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• how to stream live m3u8 file using ffmpeg to youtube rtmp

  18 juillet 2018, par Dark webmaster

  i want to restream a live m3u8 file to youtube
  .i used following code

  fmpeg -re -i /mypanel.tv:8080/live/****/slyv0955k9/14131.m3u8
  
  > -c:v copy -c:a aac -ar 44100 -ab 128k -ac 2 -strict -2 -flags +global_header -bsf:a aac_adtstoasc -bufsize 3000k -f flv "/live-dfw.twitch.tv/app/{live_231566994_FS4BN0qoJMeXEuWklm6j0l1ODQj9u6}>"

  and i return i get this from my linux server

  [root@server ~]# ffmpeg -re -i http://mypanel.tv:8080/live/****/slyv0955k9/14131.m3u8
  
   -c:v copy -c:a aac -ar 44100 -ab 128k -ac 2 -strict -2 -flags +global_header -bsf:a aac_adtstoasc -bufsize 3000k -f flv "/live-dfw.twitch.tv/app/{live_23156556994_FS4BN0qoJMeXEuWklm6j0l1ODQj9u6}>"ffmpeg version 2.6.8 Copyright (c) 2000-2016 the FFmpeg developers
  
    built with gcc 4.8.5 (GCC) 20150623 (Red Hat 4.8.5-4)
  
    configuration: --prefix=/usr --bindir=/usr/bin --datadir=/usr/share/ffmpeg --incdir=/usr/include/ffmpeg --libdir=/usr/lib64 --mandir=/usr/share/man --arch=x86_64 --optflags='-O2 -g -pipe -Wall -Wp,-D_FORTIFY_SOURCE=2 -fexceptions -fstack-protector-strong --param=ssp-buffer-size=4 -grecord-gcc-switches -m64 -mtune=generic' --enable-bzlib --disable-crystalhd --enable-gnutls --enable-ladspa --enable-libass --enable-libcdio --enable-libdc1394 --enable-libfaac --enable-nonfree --enable-libfdk-aac --enable-nonfree --disable-indev=jack --enable-libfreetype --enable-libgsm --enable-libmp3lame --enable-openal --enable-libopenjpeg --enable-libopus --enable-libpulse --enable-libschroedinger --enable-libsoxr --enable-libspeex --enable-libtheora --enable-libvorbis --enable-libv4l2 --enable-libx264 --enable-libx265 --enable-libxvid --enable-x11grab --enable-avfilter --enable-avresample --enable-postproc --enable-pthreads --disable-static --enable-shared --enable-gpl --disable-debug --disable-stripping --shlibdir=/usr/lib64 --enable-runtime-cpudetect
  
    libavutil      54. 20.100 / 54. 20.100
  
    libavcodec     56. 26.100 / 56. 26.100
  
    libavformat    56. 25.101 / 56. 25.101
  
    libavdevice    56.  4.100 / 56.  4.100
  
    libavfilter     5. 11.102 /  5. 11.102
  
    libavresample   2.  1.  0 /  2.  1.  0
  
    libswscale      3.  1.101 /  3.  1.101
  
    libswresample   1.  1.100 /  1.  1.100
  
    libpostproc    53.  3.100 / 53.  3.100
  
  [h264 @ 0x1029ba0] non-existing SPS 0 referenced in buffering period
  
      Last message repeated 1 times
  
  [h264 @ 0x1073680] non-existing SPS 0 referenced in buffering period
  
  Input #0, hls,applehttp, from 'http://mypanel.tv:8080/live/***/slyv0955k9/14131.m3u8':
  
    Duration: N/A, start: 39062.400000, bitrate: N/A
  
    Program 0
  
      Metadata:
  
        variant_bitrate : 0
  
      Stream #0:0: Video: h264 (Main) ([27][0][0][0] / 0x001B), yuv420p, 1280x720 [SAR 1:1 DAR 16:9], 25 fps, 25 tbr, 90k tbn, 50 tbc
  
      Stream #0:1: Audio: aac (LC) ([15][0][0][0] / 0x000F), 48000 Hz, stereo, fltp, 133 kb/s
  
  At least one output file must be specified

  so can anyone help me with this ?please note i’m not an expert in linux so please give me specific commands to restream a live m3u8 file

• Unity : Converting Texture2D to YUV420P using FFmpeg

  23 juillet 2021, par strong_kobayashi

  I'm trying to create a game in Unity where each frame is rendered into a texture and then put together into a video using FFmpeg. The output created by FFmpeg should eventually be sent over the network to a client UI. However, I'm struggling mainly with the part where a frame is caught, and passed to an unsafe method as a byte array where it should be processed further by FFmpeg. The wrapper I'm using is FFmpeg.AutoGen.

