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  • MediaSPIP 0.1 Beta version

    25 avril 2011, par

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    18 février 2011, par

    Multilang est un plugin supplémentaire qui n’est pas activé par défaut lors de l’initialisation de MediaSPIP.
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    5 mars 2010, par

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Sur d’autres sites (8850)

  • File conversion to mp3 returning failure everytime using flutter package ffmpeg_kit_flutter

    6 novembre 2024, par Sanath balthar

    I am trying to convert a .wav audio file generated from a flutter's text to speech package - "flutter_tts" to mp3 file but it is failing everytime.
I have written the below code for file conversion. I have imported the package ffmpeg_kit_flutter. It doesnt even show why the conversion is failing.
I have looked up in stackoverflow and other sites but could not find any relevant solutions. I am using vscode as editor. I have attached flutter doctor output below as well. Could anyone please guide me ? Let me know if you need more information.

    


    List<string> command = [&#xA;              &#x27;-i&#x27;, &#x27;$filePath/998tts.wav&#x27;,&#xA;              &#x27;-c:a&#x27;, &#x27;mp3&#x27;,&#xA;              &#x27;$filePath/998.mp3&#x27;&#xA;            ];&#xA;&#xA; await FFmpegKitConfig.enableLogs();&#xA;            FFmpegKitConfig.enableLogCallback((log) =>print(&#x27;FFmpeg log: $log&#x27;));        &#xA;          FFmpegSession result = await FFmpegKit.executeWithArguments(command);&#xA;          dynamic resultcode = await result.getReturnCode();&#xA;          dynamic resultlogs = await result.getLogsAsString();&#xA;          // FFmpegKitConfig.setLogLevel(logLevel)&#xA;          if(ReturnCode.isSuccess(resultcode)){&#xA;          print("file saved after conversion at $filePath/998.mp3 and result : Success and logs : $resultlogs");&#xA;          }&#xA;          else{&#xA;            print("Result : failure and logs : $resultlogs");&#xA;          }&#xA;&#xA;Flutter doctor output:&#xA;[√] Flutter (Channel stable, 3.19.6, on Microsoft Windows [Version 10.0.22631.3296], locale en-IN)&#xA;[√] Windows Version (Installed version of Windows is version 10 or higher)&#xA;[√] Android toolchain - develop for Android devices (Android SDK version 34.0.0)&#xA;[√] Chrome - develop for the web&#xA;[!] Visual Studio - develop Windows apps (Visual Studio Community 2022 17.9.5)&#xA;X Visual Studio is missing necessary components. Please re-run the Visual Studio installer for the "Desktop development with C&#x2B;&#x2B;"&#xA;workload, and include these components:&#xA;MSVC v142 - VS 2019 C&#x2B;&#x2B; x64/x86 build tools&#xA;- If there are multiple build tool versions available, install the latest&#xA;C&#x2B;&#x2B; CMake tools for Windows&#xA;Windows 10 SDK&#xA;[√] Android Studio (version 2023.2)&#xA;[√] VS Code (version 1.89.0)&#xA;[√] Connected device (3 available)&#xA;[√] Network resources&#xA;&#xA;! Doctor found issues in 1 category.&#xA;&#xA;</string>

    &#xA;

    Edit : Attaching error logs :

    &#xA;

