Recherche avancée

Médias (1)

Mot : - Tags -/Rennes

Autres articles (22)

  • Personnaliser en ajoutant son logo, sa bannière ou son image de fond

    5 septembre 2013, par

    Certains thèmes prennent en compte trois éléments de personnalisation : l’ajout d’un logo ; l’ajout d’une bannière l’ajout d’une image de fond ;

  • Ecrire une actualité

    21 juin 2013, par

    Présentez les changements dans votre MédiaSPIP ou les actualités de vos projets sur votre MédiaSPIP grâce à la rubrique actualités.
    Dans le thème par défaut spipeo de MédiaSPIP, les actualités sont affichées en bas de la page principale sous les éditoriaux.
    Vous pouvez personnaliser le formulaire de création d’une actualité.
    Formulaire de création d’une actualité Dans le cas d’un document de type actualité, les champs proposés par défaut sont : Date de publication ( personnaliser la date de publication ) (...)

  • Publier sur MédiaSpip

    13 juin 2013

    Puis-je poster des contenus à partir d’une tablette Ipad ?
    Oui, si votre Médiaspip installé est à la version 0.2 ou supérieure. Contacter au besoin l’administrateur de votre MédiaSpip pour le savoir

Sur d’autres sites (5166)

  • Why does my script return 0 when it fails ? Bash, Whisper, ffmpeg

    30 avril 2024, par d-b

    I have a bash script that executes whisper on all sound files in a directory. Whisper uses ffmpeg to decode sound files to a format it can handle. One of the files in the directory was corrupt and caused ffmpeg to fail. The first time I executed my script it returned 1 when it hit that file but when I try to reproduce the error it always returns 0.

    


    Here is the log from the first run when 1 was returned :

    


    Traceback (most recent call last):&#xA;  File "/Users/db/Library/Python/3.11/lib/python/site-packages/whisper/audio.py", line 48, in load_audio&#xA;    .run(cmd=["ffmpeg", "-nostdin"], capture_stdout=True, capture_stderr=True)&#xA;     ^^^^^^^^^^^^^^^^^^^^^^^^^^^^^^^^^^^^^^^^^^^^^^^^^^^^^^^^^^^^^^^^^^^^^^^^^&#xA;  File "/Users/db/Library/Python/3.11/lib/python/site-packages/ffmpeg/_run.py", line 325, in run&#xA;    raise Error(&#x27;ffmpeg&#x27;, out, err)&#xA;ffmpeg._run.Error: ffmpeg error (see stderr output for detail)&#xA;&#xA;The above exception was the direct cause of the following exception:&#xA;&#xA;Traceback (most recent call last):&#xA;  File "/Users/db/Library/Python/3.11/bin/whisper", line 8, in <module>&#xA;    sys.exit(cli())&#xA;             ^^^^^&#xA;  File "/Users/db/Library/Python/3.11/lib/python/site-packages/whisper/transcribe.py", line 437, in cli&#xA;    result = transcribe(model, audio_path, temperature=temperature, **args)&#xA;             ^^^^^^^^^^^^^^^^^^^^^^^^^^^^^^^^^^^^^^^^^^^^^^^^^^^^^^^^^^^^^^&#xA;  File "/Users/db/Library/Python/3.11/lib/python/site-packages/whisper/transcribe.py", line 121, in transcribe&#xA;    mel = log_mel_spectrogram(audio, padding=N_SAMPLES)&#xA;          ^^^^^^^^^^^^^^^^^^^^^^^^^^^^^^^^^^^^^^^^^^^^^&#xA;  File "/Users/db/Library/Python/3.11/lib/python/site-packages/whisper/audio.py", line 130, in log_mel_spectrogram&#xA;    audio = load_audio(audio)&#xA;            ^^^^^^^^^^^^^^^^^&#xA;  File "/Users/db/Library/Python/3.11/lib/python/site-packages/whisper/audio.py", line 51, in load_audio&#xA;    raise RuntimeError(f"Failed to load audio: {e.stderr.decode()}") from e&#xA;RuntimeError: Failed to load audio: ffmpeg version 4.4.4 Copyright (c) 2000-2023 the FFmpeg developers&#xA;  built with Apple clang version 15.0.0 (clang-1500.1.0.2.5)&#xA;  configuration: --prefix=/opt/local --cc=/usr/bin/clang --mandir=/opt/local/share/man --enable-audiotoolbox --disable-indev=jack --disable-libjack --disable-libopencore-amrnb --disable-libopencore-amrwb --disable-libxcb --disable-libxcb-shm --disable-libxcb-xfixes --enable-opencl --disable-outdev=xv --enable-sdl2 --disable-securetransport --enable-videotoolbox --enable-avfilter --enable-avresample --enable-fontconfig --enable-gnutls --enable-libass --enable-libbluray --enable-libdav1d --enable-libfreetype --enable-libfribidi --enable-libmodplug --enable-libmp3lame --enable-libopenjpeg --enable-libopus --enable-librsvg --enable-libsoxr --enable-libspeex --enable-libtheora --enable-libvorbis --enable-libvpx --enable-libzimg --enable-libzvbi --enable-lzma --enable-pthreads --enable-shared --enable-swscale --enable-zlib --enable-libaom --enable-libsvtav1 --arch=x86_64 --enable-x86asm --enable-gpl --enable-libvidstab --enable-libx264 --enable-libx265 --enable-libxvid --enable-postproc&#xA;  libavutil      56. 70.100 / 56. 70.100&#xA;  libavcodec     58.134.100 / 58.134.100&#xA;  libavformat    58. 76.100 / 58. 76.100&#xA;  libavdevice    58. 13.100 / 58. 13.100&#xA;  libavfilter     7.110.100 /  7.110.100&#xA;  libavresample   4.  0.  0 /  4.  0.  0&#xA;  libswscale      5.  9.100 /  5.  9.100&#xA;  libswresample   3.  9.100 /  3.  9.100&#xA;  libpostproc    55.  9.100 / 55.  9.100&#xA;/soundfile1.opus: Invalid data found when processing input&#xA;&#xA;&#xA;exit: 1 //this is my own output&#xA;</module>

