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• Troubleshooting ffmpeg/ffplay client RTSP RTP UDP * multicast * issue

  6 novembre 2020, par MAXdB

  I'm having problem with using udp_multicast transport method using ffmpeg or ffplay as a client to a webcam.

  


  TCP transport works :

  


  ffplay -rtsp_transport tcp rtsp://192.168.1.100/videoinput_1/mjpeg_3/media.stm


  


  UDP transport works :

  


  ffplay -rtsp_transport udp rtsp://192.168.1.100/videoinput_1/mjpeg_3/media.stm


  


  Multicast transport does not work :

  


  ffplay -rtsp_transport udp_multicast rtsp://192.168.1.100/videoinput_1/mjpeg_3/media.stm


  


  The error message when udp_multicast is chosen reads :

  


  [rtsp @ 0x7fd6a8000b80] Could not find codec parameters for stream 0 (Video: mjpeg, none(bt470bg/unknown/unknown)): unspecified size


  


  Run with -v debug : Observe that the UDP multicast information appears in the SDP even though the chosen transport is unicast for this run. The SDP content is unchanged for unicast or multicast.

  


  [tcp @ 0x7f648c002f40] Starting connection attempt to 192.168.1.100 port 554
[tcp @ 0x7f648c002f40] Successfully connected to 192.168.1.100 port 554
[rtsp @ 0x7f648c000b80] SDP:
v=0
o=- 621355968671884050 621355968671884050 IN IP4 192.168.1.100
s=/videoinput_1:0/mjpeg_3/media.stm
c=IN IP4 0.0.0.0
m=video 40004 RTP/AVP 26
c=IN IP4 237.0.0.3/1
a=control:trackID=1
a=range:npt=0-
a=framerate:25.0

