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  • Inappropriate ioctl for device in zoneminder - zmc_dvideo0

    9 juin 2024, par Adam

    I recently needed to rebuild my zoneminder local camera system, running on an RPi. I switched OS from Debian to Ubuntu and now I'm struggling to get video4linux to decode the video input. The result is either a black or green screen depending on the compression palette I select.

    


    I'm using /dev/video0 as my input.

    


    v4l2-ctl --list-devices

unicam (platform:fe801000.csi):
        /dev/video0
        /dev/media3


    


    In the logs I'm getting :

    


    zmc_dvideo0 10907   ERR Unable to query control: Inappropriate ioctl for device 


    


    I don't understand the function of the /dev/media3 device, presumably created by the bcm2835-v4l2 module. This site suggests that /dev/video0 needs to extract the ioctl from or link in some way to /dev/media3. Perhaps a red herring.

    


    If I select /dev/media3 as the source, I get :

    


    : FAT [zmc_dmedia3] [Failed to query video device: Inappropriate ioctl for device]
: ERR ['zmc -d /dev/media3' exited abnormally, exit status 255]


    


    I'm using

    


      

    • PAL60 encoding
    • 


    • YUV422p compression
    • 


    • 1280 x 720 Capture Resolution
    • 


    


    # lsb_release -a
No LSB modules are available.
Distributor ID: Ubuntu
Description:    Ubuntu 24.04 LTS
Release:        24.04
Codename:       noble

# uname -a
Linux ubuntu 6.8.0-1005-raspi #5-Ubuntu SMP PREEMPT_DYNAMIC Wed May 22 15:01:35 UTC 2024 aarch64 aarch64 aarch64 GNU/Linux


    


    Using ffmpeg from the cli, I get a similar output, resulting in a green display. This happens both with and without the hardware acceleration flag.

    


    sudo -u www-data ffmpeg -f v4l2 -pix_fmt yuyv422 -framerate 30 -video_size 1280x720 -i /dev/video0 -codec:v h264_v4l2m2m video0_out.mp4


    


    I've set debug to 6 in the ZM options - Any clues on troublshooting steps I can take to further understand why the camera stream is not being shown and how to fix it ?

    


  • Amplification of recorded audio in flutter app using FFMPEG not working correctly

    20 mai 2024, par Noman khanbhai

    In my app I need to record audio and send it to server, server then sends the file to a hardware using mqtt and then file gets played on the hardware. I am using flutter to build app and using record 5.0.5 package for audio recording and for amplification ffmpeg_kit_flutter 6.0.3 package to do the amplification.

    


    The issue is it doesnt seems like there is much change in amplitude, I used different values for amplification factor but audio remains same.

    


    Here is the code for amplification

    


    Future<string>? amplifyAudio(&#xA;      String inputPath, String outputPath) async {&#xA;&#xA;    // Build FFmpeg command to amplify audio&#xA;    outputPath = await modifyOutputPath(inputPath)!;&#xA;    String audioFilter = &#x27;volume=${amplificationFactor}dB&#x27;; &#xA;    //-c:a aac&#xA;    String command = &#x27;-i $inputPath -af $audioFilter $outputPath&#x27;;&#xA;&#xA;    // Execute FFmpeg command&#xA;    await FFmpegKit.executeAsync(command).then((session) async {&#xA;      debugPrint("After executeAsync session ${session.toString()}");&#xA;      debugPrint(&#xA;          "After executeAsync returncode ${await session.getReturnCode()}");&#xA;      debugPrint("After executeAsync command ${session.getCommand()}");&#xA;      log("After executeAsync alllogs ${await session.getAllLogs()}");&#xA;      log("After executeAsync alllogstring ${await session.getAllLogsAsString()}");&#xA;      log("After executeAsync failStackTrace ${await session.getFailStackTrace()}");&#xA;    }).onError((error, stackTrace) {&#xA;      debugPrint("After executeAsync error ${error.toString()}");&#xA;    });&#xA;&#xA;    return outputPath;&#xA;  }&#xA;&#xA;</string>

    &#xA;

    This are the logs when above method gets executed.