    &#xA;

    and here is an example of when it returns 0 despite the input file is corrupt :

    &#xA;

    % whisper soundfile1.opus&#xA;&#xA;Traceback (most recent call last):&#xA;  File "/Users/db/Library/Python/3.11/lib/python/site-packages/whisper/audio.py", line 58, in load_audio&#xA;    out = run(cmd, capture_output=True, check=True).stdout&#xA;          ^^^^^^^^^^^^^^^^^^^^^^^^^^^^^^^^^^^^^^^^^&#xA;  File "/opt/local/Library/Frameworks/Python.framework/Versions/3.11/lib/python3.11/subprocess.py", line 571, in run&#xA;    raise CalledProcessError(retcode, process.args,&#xA;subprocess.CalledProcessError: Command &#x27;[&#x27;ffmpeg&#x27;, &#x27;-nostdin&#x27;, &#x27;-threads&#x27;, &#x27;0&#x27;, &#x27;-i&#x27;, &#x27;soundfile1.opus&#x27;, &#x27;-f&#x27;, &#x27;s16le&#x27;, &#x27;-ac&#x27;, &#x27;1&#x27;, &#x27;-acodec&#x27;, &#x27;pcm_s16le&#x27;, &#x27;-ar&#x27;, &#x27;16000&#x27;, &#x27;-&#x27;]&#x27; returned non-zero exit status 1.&#xA;&#xA;The above exception was the direct cause of the following exception:&#xA;&#xA;Traceback (most recent call last):&#xA;  File "/Users/db/Library/Python/3.11/lib/python/site-packages/whisper/transcribe.py", line 597, in cli&#xA;    result = transcribe(model, audio_path, temperature=temperature, **args)&#xA;             ^^^^^^^^^^^^^^^^^^^^^^^^^^^^^^^^^^^^^^^^^^^^^^^^^^^^^^^^^^^^^^&#xA;  File "/Users/db/Library/Python/3.11/lib/python/site-packages/whisper/transcribe.py", line 133, in transcribe&#xA;    mel = log_mel_spectrogram(audio, model.dims.n_mels, padding=N_SAMPLES)&#xA;          ^^^^^^^^^^^^^^^^^^^^^^^^^^^^^^^^^^^^^^^^^^^^^^^^^^^^^^^^^^^^^^^^&#xA;  File "/Users/db/Library/Python/3.11/lib/python/site-packages/whisper/audio.py", line 140, in log_mel_spectrogram&#xA;    audio = load_audio(audio)&#xA;            ^^^^^^^^^^^^^^^^^&#xA;  File "/Users/db/Library/Python/3.11/lib/python/site-packages/whisper/audio.py", line 60, in load_audio&#xA;    raise RuntimeError(f"Failed to load audio: {e.stderr.decode()}") from e&#xA;RuntimeError: Failed to load audio: ffmpeg version 4.4.4 Copyright (c) 2000-2023 the FFmpeg developers&#xA;  built with Apple clang version 15.0.0 (clang-1500.1.0.2.5)&#xA;  configuration: --prefix=/opt/local --cc=/usr/bin/clang --mandir=/opt/local/share/man --enable-audiotoolbox --disable-indev=jack --disable-libjack --disable-libopencore-amrnb --disable-libopencore-amrwb --disable-libxcb --disable-libxcb-shm --disable-libxcb-xfixes --enable-opencl --disable-outdev=xv --enable-sdl2 --disable-securetransport --enable-videotoolbox --enable-avfilter --enable-avresample --enable-fontconfig --enable-gnutls --enable-libass --enable-libbluray --enable-libdav1d --enable-libfreetype --enable-libfribidi --enable-libmodplug --enable-libmp3lame --enable-libopenjpeg --enable-libopus --enable-librsvg --enable-libsoxr --enable-libspeex --enable-libtheora --enable-libvorbis --enable-libvpx --enable-libzimg --enable-libzvbi --enable-lzma --enable-pthreads --enable-shared --enable-swscale --enable-zlib --enable-libaom --enable-libsvtav1 --arch=x86_64 --enable-x86asm --enable-gpl --enable-libvidstab --enable-libx264 --enable-libx265 --enable-libxvid --enable-postproc&#xA;  libavutil      56. 