Failed to parse interval end specification ''
[rtp @ 0x7f648c008e00] No default whitelist set
[udp @ 0x7f648c009900] No default whitelist set
[udp @ 0x7f648c009900] end receive buffer size reported is 425984
[udp @ 0x7f648c019c80] No default whitelist set
[udp @ 0x7f648c019c80] end receive buffer size reported is 425984
[rtsp @ 0x7f648c000b80] setting jitter buffer size to 500
[rtsp @ 0x7f648c000b80] hello state=0
Failed to parse interval end specification ''
[mjpeg @ 0x7f648c0046c0] marker=d8 avail_size_in_buf=145103 
[mjpeg @ 0x7f648c0046c0] marker parser used 0 bytes (0 bits)
[mjpeg @ 0x7f648c0046c0] marker=e0 avail_size_in_buf=145101
[mjpeg @ 0x7f648c0046c0] marker parser used 16 bytes (128 bits)
[mjpeg @ 0x7f648c0046c0] marker=db avail_size_in_buf=145083
[mjpeg @ 0x7f648c0046c0] index=0
[mjpeg @ 0x7f648c0046c0] qscale[0]: 5
[mjpeg @ 0x7f648c0046c0] index=1
[mjpeg @ 0x7f648c0046c0] qscale[1]: 10
[mjpeg @ 0x7f648c0046c0] marker parser used 132 bytes (1056 bits)
[mjpeg @ 0x7f648c0046c0] marker=c4 avail_size_in_buf=144949
[mjpeg @ 0x7f648c0046c0] marker parser used 0 bytes (0 bits)
[mjpeg @ 0x7f648c0046c0] marker=c0 avail_size_in_buf=144529
[mjpeg @ 0x7f648c0046c0] Changing bps from 0 to 8
[mjpeg @ 0x7f648c0046c0] sof0: picture: 1920x1080
[mjpeg @ 0x7f648c0046c0] component 0 2:2 id: 0 quant:0
[mjpeg @ 0x7f648c0046c0] component 1 1:1 id: 1 quant:1
[mjpeg @ 0x7f648c0046c0] component 2 1:1 id: 2 quant:1
[mjpeg @ 0x7f648c0046c0] pix fmt id 22111100
[mjpeg @ 0x7f648c0046c0] Format yuvj420p chosen by get_format().
[mjpeg @ 0x7f648c0046c0] marker parser used 17 bytes (136 bits)
[mjpeg @ 0x7f648c0046c0] escaping removed 676 bytes
[mjpeg @ 0x7f648c0046c0] marker=da avail_size_in_buf=144510
[mjpeg @ 0x7f648c0046c0] marker parser used 143834 bytes (1150672 bits)
[mjpeg @ 0x7f648c0046c0] marker=d9 avail_size_in_buf=2
[mjpeg @ 0x7f648c0046c0] decode frame unused 2 bytes
[rtsp @ 0x7f648c000b80] All info found vq=    0KB sq=    0B f=0/0
[rtsp @ 0x7f648c000b80] rfps: 24.416667 0.018101
    Last message repeated 1 times
[rtsp @ 0x7f648c000b80] rfps: 24.500000 0.013298
    Last message repeated 1 times
[rtsp @ 0x7f648c000b80] rfps: 24.583333 0.009235
    Last message repeated 1 times
[rtsp @ 0x7f648c000b80] rfps: 24.666667 0.005910
    Last message repeated 1 times
[rtsp @ 0x7f648c000b80] rfps: 24.750000 0.003324
    Last message repeated 1 times
[rtsp @ 0x7f648c000b80] rfps: 24.833333 0.001477
    Last message repeated 1 times
[rtsp @ 0x7f648c000b80] rfps: 24.916667 0.000369
    Last message repeated 1 times
[rtsp @ 0x7f648c000b80] rfps: 25.000000 0.000000
[rtsp @ 0x7f648c000b80] rfps: 25.083333 0.000370
    Last message repeated 1 times
[rtsp @ 0x7f648c000b80] rfps: 25.166667 0.001478
    Last message repeated 1 times
[rtsp @ 0x7f648c000b80] rfps: 25.250000 0.003326
    Last message repeated 1 times
[rtsp @ 0x7f648c000b80] rfps: 25.333333 0.005912
    Last message repeated 1 times
[rtsp @ 0x7f648c000b80] rfps: 25.416667 0.009238
    Last message repeated 1 times
[rtsp @ 0x7f648c000b80] rfps: 25.500000 0.013302
    Last message repeated 1 times
[rtsp @ 0x7f648c000b80] rfps: 25.583333 0.018105
    Last message repeated 1 times
[rtsp @ 0x7f648c000b80] rfps: 50.000000 0.000000
[rtsp @ 0x7f648c000b80] Setting avg frame rate based on r frame rate
Input #0, rtsp, from 'rtsp://192.168.1.100/videoinput_1/mjpeg_3/media.stm':
  Metadata:
    title           : /videoinput_1:0/mjpeg_3/media.stm
  Duration: N/A, start: 0.000000, bitrate: N/A
    Stream #0:0, 21, 1/90000: Video: mjpeg (Baseline), 1 reference frame, yuvj420p(pc, bt470bg/unknown/unknown, center), 1920x1080 [SAR 1:1 DAR 16:9], 0/1, 25 fps, 25 tbr, 90k tbn, 90k tbc
[mjpeg @ 0x7f648c02ad80] marker=d8 avail_size_in_buf=145103


  


  Here is the same debug section when using udp_multicast. The SDP is identical as mentioned, and the block after the SDP containing [mjpeg] codec info is entirely missing (beginning with marker=d8)—the stream is never identified. This happens (to the eye) instantaneously, there's no indication of a timeout waiting unsuccessfully for an RTP packet, though this, too, could just be insufficient debug info in the driver. Also note that ffmpeg knows that the frames are MJPEG frames and the color primaries are PAL, it just doesn't know the size. Also curious, but not relevant to the problem, the unicast UDP transport destination port utilized for the stream does not appear in the ffmpeg debug dump shown above, meaning part of the RTSP/RTP driver is hiding important information under the kimono, that port number and how it knows that the frames will be MJPEG.