    &#xA;

    FFMpeg command -> `-i /data/user/0/com.orgname.flutter.appname/app_flutter/1716209206469.aac -af volume=10.0dB /storage/emulated/0/Download/1716209213238_amplified.aac`&#xA;&#xA;> Logs&#xA;> After executeAsync alllogstring ffmpeg version n6.0 Copyright (c) 2000-2023 the FFmpeg developers&#xA;> built with Android (7155654, based on r399163b1) clang version 11.0.5 (https://android.googlesource.com/toolchain/llvm-project 87f1315dfbea7c137aa2e6d362dbb457e388158d)&#xA;> 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 --enable-pic --enable-jni --enable-optimizations --enable-swscale --disable-static --enable-shared --enable-pthreads --enable-v4l2-m2m --disable-outdev=fbdev --disable-indev=fbdev --enable-small --disable-xmm-clobber-test --disable-debug --enable-lto --disable-neon-clobber-test --disable-programs --disable-postproc --disable-doc --disable-htmlpages --disable-manpages --disable-podpages --disable-txtpages --disable-sndio --disable-schannel --disable-securetransport --disable-xlib --disable-cuda --disable-cuvid --disable-nvenc --disable-vaapi --disable-vdpau --disable-videotoolbox --disable-audiotoolbox --disable-appkit --disable-alsa --disable-cuda --disable-cuvid --disable-nvenc --disable-vaapi --disable-vdpau --enable-gmp --enable-gnutls --enable-iconv --disable-sdl2 --disable-openssl --enable-zlib --enable-mediacodec&#xA;> libavutil      58.  2.100 / 58.  2.100&#xA;> libavcodec     60.  3.100 / 60.  3.100&#xA;> libavformat    60.  3.100 / 60.  3.100&#xA;> libavdevice    60.  1.100 / 60.  1.100&#xA;> libavfilter     9.  3.100 /  9.  3.100&#xA;> libswscale      7.  1.100 /  7.  1.100&#xA;> libswresample   4. 10.100 /  4. 10.100&#xA;> Input #0, mov,mp4,m4a,3gp,3g2,mj2, from &#x27;/data/user/0/com.orgname.flutter.appname/app_flutter/1716209206469.aac&#x27;:&#xA;> Metadata:&#xA;> major_brand     : mp42&#xA;> minor_version   : 0&#xA;> compatible_brands: isommp42&#xA;> creation_time   : 2024-05-20T12:46:52.000000Z&#xA;> com.android.version: 12&#xA;> Duration: 00:00:04.76, start: 0.000000, bitrate: 131 kb/s&#xA;> Stream #0:0[0x1](eng): Audio: aac (mp4a / 0x6134706D), 44100 Hz, stereo, fltp, 128 kb/s (default)&#xA;> Metadata:&#xA;> creation_time   : 2024-05-20T12:46:52.000000Z&#xA;> handler_name    : SoundHandle&#xA;> vendor_id       : [0][0][0][0]&#xA;> Stream mapping:&#xA;> Stream #0:0 -> #0:0 (aac (native) -> aac (native))&#xA;> Press [q] to stop, [?] for help&#xA;

    &#xA;

    Note - I am also playing the audio after recording and before amplification in app, and also saving in download. to make sure audio file is correct.

    &#xA;

    Amplified file also gets saved but there is almost no difference.

    &#xA;

    I have also searched/googled/ and also done chatgpt to resolve issue. but nothing worked.

    &#xA;

  • ffprobe newer version detect audio codec incorrectly

    16 janvier, par alancc

    I find a strange problem.

    &#xA;

    I have a test video with h264 video codec and aac audio codec. It is at https://drive.google.com/file/d/1YAyz5cO0kb9r0MgahCpISR4bZ_1_n8PL/view?usp=sharing

    &#xA;

    I build a ffmpeg version by myself, its version is :

    &#xA;

    ffprobe version 7.0.2 Copyright (c) 2007-2024 the FFmpeg developers&#xA;  built with gcc 14.1.0 (Rev3, Built by MSYS2 project)&#xA;  configuration: --enable-shared&#xA;  libavutil      59.  8.100 / 59.  8.100&#xA;  libavcodec     61.  3.100 / 61.  3.100&#xA;  libavformat    61.  1.100 / 61.  1.100&#xA;  libavdevice    61.  1.100 / 61.  1.100&#xA;  libavfilter    10.  1.100 / 10.  1.100&#xA;  libswscale      8.  1.100 /  8.  1.100&#xA;  libswresample   5.  1.100 /  5.  1.100&#xA;

    &#xA;

    I then use ffprobe to get its info :

    &#xA;

    ffprobe -v quiet -print_format ini -show_streams -show_packets test_h264.mp4 > test_h264.ini&#xA;

    &#xA;

    Then I get an ini file which shows the audio codec as MP2 :

    &#xA;

    [streams.stream.0]&#xA;index=0&#xA;codec_name=mp2&#xA;codec_long_name=MP2 (MPEG audio layer 2)&#xA;profile=unknown&#xA;codec_type=audio&#xA;codec_tag_string=mp4a&#xA;codec_tag=0x6134706d&#xA;sample_fmt=fltp&#xA;sample_rate=44100&#xA;channels=2&#xA;channel_layout=stereo&#xA;bits_per_sample=0&#xA;initial_padding=0&#xA;id=0x1&#xA;r_frame_rate=0/0&#xA;avg_frame_rate=0/0&#xA;time_base=1/44100&#xA;start_pts=2788&#xA;start_time=0.063220&#xA;duration_ts=435455&#xA;duration=9.874263&#xA;bit_rate=127706&#xA;max_bit_rate=N/A&#xA;bits_per_raw_sample=N/A&#xA;nb_frames=378&#xA;nb_read_frames=N/A&#xA;nb_read_packets=378&#xA;

