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• FFmpeg (merge two audio)

  10 septembre 2017, par Ameer Alkateeb

  I have two long audio files. I want to merge cut some part from each video
  and merge both parts in one file.

  In below command the problem there is no audio in the second audio part. It contain the first part and the second is empty.

  What is the problem ?

  ffmpeg -f lavfi -i color=c=black 
  
  -ss 157.824 -t 99.818 
  
  -i "file1.mp4" 
  
  -ss 315.764 -t 50.308 
  
  -i "file2.mp4" 
  
  -s 854x480 
  
  -aspect 1.779167
  
  -r 25 
  
  -c:v libx264 
  
  -b:v 800k 
  
  -c:a aac 
  
  -strict experimental 
  
  -b:a 128k 
  
  -f mp4 
  
  -t 150.126 -async 1 
  
  -y "output.mp4"

• Stream RTP to FFMPEG using SDP

  9 avril 2021, par Johnathan Kanarek

  I get RTP stream from WebRTC server (I used mediasoup) using node.js and I get the decrypted RTP packets raw data from the stream.
I want to forward this RTP data to ffmpeg and from there I can save it to file, or push it as RTMP stream to other media servers.
I guess that the best way would be to create SDP file that describes both the audio and video streams and send the packets through new sockets.

  



  The ffmpeg command is :

  



  ffmpeg -loglevel debug -protocol_whitelist file,crypto,udp,rtp -re -vcodec libvpx -acodec opus -i test.sdp -vcodec libx264 -acodec aac -y output.mp4

  



  I tried to send the packets through UDP :

  



  v=0
o=mediasoup 7199daf55e496b370e36cd1d25b1ef5b9dff6858 0 IN IP4 192.168.193.182
s=7199daf55e496b370e36cd1d25b1ef5b9dff6858
c=IN IP4 192.168.193.182
t=0 0
m=audio 33301 RTP/AVP 111
a=rtpmap:111 /opus/48000
a=fmtp:111 minptime=10;useinbandfec=1
a=rtcp-fb:111 transport-cc
a=sendrecv
m=video 33302 RTP/AVP 100
a=rtpmap:100 /VP8/90000
a=rtcp-fb:100 ccm fir
a=rtcp-fb:100 nack
a=rtcp-fb:100 nack pli
a=rtcp-fb:100 goog-remb
a=rtcp-fb:100 transport-cc
a=sendrecv


  



  But I always get (removed the boring parts) :

  



  Opening an input file: test.sdp.

[sdp @ 0x103dea0]
Format sdp probed with size=2048 and score=50
[sdp @ 0x103dea0] audio codec set to: (null)
[sdp @ 0x103dea0] audio samplerate set to: 44100
[sdp @ 0x103dea0] audio channels set to: 1
[sdp @ 0x103dea0] video codec set to: (null)
[udp @ 0x10402e0] end receive buffer size reported is 131072
[udp @ 0x10400c0] end receive buffer size reported is 131072
[sdp @ 0x103dea0] setting jitter buffer size to 500
[udp @ 0x1040740] bind failed: Address already in use
[AVIOContext @ 0x1046980] Statistics: 473 bytes read, 0 seeks
test.sdp: Invalid data found when processing input


  



  Note that I get it even if I don't open socket at all or send anything to this port, as if the ffmpeg itself tries to open these ports more than once.

  



  I tried also to open two (video and audio) TCP servers and define SDP with TCP :

  



  v=0
o=mediasoup 7199daf55e496b370e36cd1d25b1ef5b9dff6858 0 IN IP4 192.168.193.182
s=7199daf55e496b370e36cd1d25b1ef5b9dff6858
c=IN IP4 192.168.193.182
t=0 0
m=audio 33301 TCP 111
a=rtpmap:111 /opus/48000
a=fmtp:111 minptime=10;useinbandfec=1
a=rtcp-fb:111 transport-cc
a=setup:active
a=connection:new
a=sendrecv
m=video 33302 TCP 100
a=rtpmap:100 /VP8/90000
a=rtcp-fb:100 ccm fir
a=rtcp-fb:100 nack
a=rtcp-fb:100 nack pli
a=rtcp-fb:100 goog-remb
a=rtcp-fb:100 transport-cc
a=setup:active
a=connection:new
a=sendrecv


  



  However I don't see any incoming connection into my TCP servers and I get the following from ffmpeg :

  



  Opening an input file: test.sdp.