    I/flutter (25865): Loading ffmpeg-kit-flutter.&#xA;D/ffmpeg-kit-flutter(25865): FFmpegKitFlutterPlugin com.arthenica.ffmpegkit.flutter.FFmpegKitFlutterPlugin@a5d9788 started listening to events on io.flutter.plugin.common.EventChannel$IncomingStreamRequestHandler$EventSinkImplementation@4cfb5f2.&#xA;I/flutter (25865): Loaded ffmpeg-kit-flutter-android-audio-arm64-v8a-6.0.3.&#xA;I/flutter (25865): Result : failure and logs : ffmpeg version n6.0 Copyright (c) 2000-2023 the FFmpeg developers&#xA;I/flutter (25865):   built with Android (7155654, based on r399163b1) clang version 11.0.5 (https://android.googlesource.com/toolchain/llvm-project 87f1315dfbea7c137aa2e6d362dbb457e388158d)&#xA;&#xA;I/flutter (25865):   configuration: --cross-prefix=aarch64-linux-android- --sysroot=/Users/sue/Library/Android/sdk/ndk/22.1.7171670/toolchains/llvm/prebuilt/darwin-x86_64/sysroot --prefix=/Users/sue/Projects/arthenica/ffmpeg-kit/prebuilt/android-arm64/ffmpeg --pkg-config=/opt/homebrew/bin/pkg-config --enable-version3 --arch=aarch64 --cpu=armv8-a --target-os=android --enable-neon --enable-asm --enable-inline-asm --ar=aarch64-linux-android-ar --cc=aarch64-linux-android24-clang --cxx=aarch64-linux-android24-clang&#x2B;&#x2B; --ranlib=aarch64-linux-android-ranlib --strip=aarch64-linux-android-strip --nm=aarch64-linux-android-nm --extra-libs=&#x27;-L/Users/sue/Projects/arthenica/ffmpeg-kit/prebuilt/android-arm64/cpu-features/lib -lndk_compat&#x27; --disable-autodetect --enable-cross-compile &#xA;

    &#xA;

  • How can I increase the performance of this video watermarking call which seems to be only using 6% of my system's CPU ? Currently using ffmpeg

    24 août 2023, par Codemonkey

    I have a php script to add a watermark to an uploaded video, using ffmpeg

    &#xA;

    shell_exec("ffmpeg -y -i input.mp4 -i watermark.png -filter_complex $filter output.mp4")&#xA;

    &#xA;

    The video is 36 seconds long, 33MB and 1080x1920, 24fps. The output video comes in at 8MB, scaled down to 540x960 with a PNG overlaid as a watermark.

    &#xA;

    The script takes about 3 seconds, averaging 320fps or so. For a one-off this would be ok, but I'm going to be processing hundreds/thousands of these at a time.

    &#xA;

    I have 128GB of ram and an EPYC 7502p (32cores/64threads), and both input and output files are on NVMe drives.

    &#xA;

    If I monitor top showing all my cores/threads in another window, with refresh delay set to a variety of things from 0.1 to 2 seconds, I'd guesstimate that this hits about 1/4 of my available threads, at about 25% utilisation each.

    &#xA;

    So it kind of feels like I COULD see a 16x speed increase if it hit all of them at 100%. That actually wouldn't be ideal, locking up the server of course, but when my CPU is sat at 99% idle 99% of the time, I feel it could do better - it'd be nice to see it hitting at least half the cores or utilising at least 50% of them when it does...

    &#xA;

    So, is there anything I can change with ffmpeg's config/build or the way I call it ? If this is as good as ffmpeg can do, are there better tools for the job ?

    &#xA;

    Thank you

    &#xA;


    &#xA;

    EDIT :

    &#xA;

    A few things I've learnt since posting :

    &#xA;

      &#xA;
    • -c:v libx264 -preset ultrafast - marginal help
    • &#xA;

    • nice -n -10 - marginal help (changing to -20 is makes negligible difference)
    • &#xA;

    • -c:a copy - significant help, which I found surprising
    • &#xA;

    &#xA;

    FPS up from 320 to 540 using those 3 combined. I still feel there should be significantly more gains somewhere though...