  


  [tcp @ 0x7effe0002f40] Starting connection attempt to 192.168.1.100 port 554
[tcp @ 0x7effe0002f40] Successfully connected to 192.168.1.100 port 554
[rtsp @ 0x7effe0000b80] SDP:aq=    0KB vq=    0KB sq=    0B f=0/0
v=0
o=- 621355968671884050 621355968671884050 IN IP4 192.168.1.100
s=/videoinput_1:0/mjpeg_3/media.stm
c=IN IP4 0.0.0.0
m=video 40004 RTP/AVP 26
c=IN IP4 237.0.0.3/1
a=control:trackID=1
a=range:npt=0-
a=framerate:25.0

Failed to parse interval end specification ''
[rtp @ 0x7effe0008e00] No default whitelist set
[udp @ 0x7effe0009900] No default whitelist set
[udp @ 0x7effe0009900] end receive buffer size reported is 425984
[udp @ 0x7effe0019c40] No default whitelist set
[udp @ 0x7effe0019c40] end receive buffer size reported is 425984
[rtsp @ 0x7effe0000b80] setting jitter buffer size to 500
[rtsp @ 0x7effe0000b80] hello state=0
Failed to parse interval end specification '' 
[rtsp @ 0x7effe0000b80] Could not find codec parameters for stream 0 (Video: mjpeg, 1 reference frame, none(bt470bg/unknown/unknown, center)): unspecified size
Consider increasing the value for the 'analyzeduration' (0) and 'probesize' (5000000) options
Input #0, rtsp, from 'rtsp://192.168.1.100/videoinput_1/mjpeg_3/media.stm':
  Metadata:
    title           : /videoinput_1:0/mjpeg_3/media.stm
  Duration: N/A, start: 0.000000, bitrate: N/A
    Stream #0:0, 0, 1/90000: Video: mjpeg, 1 reference frame, none(bt470bg/unknown/unknown, center), 90k tbr, 90k tbn, 90k tbc
    nan M-V:    nan fd=   0 aq=    0KB vq=    0KB sq=    0B f=0/0


  


  This is the TCPDUMP of the traffic. The information in both streams appears identical.

  


  19:21:30.703599 IP 192.168.1.100.64271 > 192.168.1.98.5239: UDP, length 60
19:21:30.703734 IP 192.168.1.100.64270 > 192.168.1.98.5238: UDP, length 1400
19:21:30.703852 IP 192.168.1.100.64270 > 192.168.1.98.5238: UDP, length 1400
19:21:30.704326 IP 192.168.1.100.64270 > 192.168.1.98.5238: UDP, length 1400
19:21:30.704326 IP 192.168.1.100.64270 > 192.168.1.98.5238: UDP, length 1400
19:21:30.704327 IP 192.168.1.100.64270 > 192.168.1.98.5238: UDP, length 1400
19:21:30.704327 IP 192.168.1.100.64270 > 192.168.1.98.5238: UDP, length 1400
19:21:30.704504 IP 192.168.1.100.64270 > 192.168.1.98.5238: UDP, length 1400
19:21:30.704813 IP 192.168.1.100.64270 > 192.168.1.98.5238: UDP, length 1400
19:21:30.704814 IP 192.168.1.100.64270 > 192.168.1.98.5238: UDP, length 1400
19:21:30.704872 IP 192.168.1.100.64270 > 192.168.1.98.5238: UDP, length 732
19:21:30.704873 IP 192.168.1.100.59869 > 237.0.0.3.40005: UDP, length 60
19:21:30.705513 IP 192.168.1.100.59868 > 237.0.0.3.40004: UDP, length 1400
19:21:30.705513 IP 192.168.1.100.59868 > 237.0.0.3.40004: UDP, length 1400
19:21:30.705513 IP 192.168.1.100.59868 > 237.0.0.3.40004: UDP, length 1400
19:21:30.705513 IP 192.168.1.100.59868 > 237.0.0.3.40004: UDP, length 1400
19:21:30.705594 IP 192.168.1.100.59868 > 237.0.0.3.40004: UDP, length 1400
19:21:30.705774 IP 192.168.1.100.59868 > 237.0.0.3.40004: UDP, length 1400
19:21:30.706236 IP 192.168.1.100.59868 > 237.0.0.3.40004: UDP, length 1400
19:21:30.706236 IP 192.168.1.100.59868 > 237.0.0.3.40004: UDP, length 1400
19:21:30.706236 IP 192.168.1.100.59868 > 237.0.0.3.40004: UDP, length 1400
19:21:30.706236 IP 192.168.1.100.59868 > 237.0.0.3.40004: UDP, length 732