[sdp @ 0xdddea0]
Format sdp probed with size=2048 and score=50

[sdp @ 0xdddea0]
audio codec set to: (null)

[sdp @ 0xdddea0]
audio samplerate set to: 44100
[sdp @ 0xdddea0] audio channels set to: 1
[sdp @ 0xdddea0] video codec set to: (null)
[udp @ 0xde02e0] end receive buffer size reported is 131072
[udp @ 0xde00c0] end receive buffer size reported is 131072
[sdp @ 0xdddea0] setting jitter buffer size to 500
[udp @ 0xde0740] end receive buffer size reported is 131072

[udp @ 0xde0180] end receive buffer size reported is 131072
[sdp @ 0xdddea0] setting jitter buffer size to 500
[sdp @ 0xdddea0] Before avformat_find_stream_info() pos: 593 bytes read:593 seeks:0 nb_streams:2
[libvpx @ 0xdeea80] v1.3.0
[libvpx @ 0xdeea80] --target=x86_64-linux-gcc --enable-pic --disable-install-srcs --as=nasm --enable-shared --prefix=/usr --libdir=/usr/lib64

[sdp @ 0xdddea0] Could not find codec parameters for stream 1 (Video: vp8, 1 reference frame, none): unspecified size
Consider increasing the value for the 'analyzeduration' and 'probesize' options
[sdp @ 0xdddea0] After avformat_find_stream_info() pos: 593 bytes read:593 seeks:0 frames:0
Input #0, sdp, from 'test.sdp':
  Metadata:
    title           : 7199daf55e496b370e36cd1d25b1ef5b9dff6858
  Duration: N/A, bitrate: N/A
    Stream #0:0, 0, 1/90000: Audio: opus, 48000 Hz, mono, fltp
    Stream #0:1, 0, 1/90000: Video: vp8, 1 reference frame, none, 90k tbr, 90k tbn, 90k tbc
Successfully opened the file.
Parsing a group of options: output file output.mp4.
Successfully parsed a group of options.
Opening an output file: output.mp4.
[file @ 0xde3660] Setting default whitelist 'file,crypto'
Successfully opened the file.

detected 1 logical cores
[graph 0 input from stream 0:0 @ 0xde3940] Setting 'time_base' to value '1/48000'
[graph 0 input from stream 0:0 @ 0xde3940] Setting 'sample_rate' to value '48000'
[graph 0 input from stream 0:0 @ 0xde3940] Setting 'sample_fmt' to value 'fltp'
[graph 0 input from stream 0:0 @ 0xde3940] Setting 'channel_layout' to value '0x4'
[graph 0 input from stream 0:0 @ 0xde3940] tb:1/48000 samplefmt:fltp samplerate:48000 chlayout:0x4
[audio format for output stream 0:0 @ 0xe37900] Setting 'sample_fmts' to value 'fltp'
[audio format for output stream 0:0 @ 0xe37900] Setting 'sample_rates' to value '96000|88200|64000|48000|44100|32000|24000|22050|16000|12000|11025|8000|7350'
[AVFilterGraph @ 0xde0220] query_formats: 4 queried, 9 merged, 0 already done, 0 delayed

Output #0, mp4, to 'output.mp4':

  Metadata:

    title           :
7199daf55e496b370e36cd1d25b1ef5b9dff6858


    encoder         :
Lavf57.56.100


    Stream #0:0
, 0, 1/48000
: Audio: aac (LC) ([64][0][0][0] / 0x0040), 48000 Hz, mono, fltp, delay 1024, 69 kb/s


    Metadata:

      encoder         :
Lavc57.64.100 aac


Stream mapping:

