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Somos millones 1
21 juillet 2014, par
Mis à jour : Juin 2015
Langue : français
Type : Video
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Using ffserver to stream older IP cam MJPEG to RTSP
26 mai 2016, par tmar89I have an older Sony IP camera that has an MJPEG stream. I need to connect this to an NVR that only takes ONVIP or RTSP and I’m trying to use ffserver and ffmpeg to convert the MJPEG stream to RTSP but it’s not working. Any have some idea of what I may be doing wrong ? Saw an error in the attempted playback about an unsupported Protocol.
Here is my ffserver config :Port 8090
RTSPPort 5544
BindAddress 0.0.0.0
RTSPBindAddress 0.0.0.0
MaxClients 100
MaxBandwidth 10000
<feed>
File /tmp/feed27.ffm
FileMaxSize 5M
ACL allow 127.0.0.1
</feed>
<stream>
Format rtp
Feed feed27.ffm
NoAudio
VideoCodec mjpeg
VideoFrameRate 30
VideoSize 736x480
</stream>And here is the ffmpeg command I am using :
[tm@tele ffserver-rtsp]# ffmpeg -f mjpeg -r 30 -s 736x480 -i http://[CAMIP]/image http://localhost:8090/feed27.ffm
FFmpeg version 0.6.5, Copyright (c) 2000-2010 the FFmpeg developers
built on Jan 29 2012 17:52:15 with gcc 4.4.5 20110214 (Red Hat 4.4.5-6)
configuration: --prefix=/usr --libdir=/usr/lib64 --shlibdir=/usr/lib64 --mandir=/usr/share/man --incdir=/usr/include --disable-avisynth --extra-cflags='-O2 -g -pipe -Wall -Wp,-D_FORTIFY_SOURCE=2 -fexceptions -fstack-protector --param=ssp-buffer-size=4 -m64 -mtune=generic -fPIC' --enable-avfilter --enable-avfilter-lavf --enable-libdc1394 --enable-libdirac --enable-libfaac --enable-libfaad --enable-libfaadbin --enable-libgsm --enable-libmp3lame --enable-libopencore-amrnb --enable-libopencore-amrwb --enable-librtmp --enable-libschroedinger --enable-libspeex --enable-libtheora --enable-libx264 --enable-gpl --enable-nonfree --enable-postproc --enable-pthreads --enable-shared --enable-swscale --enable-vdpau --enable-version3 --enable-x11grab
libavutil 50.15. 1 / 50.15. 1
libavcodec 52.72. 2 / 52.72. 2
libavformat 52.64. 2 / 52.64. 2
libavdevice 52. 2. 0 / 52. 2. 0
libavfilter 1.19. 0 / 1.19. 0
libswscale 0.11. 0 / 0.11. 0
libpostproc 51. 2. 0 / 51. 2. 0
[mjpeg @ 0x1ece670]Estimating duration from bitrate, this may be inaccurate
Input #0, mjpeg, from 'http://[CAMIP]/image':
Duration: N/A, bitrate: N/A
Stream #0.0: Video: mjpeg, yuvj422p, 736x480, 30 fps, 30 tbr, 1200k tbn, 30 tbc
Output #0, ffm, to 'http://localhost:8090/feed27.ffm':
Metadata:
encoder : Lavf52.64.2
Stream #0.0: Video: mjpeg, yuvj420p, 736x480, q=2-31, 200 kb/s, 1000k tbn, 30 tbc
Stream mapping:
Stream #0.0 -> #0.0
Press [q] to stop encoding
[mjpeg @ 0x222d110]rc buffer underflow
frame= 640 fps= 17 q=31.4 size= 12884kB time=21.33 bitrate=4947.5kbits/sWhen I use VLC to open the stream, it cannot be found :
Your input can't be opened:
VLC is unable to open the MRL 'rtsp://localhost:5544/stream27.mpg'. Check the log for details.Finally, using ffplay on the same machine :
[tm@tele tmp]# ffplay rtsp://localhost:5544/stream27.sdp
FFplay version 0.6.5, Copyright (c) 2003-2010 the FFmpeg developers
built on Jan 29 2012 17:52:15 with gcc 4.4.5 20110214 (Red Hat 4.4.5-6)