  


  I hope this is a configuration problem, that I can fix this in my ffplay/ffmpeg line, and it's not a bug in ffmpeg. Thanks for any tips.

  


• Files created with "ffmpeg hevc_nvenc" do not play on TV. (with video codec SDK 9.1 of nvidia)

  29 janvier 2020, par Dashhh

  Problem

  • Files created with hevc_nvenc do not play on TV. (samsung smart tv, model unknown)
    Related to my ffmpeg build is below.

  FFmpeg build conf

  $ ffmpeg -buildconf
  
      --enable-cuda
  
      --enable-cuvid
  
      --enable-nvenc
  
      --enable-nonfree
  
      --enable-libnpp
  
      --extra-cflags=-I/path/cuda/include
  
      --extra-ldflags=-L/path/cuda/lib64
  
      --prefix=/prefix/ffmpeg_build
  
      --pkg-config-flags=--static
  
      --extra-libs='-lpthread -lm'
  
      --extra-cflags=-I/prefix/ffmpeg_build/include
  
      --extra-ldflags=-L/prefix/ffmpeg_build/lib
  
      --enable-gpl
  
      --enable-nonfree
  
      --enable-version3
  
      --disable-stripping
  
      --enable-avisynth
  
      --enable-libass
  
      --enable-libfontconfig
  
      --enable-libfreetype
  
      --enable-libfribidi
  
      --enable-libgme
  
      --enable-libgsm
  
      --enable-librubberband
  
      --enable-libshine
  
      --enable-libsnappy
  
      --enable-libssh
  
      --enable-libtwolame
  
      --enable-libwavpack
  
      --enable-libzvbi
  
      --enable-openal
  
      --enable-sdl2
  
      --enable-libdrm
  
      --enable-frei0r
  
      --enable-ladspa
  
      --enable-libpulse
  
      --enable-libsoxr
  
      --enable-libspeex
  
      --enable-avfilter
  
      --enable-postproc
  
      --enable-pthreads
  
      --enable-libfdk-aac
  
      --enable-libmp3lame
  
      --enable-libopus
  
      --enable-libtheora
  
      --enable-libvorbis
  
      --enable-libvpx
  
      --enable-libx264
  
      --enable-libx265
  
      --disable-ffplay
  
      --enable-libopenjpeg
  
      --enable-libwebp
  
      --enable-libxvid
  
      --enable-libvidstab
  
      --enable-libopenh264
  
      --enable-zlib
  
      --enable-openssl

  ---

  ffmpeg Command

  • Command about FFmpeg encoding

  ffmpeg -ss 1800 -vsync 0 -hwaccel cuvid -hwaccel_device 0 \
  
  -c:v h264_cuvid -i /data/input.mp4 -t 10 \
  
  -filter_complex "\
  
  [0:v]hwdownload,format=nv12,format=yuv420p,\
  
  scale=iw*2:ih*2" -gpu 0 -c:v hevc_nvenc -pix_fmt yuv444p16le -preset slow -rc cbr_hq -b:v 5000k -maxrate 7000k -bufsize 1000k -acodec aac -ac 2 -dts_delta_threshold 1000 -ab 128k -flags global_header ./makevideo_nvenc_hevc.mp4

  Full log about This Command - check this full log

  The reason for adding "-color_ " in the command is as follows.

  • HDR video after creating bt2020 + smpte2084 video using nvidia hardware accelerator. (I’m studying to make HDR videos. I’m not sure if this is right.)