    &#xA;

    Another developer he uses his version of ffprobe :

    &#xA;

    ffprobe version 2023-02-22-git-d5cc7acff1-full_build-www.gyan.dev Copyright (c) 2007-2023 the FFmpeg developers  &#xA;

    &#xA;

    Based on the year, my version(2024) should be newer than his(2023), but his version of ffprobe can get the audio codec properly :

    &#xA;

    [streams.stream.1]&#xA;index=1&#xA;codec_name=aac&#xA;codec_long_name=AAC (Advanced Audio Coding)&#xA;profile=LC&#xA;codec_type=audio&#xA;codec_tag_string=mp4a&#xA;codec_tag=0x6134706d&#xA;sample_fmt=fltp&#xA;sample_rate=44100&#xA;channels=2&#xA;channel_layout=stereo&#xA;bits_per_sample=0&#xA;initial_padding=0&#xA;id=0x2&#xA;r_frame_rate=0/0&#xA;avg_frame_rate=0/0&#xA;time_base=1/44100&#xA;start_pts=1764&#xA;start_time=0.040000&#xA;duration_ts=436480&#xA;duration=9.897506&#xA;bit_rate=111733&#xA;max_bit_rate=N/A&#xA;bits_per_raw_sample=N/A&#xA;nb_frames=427&#xA;nb_read_frames=N/A&#xA;nb_read_packets=427&#xA;extradata_size=5&#xA;

    &#xA;

    Why ?

    &#xA;

    I also tried a ffprobe version on ubuntu with the following version :

    &#xA;

    ffprobe version 6.1.1-3ubuntu5 Copyright (c) 2007-2023 the FFmpeg developers&#xA;  built with gcc 13 (Ubuntu 13.2.0-23ubuntu3)&#xA;  configuration: --prefix=/usr --extra-version=3ubuntu5 --toolchain=hardened --libdir=/usr/lib/x86_64-linux-gnu --incdir=/usr/include/x86_64-linux-gnu --arch=amd64 --enable-gpl --disable-stripping --disable-omx --enable-gnutls --enable-libaom --enable-libass --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-libharfbuzz --enable-libmp3lame --enable-libmysofa --enable-libopenjpeg --enable-libopenmpt --enable-libopus --enable-librubberband --enable-libshine --enable-libsnappy --enable-libsoxr --enable-libspeex --enable-libtheora --enable-libtwolame --enable-libvidstab --enable-libvorbis --enable-libvpx --enable-libwebp --enable-libx265 --enable-libxml2 --enable-libxvid --enable-libzimg --enable-openal --enable-opencl --enable-opengl --disable-sndio --enable-libvpl --disable-libmfx --enable-libdc1394 --enable-libdrm --enable-libiec61883 --enable-chromaprint --enable-frei0r --enable-ladspa --enable-libbluray --enable-libjack --enable-libpulse --enable-librabbitmq --enable-librist --enable-libsrt --enable-libssh --enable-libsvtav1 --enable-libx264 --enable-libzmq --enable-libzvbi --enable-lv2 --enable-sdl2 --enable-libplacebo --enable-librav1e --enable-pocketsphinx --enable-librsvg --enable-libjxl --enable-shared&#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;

    &#xA;

    It will detect the audio as aac properly, but with different parameters, for example, bit_rate is 111733(developer) but 110399(ubuntu). But this parameter comes from the same file so should be the same.

    &#xA;

    [streams.stream.1]&#xA;index=1&#xA;codec_name=aac&#xA;codec_long_name=AAC (Advanced Audio Coding)&#xA;profile=LC&#xA;codec_type=audio&#xA;codec_tag_string=mp4a&#xA;codec_tag=0x6134706d&#xA;sample_fmt=fltp&#xA;sample_rate=44100&#xA;channels=2&#xA;channel_layout=stereo&#xA;bits_per_sample=0&#xA;initial_padding=0&#xA;id=0x2&#xA;r_frame_rate=0/0&#xA;avg_frame_rate=0/0&#xA;time_base=1/44100&#xA;start_pts=0&#xA;start_time=0.000000&#xA;duration_ts=441353&#xA;duration=10.008005&#xA;bit_rate=110399&#xA;max_bit_rate=N/A&#xA;bits_per_raw_sample=N/A&#xA;nb_frames=432&#xA;nb_read_frames=N/A&#xA;nb_read_packets=432&#xA;extradata_size=5&#xA;&#xA;

    &#xA;