  Stream #0:0 -> #0:0 (opus (native) -> aac (native))
Press [q] to stop, [?] for help
cur_dts is invalid (this is harmless if it occurs once at the start per stream)

test.sdp: Connection timed out
cur_dts is invalid (this is harmless if it occurs once at the start per stream)
cur_dts is invalid (this is harmless if it occurs once at the start per stream)
[output stream 0:0 @ 0xde3b40] EOF on sink link output stream 0:0:default.
No more output streams to write to, finishing.
[aac @ 0xde2b00] Trying to remove 1024 samples, but the queue is empty
[aac @ 0xde2b00] Trying to remove 1024 more samples than there are in the queue
[mp4 @ 0xe6a540] 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
[mp4 @ 0xe6a540] Encoder did not produce proper pts, making some up.
[aac @ 0xde2b00] Trying to remove 1024 samples, but the queue is empty
[aac @ 0xde2b00] Trying to remove 1024 more samples than there are in the queue
size=       1kB time=00:00:00.04 bitrate= 157.9kbits/s speed=0.00426x
video:0kB audio:0kB subtitle:0kB other streams:0kB global headers:0kB muxing overhead: 3268.000000%
Input file #0 (test.sdp):
  Input stream #0:0 (audio): 0 packets read (0 bytes); 0 frames decoded (0 samples);
  Input stream #0:1 (video): 0 packets read (0 bytes);
  Total: 0 packets (0 bytes) demuxed
Output file #0 (output.mp4):
  Output stream #0:0 (audio): 0 frames encoded (0 samples); 2 packets muxed (25 bytes);
  Total: 2 packets (25 bytes) muxed
0 frames successfully decoded, 0 decoding errors
[AVIOContext @ 0xde37a0] Statistics: 30 seeks, 25 writeouts
[aac @ 0xde2b00] Qavg: 47249.418

[AVIOContext @ 0xde6980] Statistics: 593 bytes read, 0 seeks


  



  Note to the "Connection timed out" in the log above.

  



  I guess that both my SDPs are wrong, any suggestions ?

  



  Alternatives to SDP are also most welcomed.

  


• FFmpeg mux video use libavformat avcodec but output couldn't be played

  10 août 2017, par tqn

  I’m trying write a app that take an input video and crop it to square video and ignore audio stream. Because bad performance if using command, I’m trying to use libavcodec and libavformat to do it. But the output isn’t playable by any video player and duration is 0 although I wrote all frame. Here are my code.

  void convert_video(char* input) {
  
      AVFormatContext *pFormatCtx = NULL;
  
      int             i, videoStreamIndex;
  
      AVCodecContext  *pCodecCtx = NULL;
  
      AVCodec         *pCodec = NULL;
  
      AVFrame         *pFrame = NULL;
  
      AVFrame         *pFrameSquare = NULL;
  
      AVPacket        packet, outPacket;
  
      int             frameFinished;
  
      int             numBytes;
  
      uint8_t         *buffer = NULL;
  
      AVCodec         *pEncodec = NULL;
  
      AVFormatContext *poFormatCxt = NULL;
  
      MuxOutputStream    videoStream = {0}, audioStream = {0};
  
      int tar_w, tar_h;
  
  
  
      const enum AVPixelFormat pic_format = AV_PIX_FMT_YUV420P;
  
      const enum AVCodecID codec_id = AV_CODEC_ID_H264;
  
      AVDictionary    *optionsDict = NULL;
  
      char output[50];
  
      sprintf(output, "%soutput.mp4", ANDROID_SDCARD);
  
  
  
      // Register all formats and codecs
  
      av_register_all();
  
  
  
      // Open video file
  
      if(avformat_open_input(&pFormatCtx, input, NULL, NULL)!=0)
  
          return; // Couldn't open file
  
      avformat_alloc_output_context2(&poFormatCxt, NULL, NULL, output);
  
  
  
      // Retrieve stream information
  
      if(avformat_find_stream_info(pFormatCtx, NULL)<0)
  
          return; // Couldn't find stream information
  
  
  
      // Find the first video stream
  
      videoStreamIndex=-1;
  
      for(i=0; inb_streams; i++)
  
          if(pFormatCtx->streams[i]->codec->codec_type==AVMEDIA_TYPE_VIDEO) {
  
              videoStreamIndex=i;
  
              break;
  