  How can I make a video using ffmpeg hevc_nvenc and have it play on TV ?

  ---

  Things i’ve done

  Here’s what I’ve researched about why it doesn’t work.
   The header information is not properly included in the resulting video file. So I used a program called nvhsp to add SEI and VUI information inside the video. See below for the commands and logs used.

  nvhsp is open source for writing VUI and SEI bitstrings in raw video. nvhsp link

  # make rawvideo for nvhsp
  
  $  ffmpeg -vsync 0 -hwaccel cuvid -hwaccel_device 0 -c:v h264_cuvid \
  
  -i /data/input.mp4 -t 10 \
  
  -filter_complex "[0:v]hwdownload,format=nv12,\
  
  format=yuv420p,scale=iw*2:ih*2" \
  
  -gpu 0 -c:v hevc_nvenc -f rawvideo output_for_nvhsp.265
  
  
  
  # use nvhsp
  
  $ python nvhsp.py ./output_for_nvhsp.265 -colorprim bt2020 \
  
  -transfer smpte-st-2084 -colormatrix bt2020nc \
  
  -maxcll "1000,300" -videoformat ntsc -full_range tv \
  
  -masterdisplay "G (13250,34500) B (7500,3000 ) R (34000,16000) WP (15635,16450) L (10000000,1)" \
  
  ./after_nvhsp_proc_output.265
  
  
  
  Parsing the infile:
  
  
  
  ==========================
  
  
  
  Prepending SEI data
  
  Starting new SEI NALu ...
  
  SEI message with MaxCLL = 1000 and MaxFall = 300 created in SEI NAL
  
  SEI message Mastering Display Data G (13250,34500) B (7500,3000) R (34000,16000) WP (15635,16450) L (10000000,1) created in SEI NAL
  
  Looking for SPS ......... [232, 22703552]
  
  SPS_Nals_addresses [232, 22703552]
  
  SPS NAL Size 488
  
  Starting reading SPS NAL contents
  
  Reading of SPS NAL finished. Read 448 of SPS NALu data.
  
  
  
  Making modified SPS NALu ...
  
  Made modified SPS NALu-OK
  
  New SEI prepended
  
  Writing new stream ...
  
  Progress: 100%
  
  =====================
  
  Done!
  
  
  
  File nvhsp_after_output.mp4 created.
  
  
  
  # after process
  
  $ ffmpeg -y -f rawvideo -r 25 -s 3840x2160 -pix_fmt yuv444p16le -color_primaries bt2020 -color_trc smpte2084  -colorspace bt2020nc -color_range tv -i ./1/after_nvhsp_proc_output.265 -vcodec copy  ./1/result.mp4 -hide_banner
  
  
  
  Truncating packet of size 49766400 to 3260044
  
  [rawvideo @ 0x40a6400] Estimating duration from bitrate, this may be inaccurate
  
  Input #0, rawvideo, from './1/nvhsp_after_output.265':
  
    Duration: N/A, start: 0.000000, bitrate: 9953280 kb/s
  
      Stream #0:0: Video: rawvideo (Y3[0][16] / 0x10003359), yuv444p16le(tv, bt2020nc/bt2020/smpte2084), 3840x2160, 9953280 kb/s, 25 tbr, 25 tbn, 25 tbc
  
  [mp4 @ 0x40b0440] Could not find tag for codec rawvideo in stream #0, codec not currently supported in container
  
  Could not write header for output file #0 (incorrect codec parameters ?): Invalid argument
  
  Stream mapping:
  
    Stream #0:0 -> #0:0 (copy)
  
      Last message repeated 1 times

  Goal

  • I want to generate matadata normally when encoding a video through hevc_nvenc.

  • I want to create a video through hevc_nvenc and play HDR Video on smart tv with 10bit color depth support.

  ---

  Additional

  • Is it normal for ffmpeg hevc_nvenc not to generate metadata in the resulting video file ? or is it a bug ?

