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• FFMPEG rtmp live stream with android fps is too low

  27 novembre 2015, par 蔡明原

  I’m interesting live stream from android to youtube.
  I have compile ffmpeg and write jni to live stream with android.
  but the fps is too low, it always has 3 5 fps, the user view is not good.
  how can i do to change the fps.
  this is my jni code with ffmpeg init.

  AVCodecContext *video_codec_ctx = video_stream->codec;
  
    video_codec_ctx->codec_id = video_codec->id;
  
    video_codec_ctx->codec_type = AVMEDIA_TYPE_VIDEO;
  
    video_codec_ctx->level = 31;
  
  
  
    video_codec_ctx->width = width;
  
    video_codec_ctx->height = height;
  
    video_codec_ctx->pix_fmt = PIX_FMT_YUV420P;
  
  
  
    video_codec_ctx->rc_max_rate = 0;
  
    video_codec_ctx->rc_buffer_size = 0;
  
    video_codec_ctx->gop_size = 30;
  
    video_codec_ctx->max_b_frames = 0;
  
    video_codec_ctx->slices = 8;   
  
    video_codec_ctx->b_frame_strategy = 1;
  
    video_codec_ctx->coder_type = 0;
  
    video_codec_ctx->me_cmp = 1;
  
    video_codec_ctx->me_range = 16;
  
    video_codec_ctx->qmin = 10;
  
    video_codec_ctx->qmax = 51;
  
    video_codec_ctx->keyint_min = 25;
  
    video_codec_ctx->refs= 3;   
  
    video_codec_ctx->trellis = 0;
  
    video_codec_ctx->scenechange_threshold = 40;
  
    video_codec_ctx->flags |= CODEC_FLAG_LOOP_FILTER;
  
    video_codec_ctx->me_method = ME_HEX;
  
    video_codec_ctx->me_subpel_quality = 6;
  
    video_codec_ctx->i_quant_factor = 0.71;
  
    video_codec_ctx->qcompress = 0.6;
  
    video_codec_ctx->max_qdiff = 4;
  
    video_codec_ctx->time_base.den = 30;
  
    video_codec_ctx->time_base.num = 1;
  
    video_codec_ctx->bit_rate = 5200*1000;
  
    video_codec_ctx->bit_rate_tolerance = 0;
  
    video_codec_ctx->flags2 |= 0x00000100;
  
  
  
    fmt_context->bit_rate = 6000*1000;
  
  
  
    av_opt_set(video_codec_ctx, "partitions", "i8x8,i4x4,p8x8,b8x8", 0);
  
  
  
    av_opt_set_int(video_codec_ctx, "direct-pred", 1, 0);
  
    av_opt_set_int(video_codec_ctx, "rc-lookahead", 0, 0);
  
    av_opt_set_int(video_codec_ctx, "fast-pskip", 1, 0);
  
    av_opt_set_int(video_codec_ctx, "mixed-refs", 1, 0);
  
    av_opt_set_int(video_codec_ctx, "8x8dct", 0, 0);
  
    av_opt_set_int(video_codec_ctx, "weightb", 0, 0);
  
  
  
    if(fmt_context->oformat->flags & AVFMT_GLOBALHEADER)
  
        video_codec_ctx->flags |= CODEC_FLAG_GLOBAL_HEADER;
  
  
  
    LOGI("Opening video codec");
  
    AVDictionary *vopts = NULL;
  
    av_dict_set(&vopts, "profile", "main", 0);
  
    //av_dict_set(&vopts, "vprofile", "main", 0);
  
    av_dict_set(&vopts, "rc-lookahead", 0, 0);
  
    av_dict_set(&vopts, "tune", "film", 0);  
  
    av_dict_set(&vopts, "preset", "ultrafast", 0);  
  
    av_opt_set(video_codec_ctx->priv_data,"tune","film",0);
  
    av_opt_set(video_codec_ctx->priv_data,"preset","ultrafast",0);
  
    av_opt_set(video_codec_ctx->priv_data,"tune","film",0);
  
    int open_res = avcodec_open2(video_codec_ctx, video_codec, &vopts);

• Syncronising Video/audio with two different sources

  22 mai 2016, par Indy

  I have an issue with an very old film that I’m trying to recover. I have two source of the same film.

  Source 1 (from ffmeg -i) :

  Duration: 02:43:35.77, start: 0.000000, bitrate: 1342 kb/s
  
  Stream #0:0: Video: mpeg4 (Simple Profile) (DIVX / 0x58564944), yuv420p, 720x480 [SAR 1:1 DAR 3:2], 1200 kb/s, 29.97 fps, 29.97 tbr, 29.97 tbn, 30k tbc
  
  Stream #0:1: Audio: mp3 (U[0][0][0] / 0x0055), 48000 Hz, stereo, s16p, 128 kb/s

  Source 2 (from ffmeg -i) :

  Duration: 02:46:11.04, start: 0.000000, bitrate: 1625 kb/s
  
  Stream #0:0(und): Video: h264 (High) (avc1 / 0x31637661), yuv420p, 1200x720 [SAR 1:1 DAR 5:3], 1431 kb/s, 30 fps, 30 tbr, 30 tbn, 60 tbc (default)
  
  Metadata:
  
    handler_name    : VideoHandler
  
  Stream #0:1(und): Audio: aac (LC) (mp4a / 0x6134706D), 44100 Hz, stereo, fltp, 191 kb/s (default)
  
  Metadata:
  
    creation_time   : 2014-06-22 12:06:22
  
    handler_name    : IsoMedia File Produced by Google, 5-11-2011

  What I would like is to take the audio from source 1 and video from source 2 and create a new file with them.

  I have tried changing the source 1 to 30 fps and then using its audio with source 2’s video but the audio ’fps’ is really after sync after sometime. Perhaps my approach is wrong.

  what’s the best way to take the audio from source 1 synced with source 2’s video ?

  Any help is appreciated. Thanks,
  Indy

• Stream from WebRTC input to HLS

  9 octobre 2016, par acroca

  I need to do video streaming where the recording is happening on a browser and has to be consumed as HLS on mobile devices.

  It seems I need some media server and send the camera stream as WebRTC to it. It seems Janus and Kurento can do that task and output RTP streams. I’m not sure now how to convert that RTP stream to HLS, I guess I could hook ffmpeg on that RTP stream and generate HLS with MpegTS, but I’m not sure this is the right direction.

  Is that the obvious way to do it ? Is there any existing service to do all this ?

  Thank you.