  



  The render to texture method :

  



  private IEnumerator CaptureFrame()
{
    yield return new WaitForEndOfFrame();

    RenderTexture.active = rt;
    frame.ReadPixels(rect, 0, 0);
    frame.Apply();

    bytes = frame.GetRawTextureData();

    EncodeAndWrite(bytes, bytes.Length);
}


  



  The unsafe encoding method so far :

  



  private unsafe void EncodeAndWrite(byte[] bytes, int size)
{
    GCHandle pinned = GCHandle.Alloc(bytes, GCHandleType.Pinned);
    IntPtr address = pinned.AddrOfPinnedObject();

    sbyte** inData = (sbyte**)address;
    fixed(int* lineSize = new int[1])
    {
        lineSize[0] = 4 * textureWidth;
        // Convert RGBA to YUV420P
        ffmpeg.sws_scale(sws, inData, lineSize, 0, codecContext->width, inputFrame->extended_data, inputFrame->linesize);
    }

    inputFrame->pts = frameCounter++;

    if(ffmpeg.avcodec_send_frame(codecContext, inputFrame) < 0)
        throw new ApplicationException("Error sending a frame for encoding!");

    pkt = new AVPacket();
    fixed(AVPacket* packet = &pkt)
        ffmpeg.av_init_packet(packet);
    pkt.data = null;
    pkt.size = 0;

    pinned.Free();
    ...
}


  



  sws_scale takes a sbyte** as the second parameter, therefore I'm trying to convert the input byte array to sbyte** by first pinning it with GCHandle and doing an explicit type conversion afterwards. I don't know if that's the correct way, though.

  



  Moreover, the condition if(ffmpeg.avcodec_send_frame(codecContext, inputFrame) < 0) alwasy throws an ApplicationException, where I also really don't know why this happens. codecContext and inputFrame are my AVCodecContext and AVFrame objects, respectively, and the fields are defined as the following :

  



  codecContext

  



  codecContext = ffmpeg.avcodec_alloc_context3(codec);
codecContext->bit_rate = 400000;
codecContext->width = textureWidth;
codecContext->height = textureHeight;

AVRational timeBase = new AVRational();
timeBase.num = 1;
timeBase.den = (int)fps;
codecContext->time_base = timeBase;
videoAVStream->time_base = timeBase;

AVRational frameRate = new AVRational();
frameRate.num = (int)fps;
frameRate.den = 1;
codecContext->framerate = frameRate;

codecContext->gop_size = 10;
codecContext->max_b_frames = 1;
codecContext->pix_fmt = AVPixelFormat.AV_PIX_FMT_YUV420P;


  



  inputFrame

  



  inputFrame = ffmpeg.av_frame_alloc();
inputFrame->format = (int)codecContext->pix_fmt;
inputFrame->width = textureWidth;
inputFrame->height = textureHeight;
inputFrame->linesize[0] = inputFrame->width;


  



  Any help in fixing the issue would be greatly appreciated :)

  


• FFMPEG- Streaming Stops after few seconds

  25 mars 2019, par Manjot Singh Kalsi

  Hi there dear community

  This is the thing that I am dealing with, since last few days. After thorough search in the ffmpeg community, I was unable to find the solution. I am unable to Stream local flv, to facebook rtmp server.

  I am running the following command to stream my local flv video to the rtmp server of Facebook for Live-Streaming of my local File.

  ```ffmpeg -re -i SampleM.flv -acodec libmp3lame  -ar 44100 -b:a 128k -pix_fmt yuv420p -profile:v baseline -s 426x240 -bufsize 6000k -vb 400k -maxrate 1500k -deinterlace -vcodec libx264 -preset veryfast -g 30 -r 30 -f flv "rtmp://live-api.facebook.com:80/rtmp/my_key"```

  It has been a misfortune-situation that even after reading ffmpeg documentation, I have been unable to find the issue that is leading me to this situation as follows.

  I am still missing something that i need to know.

  Following is the log to the execution of the above command.

