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• Intel QuickSync error : Error while opening encoder for output stream #0:0 - maybe incorrect parameters such as bit_rate, rate, width or height

  30 novembre 2018, par Arie Syamsudin

  I want to make IPTV with a topology like this :

  Restream / source -----> IPTV server -----> User

  The problem at the moment is the question of bandwidth that is still large.

  Finally I announced by IPTV software providers to transcode with Intel QuickSync video and FFmpeg.

  I followed the steps given by the software provider at the following link :
  http://www.ezhometech.com/document/intel_ffmpeg.txt

  Which comes from the link below :
  https://www.intel.com/content/dam/www/public/us/en/documents/white-paper...

  I installed IPTV software on a laptop with an Intel processor, chipset and VGA.

  [root@iptv ffmpeg-4.1]# **lspci**
  
  00:00.0 Host bridge: Intel Corporation 2nd Generation Core Processor Family DRAM Controller (rev 09)
  
  00:02.0 VGA compatible controller: Intel Corporation 2nd Generation Core Processor Family Integrated Graphics Controller (rev 09)
  
  00:16.0 Communication controller: Intel Corporation 6 Series/C200 Series Chipset Family MEI Controller #1 (rev 04)
  
  00:1a.0 USB controller: Intel Corporation 6 Series/C200 Series Chipset Family USB Enhanced Host Controller #2 (rev 05)
  
  00:1b.0 Audio device: Intel Corporation 6 Series/C200 Series Chipset Family High Definition Audio Controller (rev 05)
  
  00:1c.0 PCI bridge: Intel Corporation 6 Series/C200 Series Chipset Family PCI Express Root Port 1 (rev b5)
  
  00:1c.1 PCI bridge: Intel Corporation 6 Series/C200 Series Chipset Family PCI Express Root Port 2 (rev b5)
  
  00:1d.0 USB controller: Intel Corporation 6 Series/C200 Series Chipset Family USB Enhanced Host Controller #1 (rev 05)
  
  00:1f.0 ISA bridge: Intel Corporation HM65 Express Chipset Family LPC Controller (rev 05)
  
  00:1f.2 SATA controller: Intel Corporation 6 Series/C200 Series Chipset Family 6 port Mobile SATA AHCI Controller (rev 05)
  
  00:1f.3 SMBus: Intel Corporation 6 Series/C200 Series Chipset Family SMBus Controller (rev 05)
  
  01:00.0 Ethernet controller: Qualcomm Atheros AR8152 v2.0 Fast Ethernet (rev c1)
  
  02:00.0 Network controller: Qualcomm Atheros AR9285 Wireless Network Adapter (PCI-Express) (rev 01)
  
  
  
  [root@iptv ffmpeg-4.1]# **lspci -nn -s 0:02.0**
  
  00:02.0 VGA compatible controller [0300]: Intel Corporation 2nd Generation Core Processor Family Integrated Graphics Controller [8086:0116] (rev 09)

  but when I do a transcoding test, an error always appears :
  Error initializing output stream 0:0 — Error while opening encoder for output stream #0:0 - maybe incorrect parameters such as bit_rate, rate, width or height.

  Error details :
  https://paste.fedoraproject.org/paste/vp9XIzbeOuyXRuLBlLfSng

  FFmpeg log :

  ./ffmpeg  -hwaccel qsv -i http://premium-play.org:8000/domm0H56Cj/q3ODLP6C16/2826 -vcodec h264_qsv -acodec copy -b:v 1200K -f mpegts udp://127.0.0.1:9011?pkt_size=1316
  
  ffmpeg version 3.2.12 Copyright (c) 2000-2018 the FFmpeg developers
  
    built with gcc 4.8.5 (GCC) 20150623 (Red Hat 4.8.5-28)
  
    configuration: --enable-libmfx --enable-nonfree
  
    libavutil      55. 34.101 / 55. 34.101
  
    libavcodec     57. 64.101 / 57. 64.101
  
    libavformat    57. 56.101 / 57. 56.101
  
    libavdevice    57.  1.100 / 57.  1.100
  
    libavfilter     6. 65.100 /  6. 65.100
  
    libswscale      4.  2.100 /  4.  2.100
  
    libswresample   2.  3.100 /  2.  3.100
  
  Input #0, mpegts, from 'http://premium-play.org:8000/domm0H56Cj/q3ODLP6C16/2826':
  
    Duration: N/A, start: 30200.274878, bitrate: N/A
  
    Program 1
  
      Metadata:
  
        service_name    : Service01
  
        service_provider: FFmpeg
  
      Stream #0:0[0x100]: Video: h264 (High) ([27][0][0][0] / 0x001B), yuv420p(progressive), 1920x1080 [SAR 1:1 DAR 16:9], 50 fps, 50 tbr, 90k tbn, 100 tbc
  
      Stream #0:1[0x101]: Audio: aac (LC) ([15][0][0][0] / 0x000F), 48000 Hz, stereo, fltp, 125 kb/s
  
  libva info: VA-API version 1.0.0
  
  libva info: va_getDriverName() returns 0
  
  libva info: User requested driver 'iHD'
  
  libva info: Trying to open /opt/intel/mediasdk/lib64/iHD_drv_video.so
  
  libva info: Found init function __vaDriverInit_1_0
  
  libva info: va_openDriver() returns 0
  
  [h264_qsv @ 0x2a89500] Selected ratecontrol mode is not supported by the QSV runtime. Choose a different mode.
  
