47 localhost.localdomain ffmpeg[49525]:       encoder         : Lavc58.54.100 aac
  
      Feb 07 17:52:47 localhost.localdomain ffmpeg[49525]:     Metadata:
  
      Feb 07 17:52:47 localhost.localdomain ffmpeg[49525]:     Stream #0:1: Audio: aac (LC) ([10][0][0][0] / 0x000A), 44100 Hz, stereo, fltp, 128 kb/s
  
      Feb 07 17:52:47 localhost.localdomain ffmpeg[49525]:     Stream #0:0: Video: h264 (Constrained Baseline) ([7][0][0][0] / 0x0007), yuv420p(progressive), 1024x768, q=2-31, 12.50 tbr, 1k tbn, 90k tbc
  
      Feb 07 17:52:47 localhost.localdomain ffmpeg[49525]:     encoder         : Lavf58.29.100
  
      Feb 07 17:52:47 localhost.localdomain ffmpeg[49525]:   Metadata:
  
      Feb 07 17:52:47 localhost.localdomain ffmpeg[49525]: Output #0, flv, to 'rtmp://a.rtmp.youtube.com/live2/':
  
      Feb 07 17:52:47 localhost.localdomain ffmpeg[49525]:   Stream #0:0 -> #0:1 (pcm_u8 (native) -> aac (native))
  
      Feb 07 17:52:47 localhost.localdomain ffmpeg[49525]:   Stream #1:0 -> #0:0 (copy)
  
      Feb 07 17:52:47 localhost.localdomain ffmpeg[49525]: Stream mapping:
  
      Feb 07 17:52:46 localhost.localdomain ffmpeg[49525]:     Stream #1:0: Video: h264 (Constrained Baseline), yuv420p(progressive), 1024x768, 12.50 tbr, 90k tbn, 180k tbc
  
      Feb 07 17:52:46 localhost.localdomain ffmpeg[49525]:   Duration: N/A, start: 136.139322, bitrate: N/A
  
      Feb 07 17:52:46 localhost.localdomain ffmpeg[49525]:     comment         : mobotix.h264
  
      Feb 07 17:52:46 localhost.localdomain ffmpeg[49525]:     title           : Mobotix IP-Camera (H.264, unicast)
  
      Feb 07 17:52:46 localhost.localdomain ffmpeg[49525]:   Metadata:
  
      Feb 07 17:52:46 localhost.localdomain ffmpeg[49525]: Input #1, rtsp, from 'rtsp://@192.168.1.1:554/mobotix.h264':
  
      Feb 07 17:52:44 localhost.localdomain ffmpeg[49525]: [h264 @ 0x559ee7857940] concealing 2400 DC, 2400 AC, 2400 MV errors in P frame
  
      Feb 07 17:52:43 localhost.localdomain ffmpeg[49525]:     Stream #0:0: Audio: pcm_u8, 44100 Hz, stereo, u8, 705 kb/s
  
      Feb 07 17:52:43 localhost.localdomain ffmpeg[49525]:   Duration: N/A, start: 0.000000, bitrate: 705 kb/s
  
      Feb 07 17:52:43 localhost.localdomain ffmpeg[49525]: Input #0, lavfi, from 'anullsrc':
  
      Feb 07 17:52:43 localhost.localdomain ffmpeg[49525]:   libpostproc    55.  5.100 / 55.  5.100
  
      Feb 07 17:52:43 localhost.localdomain ffmpeg[49525]:   libswresample   3.  5.100 /  3.  5.100
  
      Feb 07 17:52:43 localhost.localdomain ffmpeg[49525]:   libswscale      5.  5.100 /  5.  5.100
  
      Feb 07 17:52:43 localhost.localdomain ffmpeg[49525]:   libavresample   4.  0.  0 /  4.  0.  0
  
      Feb 07 17:52:43 localhost.localdomain ffmpeg[49525]:   libavfilter     7. 57.100 /  7. 57.100
  
      Feb 07 17:52:43 localhost.localdomain ffmpeg[49525]:   libavdevice    58.  8.100 / 58.  8.100
  