  • Please refer to the image below. (*’알 수 없음’ meaning ’unknown’)

    • if you need more detail file info, check this Gist Link (by ffprobe)
      

  • However, if you encode a file in libx265, the attribute information is entered correctly as shown below.

    • if you need more detail file info, check this Gist Link
      

  However, when using hevc_nvenc, all information is missing.

  • i used option -show_streams -show_programs -show_format -show_data -of json -show_frames -show_log 56 at ffprobe

• ffmpeg pipe process ends right after writing first buffer data to input stream and does not keep running

  6 mai, par Taketo Matsunaga

  I have been trying to convert 16bit PCM (s16le) audio data to webm using ffmpeg in C#.
But the process ends right after the writing the first buffer data to standard input.
I has exited with the status 0, meaning success. But do not know why....
Could anyone tell me why ?

  


  I apprecite it if you could support me.

  


      public class SpeechService : ISpeechService&#xA;    {&#xA;        &#xA;        /// <summary>&#xA;        /// Defines the _audioInputStream&#xA;        /// </summary>&#xA;        private readonly MemoryStream _audioInputStream = new MemoryStream();&#xA;&#xA;        public async Task SendPcmAsWebmViaWebSocketAsync(&#xA;            MemoryStream pcmAudioStream,&#xA;            int sampleRate,&#xA;            int channels) &#xA;        {&#xA;            string inputFormat = "s16le";&#xA;&#xA;            var ffmpegProcessInfo = new ProcessStartInfo&#xA;            {&#xA;                FileName = _ffmpegPath,&#xA;                Arguments =&#xA;                    $"-f {inputFormat} -ar {sampleRate} -ac {channels} -i pipe:0 " &#x2B;&#xA;                    $"-f webm pipe:1",&#xA;                RedirectStandardInput = true,&#xA;                RedirectStandardOutput = true,&#xA;                RedirectStandardError = true,&#xA;                UseShellExecute = false,&#xA;                CreateNoWindow = true,&#xA;            };&#xA;&#xA;            _ffmpegProcess = new Process { StartInfo = ffmpegProcessInfo };&#xA;&#xA;            Console.WriteLine("Starting FFmpeg process...");&#xA;            try&#xA;            {&#xA;&#xA;                if (!await Task.Run(() => _ffmpegProcess.Start()))&#xA;                {&#xA;                    Console.Error.WriteLine("Failed to start FFmpeg process.");&#xA;                    return;&#xA;                }&#xA;                Console.WriteLine("FFmpeg process started.");&#xA;&#xA;            }&#xA;            catch (Exception ex)&#xA;            {&#xA;                Console.Error.WriteLine($"Error starting FFmpeg process: {ex.Message}");&#xA;                throw;&#xA;            }&#xA;&#xA;            var encodeAndSendTask = Task.Run(async () =>&#xA;            {&#xA;                try&#xA;                {&#xA;                    using var ffmpegOutputStream = _ffmpegProcess.StandardOutput.BaseStream;&#xA;                    byte[] buffer = new byte[8192]; // Temporary buffer to read data&#xA;                    byte[] sendBuffer = new byte[8192]; // Buffer to accumulate data for sending&#xA;                    int sendBufferIndex = 0; // Tracks the current size of sendBuffer&#xA;                    int bytesRead;&#xA;&#xA;                    Console.WriteLine("Reading WebM output from FFmpeg and sending via WebSocket...");&#xA;                    while (true)&#xA;                    {&#xA;                        if ((bytesRead = await ffmpegOutputStream.ReadAsync(buffer, 0, buffer.Length)) > 