70.100 / 56. 70.100&#xA;  libavcodec     58.134.100 / 58.134.100&#xA;  libavformat    58. 76.100 / 58. 76.100&#xA;  libavdevice    58. 13.100 / 58. 13.100&#xA;  libavfilter     7.110.100 /  7.110.100&#xA;  libavresample   4.  0.  0 /  4.  0.  0&#xA;  libswscale      5.  9.100 /  5.  9.100&#xA;  libswresample   3.  9.100 /  3.  9.100&#xA;  libpostproc    55.  9.100 / 55.  9.100&#xA;soundfile1.opus: Invalid data found when processing input&#xA;&#xA;Skipping soundfile1.opus due to RuntimeError: Failed to load audio: ffmpeg version 4.4.4 Copyright (c) 2000-2023 the FFmpeg developers&#xA;  built with Apple clang version 15.0.0 (clang-1500.1.0.2.5)&#xA;  configuration: --prefix=/opt/local --cc=/usr/bin/clang --mandir=/opt/local/share/man --enable-audiotoolbox --disable-indev=jack --disable-libjack --disable-libopencore-amrnb --disable-libopencore-amrwb --disable-libxcb --disable-libxcb-shm --disable-libxcb-xfixes --enable-opencl --disable-outdev=xv --enable-sdl2 --disable-securetransport --enable-videotoolbox --enable-avfilter --enable-avresample --enable-fontconfig --enable-gnutls --enable-libass --enable-libbluray --enable-libdav1d --enable-libfreetype --enable-libfribidi --enable-libmodplug --enable-libmp3lame --enable-libopenjpeg --enable-libopus --enable-librsvg --enable-libsoxr --enable-libspeex --enable-libtheora --enable-libvorbis --enable-libvpx --enable-libzimg --enable-libzvbi --enable-lzma --enable-pthreads --enable-shared --enable-swscale --enable-zlib --enable-libaom --enable-libsvtav1 --arch=x86_64 --enable-x86asm --enable-gpl --enable-libvidstab --enable-libx264 --enable-libx265 --enable-libxvid --enable-postproc&#xA;  libavutil      56. 70.100 / 56. 70.100&#xA;  libavcodec     58.134.100 / 58.134.100&#xA;  libavformat    58. 76.100 / 58. 76.100&#xA;  libavdevice    58. 13.100 / 58. 13.100&#xA;  libavfilter     7.110.100 /  7.110.100&#xA;  libavresample   4.  0.  0 /  4.  0.  0&#xA;  libswscale      5.  9.100 /  5.  9.100&#xA;  libswresample   3.  9.100 /  3.  9.100&#xA;  libpostproc    55.  9.100 / 55.  9.100&#xA;soundfile1.opus: Invalid data found when processing input&#xA;&#xA;% echo $?&#xA;0&#xA;

    &#xA;

    When I try to reproduce the behaviour I have both tried to execute the script as well as just running the actual line of code from the command prompt. The actual script looks like this :

    &#xA;

    if whisper "$file" >> "${directory}/../${transdir}/${name}.txt"; then&#xA;    whispersuccess=$?&#xA;    echo "exit code $whispersuccess"&#xA;else&#xA;    whispersuccess=$?&#xA;    echo "exit code: $whispersuccess"&#xA;

    &#xA;

    I want it to return ¬0 when something like this happens. How do I achieve this ?

