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  • 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

  • Use, discuss, criticize

    13 avril 2011, par

    Talk to people directly involved in MediaSPIP’s development, or to people around you who could use MediaSPIP to share, enhance or develop their creative projects.
    The bigger the community, the more MediaSPIP’s potential will be explored and the faster the software will evolve.
    A discussion list is available for all exchanges between users.

  • Changer son thème graphique

    22 février 2011, par

    Le thème graphique ne touche pas à la disposition à proprement dite des éléments dans la page. Il ne fait que modifier l’apparence des éléments.
    Le placement peut être modifié effectivement, mais cette modification n’est que visuelle et non pas au niveau de la représentation sémantique de la page.
    Modifier le thème graphique utilisé
    Pour modifier le thème graphique utilisé, il est nécessaire que le plugin zen-garden soit activé sur le site.
    Il suffit ensuite de se rendre dans l’espace de configuration du (...)

Sur d’autres sites (5995)

  • ffprobe newer version detect audio codec incorrectly

    16 janvier, par alancc

    I find a strange problem.

    


    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

    


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

    


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


    


    I then use ffprobe to get its info :

    


    ffprobe -v quiet -print_format ini -show_streams -show_packets test_h264.mp4 > test_h264.ini


    


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

    


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


    


    Another developer he uses his version of ffprobe :

    


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


    


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

    


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


    


    Why ?

    


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

    


    ffprobe version 6.1.1-3ubuntu5 Copyright (c) 2007-2023 the FFmpeg developers
  built with gcc 13 (Ubuntu 13.2.0-23ubuntu3)
  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
  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


    


    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.

    


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



    


  • How do I get drawtext filter in ffmpeg to work on ubuntu 22.04 ?

    3 décembre 2024, par chovy

    Here's my script, but no matter what I try the drawtext filter is not enabled :

    


    #!/bin/bash

set -e  # Exit on any error

# Define installation directories
INSTALL_DIR="$HOME/ffmpeg_build"
BIN_DIR="$HOME/bin"
SOURCE_DIR="$HOME/ffmpeg_sources"
NUM_CORES=$(nproc)

echo "Creating necessary directories..."
mkdir -p "$INSTALL_DIR" "$BIN_DIR" "$SOURCE_DIR"

# Install required tools and dependencies
echo "Installing build tools and essential libraries..."
sudo apt-get update
sudo apt-get install -y \
  autoconf automake build-essential cmake git-core libass-dev \
  libfreetype6-dev libsdl2-dev libtool libva-dev libvdpau-dev \
  libvorbis-dev libxcb1-dev libxcb-shm0-dev libxcb-xfixes0-dev \
  meson ninja-build pkg-config texinfo wget yasm zlib1g-dev \
  nasm libnuma-dev libfdk-aac-dev libmp3lame-dev libopus-dev \
  libfreetype6 libdrm-dev mercurial

# Remove system-installed x264 and x265 to prevent conflicts
sudo apt-get remove -y libx264-dev libx265-dev x264 x265

# Build dependencies
cd "$SOURCE_DIR"

# Install libx264 (static)
if [ ! -d "$SOURCE_DIR/x264" ]; then
  echo "Building and installing libx264..."
  git clone --branch stable --depth 1 https://code.videolan.org/videolan/x264.git
  cd x264
  make distclean || true
  ./configure --prefix="$INSTALL_DIR" --enable-static --disable-opencl
  make -j$NUM_CORES
  make install
  cd "$SOURCE_DIR"
fi

# Install libx265 (static)
if [ ! -d "$SOURCE_DIR/x265" ]; then
  echo "Building and installing libx265..."
  git clone --depth 1 https://github.com/videolan/x265.git
  cd x265/build/linux
  cmake -G "Unix Makefiles" -DCMAKE_INSTALL_PREFIX="$INSTALL_DIR" \
    -DENABLE_SHARED=OFF -DENABLE_PIC=ON -DENABLE_PKGCONFIG=ON ../../source
  make -j$NUM_CORES
  make install
  cd "$SOURCE_DIR"
fi