      Feb 07 17:52:43 localhost.localdomain ffmpeg[49525]:   libavformat    58. 29.100 / 58. 29.100
  
      Feb 07 17:52:43 localhost.localdomain ffmpeg[49525]:   libavcodec     58. 54.100 / 58. 54.100
  
      Feb 07 17:52:43 localhost.localdomain ffmpeg[49525]:   libavutil      56. 31.100 / 56. 31.100
  
      Feb 07 17:52:43 localhost.localdomain ffmpeg[49525]:   configuration: --prefix=/usr --bindir=/usr/bin --datadir=/usr/share/ffmpeg --docdir=/usr/share/doc/ffmpeg --incdir=/usr/include/ffmpeg --libdir=/usr/lib64 --mandir=/usr/share/man --arch=x86_64 --optflags='-O2 >
  
      Feb 07 17:52:43 localhost.localdomain ffmpeg[49525]:   built with gcc 8 (GCC)
  
      Feb 07 17:52:43 localhost.localdomain ffmpeg[49525]: ffmpeg version 4.2.1 Copyright (c) 2000-2019 the FFmpeg developers

  Running the command manually gave me logs and exited, whereas the systemd service keeps running.

      [flv @ 0x560541a8b3c0] Timestamps are unset in a packet for stream 0. This is deprecated and will stop working in the future. Fix your code to set the timestamps properly
  
      av_interleaved_write_frame(): Broken pipekB time=03:35:18.84 bitrate=3131.8kbits/s speed=0.999x    
  
          Last message repeated 1 times
  
      [flv @ 0x560541a8b3c0] Failed to update header with correct duration.
  
      [flv @ 0x560541a8b3c0] Failed to update header with correct filesize.
  
      Error writing trailer of rtmp://a.rtmp.youtube.com/live2/: Broken pipe
  
      frame=134427 fps= 10 q=-1.0 Lsize= 4938869kB time=03:35:18.96 bitrate=3131.8kbits/s speed=0.997x    
  
      video:4923745kB audio:3296kB subtitle:0kB other streams:0kB global headers:0kB muxing overhead: 0.240068%
  
      [aac @ 0x560541a6d680] Qavg: 65536.000
  
      Conversion failed!

• FFmpeg C/C++ audio video streams (mic, webcam) sync to mp4

  21 février 2020, par NIAZ

  I am trying to capture audio and video using a microphone and a webcam into an mp4 file. The recorded file is playable but over the time the audio starts drifting away from video and for a longer period of time the gap increases. Both the audio and video is handled in separate threads, for the audio I am using audiofifo adapted from the transcode_acc.c example https://github.com/FFmpeg/FFmpeg/blob/master/doc/examples/transcode_aac.c
  Here is how I am setting up the streams
  Video :

  video_output_codec_ctx = video_stream->codec;
  
  video_output_codec_ctx->bit_rate = 2000000;
  
  video_output_codec_ctx->codec_id = AV_CODEC_ID_MPEG4;
  
  video_output_codec_ctx->width = 640;
  
  video_output_codec_ctx->height = 480;
  
  video_stream->time_base = (AVRational){1, fps};
  
  video_output_codec_ctx->time_base = video_stream->time_base;
  
  video_output_codec_ctx->pix_fmt = AV_PIX_FMT_YUV420P;
  
  video_output_codec_ctx->codec_type = AVMEDIA_TYPE_VIDEO;

  Audio :

  audio_output_codec_ctx->channels       = OUTPUT_CHANNELS; // 2
  
  audio_output_codec_ctx->channel_layout = av_get_default_channel_layout(OUTPUT_CHANNELS);
  
  audio_output_codec_ctx->sample_rate    = audio_input_codec_ctx->sample_rate;
  
  audio_output_codec_ctx->sample_fmt     = audio_output_codec->sample_fmts[0];
  
  audio_output_codec_ctx->bit_rate       = OUTPUT_BIT_RATE; // 96000
  
  audio_output_codec_ctx->strict_std_compliance = FF_COMPLIANCE_EXPERIMENTAL;
  