0)&#xA;                        {&#xA;                            // Copy data to sendBuffer&#xA;                            Array.Copy(buffer, 0, sendBuffer, sendBufferIndex, bytesRead);&#xA;                            sendBufferIndex &#x2B;= bytesRead;&#xA;&#xA;                            // If sendBuffer is full, send it via WebSocket&#xA;                            if (sendBufferIndex >= sendBuffer.Length)&#xA;                            {&#xA;                                var segment = new ArraySegment<byte>(sendBuffer, 0, sendBuffer.Length);&#xA;                                _ws.SendMessage(segment);&#xA;                                sendBufferIndex = 0; // Reset the index after sending&#xA;                            }&#xA;                        }&#xA;                    }&#xA;                }&#xA;                catch (OperationCanceledException)&#xA;                {&#xA;                    Console.WriteLine("Encode/Send operation cancelled.");&#xA;                }&#xA;                catch (IOException ex) when (ex.InnerException is ObjectDisposedException)&#xA;                {&#xA;                    Console.WriteLine("Stream was closed, likely due to process exit or cancellation.");&#xA;                }&#xA;                catch (Exception ex)&#xA;                {&#xA;                    Console.Error.WriteLine($"Error during encoding/sending: {ex}");&#xA;                }&#xA;            });&#xA;&#xA;            var errorReadTask = Task.Run(async () =>&#xA;            {&#xA;                Console.WriteLine("Starting to read FFmpeg stderr...");&#xA;                using var errorReader = _ffmpegProcess.StandardError;&#xA;                try&#xA;                {&#xA;                    string? line;&#xA;                    while ((line = await errorReader.ReadLineAsync()) != null) &#xA;                    {&#xA;                        Console.WriteLine($"[FFmpeg stderr] {line}");&#xA;                    }&#xA;                }&#xA;                catch (OperationCanceledException) { Console.WriteLine("FFmpeg stderr reading cancelled."); }&#xA;                catch (TimeoutException) { Console.WriteLine("FFmpeg stderr reading timed out (due to cancellation)."); }&#xA;                catch (Exception ex) { Console.Error.WriteLine($"Error reading FFmpeg stderr: {ex.Message}"); }&#xA;                Console.WriteLine("Finished reading FFmpeg stderr.");&#xA;            });&#xA;&#xA;        }&#xA;&#xA;        public async Task AppendAudioBuffer(AudioMediaBuffer audioBuffer)&#xA;        {&#xA;            try&#xA;            {&#xA;                // audio for a 1:1 call&#xA;                var bufferLength = audioBuffer.Length;&#xA;                if (bufferLength > 0)&#xA;                {&#xA;                    var buffer = new byte[bufferLength];&#xA;                    Marshal.Copy(audioBuffer.Data, buffer, 0, (int)bufferLength);&#xA;&#xA;                    _logger.Info("_ffmpegProcess.HasExited:" &#x2B; _ffmpegProcess.HasExited);&#xA;                    using var ffmpegInputStream = _ffmpegProcess.StandardInput.BaseStream;&#xA;                    await ffmpegInputStream.WriteAsync(buffer, 0, buffer.Length);&#xA;                    await ffmpegInputStream.FlushAsync(); // バッファをフラッシュ&#xA;                    _logger.Info("Wrote buffer data.");&#xA;&#xA;                }&#xA;            }&#xA;            catch (Exception e)&#xA;            {&#xA;                _logger.Error(e, "Exception happend writing to input stream");&#xA;            }&#xA;        }&#xA;&#xA;</byte>