configuration: --prefix=/usr --libdir=/usr/lib64 --shlibdir=/usr/lib64 --mandir=/usr/share/man --incdir=/usr/include --disable-avisynth --extra-cflags='-O2 -g -pipe -Wall -Wp,-D_FORTIFY_SOURCE=2 -fexceptions -fstack-protector --param=ssp-buffer-size=4 -m64 -mtune=generic -fPIC' --enable-avfilter --enable-avfilter-lavf --enable-libdc1394 --enable-libdirac --enable-libfaac --enable-libfaad --enable-libfaadbin --enable-libgsm --enable-libmp3lame --enable-libopencore-amrnb --enable-libopencore-amrwb --enable-librtmp --enable-libschroedinger --enable-libspeex --enable-libtheora --enable-libx264 --enable-gpl --enable-nonfree --enable-postproc --enable-pthreads --enable-shared --enable-swscale --enable-vdpau --enable-version3 --enable-x11grab
libavutil 50.15. 1 / 50.15. 1
libavcodec 52.72. 2 / 52.72. 2
libavformat 52.64. 2 / 52.64. 2
libavdevice 52. 2. 0 / 52. 2. 0
libavfilter 1.19. 0 / 1.19. 0
libswscale 0.11. 0 / 0.11. 0
libpostproc 51. 2. 0 / 51. 2. 0
ALSA lib pulse.c:229:(pulse_connect) PulseAudio: Unable to connect: Connection refused
rtsp://localhost:5544/stream27.sdp: Protocol not supportedAnd here was the log from ffserver :
127.0.0.1:5000 - - "PLAY stream27.mpg/streamid=0 RTP/UDP"
[rtp @ 0x721dc0]Unsupported codec 8
127.0.0.1:0 - - "PLAY stream27.mpg/streamid=0 RTP/TCP"
[rtp @ 0x728cb0]Unsupported codec 8
127.0.0.1 - - [SETUP] "rtsp://localhost:5544/stream27.mpg/streamid=0 RTSP/1.0" 200 641 -
How to stop ffmpeg when there's no incoming rtmp stream
5 juillet 2016, par M. IrichI use ffmpeg together with nginx-rtmp.
The thing is ffmpeg doesn’t finish the process when the stream’s finishedI use the following command :
ffmpeg -i 'rtmp://localhost:443/live/test' -loglevel debug -c:a libfdk_aac -b:a 192k -c:v libx264 -profile baseline -preset superfast -tune zerolatency -b:v 2500k -maxrate 4500k -minrate 1500k -bufsize 9000k -keyint_min 15 -g 15 -f dash -use_timeline 1 -use_template 1 -min_seg_duration 5000 -y /tmp/dash/test/test.mpdbut even the stream’s not running ffmpeg still can’t finish the process and is waiting for the rtmp stream
Successfully parsed a group of options.
Opening an input file: rtmp://localhost:443/live/test.
[rtmp @ 0x2ba2160] No default whitelist set
[tcp @ 0x2ba2720] No default whitelist set
[rtmp @ 0x2ba2160] Handshaking...
[rtmp @ 0x2ba2160] Type answer 3
[rtmp @ 0x2ba2160] Server version 13.14.10.13
[rtmp @ 0x2ba2160] Proto = rtmp, path = /live/test, app = live, fname = test
[rtmp @ 0x2ba2160] Server bandwidth = 5000000
[rtmp @ 0x2ba2160] Client bandwidth = 5000000
[rtmp @ 0x2ba2160] New incoming chunk size = 4096
[rtmp @ 0x2ba2160] Creating stream...
[rtmp @ 0x2ba2160] Sending play command for 'test'Is it possible to limit the latency time to several seconds ?
Sorry for any possible mistakes - English’s not my native language.
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ffmpeg Audiosegment error in get audio chunks in socketIo server in python
26 janvier 2024, par a_crszkvc30Last_NameColI want to send each audio chunk every minute.
this is the test code and i want to save audiofile and audio chunk file.
then, i will combine two audio files stop button was worked correctly but with set time function is not worked in python server.
there is python server code with socketio