  
  
  /* Set the sample rate for the container. */
  
  audio_stream->time_base.den = audio_input_codec_ctx->sample_rate;
  
  audio_stream->time_base.num = 1;

  For the video pts, I increment the index by 1 once the frame is encoded and before sending the frame to the encoder I use rescale and also after receiving the frame,afterwards the packets are written via av_interleaved_write_frame().

  output_frame->pts = av_rescale_q(video_frame_index, video_output_codec_ctx->time_base, video_input_format_ctx->streams[0]->time_base);
  
  
  
  error = avcodec_send_frame(video_output_codec_ctx, output_frame);
  
  
  
  error = avcodec_receive_packet(video_output_codec_ctx, &output_packet);
  
  
  
  output_packet.stream_index = video_index;
  
  
  
  output_packet.pts = av_rescale_q_rnd(output_packet.pts, video_input_format_ctx->streams[0]->time_base, output_format_context->streams[video_index]->time_base, (AVRounding)(AV_ROUND_NEAR_INF|AV_ROUND_PASS_MINMAX));
  
  
  
  output_packet.dts = av_rescale_q_rnd(output_packet.dts, video_input_format_ctx->streams[0]->time_base, output_format_context->streams[video_index]->time_base, (AVRounding)(AV_ROUND_NEAR_INF|AV_ROUND_PASS_MINMAX));
  
  
  
  output_packet.duration = ((output_format_context->streams[0]->time_base.den / output_format_context->streams[0]->time_base.num) / video_output_codec_ctx->time_base.den);
  
  
  
  output_packet.pos = -1;  
  
  
  
  video_frame_index++;

  For the audio pts, I increment by frame->nb_samples once the frame is encoded I use rescale, afterwards the packets are written via av_interleaved_write_frame().

  frame->pts = aud_pts;
  
  aud_pts += frame->nb_samples;
  
  
  
  error = avcodec_send_frame(audio_output_codec_ctx, frame);
  
  
  
  error = avcodec_receive_packet(audio_output_codec_ctx, &output_packet);
  
  
  
  output_packet.stream_index = audio_index;
  
  
  
  output_packet.pts = av_rescale_q_rnd(output_packet.pts, audio_output_codec_ctx->time_base, output_format_context->streams[audio_index]->time_base, (AVRounding)(AV_ROUND_NEAR_INF|AV_ROUND_PASS_MINMAX));
  
  
  
  output_packet.dts = av_rescale_q_rnd(output_packet.dts, audio_output_codec_ctx->time_base, output_format_context->streams[audio_index]->time_base, (AVRounding)(AV_ROUND_NEAR_INF|AV_ROUND_PASS_MINMAX));
  
  
  
  output_packet.duration = av_rescale_q(output_packet.duration, audio_output_codec_ctx->time_base, output_format_context->streams[audio_index]->time_base);

  I am new to the FFmpeg C API, have tried various resources / posts over the internet but still haven’t been able to sync the audio and video in a robust way. Here are few question that I want to understand which will help me to resolve this. Any thoughts are really appreciated on this.

  1. Can the FFmpeg C API handle sync internally or this is something which needs to be handled from the caller side ?

  2. Am I setting up the PTS correctly both for audio and video in the first place ? I have noticed that when I use fps lower than 20, I get Invalid pts (66667) <= last (66667) Operation not permitted from the encoder. This must be something wrong with the way I am currently setting the video PTS. How can I set up the video PTS to handle lower fps ?

  3. I am also trying to adopt the idea of clocks sync from dranger’s tutorial, not sure whether this would be suitable for my use case, things like where to setup the audio and video clocks as he used only decoders, for the audio I am using fifo and not sure how to adjust the samples based on the clocks sync, also the way to call and setup the refresh timer ?

  4. Is there a better mechanism for creating a robust sync for my use case which can handle both audio and video if they go out of sync, would be great to have an idea about the samples and frames adjustment based on that ?