          }
  
      if(videoStreamIndex==-1)
  
          return; // Didn't find a video stream
  
  
  
      // Get a pointer to the codec context for the video stream
  
      pCodecCtx = pFormatCtx->streams[videoStreamIndex]->codec;
  
      tar_w = pCodecCtx->width > pCodecCtx->height ? pCodecCtx->height : pCodecCtx->width;
  
      tar_h = tar_w;
  
  
  
      // Find the decoder for the video stream
  
      pCodec=avcodec_find_decoder(pCodecCtx->codec_id);
  
      pEncodec = avcodec_find_encoder(codec_id);
  
  
  
      add_stream_mux(&videoStream, poFormatCxt, &pEncodec, codec_id, tar_w, tar_h);
  
      videoStream.st[0].time_base = pFormatCtx->streams[videoStreamIndex]->time_base;
  
      videoStream.st[0].codec->time_base = videoStream.st[0].time_base;
  
      videoStream.st[0].codec->time_base.den *= videoStream.st[0].codec->ticks_per_frame;
  
  //    add_stream(&audioStream, poFormatCxt, &)
  
      open_video(poFormatCxt, pEncodec, &videoStream, optionsDict);
  
      int ret = avio_open(&poFormatCxt->pb, output, AVIO_FLAG_WRITE);
  
  
  
      // Open codec
  
      if(avcodec_open2(pCodecCtx, pCodec, &optionsDict) < 0)
  
          return; // Could not open codec
  
  
  
      ret = avformat_write_header(poFormatCxt, &optionsDict);
  
      if (ret != 0) {
  
          ANDROID_LOG("Died");
  
      }
  
  
  
      // Allocate video frame
  
      pFrame=av_frame_alloc();
  
      pFrame->format = videoStream.st->codec->pix_fmt;
  
      pFrame->width = pCodecCtx->width;
  
      pFrame->height = pCodecCtx->height;
  
      av_frame_get_buffer(pFrame, 32);
  
  
  
      // Allocate an AVFrame structure
  
      pFrameSquare=av_frame_alloc();
  
      if(pFrameSquare==NULL)
  
          return;
  
  
  
      // Determine required buffer size and allocate buffer
  
      numBytes=avpicture_get_size(pic_format, tar_w,
  
                                  tar_h);
  
      buffer = (uint8_t *)av_malloc(numBytes*sizeof(uint8_t));
  
  
  
      // Assign appropriate parts of buffer to image planes in pFrameSquare
  
      // Note that pFrameSquare is an AVFrame, but AVFrame is a superset
  
      // of AVPicture
  
      ret = avpicture_fill((AVPicture *)pFrameSquare, buffer, pic_format,
  
                     tar_w, tar_h);
  
      if (ret < 0) {
  
          ANDROID_LOG("Can't fill picture");
  
          return;
  
      }
  
  
  
      // Read frames and save first five frames to disk
  
      i=0;
  
      ret = av_read_frame(pFormatCtx, &packet);
  
      while(ret >= 0) {
  
          // Is this a packet from the video stream?
  
          if(packet.stream_index == videoStreamIndex) {
  
              // Decode video frame
  
  //            av_packet_rescale_ts(&packet, videoStream.st->time_base, videoStream.st->codec->time_base);
  
              avcodec_decode_video2(pCodecCtx, pFrame, &frameFinished,
  
                                    &packet);
  
  //            while (!frameFinished) {
  
  //                avcodec_decode_video2(videoStream.st->codec, pFrame, &frameFinished, NULL);
  
  //            }
  
              ANDROID_LOG("Trying to decode frame %d with result %d", i, frameFinished);
  
              ret = av_picture_crop((AVPicture*) pFrameSquare, (AVPicture*) pFrame, pic_format, 0, 0);
  
              if (ret < 0) {
  
                  ANDROID_LOG("Can't crop image");
  
              }
  
  //            av_frame_get_best_effort_timestamp(pFrame);
  
  //            av_rescale_q()
  
  
  
              if(frameFinished) {
  
  
  
                  // Save the frame to disk
  
                  av_init_packet(&outPacket);
  