    &#xA;

  • ffmpeg failed to load audio file

    14 avril 2024, par Vaishnav Ghenge
    Failed to load audio: ffmpeg version 5.1.4-0&#x2B;deb12u1 Copyright (c) Failed to load audio: ffmpeg version 5.1.4-0&#x2B;deb12u1 Copyright (c) 2000-2023 the FFmpeg developers&#xA;  built with gcc 12 (Debian 12.2.0-14)&#xA;  configuration: --prefix=/usr --extra-version=0&#x2B;deb12u1 --toolchain=hardened --libdir=/usr/lib/x86_64-linux-gnu --incdir=/usr/include/x86_64-linux-gnu --arch=amd64 --enable-gpl --disable-stripping --enable-gnutls --enable-ladspa --enable-libaom --enable-libass --enable-libbluray --enable-libbs2b --enable-libcaca --enable-libcdio --enable-libcodec2 --enable-libdav1d --enable-libflite --enable-libfontconfig --enable-libfreetype --enable-libfribidi --enable-libglslang --enable-libgme --enable-libgsm --enable-libjack --enable-libmp3lame --enable-libmysofa --enable-libopenjpeg --enable-libopenmpt --enable-libopus --enable-libpulse --enable-librabbitmq --enable-librist --enable-librubberband --enable-libshine --enable-libsnappy --enable-libsoxr --enable-libspeex --enable-libsrt --enable-libssh --enable-libsvtav1 --enable-libtheora --enable-libtwolame --enable-libvidstab --enable-libvorbis --enable-libvpx --enable-libwebp --enable-libx265 --enable-libxml2 --enable-libxvid --enable-libzimg --enable-libzmq --enable-libzvbi --enable-lv2 --enable-omx --enable-openal --enable-opencl --enable-opengl --enable-sdl2 --disable-sndio --enable-libjxl --enable-pocketsphinx --enable-librsvg --enable-libmfx --enable-libdc1394 --enable-libdrm --enable-libiec61883 --enable-chromaprint --enable-frei0r --enable-libx264 --enable-libplacebo --enable-librav1e --enable-shared&#xA;  libavutil      57. 28.100 / 57. 28.100&#xA;  libavcodec     59. 37.100 / 59. 37.100&#xA;  libavformat    59. 27.100 / 59. 27.100&#xA;  libavdevice    59.  7.100 / 59.  7.100&#xA;  libavfilter     8. 44.100 /  8. 44.100&#xA;  libswscale      6.  7.100 /  6.  7.100&#xA;  libswresample   4.  7.100 /  4.  7.100&#xA;  libpostproc    56.  6.100 / 56.  6.100&#xA;/tmp/tmpjlchcpdm.wav: Invalid data found when processing input&#xA;

    &#xA;

    backend :

    &#xA;