    &#xA;


    &#xA;

    Here's the output from the original script, if it helps :

    &#xA;

    ffmpeg version 4.2.9 Copyright (c) 2000-2023 the FFmpeg developers&#xA;  built with gcc 8 (GCC)&#xA;  configuration: --prefix=/usr --bindir=/usr/bin --datadir=/usr/share/ffmpeg --docdir=/usr/share/doc/ffmpeg --incdir=/usr/include/ffmpeg --libdir=/usr/lib64 --mandir=/usr/share/man --arch=x86_64 --optflags=&#x27;-O2 -g -pipe -Wall -Werror=format-security -Wp,-D_FORTIFY_SOURCE=2 -Wp,-D_GLIBCXX_ASSERTIONS -fexceptions -fstack-protector-strong -grecord-gcc-switches -specs=/usr/lib/rpm/redhat/redhat-hardened-cc1 -specs=/usr/lib/rpm/redhat/redhat-annobin-cc1 -m64 -mtune=generic -fasynchronous-unwind-tables -fstack-clash-protection -fcf-protection&#x27; --extra-ldflags=&#x27;-Wl,-z,relro -Wl,-z,now -specs=/usr/lib/rpm/redhat/redhat-hardened-ld &#x27; --extra-cflags=&#x27; &#x27; --enable-libopencore-amrnb --enable-libopencore-amrwb --enable-libvo-amrwbenc --enable-version3 --enable-bzlib --disable-crystalhd --enable-fontconfig --enable-frei0r --enable-gcrypt --enable-gnutls --enable-ladspa --enable-libaom --enable-libdav1d --enable-libass --enable-libbluray --enable-libcdio --enable-libdrm --enable-libjack --enable-libfreetype --enable-libfribidi --enable-libgsm --enable-libmp3lame --enable-nvenc --enable-openal --enable-opencl --enable-opengl --enable-libopenjpeg --enable-libopus --enable-libpulse --enable-librsvg --enable-libsrt --enable-libsoxr --enable-libspeex --enable-libssh --enable-libtheora --enable-libvorbis --enable-libv4l2 --enable-libvidstab --enable-libvmaf --enable-version3 --enable-vapoursynth --enable-libvpx --enable-libx264 --enable-libx265 --enable-libxvid --enable-libzimg --enable-libzvbi --enable-avfilter --enable-avresample --enable-libmodplug --enable-postproc --enable-pthreads --disable-static --enable-shared --enable-gpl --disable-debug --disable-stripping --shlibdir=/usr/lib64 --enable-libmfx --enable-runtime-cpudetect&#xA;  libavutil      56. 31.100 / 56. 31.100&#xA;  libavcodec     58. 54.100 / 58. 54.100&#xA;  libavformat    58. 29.100 / 58. 29.100&#xA;  libavdevice    58.  8.100 / 58.  8.100&#xA;  libavfilter     7. 57.100 /  7. 57.100&#xA;  libavresample   4.  0.  0 /  4.  0.  0&#xA;  libswscale      5.  5.100 /  5.  5.100&#xA;  libswresample   3.  5.100 /  3.  5.100&#xA;  libpostproc    55.  5.100 / 55.  5.100&#xA;Input #0, mov,mp4,m4a,3gp,3g2,mj2, from &#x27;input.mp4&#x27;:&#xA;  Metadata:&#xA;    major_brand     : M4V&#xA;    minor_version   : 1&#xA;    compatible_brands: isomavc1mp42&#xA;    creation_time   : 2023-07-03T14:55:37.000000Z&#xA;  Duration: 00:00:36.50, start: 0.000000, bitrate: 7315 kb/s&#xA;    Stream #0:0(und): Video: h264 (High) (avc1 / 0x31637661), yuv420p(tv, bt709/bt709/unknown), 1080x1920 [SAR 1:1 DAR 9:16], 7213 kb/s, 24 fps, 24 tbr, 24k tbn, 48 tbc (default)&#xA;    Metadata:&#xA;      creation_time   : 2023-07-03T14:55:37.000000Z&#xA;      handler_name    : ETI ISO Video Media Handler&#xA;      encoder         : Elemental H.264&#xA;    Stream #0:1(und): Audio: aac (LC) (mp4a / 0x6134706D), 48000 Hz, stereo, fltp, 96 kb/s (default)&#xA;    Metadata:&#xA;      creation_time   : 2023-07-03T14:55:37.000000Z&#xA;      handler_name    : ETI ISO Audio Media Handler&#xA;Input #1, png_pipe, from &#x27;watermark.png&#x27;:&#xA;  