  //                av_packet_rescale_ts(&outPacket, videoStream.st->codec->time_base, videoStream.st->time_base);
  
                  pFrameSquare->width = tar_w;
  
                  pFrameSquare->height = tar_h;
  
                  pFrameSquare->format = pic_format;
  
                  pFrameSquare->pts = ++videoStream.next_pts;
  
                  ret = avcodec_encode_video2(videoStream.st->codec, &outPacket, pFrameSquare, &frameFinished);
  
  
  
  //                int count = 0;
  
  //                while (!frameFinished && count++ < 6) {
  
  //                    ret = avcodec_encode_video2(videoStream.st->codec, &outPacket, NULL, &frameFinished);
  
  //                }
  
                  if (frameFinished) {
  
                      ANDROID_LOG("Writing frame %d", i);
  
                      outPacket.stream_index = videoStreamIndex;
  
                      av_interleaved_write_frame(poFormatCxt, &outPacket);
  
                  }
  
                  av_free_packet(&outPacket);
  
              }
  
          }
  
  
  
          // Free the packet that was allocated by av_read_frameav_free_packet(&packet);
  
          ret = av_read_frame(pFormatCtx, &packet);
  
      }
  
  
  
      ret = av_write_trailer(poFormatCxt);
  
      if (ret < 0) {
  
          ANDROID_LOG("Couldn't write trailer");
  
      } else {
  
          ANDROID_LOG("Video convert finished");
  
      }
  
  
  
      // Free the RGB image
  
      av_free(buffer);
  
      av_free(pFrameSquare);
  
  
  
      // Free the YUV frame
  
      av_free(pFrame);
  
  
  
      // Close the codec
  
      avcodec_close(pCodecCtx);
  
  //    avcodec_close(pEncodecCtx);
  
  
  
      // Close the video file
  
      avformat_close_input(&pFormatCtx);
  
  
  
      return;
  
  }

  Helper

  #define STREAM_DURATION   10.0
  
  #define STREAM_FRAME_RATE 25 /* 25 images/s */
  
  #define STREAM_PIX_FMT    AV_PIX_FMT_YUV420P /* default pix_fmt */
  
  
  
  /* Add an output stream. */
  
  void add_stream_mux(MuxOutputStream *ost, AVFormatContext *oc,
  
                         AVCodec **codec,
  
                         enum AVCodecID codec_id, int width, int heigh)
  
  {
  
      AVCodecContext *codecCtx;
  
      int i;
  
      /* find the encoder */
  
      *codec = avcodec_find_encoder(codec_id);
  
      if (!(*codec)) {
  
          fprintf(stderr, "Could not find encoder for '%s'\n",
  
                  avcodec_get_name(codec_id));
  
          exit(1);
  
      }
  
      ost->st = avformat_new_stream(oc, *codec);
  
      if (!ost->st) {
  
          fprintf(stderr, "Could not allocate stream\n");
  
          exit(1);
  
      }
  
      ost->st->id = oc->nb_streams-1;
  
      codecCtx = ost->st->codec;
  
      switch ((*codec)->type) {
  
          case AVMEDIA_TYPE_AUDIO:
  
              codecCtx->sample_fmt  = (*codec)->sample_fmts ?
  
                               (*codec)->sample_fmts[0] : AV_SAMPLE_FMT_FLTP;
  
              codecCtx->bit_rate    = 64000;
  
              codecCtx->sample_rate = 44100;
  
              if ((*codec)->supported_samplerates) {
  
                  codecCtx->sample_rate = (*codec)->supported_samplerates[0];
  
                  for (i = 0; (*codec)->supported_samplerates[i]; i++) {
  
                      if ((*codec)->supported_samplerates[i] == 44100)
  
                          codecCtx->sample_rate = 44100;
  
                  }
  
              }
  
              codecCtx->channels        = av_get_channel_layout_nb_channels(codecCtx->channel_layout);
  
              codecCtx->channel_layout = AV_CH_LAYOUT_STEREO;
  
              if ((*codec)->channel_layouts) {
  
                  codecCtx->channel_layout = (*codec)->channel_layouts[0];
  
                  for (i = 0; (*codec)->channel_layouts[i]; i++) {
  
                      if ((*codec)->channel_layouts[i] == AV_CH_LAYOUT_STEREO)
  
