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• ffmpeg alsa buffer xrun causes skipping for 30 seconds

  26 juillet 2013, par Brianjs

  I am running the following command :

  ffmpeg -y -f rtsp -async 1 -itsoffset -00:00:00.5 -rtsp_transport tcp -i rtsp://root:password@192.168.1.100?streamprofile=Quality -f alsa -ac 2 -ar 44100 -i hw:0 -c:a aac -strict -2 -q:v 4 -q:a 5 -g 15 -threads 10 -s 1024x576 -t 232 -b:v 800k -b:a 64k -c:v flv -pix_fmt yuv420p http://localhost:8090/5flv.ffm \-strict -2 -q:v 4 -q:a 5 -g 15  -threads 10 -r 25 -s 1024x576 -t 232 -b:v 800k -b:a 64k -c:v flv -pix_fmt yuv420p /home/user/videos/7_25_2013_5_1374805807.flv

  Depending on the video quality, for the first 10-35 seconds I keep getting alsa buffer xrun every few seconds or so. This causes 2-4 second skips in the live and recorded stream. After the 30 or so seconds it seams to fix itself, but I cant have it doing that. Is there a fix anyone sees ? Or perhaps a way not to record the first 30 seconds ? I could care less about the live stream.

• Muxing with libav

  14 février 2014, par LordDoskias

  I have a program which is supposed to demux input mpeg-ts, transcode the mpeg2 into h264 and then mux the audio alongside the transcoded video. When I open the resulting muxed file with VLC I neither get audio nor video. Here is the relevant code.

  My main worker loop is as follows :

  void
  
  *writer_thread(void *thread_ctx) {
  
  
  
      struct transcoder_ctx_t *ctx = (struct transcoder_ctx_t *) thread_ctx;
  
      AVStream *video_stream = NULL, *audio_stream = NULL;
  
      AVFormatContext *output_context = init_output_context(ctx, &video_stream, &audio_stream);
  
      struct mux_state_t mux_state = {0};
  
  
  
      //from omxtx
  
      mux_state.pts_offset = av_rescale_q(ctx->input_context->start_time, AV_TIME_BASE_Q, output_context->streams[ctx->video_stream_index]->time_base);
  
  
  
      //write stream header if any
  
      avformat_write_header(output_context, NULL);
  
  
  
      //do not start doing anything until we get an encoded packet
  
      pthread_mutex_lock(&ctx->pipeline.video_encode.is_running_mutex);
  
      while (!ctx->pipeline.video_encode.is_running) {
  
          pthread_cond_wait(&ctx->pipeline.video_encode.is_running_cv, &ctx->pipeline.video_encode.is_running_mutex);
  
      }
  
  
  
      while (!ctx->pipeline.video_encode.eos || !ctx->processed_audio_queue->queue_finished) {
  
          //FIXME a memory barrier is required here so that we don't race 
  
          //on above variables 
  
  
  
          //fill a buffer with video data 
  
          OERR(OMX_FillThisBuffer(ctx->pipeline.video_encode.h, omx_get_next_output_buffer(&ctx->pipeline.video_encode)));
  
  
  
          write_audio_frame(output_context, audio_stream, ctx); //write full audio frame 
  
          //FIXME no guarantee that we have a full frame per packet?
  
          write_video_frame(output_context, video_stream, ctx, &mux_state); //write full video frame
  
          //encoded_video_queue is being filled by the previous command
  
  
  
      }
  
  
  
      av_write_trailer(output_context);
  
  
  
      //free all the resources
  
      avcodec_close(video_stream->codec);
  
      avcodec_close(audio_stream->codec);
  
      /* Free the streams. */
  
      for (int i = 0; i < output_context->nb_streams; i++) {
  
          av_freep(&output_context->streams[i]->codec);
  
          av_freep(&output_context->streams[i]);
  
      }
  
  
  
      if (!(output_context->oformat->flags & AVFMT_NOFILE)) {
  
          /* Close the output file. */
  
          avio_close(output_context->pb);
  
      }
  
  
  
  
  
      /* free the stream */
  
      av_free(output_context);
  
      free(mux_state.pps);
  
      free(mux_state.sps);
  
  }

  The code for initialising libav output context is this :

  static 
  
  AVFormatContext *
  
  init_output_context(const struct transcoder_ctx_t *ctx, AVStream **video_stream, AVStream **audio_stream) {
  
      AVFormatContext *oc;
  
      AVOutputFormat *fmt;
  
      AVStream *input_stream, *output_stream;
  
      AVCodec *c;
  
      AVCodecContext *cc;
  
      int audio_copied = 0; //copy just 1 stream
  
  
  
      fmt = av_guess_format("mpegts", NULL, NULL);
  
      if (!fmt) {
  
          fprintf(stderr, "[DEBUG] Error guessing format, dying\n");
  
          exit(199);
  
      }
  
  
  
      oc = avformat_alloc_context();
  
      if (!oc) {
  
          fprintf(stderr, "[DEBUG] Error allocating context, dying\n");
  
          exit(200);
  
      }
  
  
  
      oc->oformat = fmt;
  
      snprintf(oc->filename, sizeof(oc->filename), "%s", ctx->output_filename);
  
      oc->debug = 1;
  
      oc->start_time_realtime = ctx->input_context->start_time;
  
      oc->start_time = ctx->input_context->start_time;
  
      oc->duration = 0;
  
      oc->bit_rate = 0;
  