def handle_voice(sid,data): # blob 으로 들어온 데이터 
 # BytesIO를 사용하여 메모리 상에서 오디오 데이터를 로드
 audio_segment = AudioSegment.from_file(BytesIO(data), format="webm")
 directory = "dddd"
 # 오디오 파일로 저장
 #directory = str(names_sid.get(sid))
 if not os.path.exists(directory):
 os.makedirs(directory)
 
 # 오디오 파일로 저장
 file_path = os.path.join(directory, f'{sid}.wav')
 audio_segment.export(file_path, format='wav') 
 print('오디오 파일 저장 완료')`
 


and there is client






 
 
 <code class="echappe-js"><script src="https://cdnjs.cloudflare.com/ajax/libs/socket.io/4.5.2/socket.io.js"></script>



 






<script>&#xA; var socket = io(&#x27;http://127.0.0.1:5000&#x27;);&#xA; const record = document.getElementById("record")&#xA; const stop = document.getElementById("stop")&#xA; const soundClips = document.getElementById("sound-clips")&#xA; const chkHearMic = document.getElementById("chk-hear-mic")&#xA;&#xA; const audioCtx = new(window.AudioContext || window.webkitAudioContext)() // 오디오 컨텍스트 정의&#xA;&#xA; const analyser = audioCtx.createAnalyser()&#xA; // const distortion = audioCtx.createWaveShaper()&#xA; // const gainNode = audioCtx.createGain()&#xA; // const biquadFilter = audioCtx.createBiquadFilter()&#xA;&#xA; function makeSound(stream) {&#xA; const source = audioCtx.createMediaStreamSource(stream)&#xA; socket.connect()&#xA; source.connect(analyser)&#xA; // analyser.connect(distortion)&#xA; // distortion.connect(biquadFilter)&#xA; // biquadFilter.connect(gainNode)&#xA; // gainNode.connect(audioCtx.destination) // connecting the different audio graph nodes together&#xA; analyser.connect(audioCtx.destination)&#xA;&#xA; }&#xA;&#xA; if (navigator.mediaDevices) {&#xA; console.log(&#x27;getUserMedia supported.&#x27;)&#xA;&#xA; const constraints = {&#xA; audio: true&#xA; }&#xA; let chunks = []&#xA;&#xA; navigator.mediaDevices.getUserMedia(constraints)&#xA; .then(stream => {&#xA;&#xA; const mediaRecorder = new MediaRecorder(stream)&#xA; &#xA; chkHearMic.onchange = e => {&#xA; if(e.target.checked == true) {&#xA; audioCtx.resume()&#xA; makeSound(stream)&#xA; } else {&#xA; audioCtx.suspend()&#xA; }&#xA; }&#xA; &#xA; record.onclick = () => {&#xA; mediaRecorder.start(1000)&#xA; console.log(mediaRecorder.state)&#xA; console.log("recorder started")&#xA; record.style.background = "red"&#xA; record.style.color = "black"&#xA; }&#xA;&#xA; stop.onclick = () => {&#xA; mediaRecorder.stop()&#xA; console.log(mediaRecorder.state)&#xA; console.log("recorder stopped")&#xA; record.style.background = ""&#xA; record.style.color = ""&#xA; }&#xA;&#xA; mediaRecorder.onstop = e => {&#xA; console.log("data available after MediaRecorder.stop() called.")&#xA; const bb = new Blob(chunks, { &#x27;type&#x27; : &#x27;audio/wav&#x27; })&#xA; socket.emit(&#x27;voice&#x27;,bb)&#xA; const clipName = prompt("오디오 파일 제목을 입력하세요.", new Date())&#xA;&#xA; const clipContainer = document.createElement(&#x27;article&#x27;)&#xA; const clipLabel = document.createElement(&#x27;p&#x27;)&#xA; const audio = document.createElement(&#x27;audio&#x27;)&#xA; const deleteButton = document.createElement(&#x27;button&#x27;)&#xA;&#xA; clipContainer.classList.add(&#x27;clip&#x27;)&#xA; audio.setAttribute(&#x27;controls&#x27;, &#x27;&#x27;)&#xA; deleteButton.innerHTML = "삭제"&#xA; clipLabel.innerHTML = clipName&#xA;&#xA; clipContainer.appendChild(audio)&#xA; clipContainer.appendChild(clipLabel)&#xA; clipContainer.appendChild(deleteButton)&#xA; soundClips.appendChild(clipContainer)&#xA;&#xA; audio.controls = true&#xA; const blob = new Blob(chunks, {&#xA; &#x27;type&#x27;: &#x27;audio/ogg codecs=opus&#x27;&#xA; })&#xA;&#xA; chunks = []&#xA; const audioURL = URL.createObjectURL(blob)&#xA; audio.src = audioURL&#xA; console.log("recorder stopped")&#xA;&#xA; deleteButton.onclick = e => {&#xA; evtTgt = e.target&#xA; evtTgt .parentNode.parentNode.removeChild(evtTgt.parentNode)&#xA; }&#xA; }&#xA;&#xA; mediaRecorder.ondataavailable = function(e) {&#xA; chunks.push(e.data)&#xA; if (chunks.length >= 5)&#xA; {&#xA; const bloddb = new Blob(chunks, { &#x27;type&#x27; : &#x27;audio/wav&#x27; })&#xA; socket.emit(&#x27;voice&#x27;, bloddb)&#xA; &#xA; chunks = []&#xA; }&#xA; mediaRecorder.sendData = function(buffer) {&#xA; const bloddb = new Blob(buffer, { &#x27;type&#x27; : &#x27;audio/wav&#x27; })&#xA; socket.emit(&#x27;voice&#x27;, bloddb)&#xA;}&#xA;};&#xA; })&#xA; .catch(err => {&#xA; console.log(&#x27;The following error occurred: &#x27; &#x2B; err)&#xA; })&#xA; }&#xA; </script>