  ```ffmpeg version N-91024-g293a6e8332 Copyright (c) 2000-2018 the FFmpeg developers
  
    built with gcc 7.3.0 (GCC)
  
    configuration: --enable-gpl --enable-version3 --enable-sdl2 --enable-bzlib --enable-fontconfig --enable-gnutls --enable-iconv --enable-libass --enable-libbluray --enable-libfreetype --enable-libmp3lame --enable-libopencore-amrnb --enable-libopencore-amrwb --enable-libopenjpeg --enable-libopus --enable-libshine --enable-libsnappy --enable-libsoxr --enable-libtheora --enable-libtwolame --enable-libvpx --enable-libwavpack --enable-libwebp --enable-libx264 --enable-libx265 --enable-libxml2 --enable-libzimg --enable-lzma --enable-zlib --enable-gmp --enable-libvidstab --enable-libvorbis --enable-libvo-amrwbenc --enable-libmysofa --enable-libspeex --enable-libxvid --enable-libaom --enable-libmfx --enable-amf --enable-ffnvcodec --enable-cuvid --enable-d3d11va --enable-nvenc --enable-nvdec --enable-dxva2 --enable-avisynth
  
    libavutil      56. 18.100 / 56. 18.100
  
    libavcodec     58. 19.101 / 58. 19.101
  
    libavformat    58. 13.102 / 58. 13.102
  
    libavdevice    58.  4.100 / 58.  4.100
  
    libavfilter     7. 21.100 /  7. 21.100
  
    libswscale      5.  2.100 /  5.  2.100
  
    libswresample   3.  2.100 /  3.  2.100
  
    libpostproc    55.  2.100 / 55.  2.100
  
  Input #0, flv, from '.\video.flv':
  
    Metadata:
  
      audiodelay      : 0
  
      canSeekToEnd    : 1
  
      creationdate    : Fri Feb 03 11:52:46 2006
  
                      :
  
    Duration: 00:00:16.92, start: 0.000000, bitrate: 316 kb/s
  
      Stream #0:0: Audio: mp3, 22050 Hz, stereo, fltp, 40 kb/s
  
      Stream #0:1: Video: vp6f, 1 reference frame, yuv420p, 360x288 (368x288), 266 kb/s, 25 fps, 25 tbr, 1k tbn
  
  Stream mapping:
  
    Stream #0:1 -> #0:0 (vp6f (native) -> h264 (libx264))
  
    Stream #0:0 -> #0:1 (mp3 (mp3float) -> mp3 (libmp3lame))
  
  Press [q] to stop, [?] for help
  
  [graph_1_in_0_0 @ 0000020c0bcc4200] tb:1/22050 samplefmt:fltp samplerate:22050 chlayout:0x3
  
  [format_out_0_1 @ 0000020c0bca2cc0] auto-inserting filter 'auto_resampler_0' between the filter 'Parsed_anull_0' and the filter 'format_out_0_1'
  
  [auto_resampler_0 @ 0000020c0bca5140] ch:2 chl:stereo fmt:fltp r:22050Hz -> ch:2 chl:stereo fmt:fltp r:44100Hz
  
  [graph 0 input from stream 0:1 @ 0000020c0c4f4600] w:360 h:288 pixfmt:yuv420p tb:1/1000 fr:25/1 sar:0/1 sws_param:flags=2
  
  [scaler_out_0_0 @ 0000020c0c4f73c0] w:426 h:240 flags:'bicubic' interl:0
  
  [scaler_out_0_0 @ 0000020c0c4f73c0] w:360 h:288 fmt:yuv420p sar:0/1 -> w:426 h:240 fmt:yuv420p sar:0/1 flags:0x4
  
  [libx264 @ 0000020c0bc80600] using cpu capabilities: MMX2 SSE2Fast SSSE3 SSE4.2 AVX FMA3 BMI2 AVX2
  
  [libx264 @ 0000020c0bc80600] profile Constrained Baseline, level 3.0
  
  [libx264 @ 0000020c0bc80600] 264 - core 155 r2901 7d0ff22 - H.264/MPEG-4 AVC codec - Copyleft 2003-2018 - http://www.videolan.org/x264.html - options: cabac=0 ref=1 deblock=1:0:0 analyse=0x1:0x111 me=hex subme=2 psy=1 psy_rd=1.00:0.00 mixed_ref=0 me_range=16 chroma_me=1 trellis=0 8x8dct=0 cqm=0 deadzone=21,11 fast_pskip=1 chroma_qp_offset=0 threads=7 lookahead_threads=1 sliced_threads=0 nr=0 decimate=1 interlaced=0 bluray_compat=0 constrained_intra=0 bframes=0 weightp=0 keyint=30 keyint_min=3 scenecut=40 intra_refresh=0 rc_lookahead=10 rc=abr mbtree=1 bitrate=400 ratetol=1.0 qcomp=0.60 qpmin=0 qpmax=69 qpstep=4 vbv_maxrate=1500 vbv_bufsize=6000 nal_hrd=none filler=0 ip_ratio=1.40 aq=1:1.00
  