  Stream mapping:
  
    Stream #0:0 -> #0:0 (h264 (native) -> h264 (h264_qsv))
  
    Stream #0:1 -> #0:1 (copy)
  
  Error while opening encoder for output stream #0:0 - maybe incorrect parameters such as bit_rate, rate, width or height

• FFMpeg : write h264 stream to mp4 container without changes

  3 mars 2017, par Bumblebee

  Good day.

  For brevity, the code omits error handling and memory management.

  I want to capture h264 video stream and pack it to mp4 container without changes. Since i don’t control the source of stream, i can not make assumptions about stream structure. In this way i must probe input stream.

      AVProbeData probeData;
  
      probeData.buf_size = s->BodySize();
  
      probeData.buf = s->GetBody();
  
      probeData.filename = "";
  
  
  
      AVInputFormat* inFormat = av_probe_input_format(&probeData, 1);  

  This code correctly defines h264 stream.

  Next, i create input format context,

      unsigned char* avio_input_buffer = reinterpret_cast<unsigned> (av_malloc(AVIO_BUFFER_SIZE));
  
  
  
      AVIOContext* avio_input_ctx = avio_alloc_context(avio_input_buffer, AVIO_BUFFER_SIZE,
  
          0, this, &amp;read_packet, NULL, NULL);
  
  
  
      AVFormatContext* ifmt_ctx = avformat_alloc_context();
  
      ifmt_ctx->pb = avio_input_ctx;
  
  
  
      int ret = avformat_open_input(&amp;ifmt_ctx, NULL, inFormat, NULL);
  
  </unsigned>

  set image size,

      ifmt_ctx->streams[0]->codec->width = ifmt_ctx->streams[0]->codec->coded_width = width;
  
      ifmt_ctx->streams[0]->codec->height = ifmt_ctx->streams[0]->codec->coded_height = height;

  create output format context,

      unsigned char* avio_output_buffer = reinterpret_cast<unsigned>(av_malloc(AVIO_BUFFER_SIZE));
  
  
  
      AVIOContext* avio_output_ctx = avio_alloc_context(avio_output_buffer, AVIO_BUFFER_SIZE,
  
          1, this, NULL, &amp;write_packet, NULL);
  
  
  
      AVFormatContext* ofmt_ctx = nullptr;
  
      avformat_alloc_output_context2(&amp;ofmt_ctx, NULL, "mp4", NULL);
  
      ofmt_ctx->pb = avio_output_ctx;
  
  
  
      AVDictionary* dict = nullptr;
  
      av_dict_set(&amp;dict, "movflags", "faststart", 0);
  
      av_dict_set(&amp;dict, "movflags", "frag_keyframe+empty_moov", 0);
  
  
  
      AVStream* outVideoStream = avformat_new_stream(ofmt_ctx, nullptr);
  
  
  
      avcodec_copy_context(outVideoStream->codec, ifmt_ctx->streams[0]->codec);
  
  
  
      ret = avformat_write_header(ofmt_ctx, &amp;dict);
  
  </unsigned>

  Initialization is done. Further there is a shifting packets from h264 stream to mp4 container. I dont calculate pts and dts, because source packet has AV_NOPTS_VALUE in them.

      AVPacket pkt;
  
      while (...)
  
      {
  
          ret = av_read_frame(ifmt_ctx, &pkt);
  
          ret = av_interleaved_write_frame(ofmt_ctx, &pkt);
  
          av_free_packet(&pkt);
  
      }

  Further i write trailer and free allocated memory. That is all. Code works and i got playable mp4 file.

  Now the problem : the stream characteristics of the resulting file is not completely consisent with the characteristics of the source stream. In particular, fps and bitrate is higher than it should be.

  As sample, below is output ffplay.exe for source stream

   Input #0, mov,mp4,m4a,3gp,3g2,mj2, from 'd:/movies/source.mp4':0/0
  
   Metadata:
  
       major_brand     : isom
  
       minor_version   : 1
  
       compatible_brands: isom
  
       creation_time   : 2014-04-14T13:03:54.000000Z
  
   Duration: 00:00:58.08, start: 0.000000, bitrate: 12130 kb/s
  
   Stream #0:0(und): Video: h264 (Constrained Baseline) (avc1 / 0x31637661),yuv420p, 1920x1080, 12129 kb/s, 25 fps, 25 tbr, 25 tbn, 50 tbc (default)
  
   Metadata:
  
       handler_name    : VideoHandler
  
   Switch subtitle stream from #-1 to #-1 vq= 1428KB sq=    0B f=0/0
  
   Seek to 49% ( 0:00:28) of total duration ( 0:00:58)       B f=0/0
  
       30.32 M-V: -0.030 fd=  87 aq=    0KB vq= 1360KB sq=    0B f=0/0  

  and for resulting stream (contains part of source stream)