Duration: N/A, bitrate: N/A&#xA;    Stream #1:0: Video: png, rgba(pc), 1232x804 [SAR 11811:11811 DAR 308:201], 25 tbr, 25 tbn, 25 tbc&#xA;Stream mapping:&#xA;  Stream #0:0 (h264) -> scale (graph 0)&#xA;  Stream #1:0 (png) -> scale (graph 0)&#xA;  overlay (graph 0) -> Stream #0:0 (libx264)&#xA;  Stream #0:1 -> #0:1 (aac (native) -> aac (native))&#xA;Press [q] to stop, [?] for help&#xA;[libx264 @ 0x56510d488e00] using SAR=1/1&#xA;[libx264 @ 0x56510d488e00] using cpu capabilities: MMX2 SSE2Fast SSSE3 SSE4.2 AVX FMA3 BMI2 AVX2&#xA;[libx264 @ 0x56510d488e00] profile High, level 3.1, 4:2:0, 8-bit&#xA;[libx264 @ 0x56510d488e00] 264 - core 157 r2980 34c06d1 - H.264/MPEG-4 AVC codec - Copyleft 2003-2019 - 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=-2 threads=30 lookahead_threads=5 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=24 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, mp4, to &#x27;output.mp4&#x27;:&#xA;  Metadata:&#xA;    major_brand     : M4V&#xA;    minor_version   : 1&#xA;    compatible_brands: isomavc1mp42&#xA;    encoder         : Lavf58.29.100&#xA;    Stream #0:0: Video: h264 (libx264) (avc1 / 0x31637661), yuv420p, 540x960 [SAR 1:1 DAR 9:16], q=-1--1, 24 fps, 12288 tbn, 24 tbc (default)&#xA;    Metadata:&#xA;      encoder         : Lavc58.54.100 libx264&#xA;    Side data:&#xA;      cpb: bitrate max/min/avg: 0/0/0 buffer size: 0 vbv_delay: -1&#xA;    Stream #0:1(und): Audio: aac (LC) (mp4a / 0x6134706D), 48000 Hz, stereo, fltp, 128 kb/s (default)&#xA;    Metadata:&#xA;      creation_time   : 2023-07-03T14:55:37.000000Z&#xA;      handler_name    : ETI ISO Audio Media Handler&#xA;      encoder         : Lavc58.54.100 aac&#xA;frame=  876 fps=324 q=-1.0 Lsize=    7768kB time=00:00:36.45 bitrate=1745.5kbits/s speed=13.5x&#xA;video:7177kB audio:566kB subtitle:0kB other streams:0kB global headers:0kB muxing overhead: 0.336181%&#xA;[libx264 @ 0x56510d488e00] frame I:9     Avg QP:21.46  size: 37765&#xA;[libx264 @ 0x56510d488e00] frame P:237   Avg QP:23.78  size: 17862&#xA;[libx264 @ 0x56510d488e00] frame B:630   Avg QP:28.03  size:  4405&#xA;[libx264 @ 0x56510d488e00] consecutive B-frames:  2.9%  2.5%  3.8% 90.9%&#xA;[libx264 @ 0x56510d488e00] mb I  I16..4: 13.6% 65.9% 20.5%&#xA;[libx264 @ 0x56510d488e00] mb P  I16..4:  5.2% 16.6%  4.1%  P16..4: 32.8% 18.0% 12.5%  0.0%  0.0%    skip:10.8%&#xA;[libx264 @ 0x56510d488e00] mb B  I16..4:  1.1%  2.7%  0.2%  B16..8: 35.3%  7.2%  2.5%  direct: 3.3%  skip:47.7%  L0:37.0% L1:52.2% BI:10.8%&#xA;[libx264 @ 0x56510d488e00] 8x8 transform intra:65.0% inter:71.5%&#xA;[libx264 @ 0x56510d488e00] coded y,uvDC,uvAC intra: 54.7% 62.1% 11.5% inter: 17.9% 12.9% 0.5%&#xA;[libx264 @ 0x56510d488e00] i16 v,h,dc,p: 16% 28%  8% 47%&#xA;[libx264 @ 0x56510d488e00] i8 v,h,dc,ddl,ddr,vr,hd,vl,hu: 21% 23% 18%  6%  6%  7%  7%  7%  7%&#xA;[libx264 @ 0x56510d488e00] i4 v,h,dc,ddl,ddr,vr,hd,vl,hu: 23% 16% 11%  7%  9% 10%  8%  8%  8%&#xA;[libx264 @ 0x56510d488e00] i8c dc,h,v,p: 55% 20% 20%  5%&#xA;[libx264 @ 0x56510d488e00] Weighted P-Frames: Y:5.5% UV:3.4%&#xA;[libx264 @ 0x56510d488e00] ref P L0: 65.0% 18.8% 13.0%  3.0%  0.2%&#xA;[libx264 @ 0x56510d488e00] ref B L0: 95.0%  4.0%  1.1%&#xA;[libx264 @ 0x56510d488e00] ref B L1: 98.2%  1.8%&#xA;[libx264 @ 0x56510d488e00] kb/s:1610.57&#xA;[aac @ 0x56510d4865c0] Qavg: 22331.562&#xA;