  


  Starting FFmpeg process...
FFmpeg process started.
Starting to read FFmpeg stderr...
Reading WebM output from FFmpeg and sending via WebSocket...
[FFmpeg stderr] ffmpeg version 7.1.1-essentials_build-www.gyan.dev Copyright (c) 2000-2025 the FFmpeg developers
[FFmpeg stderr]   built with gcc 14.2.0 (Rev1, Built by MSYS2 project)
[FFmpeg stderr]   configuration: --enable-gpl --enable-version3 --enable-static --disable-w32threads --disable-autodetect --enable-fontconfig --enable-iconv --enable-gnutls --enable-libxml2 --enable-gmp --enable-bzlib --enable-lzma --enable-zlib --enable-libsrt --enable-libssh --enable-libzmq --enable-avisynth --enable-sdl2 --enable-libwebp --enable-libx264 --enable-libx265 --enable-libxvid --enable-libaom --enable-libopenjpeg --enable-libvpx --enable-mediafoundation --enable-libass --enable-libfreetype --enable-libfribidi --enable-libharfbuzz --enable-libvidstab --enable-libvmaf --enable-libzimg --enable-amf --enable-cuda-llvm --enable-cuvid --enable-dxva2 --enable-d3d11va --enable-d3d12va --enable-ffnvcodec --enable-libvpl --enable-nvdec --enable-nvenc --enable-vaapi --enable-libgme --enable-libopenmpt --enable-libopencore-amrwb --enable-libmp3lame --enable-libtheora --enable-libvo-amrwbenc --enable-libgsm --enable-libopencore-amrnb --enable-libopus --enable-libspeex --enable-libvorbis --enable-librubberband
[FFmpeg stderr]   libavutil      59. 39.100 / 59. 39.100
[FFmpeg stderr]   libavcodec     61. 19.101 / 61. 19.101
[FFmpeg stderr]   libavformat    61.  7.100 / 61.  7.100
[FFmpeg stderr]   libavdevice    61.  3.100 / 61.  3.100
[FFmpeg stderr]   libavfilter    10.  4.100 / 10.  4.100
[FFmpeg stderr]   libswscale      8.  3.100 /  8.  3.100
[FFmpeg stderr]   libswresample   5.  3.100 /  5.  3.100
[FFmpeg stderr]   libpostproc    58.  3.100 / 58.  3.100

[2025-05-06 15:44:43,598][INFO][XbLogger.cs:85] _ffmpegProcess.HasExited:False
[2025-05-06 15:44:43,613][INFO][XbLogger.cs:85] Wrote buffer data.
[2025-05-06 15:44:43,613][INFO][XbLogger.cs:85] Wrote buffer data.
[FFmpeg stderr] [aist#0:0/pcm_s16le @ 0000025ec8d36040] Guessed Channel Layout: mono
[FFmpeg stderr] Input #0, s16le, from 'pipe:0':
[FFmpeg stderr]   Duration: N/A, bitrate: 256 kb/s
[FFmpeg stderr]   Stream #0:0: Audio: pcm_s16le, 16000 Hz, mono, s16, 256 kb/s
[FFmpeg stderr] Stream mapping:
[FFmpeg stderr]   Stream #0:0 -> #0:0 (pcm_s16le (native) -> opus (libopus))
[FFmpeg stderr] [libopus @ 0000025ec8d317c0] No bit rate set. Defaulting to 64000 bps.
[FFmpeg stderr] Output #0, webm, to 'pipe:1':
[FFmpeg stderr]   Metadata:
[FFmpeg stderr]     encoder         : Lavf61.7.100
[FFmpeg stderr]   Stream #0:0: Audio: opus, 16000 Hz, mono, s16, 64 kb/s
[FFmpeg stderr]       Metadata:
[FFmpeg stderr]         encoder         : Lavc61.19.101 libopus
[FFmpeg stderr] [out#0/webm @ 0000025ec8d36200] video:0KiB audio:1KiB subtitle:0KiB other streams:0KiB global headers:0KiB muxing overhead: 67.493113%
[FFmpeg stderr] size=       1KiB time=00:00:00.04 bitrate= 243.2kbits/s speed=2.81x
Finished reading FFmpeg stderr.
[2025-05-06 15:44:44,101][INFO][XbLogger.cs:85] _ffmpegProcess.HasExited:True
[2025-05-06 15:44:44,132][ERROR][XbLogger.cs:67] Exception happend writing to input stream
System.ObjectDisposedException: Cannot access a closed file.
   at System.IO.FileStream.WriteAsync(Byte[] buffer, Int32 offset, Int32 count, CancellationToken cancellationToken)
   at System.IO.Stream.WriteAsync(Byte[] buffer, Int32 offset, Int32 count)
   at EchoBot.Media.SpeechService.AppendAudioBuffer(AudioMediaBuffer audioBuffer) in C:\Users\tm068\Documents\workspace\myprj\xbridge-teams-bot\src\EchoBot\Media\SpeechService.cs:line 242


  


  I am expecting the ffmpeg process keep running.