    &#xA;@app.route("/transcribe", methods=["POST"])&#xA;def transcribe():&#xA;    # Check if audio file is present in the request&#xA;    if &#x27;audio_file&#x27; not in request.files:&#xA;        return jsonify({"error": "No file part"}), 400&#xA;    &#xA;    audio_file = request.files.get(&#x27;audio_file&#x27;)&#xA;&#xA;    # Check if audio_file is sent in files&#xA;    if not audio_file:&#xA;        return jsonify({"error": "`audio_file` is missing in request.files"}), 400&#xA;&#xA;    # Check if the file is present&#xA;    if audio_file.filename == &#x27;&#x27;:&#xA;        return jsonify({"error": "No selected file"}), 400&#xA;&#xA;    # Save the file with a unique name&#xA;    filename = secure_filename(audio_file.filename)&#xA;    unique_filename = os.path.join("uploads", str(uuid.uuid4()) &#x2B; &#x27;_&#x27; &#x2B; filename)&#xA;    # audio_file.save(unique_filename)&#xA;    &#xA;    # Read the contents of the audio file&#xA;    contents = audio_file.read()&#xA;&#xA;    max_file_size = 500 * 1024 * 1024&#xA;    if len(contents) > max_file_size:&#xA;        return jsonify({"error": "File is too large"}), 400&#xA;&#xA;    # Check if the file extension suggests it&#x27;s a WAV file&#xA;    if not filename.lower().endswith(&#x27;.wav&#x27;):&#xA;        # Delete the file if it&#x27;s not a WAV file&#xA;        os.remove(unique_filename)&#xA;        return jsonify({"error": "Only WAV files are supported"}), 400&#xA;&#xA;    print(f"\033[92m{filename}\033[0m")&#xA;&#xA;    # Call Celery task asynchronously&#xA;    result = transcribe_audio.delay(contents)&#xA;&#xA;    return jsonify({&#xA;        "task_id": result.id,&#xA;        "status": "pending"&#xA;    })&#xA;&#xA;&#xA;@celery_app.task&#xA;def transcribe_audio(contents):&#xA;    # Transcribe the audio&#xA;    try:&#xA;        # Create a temporary file to save the audio data&#xA;        with tempfile.NamedTemporaryFile(suffix=".wav", delete=False) as temp_audio:&#xA;            temp_path = temp_audio.name&#xA;            temp_audio.write(contents)&#xA;&#xA;            print(f"\033[92mFile temporary path: {temp_path}\033[0m")&#xA;            transcribe_start_time = time.time()&#xA;&#xA;            # Transcribe the audio&#xA;            transcription = transcribe_with_whisper(temp_path)&#xA;            &#xA;            transcribe_end_time = time.time()&#xA;            print(f"\033[92mTranscripted text: {transcription}\033[0m")&#xA;&#xA;            return transcription, transcribe_end_time - transcribe_start_time&#xA;&#xA;    except Exception as e:&#xA;        print(f"\033[92mError: {e}\033[0m")&#xA;        return str(e)&#xA;

    &#xA;

    frontend :

    &#xA;

        useEffect(() => {&#xA;        const init = () => {&#xA;            navigator.mediaDevices.getUserMedia({audio: true})&#xA;                .then((audioStream) => {&#xA;                    const recorder = new MediaRecorder(audioStream);&#xA;&#xA;                    recorder.ondataavailable = e => {&#xA;                        if (e.data.size > 0) {&#xA;                            setChunks(prevChunks => [...prevChunks, e.data]);&#xA;                        }&#xA;                    };&#xA;&#xA;                    recorder.onerror = (e) => {&#xA;                        console.log("error: ", e);&#xA;                    }&#xA;&#xA;                    recorder.onstart = () => {&#xA;                        console.log("started");&#xA;                    }&#xA;&#xA;                    recorder.start();&#xA;&#xA;                    setStream(audioStream);&#xA;                    setRecorder(recorder);&#xA;                });&#xA;        }&#xA;&#xA;        init();&#xA;&#xA;        return () => {&#xA;            if (recorder &amp;&amp; recorder.state === &#x27;recording&#x27;) {&#xA;                recorder.stop();&#xA;            }&#xA;&#xA;            if (stream) {&#xA;                stream.getTracks().forEach(track => track.stop());&#xA;            }&#xA;        }&#xA;    }, []);&#xA;&#xA;    useEffect(() => {&#xA;        // Send chunks of audio data to the backend at regular intervals&#xA;        const intervalId = setInterval(() => {&#xA;            if (recorder &amp;&amp; recorder.state === &#x27;recording&#x27;) {&#xA;                recorder.requestData(); // Trigger data available event&#xA;            }&#xA;        }, 8000); // Adjust the interval as needed&#xA;&#xA;&#xA;        return () => {&#xA;            if (intervalId) {&#xA;                console.log("Interval cleared");&#xA;                clearInterval(intervalId);&#xA;            }&#xA;        };&#xA;    }, [recorder]);&#xA;&#xA;    useEffect(() => {&#xA;        const processAudio = async () => {&#xA;            if (chunks.length > 0) {&#xA;                // Send the latest chunk to the server for transcription&#xA;                const latestChunk = chunks[chunks.length - 1];&#xA;&#xA;                const audioBlob = new Blob([latestChunk]);&#xA;                convertBlobToAudioFile(audioBlob);&#xA;            }&#xA;        };&#xA;&#xA;        void processAudio();&#xA;    }, [chunks]);&#xA;&#xA;    const convertBlobToAudioFile = useCallback((blob: Blob) => {&#xA;        // Convert Blob to audio file (e.g., WAV)&#xA;        // This conversion may require using a third-party library or service&#xA;        // For example, you can use the MediaRecorder API to record audio in WAV format directly&#xA;        // Alternatively, you can use a library like recorderjs to perform the conversion&#xA;        // Here&#x27;s a simplified example using recorderjs:&#xA;&#xA;        const reader = new FileReader();&#xA;        reader.onload = () => {&#xA;            const audioBuffer = reader.result; // ArrayBuffer containing audio data&#xA;&#xA;            // Send audioBuffer to Flask server or perform further processing&#xA;            sendAudioToFlask(audioBuffer as ArrayBuffer);&#xA;        };&#xA;&#xA;        reader.readAsArrayBuffer(blob);&#xA;    }, []);&#xA;&#xA;    const sendAudioToFlask = useCallback((audioBuffer: ArrayBuffer) => {&#xA;        const formData = new FormData();&#xA;        formData.append(&#x27;audio_file&#x27;, new Blob([audioBuffer]), `speech_audio.wav`);&#xA;&#xA;        console.log(formData.get("audio_file"));&#xA;&#xA;        fetch(&#x27;http://34.87.75.138:8000/transcribe&#x27;, {&#xA;            method: &#x27;POST&#x27;,&#xA;            body: formData&#xA;        })&#xA;            .then(response => response.json())&#xA;            .then((data: { task_id: string, status: string }) => {&#xA;                pendingTaskIdsRef.current.push(data.task_id);&#xA;            })&#xA;            .catch(error => {&#xA;                console.error(&#x27;Error sending audio to Flask server:&#x27;, error);&#xA;            });&#xA;    }, []);&#xA;