                          codecCtx->channel_layout = AV_CH_LAYOUT_STEREO;
  
                  }
  
              }
  
              codecCtx->channels        = av_get_channel_layout_nb_channels(codecCtx->channel_layout);
  
              ost->st->time_base = (AVRational){ 1, codecCtx->sample_rate };
  
              break;
  
          case AVMEDIA_TYPE_VIDEO:
  
              codecCtx->codec_id = codec_id;
  
              codecCtx->bit_rate = 400000;
  
              /* Resolution must be a multiple of two. */
  
              codecCtx->width    = width;
  
              codecCtx->height   = heigh;
  
              /* timebase: This is the fundamental unit of time (in seconds) in terms
  
               * of which frame timestamps are represented. For fixed-fps content,
  
               * timebase should be 1/framerate and timestamp increments should be
  
               * identical to 1. */
  
              ost->st->time_base = (AVRational){ 1, STREAM_FRAME_RATE };
  
              codecCtx->time_base       = ost->st->time_base;
  
              codecCtx->gop_size      = 12; /* emit one intra frame every twelve frames at most */
  
              codecCtx->pix_fmt       = STREAM_PIX_FMT;
  
              if (codecCtx->codec_id == AV_CODEC_ID_MPEG2VIDEO) {
  
                  /* just for testing, we also add B frames */
  
                  codecCtx->max_b_frames = 2;
  
              }
  
              if (codecCtx->codec_id == AV_CODEC_ID_MPEG1VIDEO) {
  
                  /* Needed to avoid using macroblocks in which some coeffs overflow.
  
                   * This does not happen with normal video, it just happens here as
  
                   * the motion of the chroma plane does not match the luma plane. */
  
                  codecCtx->mb_decision = 2;
  
              }
  
              break;
  
          default:
  
              break;
  
      }
  
      /* Some formats want stream headers to be separate. */
  
      if (oc->oformat->flags & AVFMT_GLOBALHEADER)
  
          codecCtx->flags |= AV_CODEC_FLAG_GLOBAL_HEADER;
  
  }
  
  static void open_video(AVFormatContext *oc, AVCodec *codec, MuxOutputStream *ost, AVDictionary *opt_arg)
  
  {
  
      int ret;
  
      AVCodecContext *c = ost->st->codec;
  
      AVDictionary *opt = NULL;
  
      av_dict_copy(&opt, opt_arg, 0);
  
      /* open the codec */
  
      ret = avcodec_open2(c, codec, &opt);
  
      av_dict_free(&opt);
  
      if (ret < 0) {
  
          fprintf(stderr, "Could not open video codec: %s\n", av_err2str(ret));
  
          exit(1);
  
      }
  
      /* allocate and init a re-usable frame */
  
      ost->frame = alloc_picture(c->pix_fmt, c->width, c->height);
  
      if (!ost->frame) {
  
          fprintf(stderr, "Could not allocate video frame\n");
  
          exit(1);
  
      }
  
      /* If the output format is not YUV420P, then a temporary YUV420P
  
       * picture is needed too. It is then converted to the required
  
       * output format. */
  
      ost->tmp_frame = NULL;
  
      if (c->pix_fmt != AV_PIX_FMT_YUV420P) {
  
          ost->tmp_frame = alloc_picture(AV_PIX_FMT_YUV420P, c->width, c->height);
  
          if (!ost->tmp_frame) {
  
              fprintf(stderr, "Could not allocate temporary picture\n");
  
              exit(1);
  
          }
  
      }
  
  }

  I’m afraid that I set wrong pts or time_base of frame, and also when decoding or encoding, I see that some first frame is lost, frameFinished is 0. See a post that I’ve to flush decoder by avcodec_decode_video2(videoStream.st->codec, pFrame, &frameFinished, NULL) but after try a few times, frameFinished still is 0, and with avcodec_encode_video2(videoStream.st->codec, &outPacket, NULL, &frameFinished) will throw error in next encode frame. So how I can get all frame that lost ? I’m using FFmpeg version 3.0.1