ask exception was never retrieved
future: <task finished="finished" coro="<InstrumentedAsyncServer._handle_event_internal()" defined="defined" at="at"> exception=CouldntDecodeError('Decoding failed. ffmpeg returned error code: 3199971767\n\nOutput from ffmpeg/avlib:\n\nffmpeg version 6.1.1-full_build-www.gyan.dev Copyright (c) 2000-2023 the FFmpeg developers\r\n built with gcc 12.2.0 (Rev10, Built by MSYS2 project)\r\n configuration: --enable-gpl --enable-version3 --enable-static --pkg-config=pkgconf --disable-w32threads --disable-autodetect --enable-fontconfig --enable-iconv --enable-gnutls --enable-libxml2 --enable-gmp --enable-bzlib --enable-lzma --enable-libsnappy --enable-zlib --enable-librist --enable-libsrt --enable-libssh --enable-libzmq --enable-avisynth --enable-libbluray --enable-libcaca --enable-sdl2 --enable-libaribb24 --enable-libaribcaption --enable-libdav1d --enable-libdavs2 --enable-libuavs3d --enable-libzvbi --enable-librav1e --enable-libsvtav1 --enable-libwebp --enable-libx264 --enable-libx265 --enable-libxavs2 --enable-libxvid --enable-libaom --enable-libjxl --enable-libopenjpeg --enable-libvpx --enable-mediafoundation --enable-libass --enable-frei0r --enable-libfreetype --enable-libfribidi --enable-libharfbuzz --enable-liblensfun --enable-libvidstab --enable-libvmaf --enable-libzimg --enable-amf --enable-cuda-llvm --enable-cuvid --enable-ffnvcodec --enable-nvdec --enable-nvenc --enable-dxva2 --enable-d3d11va --enable-libvpl --enable-libshaderc --enable-vulkan --enable-libplacebo --enable-opencl --enable-libcdio --enable-libgme --enable-libmodplug --enable-libopenmpt --enable-libopencore-amrwb --enable-libmp3lame --enable-libshine --enable-libtheora --enable-libtwolame --enable-libvo-amrwbenc --enable-libcodec2 --enable-libilbc --enable-libgsm --enable-libopencore-amrnb --enable-libopus --enable-libspeex --enable-libvorbis --enable-ladspa --enable-libbs2b --enable-libflite --enable-libmysofa --enable-librubberband --enable-libsoxr --enable-chromaprint\r\n libavutil 58. 29.100 / 58. 29.100\r\n libavcodec 60. 31.102 / 60. 31.102\r\n libavformat 60. 16.100 / 60. 16.100\r\n libavdevice 60. 3.100 / 60. 3.100\r\n libavfilter 9. 12.100 / 9. 12.100\r\n libswscale 7. 5.100 / 7. 5.100\r\n libswresample 4. 12.100 / 4. 12.100\r\n libpostproc 57. 3.100 / 57. 3.100\r\n[cache @ 000001d9828efe40] Inner protocol failed to seekback end : -40\r\n[matroska,webm @ 000001d9828efa00] EBML header parsing failed\r\n[cache @ 000001d9828efe40] Statistics, cache hits:0 cache misses:3\r\n[in#0 @ 000001d9828da3c0] Error opening input: Invalid data found when processing input\r\nError opening input file cache:pipe:0.\r\nError opening input files: Invalid data found when processing input\r\n')>
Traceback (most recent call last):
 File "f:\fastapi-socketio-wb38\.vent\Lib\site-packages\socketio\async_admin.py", line 276, in _handle_event_internal
 ret = await self.sio.__handle_event_internal(server, sid, eio_sid,
 ^^^^^^^^^^^^^^^^^^^^^^^^^^^^^^^^^^^^^^^^^^^^^^^^^^^^^^^^^^^^
 File "f:\fastapi-socketio-wb38\.vent\Lib\site-packages\socketio\async_server.py", line 597, in _handle_event_internal
 r = await server._trigger_event(data[0], namespace, sid, *data[1:])
 ^^^^^^^^^^^^^^^^^^^^^^^^^^^^^^^^^^^^^^^^^^^^^^^^^^^^^^^^^^^^^^^
 File "f:\fastapi-socketio-wb38\.vent\Lib\site-packages\socketio\async_server.py", line 635, in _trigger_event
 ret = handler(*args)
 ^^^^^^^^^^^^^^
 File "f:\fastapi-socketio-wb38\Python-Javascript-Websocket-Video-Streaming--main\poom2.py", line 153, in handle_voice
 audio_segment = AudioSegment.from_file(BytesIO(data), format="webm")
 ^^^^^^^^^^^^^^^^^^^^^^^^^^^^^^^^^^^^^^^^^^^^^^^^^^^^
 File "f:\fastapi-socketio-wb38\.vent\Lib\site-packages\pydub\audio_segment.py", line 773, in from_file
 raise CouldntDecodeError(
pydub.exceptions.CouldntDecodeError: Decoding failed. ffmpeg returned error code: 3199971767