    &#xA;

    I was trying to pass the audio from frontend to whisper model which is in flask app

    &#xA;

  • FFmpeg RTSP drop rate increases when frame rate is reduced

    13 avril 2024, par Avishka Perera

    I need to read an RTSP stream, process the images individually in Python, and then write the images back to an RTSP stream. As the RTSP server, I am using Mediamtx [1]. For streaming, I am using FFmpeg [2].

    &#xA;

    I have the following code that works perfectly fine. For simplification purposes, I am streaming three generated images.

    &#xA;

    import time&#xA;import numpy as np&#xA;import subprocess&#xA;&#xA;width, height = 640, 480&#xA;fps = 25&#xA;rtsp_server_address = f"rtsp://localhost:8554/mystream"&#xA;&#xA;ffmpeg_cmd = [&#xA;    "ffmpeg",&#xA;    "-re",&#xA;    "-f",&#xA;    "rawvideo",&#xA;    "-pix_fmt",&#xA;    "rgb24",&#xA;    "-s",&#xA;    f"{width}x{height}",&#xA;    "-i",&#xA;    "-",&#xA;    "-r",&#xA;    str(fps),&#xA;    "-avoid_negative_ts",&#xA;    "make_zero",&#xA;    "-vcodec",&#xA;    "libx264",&#xA;    "-threads",&#xA;    "4",&#xA;    "-f",&#xA;    "rtsp",&#xA;    rtsp_server_address,&#xA;]&#xA;colors = np.array(&#xA;    [&#xA;        [255, 0, 0],&#xA;        [0, 255, 0],&#xA;        [0, 0, 255],&#xA;    ]&#xA;).reshape(3, 1, 1, 3)&#xA;images = (np.ones((3, width, height, 3)) * colors).astype(np.uint8)&#xA;&#xA;if __name__ == "__main__":&#xA;&#xA;    process = subprocess.Popen(ffmpeg_cmd, stdin=subprocess.PIPE)&#xA;    start = time.time()&#xA;    exported = 0&#xA;    while True:&#xA;        exported &#x2B;= 1&#xA;        next_time = start &#x2B; exported / fps&#xA;        now = time.time()&#xA;        if next_time > now:&#xA;            sleep_dur = next_time - now&#xA;            time.sleep(sleep_dur)&#xA;&#xA;        image = images[exported % 3]&#xA;        image_bytes = image.tobytes()&#xA;&#xA;        process.stdin.write(image_bytes)&#xA;        process.stdin.flush()&#xA;&#xA;    process.stdin.close()&#xA;    process.wait()&#xA;

    &#xA;

    The issue is, that I need to run this at 10 fps because the processing step is heavy and can only afford 10 fps. Hence, as I reduce the frame rate from 25 to 10, the drop rate increases from 0% to 100%. And after a few iterations, I get a BrokenPipeError: [Errno 32] Broken pipe. Refer to the appendix for the complete log.