    &#xA;

  • python subprocess ffmpeg return code = 69

    13 juin 2023, par Tim Chen

    I try to call ffmpeg through the subprocess.run([&#x27;ffmpeg&#x27;, &#x27;-i&#x27;, file_name, output_file_name], capture_output=True, text=True) command in python to convert the audio file incoming from the front end to wav format file. The backend code is as follows, using python+fastapi :

    &#xA;

    @app.post("/api/upload/convert")&#xA;async def convert_upload_file(request: Request, file: UploadFile = File(...)):&#xA;    token = uuid.uuid4().hex&#xA;    tmpFileName = os.path.join(os.path.dirname(__file__), token)&#xA;    with open(tmpFileName, "wb") as buffer:&#xA;        buffer.write(await file.read())&#xA;    await file.seek(0)&#xA;    output_path = tmpFileName &#x2B; &#x27;-output.wav&#x27;&#xA;    command = [&#x27;ffmpeg&#x27;, &#x27;-i&#x27;, tmpFileName, output_path]&#xA;    result = subprocess.run(command, capture_output=True, text=True)&#xA;

    &#xA;

    This code usually works, but there are some scenarios where it doesn't work. The audio file is recorded by js code (specifically navigator.mediaDevices.getUserMedia({audio: true})).&#xA;The code of the audio recorded in windows chrome can run normally and get the converted wav file, but the audio recorded from ios15 safari for more than 3 seconds cannot be converted, prompting returncode=69. The error message is as follows :

    &#xA;