   Input #0, mov,mp4,m4a,3gp,3g2,mj2, from 'd:/movies/target.mp4':f=0/0
  
   Metadata:
  
       major_brand     : isom
  
       minor_version   : 512
  
       compatible_brands: isomiso2avc1iso6mp41
  
       encoder         : Lavf57.56.101
  
   Duration: 00:00:11.64, start: 0.000000, bitrate: 18686 kb/s
  
   Stream #0:0(und): Video: h264 (Constrained Baseline) (avc1 / 0x31637661), yuv420p, 1920x1080, 18683 kb/s, 38.57 fps, 40 tbr, 90k tbn, 50 tbc (default)
  
   Metadata:
  
       handler_name    : VideoHandler
  
   Switch subtitle stream from #-1 to #-1 vq= 2309KB sq=    0B f=0/0
  
       5.70 M-V:  0.040 fd= 127 aq=    0KB vq= 2562KB sq=    0B f=0/0  

  So there is a question, what i missed when copying stream ? I will be grateful for any help.

  Best regards

• Wave bytes to buffer

  24 août 2016, par Mohammad Abu Musa

  I am encoding wav input from microphone which comes in four bytes format to ogg format. I think I have a problem shifting the bytes to the correct format here is the code I am using

  To explain more I get the audio frames from Google Chrome where I get data as const8 and channels, and samples. data field is always in 4 bytes format.

  I copy the data to a vector of type int16_t then I loop uninterleave samples which I think I am doing wrong. my question is how can I make sure the data is formatted correctly for ogg encoder to handle them correctly ?

  void EncoderInstance::OnGetBuffer(int32_t result, pp::AudioBuffer buffer) {
  
      if (result != PP_OK)
  
        return;
  
  
  
      assert(buffer.GetSampleSize() == PP_AUDIOBUFFER_SAMPLESIZE_16_BITS);
  
      const char* data = static_cast<const>(buffer.GetDataBuffer());
  
      uint32_t channels = buffer.GetNumberOfChannels();
  
      uint32_t samples = buffer.GetNumberOfSamples() / channels;
  
  
  
      if (channel_count_ != channels || sample_count_ != samples) {
  
        channel_count_ = channels;
  
        sample_count_ = samples;
  
  
  
        samples_.resize(sample_count_ * channel_count_);
  
        // Try (+ 5) to ensure that we pick up a new set of samples between each
  
        // timer-generated repaint.
  
        timer_interval_ = (sample_count_ * 1000) / buffer.GetSampleRate() + 5;
  
        // Start the timer for the first buffer.
  
        if (first_buffer_) {
  
          first_buffer_ = false;
  
          ScheduleNextTimer();
  
        }
  
      }
  
  
  
      if(is_audio_recording &amp;&amp; is_audio_header_written_)
  
      {
  
          memcpy(samples_.data(), data,
  
              sample_count_ * channel_count_ * sizeof(int16_t));
  
  
  
          float **buffer=vorbis_analysis_buffer(&amp;vd,samples);
  
  
  
          /* uninterleave samples */
  
          for(i=0;i4;i++)
  
          {
  
              buffer[0][i]=((samples_.at(i*4+1)&lt;&lt;8)|
  
                        (0x00ff&amp;(int16_t)samples_.at(i*4)))/32768.f;
  
              buffer[1][i]=((samples_.at(i*4+3)&lt;&lt;8)|
  
                        (0x00ff&amp;(int16_t)samples_.at(i*4+2)))/32768.f;
  
          }
  
  
  
          vorbis_analysis_wrote(&amp;vd,i);
  
  
  
          while(vorbis_analysis_blockout(&amp;vd,&amp;vb)==1){
  
  
  
            /* analysis, assume we want to use bitrate management */
  
            vorbis_analysis(&amp;vb,NULL);
  
            vorbis_bitrate_addblock(&amp;vb);
  
  
  
            while(vorbis_bitrate_flushpacket(&amp;vd,&amp;op)){
  
  
  
              /* weld the packet into the bitstream */
  
              ogg_stream_packetin(&amp;os,&amp;op);
  
  
  
              /* write out pages (if any) */
  
              while(!eos){
  
                int result=ogg_stream_pageout(&amp;os,&amp;og);
  
                if(result==0)break;
  
                glb_app_thread.message_loop().PostWork(callback_factory_.NewCallback(&amp;EncoderInstance::writeAudioHeader));
  
                if(ogg_page_eos(&amp;og))eos=1;
  
              }
  
            }
  
          }
  
  
  
  
  
      }
  
  
  
  
  
  
  
  
  
      audio_track_.RecycleBuffer(buffer);
  
      audio_track_.GetBuffer(callback_factory_.NewCallbackWithOutput(
  
          &amp;EncoderInstance::OnGetBuffer));
  
  
  
  }
  
  </const>