    &#xA;

    As an alternative, I can use OpenCV compiled from source with GStreamer [3], but I prefer using FFmpeg to make the shipping process simple. Since compiling OpenCV from source can be tedious and dependent on the system.

    &#xA;

    References

    &#xA;

    [1] Mediamtx (formerly rtsp-simple-server) : https://github.com/bluenviron/mediamtx

    &#xA;

    [2] FFmpeg : https://github.com/FFmpeg/FFmpeg

    &#xA;

    [3] Compile OpenCV with GStreamer : https://github.com/bluenviron/mediamtx?tab=readme-ov-file#opencv

    &#xA;

    Appendix

    &#xA;

    Creating the source stream

    &#xA;

    To instantiate the unprocessed stream, I use the following command. This streams the content of my webcam as and RTSP stream.

    &#xA;

    ffmpeg -video_size 1280x720 -i /dev/video0  -avoid_negative_ts make_zero -vcodec libx264 -r 10 -f rtsp rtsp://localhost:8554/webcam&#xA;

    &#xA;

    Error log

    &#xA;

    ffmpeg version 6.1.1 Copyright (c) 2000-2023 the FFmpeg developers&#xA;  built with gcc 12.3.0 (conda-forge gcc 12.3.0-5)&#xA;  configuration: --prefix=/home/conda/feedstock_root/build_artifacts/ffmpeg_1712656518955/_h_env_placehold_placehold_placehold_placehold_placehold_placehold_placehold_placehold_placehold_placehold_placehold_placehold_placehold_placehold_placehold_placehold_placehold_placehold_plac --cc=/home/conda/feedstock_root/build_artifacts/ffmpeg_1712656518955/_build_env/bin/x86_64-conda-linux-gnu-cc --cxx=/home/conda/feedstock_root/build_artifacts/ffmpeg_1712656518955/_build_env/bin/x86_64-conda-linux-gnu-c&#x2B;&#x2B; --nm=/home/conda/feedstock_root/build_artifacts/ffmpeg_1712656518955/_build_env/bin/x86_64-conda-linux-gnu-nm --ar=/home/conda/feedstock_root/build_artifacts/ffmpeg_1712656518955/_build_env/bin/x86_64-conda-linux-gnu-ar --disable-doc --disable-openssl --enable-demuxer=dash --enable-hardcoded-tables --enable-libfreetype --enable-libharfbuzz --enable-libfontconfig --enable-libopenh264 --enable-libdav1d --enable-gnutls --enable-libmp3lame --enable-libvpx --enable-libass --enable-pthreads --enable-vaapi --enable-libopenvino --enable-gpl --enable-libx264 --enable-libx265 --enable-libaom --enable-libsvtav1 --enable-libxml2 --enable-pic --enable-shared --disable-static --enable-version3 --enable-zlib --enable-libopus --pkg-config=/home/conda/feedstock_root/build_artifacts/ffmpeg_1712656518955/_build_env/bin/pkg-config&#xA;  libavutil      58. 29.100 / 58. 29.100&#xA;  libavcodec     60. 31.102 / 60. 31.102&#xA;  libavformat    60. 16.100 / 60. 16.100&#xA;  libavdevice    60.  3.100 / 60.  3.100&#xA;  libavfilter     9. 12.100 /  9. 12.100&#xA;  libswscale      7.  5.100 /  7.  5.100&#xA;  libswresample   4. 12.100 /  4. 12.100&#xA;  libpostproc    57.  3.100 / 57.  