  
  
      for (int i = 0; i < ctx->input_context->nb_streams; i++) {
  
          input_stream = ctx->input_context->streams[i];
  
          output_stream = NULL;
  
          if (input_stream->index == ctx->video_stream_index) {
  
              //copy stuff from input video index
  
              c = avcodec_find_encoder(CODEC_ID_H264);
  
              output_stream = avformat_new_stream(oc, c);
  
              *video_stream = output_stream;
  
              cc = output_stream->codec;
  
              cc->width = input_stream->codec->width;
  
              cc->height = input_stream->codec->height;
  
              cc->codec_id = CODEC_ID_H264;
  
              cc->codec_type = AVMEDIA_TYPE_VIDEO;
  
              cc->bit_rate = ENCODED_BITRATE;
  
              cc->time_base = input_stream->codec->time_base;
  
  
  
              output_stream->avg_frame_rate = input_stream->avg_frame_rate;
  
              output_stream->r_frame_rate = input_stream->r_frame_rate;
  
              output_stream->start_time = AV_NOPTS_VALUE;
  
  
  
          } else if ((input_stream->codec->codec_type == AVMEDIA_TYPE_AUDIO) && !audio_copied)  { 
  
              /* i care only about audio */
  
              c = avcodec_find_encoder(input_stream->codec->codec_id);
  
              output_stream = avformat_new_stream(oc, c);
  
              *audio_stream = output_stream;
  
              avcodec_copy_context(output_stream->codec, input_stream->codec);
  
              /* Apparently fixes a crash on .mkvs with attachments: */
  
              av_dict_copy(&output_stream->metadata, input_stream->metadata, 0);
  
              /* Reset the codec tag so as not to cause problems with output format */
  
              output_stream->codec->codec_tag = 0;
  
              audio_copied = 1;
  
          }
  
      }
  
  
  
      for (int i = 0; i < oc->nb_streams; i++) {
  
          if (oc->oformat->flags & AVFMT_GLOBALHEADER)
  
              oc->streams[i]->codec->flags |= CODEC_FLAG_GLOBAL_HEADER;
  
          if (oc->streams[i]->codec->sample_rate == 0)
  
              oc->streams[i]->codec->sample_rate = 48000; /* ish */
  
      }
  
  
  
      if (!(fmt->flags & AVFMT_NOFILE)) {
  
          fprintf(stderr, "[DEBUG] AVFMT_NOFILE set, allocating output container\n");
  
          if (avio_open(&oc->pb, ctx->output_filename, AVIO_FLAG_WRITE) < 0) {
  
              fprintf(stderr, "[DEBUG] error creating the output context\n");
  
              exit(1);
  
          }
  
      }
  
  
  
      return oc;
  
  }

  Finally this is the code for writing audio :

  static
  
  void
  
  write_audio_frame(AVFormatContext *oc, AVStream *st, struct transcoder_ctx_t *ctx) {
  
      AVPacket pkt = {0}; // data and size must be 0;
  
      struct packet_t *source_audio;
  
      av_init_packet(&pkt);
  
  
  
      if (!(source_audio = packet_queue_get_next_item_asynch(ctx->processed_audio_queue))) {
  
          return;
  
      }
  
  
  
      pkt.stream_index = st->index;
  
      pkt.size = source_audio->data_length;
  
      pkt.data = source_audio->data;
  
      pkt.pts = source_audio->PTS;
  
      pkt.dts = source_audio->DTS;
  
      pkt.duration = source_audio->duration;
  
      pkt.destruct = avpacket_destruct;
  
      /* Write the compressed frame to the media file. */
  
      if (av_interleaved_write_frame(oc, &pkt) != 0) {
  
          fprintf(stderr, "[DEBUG] Error while writing audio frame\n");
  
      }
  
  
  
      packet_queue_free_packet(source_audio, 0);
  
  }

  A resulting mpeg4 file can be obtained from here : http://87.120.131.41/dl/mpeg4.h264

  I have ommited the write_video_frame code since it is a lot more complicated and I might be making something wrong there as I'm doing timebase conversation etc. For audio however I'm doing 1:1 copy. Each packet_t packet contains data from av_read_frame from the input mpegts container. In the worst case I'd expect that my audio is working and not my video. However I cannot get either of those to work. Seems the documentation is rather vague on making things like that - I've tried both libav and ffmpeg irc channels to no avail. Any information regarding how I can debug the issue will be greatly appreciated.

• FFmpeg, capturing video is laggy, how to eliminate that ?

  30 avril 2017, par John Smith

  Lets say I dont have a very good computer and my recordings are laggy. Two version I tried :

  ffmpeg -f dshow -framerate 30 -i video="screen-capture-recorder":audio="virtual-audio-capturer" -vcodec libx264 -crf 0 -preset ultrafast -acodec pcm_s16le output.mkv
  
  
  
  ffmpeg -f gdigrab -framerate 30 -i desktop -vcodec libx264 -crf 0 -preset ultrafast -acodec pcm_s16le output.mkv

  I tried to avoid all compressing stuff, still lagging. Although Im recording 1920x1080 video... I dont care how big the result is. Isnt there an option like "store all frames, and encoding a bit later, I dont care the realtime capturing" ?