Output from ffmpeg/avlib:

ffmpeg version 6.1.1-full_build-www.gyan.dev Copyright (c) 2000-2023 the FFmpeg developers
 built with gcc 12.2.0 (Rev10, Built by MSYS2 project)
 configuration: --enable-gpl --enable-version3 --enable-static --pkg-config=pkgconf --disable-w32threads --disable-autodetect --enable-fontconfig --enable-iconv --enable-gnutls --enable-libxml2 --enable-gmp --enable-bzlib --enable-lzma --enable-libsnappy --enable-zlib --enable-librist --enable-libsrt --enable-libssh --enable-libzmq --enable-avisynth --enable-libbluray --enable-libcaca --enable-sdl2 --enable-libaribb24 --enable-libaribcaption --enable-libdav1d --enable-libdavs2 --enable-libuavs3d --enable-libzvbi --enable-librav1e --enable-libsvtav1 --enable-libwebp --enable-libx264 --enable-libx265 --enable-libxavs2 --enable-libxvid --enable-libaom --enable-libjxl --enable-libopenjpeg --enable-libvpx --enable-mediafoundation --enable-libass --enable-frei0r --enable-libfreetype --enable-libfribidi --enable-libharfbuzz --enable-liblensfun --enable-libvidstab --enable-libvmaf --enable-libzimg --enable-amf --enable-cuda-llvm --enable-cuvid --enable-ffnvcodec --enable-nvdec --enable-nvenc --enable-dxva2 --enable-d3d11va --enable-libvpl --enable-libshaderc --enable-vulkan --enable-libplacebo --enable-opencl --enable-libcdio --enable-libgme --enable-libmodplug --enable-libopenmpt --enable-libopencore-amrwb --enable-libmp3lame --enable-libshine --enable-libtheora --enable-libtwolame --enable-libvo-amrwbenc --enable-libcodec2 --enable-libilbc --enable-libgsm --enable-libopencore-amrnb --enable-libopus --enable-libspeex --enable-libvorbis --enable-ladspa --enable-libbs2b --enable-libflite --enable-libmysofa --enable-librubberband --enable-libsoxr --enable-chromaprint
 libavutil 58. 29.100 / 58. 29.100
 libavcodec 60. 31.102 / 60. 31.102
 libavformat 60. 16.100 / 60. 16.100
 libavdevice 60. 3.100 / 60. 3.100
 libavfilter 9. 12.100 / 9. 12.100
 libswscale 7. 5.100 / 7. 5.100
 libswresample 4. 12.100 / 4. 12.100
 libpostproc 57. 3.100 / 57. 3.100
[cache @ 000001d9828efe40] Inner protocol failed to seekback end : -40
[matroska,webm @ 000001d9828efa00] EBML header parsing failed
[cache @ 000001d9828efe40] Statistics, cache hits:0 cache misses:3
[in#0 @ 000001d9828da3c0] Error opening input: Invalid data found when processing input
Error opening input file cache:pipe:0.
Error opening input files: Invalid data found when processing input
</task>

im using version of ffmpeg-6.1.1-full_build.
i dont know this error exist the stop button sent event correctly. but chunk data was not work correctly in python server.
my english was so bad. sry


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