  Output #0, flv, to 'rtmp://live-api.facebook.com:80/rtmp/my key:
  
    Metadata:
  
      audiodelay      : 0
  
      canSeekToEnd    : 1
  
      creationdate    : Fri Feb 03 11:52:46 2006
  
                      :
  
      encoder         : Lavf58.13.102
  
      Stream #0:0: Video: h264 (libx264), 1 reference frame ([7][0][0][0] / 0x0007), yuv420p, 426x240, q=-1--1, 400 kb/s, 30 fps, 1k tbn, 30 tbc
  
      Metadata:
  
        encoder         : Lavc58.19.101 libx264
  
      Side data:
  
        cpb: bitrate max/min/avg: 1500000/0/400000 buffer size: 6000000 vbv_delay: -1
  
      Stream #0:1: Audio: mp3 (libmp3lame) ([2][0][0][0] / 0x0002), 44100 Hz, stereo, fltp, delay 1105, 128 kb/s
  
      Metadata:
  
        encoder         : Lavc58.19.101 libmp3lame
  
  No more output streams to write to, finishing.e=00:00:16.51 bitrate= 533.3kbits/s speed=0.992x
  
  [flv @ 0000020c0bc97fc0] Failed to update header with correct duration.
  
  [flv @ 0000020c0bc97fc0] Failed to update header with correct filesize.
  
  frame=  424 fps= 25 q=-1.0 Lsize=    1153kB time=00:00:16.95 bitrate= 557.1kbits/s speed=0.997x
  
  video:869kB audio:265kB subtitle:0kB other streams:0kB global headers:0kB muxing overhead: 1.666647%
  
  Input file #0 (.\video.flv):
  
    Input stream #0:0 (audio): 649 packets read (84767 bytes); 649 frames decoded (373824 samples);
  
    Input stream #0:1 (video): 424 packets read (566376 bytes); 424 frames decoded;
  
    Total: 1073 packets (651143 bytes) demuxed
  
  Output file #0 (rtmp://live-api.facebook.com:80/rtmp/my key):
  
    Output stream #0:0 (video): 424 frames encoded; 424 packets muxed (889641 bytes);
  
    Output stream #0:1 (audio): 649 frames encoded (747648 samples); 650 packets muxed (271673 bytes);
  
    Total: 1074 packets (1161314 bytes) muxed
  
  [libx264 @ 0000020c0bc80600] frame I:18    Avg QP:27.75  size:  7001
  
  [libx264 @ 0000020c0bc80600] frame P:406   Avg QP:32.67  size:  1879
  
  [libx264 @ 0000020c0bc80600] mb I  I16..4: 39.3%  0.0% 60.7%
  
  [libx264 @ 0000020c0bc80600] mb P  I16..4:  8.3%  0.0%  2.3%  P16..4: 42.2% 16.2%  4.4%  0.0%  0.0%    skip:26.6%
  
  [libx264 @ 0000020c0bc80600] final ratefactor: 28.61
  
  [libx264 @ 0000020c0bc80600] coded y,uvDC,uvAC intra: 40.2% 33.7% 9.2% inter: 23.8% 7.5% 0.2%
  
  [libx264 @ 0000020c0bc80600] i16 v,h,dc,p: 24% 50% 20%  6%
  
  [libx264 @ 0000020c0bc80600] i4 v,h,dc,ddl,ddr,vr,hd,vl,hu: 13% 34% 18%  5%  5%  4%  8%  4%  8%
  
  [libx264 @ 0000020c0bc80600] i8c dc,h,v,p: 67% 24%  7%  2%
  
  [libx264 @ 0000020c0bc80600] kb/s:418.33```

  This Image shows that The stream was alive few Few Seconds on Facebook

  ```[flv @ 0000020c0bc97fc0] Failed to update header with correct duration.
  
  [flv @ 0000020c0bc97fc0] Failed to update header with correct filesize.```

  Please correct me, the above error listed in the log, seems to be the main cause that the video stops streaming after few seconds. I’ve checked for the latency issues, But they won’t help anyway.

  Please Help me to tackle this issue. I’ll be very much thankful. :’)

  Streaming ends even earlier, when I use Google compute engine, instead my own PC as streaming service.