    CompletedProcess(args=[&#x27;ffmpeg&#x27;, &#x27;-i&#x27;, &#x27;5cfb52c503a646bda0f422b517c8014a&#x27;, &#x27;5cfb52c503a646bda0f422b517c8014a-output.wav&#x27;], returncode=69, stdout=&#x27;&#x27;, stderr="&#xA;ffmpeg version 4.4.2-0ubuntu0.22.04.1 Copyright (c) 2000-2021 the FFmpeg developers&#xA;built with gcc 11 (Ubuntu 11.2.0-19ubuntu1)&#xA;configuration: --prefix=/usr --extra-version=0ubuntu0.22.04.1 --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-libgme --enable-libgsm --enable-libjack --enable-libmp3lame --enable-libmysofa --enable-libopenjpeg --enable-libopenmpt --enable-libopus --enable-libpulse --enable-librabbitmq --enable-librubberband --enable-libshine --enable-libsnappy --enable-libsoxr --enable-libspeex --enable-libsrt --enable-libssh --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 --enable-pocketsphinx --enable-librsvg --enable-libmfx --enable-libdc1394 --enable-libdrm --enable-libiec61883 --enable-chromaprint --enable-frei0r --enable-libx264 --enable-shared&#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;libswscale      5.  9.100 /  5.  9.100&#xA;libswresample   3.  9.100 /  3.  9.100&#xA;libpostproc    55.  9.100 / 55.  9.100&#xA;Input #0, mov,mp4,m4a,3gp,3g2,mj2, from &#x27;5cfb52c503a646bda0f422b517c8014a&#x27;:&#xA;  Metadata:&#xA;    major_brand     : iso5&#xA;    minor_version   : 1&#xA;    compatible_brands: isomiso5hlsf&#xA;    creation_time   : 2023-06-11T16:36:53.000000Z&#xA;  Duration: 00:00:07.06, start: 0.000000, bitrate: 187 kb/s&#xA;  Stream #0:0(und): Audio: aac (LC) (mp4a / 0x6134706D), 48000 Hz, mono, fltp, 184 kb/s (default)&#xA;    Metadata:&#xA;      creation_time   : 2023-06-11T16:36:53.000000Z&#xA;      handler_name    : Core Media Audio&#xA;      vendor_id       : [0][0][0][0]&#xA;Stream mapping:&#xA;  Stream #0:0 -> #0:0 (aac (native) -> pcm_s16le (native))&#xA;Press [q] to stop, [?] for help&#xA;Output #0, wav, to &#x27;5cfb52c503a646bda0f422b517c8014a-output.wav&#x27;:&#xA;  Metadata:&#xA;    major_brand     : iso5&#xA;    minor_version   : 1&#xA;    compatible_brands: isomiso5hlsf&#xA;    ISFT            : Lavf58.76.100&#xA;  Stream #0:0(und): Audio: pcm_s16le ([1][0][0][0] / 0x0001), 48000 Hz, mono, s16, 768 kb/s (default)&#xA;    Metadata:&#xA;      creation_time   : 2023-06-11T16:36:53.000000Z&#xA;      handler_name    : Core Media Audio&#xA;      vendor_id       : [0][0][0][0]&#xA;      encoder         : Lavc58.134.100 pcm_s16le&#xA;size=       2kB time=00:00:00.00 bitrate=N/A speed=N/A    &#xA;[aac @ 0x55f1f8f19fc0] Sample rate index in program config element does not match the sample rate index configured by the container.&#xA;[aac @ 0x55f1f8f19fc0] Too large remapped id is not implemented. Update your FFmpeg version to the newest one from Git. If the problem still occurs, it means that your file has a feature which has not been implemented.&#xA;[aac @ 0x55f1f8f19fc0] If you want to help, upload a sample of this file to https://streams.videolan.org/upload/ and contact the ffmpeg-devel mailing list. (ffmpeg-devel@ffmpeg.org)&#xA;Error while decoding stream #0:0: Not yet implemented in FFmpeg, patches welcome&#xA;[aac @ 0x55f1f8f19fc0] Multiple frames in a packet.&#xA;[aac @ 0x55f1f8f19fc0] Reserved bit set.&#xA;[aac @ 0x55f1f8f19fc0] Number of bands (18) exceeds limit (13).&#xA;Error while decoding stream #0:0: Invalid data found when processing input&#xA;[aac @ 0x55f1f8f19fc0] Reserved bit set.&#xA;[aac @ 0x55f1f8f19fc0] Prediction is not allowed in AAC-LC.&#xA;Error while decoding stream #0:0: Invalid data found when processing input&#xA;[aac @ 0x55f1f8f19fc0] Reserved bit set.&#xA;

    &#xA;

    For the abnormal code, I tried to execute ffmpeg -i input output.wav after fastapi handle request on the command line and subprocess.run([&#x27;ffmpeg&#x27;, &#x27;-i&#x27;, file_name, output_path], capture_output =True, text=True), all succeeded, which means that the final file must be normal, otherwise the subsequent verification work will get the same error.

    &#xA;

    This confuses me, is there some information I'm missing ?

    &#xA;