3.100&#xA;Input #0, rawvideo, from &#x27;fd:&#x27;:&#xA;  Duration: N/A, start: 0.000000, bitrate: 184320 kb/s&#xA;  Stream #0:0: Video: rawvideo (RGB[24] / 0x18424752), rgb24, 640x480, 184320 kb/s, 25 tbr, 25 tbn&#xA;Stream mapping:&#xA;  Stream #0:0 -> #0:0 (rawvideo (native) -> h264 (libx264))&#xA;[libx264 @ 0x5e2ef8b01340] using cpu capabilities: MMX2 SSE2Fast SSSE3 SSE4.2 AVX FMA3 BMI2 AVX2&#xA;[libx264 @ 0x5e2ef8b01340] profile High 4:4:4 Predictive, level 2.2, 4:4:4, 8-bit&#xA;[libx264 @ 0x5e2ef8b01340] 264 - core 164 r3095 baee400 - H.264/MPEG-4 AVC codec - Copyleft 2003-2022 - http://www.videolan.org/x264.html - options: cabac=1 ref=3 deblock=1:0:0 analyse=0x3:0x113 me=hex subme=7 psy=1 psy_rd=1.00:0.00 mixed_ref=1 me_range=16 chroma_me=1 trellis=1 8x8dct=1 cqm=0 deadzone=21,11 fast_pskip=1 chroma_qp_offset=4 threads=4 lookahead_threads=1 sliced_threads=0 nr=0 decimate=1 interlaced=0 bluray_compat=0 constrained_intra=0 bframes=3 b_pyramid=2 b_adapt=1 b_bias=0 direct=1 weightb=1 open_gop=0 weightp=2 keyint=250 keyint_min=10 scenecut=40 intra_refresh=0 rc_lookahead=40 rc=crf mbtree=1 crf=23.0 qcomp=0.60 qpmin=0 qpmax=69 qpstep=4 ip_ratio=1.40 aq=1:1.00&#xA;Output #0, rtsp, to &#x27;rtsp://localhost:8554/mystream&#x27;:&#xA;  Metadata:&#xA;    encoder         : Lavf60.16.100&#xA;  Stream #0:0: Video: h264, yuv444p(tv, progressive), 640x480, q=2-31, 10 fps, 90k tbn&#xA;    Metadata:&#xA;      encoder         : Lavc60.31.102 libx264&#xA;    Side data:&#xA;      cpb: bitrate max/min/avg: 0/0/0 buffer size: 0 vbv_delay: N/A&#xA;[vost#0:0/libx264 @ 0x5e2ef8b01080] Error submitting a packet to the muxer: Broken pipe   &#xA;[out#0/rtsp @ 0x5e2ef8afd780] Error muxing a packet&#xA;[out#0/rtsp @ 0x5e2ef8afd780] video:1kB audio:0kB subtitle:0kB other streams:0kB global headers:0kB muxing overhead: unknown&#xA;frame=    1 fps=0.1 q=-1.0 Lsize=N/A time=00:00:04.70 bitrate=N/A dup=0 drop=70 speed=0.389x    &#xA;[libx264 @ 0x5e2ef8b01340] frame I:16    Avg QP: 6.00  size:   147&#xA;[libx264 @ 0x5e2ef8b01340] frame P:17    Avg QP: 9.94  size:   101&#xA;[libx264 @ 0x5e2ef8b01340] frame B:17    Avg QP: 9.94  size:    64&#xA;[libx264 @ 0x5e2ef8b01340] consecutive B-frames: 50.0%  0.0% 42.0%  8.0%&#xA;[libx264 @ 0x5e2ef8b01340] mb I  I16..4: 81.3% 18.7%  0.0%&#xA;[libx264 @ 0x5e2ef8b01340] mb P  I16..4: 52.9%  0.0%  0.0%  P16..4:  0.0%  0.0%  0.0%  0.0%  0.0%    skip:47.1%&#xA;[libx264 @ 0x5e2ef8b01340] mb B  I16..4:  0.0%  5.9%  0.0%  B16..8:  0.1%  0.0%  0.0%  direct: 0.0%  skip:94.0%  L0:56.2% L1:43.8% BI: 0.0%&#xA;[libx264 @ 0x5e2ef8b01340] 8x8 transform intra:15.4% inter:100.0%&#xA;[libx264 @ 0x5e2ef8b01340] coded y,u,v intra: 0.0% 0.0% 0.0% inter: 0.0% 0.0% 0.0%&#xA;[libx264 @ 0x5e2ef8b01340] i16 v,h,dc,p: 97%  0%  3%  0%&#xA;[libx264 @ 0x5e2ef8b01340] i8 v,h,dc,ddl,ddr,vr,hd,vl,hu:  0%  0% 100%  0%  0%  0%  0%  0%  0%&#xA;[libx264 @ 0x5e2ef8b01340] Weighted P-Frames: Y:52.9% UV:52.9%&#xA;[libx264 @ 0x5e2ef8b01340] ref P L0: 88.9%  0.0%  0.0% 11.1%&#xA;[libx264 @ 0x5e2ef8b01340] kb/s:8.27&#xA;Conversion failed!&#xA;Traceback (most recent call last):&#xA;  File "/home/avishka/projects/read-process-stream/minimal-ffmpeg-error.py", line 58, in <module>&#xA;    process.stdin.write(image_bytes)&#xA;BrokenPipeError: [Errno 32] Broken pipe&#xA;</module>

    &#xA;