# Install libvpx (static)
if [ ! -d "$SOURCE_DIR/libvpx" ]; then
  echo "Building and installing libvpx..."
  git clone --depth 1 https://chromium.googlesource.com/webm/libvpx.git
  cd libvpx
  ./configure --prefix="$INSTALL_DIR" --disable-examples --disable-unit-tests \
    --enable-vp9-highbitdepth --as=yasm --enable-static --enable-pic
  make -j$NUM_CORES
  make install
  cd "$SOURCE_DIR"
fi

# Install libopus (static)
if [ ! -d "$SOURCE_DIR/opus" ]; then
  echo "Building and installing libopus..."
  git clone --depth 1 https://github.com/xiph/opus.git
  cd opus
  ./autogen.sh
  ./configure --prefix="$INSTALL_DIR" --disable-shared
  make -j$NUM_CORES
  make install
  cd "$SOURCE_DIR"
fi

# Install libaom (static)
if [ ! -d "$SOURCE_DIR/aom" ]; then
  echo "Building and installing libaom..."
  git clone --depth 1 https://aomedia.googlesource.com/aom
  mkdir -p aom_build
  cd aom_build
  cmake -G "Unix Makefiles" -DCMAKE_INSTALL_PREFIX="$INSTALL_DIR" \
    -DBUILD_SHARED_LIBS=0 -DENABLE_NASM=1 -DCMAKE_C_FLAGS="-fPIC" ../aom
  make -j$NUM_CORES
  make install
  cd "$SOURCE_DIR"
fi

# Build and install FFmpeg
echo "Building and installing FFmpeg..."
cd "$SOURCE_DIR"
if [ ! -d "$SOURCE_DIR/ffmpeg" ]; then
  git clone --depth 1 https://git.ffmpeg.org/ffmpeg.git ffmpeg
  cd ffmpeg
else
  cd ffmpeg
  git pull
fi

export PKG_CONFIG_PATH="$INSTALL_DIR/lib/pkgconfig:/usr/lib/pkgconfig:/usr/share/pkgconfig:/usr/lib/$(uname -m)-linux-gnu/pkgconfig:$PKG_CONFIG_PATH"

make distclean

./configure \
  --prefix="$INSTALL_DIR" \
  --pkg-config-flags="--static" \
  --extra-cflags="-I$INSTALL_DIR/include" \
  --extra-ldflags="-L$INSTALL_DIR/lib" \
  --extra-libs="-lpthread -lm" \
  --bindir="$BIN_DIR" \
  --enable-gpl \
  --enable-nonfree \
  --enable-libfreetype \
  --enable-libx264 \
  --enable-libvpx \
  --enable-libmp3lame \
  --enable-libopus \
  --enable-libass \
  --enable-libvorbis \
  --enable-libaom \
  --enable-libdrm \
  --enable-version3 \
  --enable-static \
  --disable-shared \
  --enable-small
  
make -j$NUM_CORES
make install

# Add ffmpeg to PATH
echo "export PATH=\"$BIN_DIR:\$PATH\"" >> "$HOME/.bashrc"
source "$HOME/.bashrc"

# Final checks
echo "FFmpeg installation complete. Verifying installation..."
ffmpeg -version



    


    Here is my buildconf which appears correct :

    


    $ ffmpeg -buildconf  

ffmpeg version N-117989-gcb27e478f7 Copyright (c) 2000-2024 the FFmpeg developers
  built with gcc 11 (Ubuntu 11.4.0-1ubuntu1~22.04)
  configuration: --prefix=/usr/local --extra-cflags=-I/home/ubuntu/src/ffmpeg_build/include --extra-ldflags=-L/home/ubuntu/src/ffmpeg_build/lib --bindir=/usr/local/bin --enable-gpl --enable-nonfree --enable-libfreetype --enable-libx264 --enable-libvpx --enable-libmp3lame --enable-libopus --enable-libass --enable-libvorbis --enable-libaom --enable-libdrm --enable-version3 --enable-shared --enable-filter=drawtext  libavutil      59. 47.101 / 59. 47.101  libavcodec     61. 26.100 / 61. 26.100  libavformat    61.  9.100 / 61.  9.100  libavdevice    61.  4.100 / 61.  4.100  libavfilter    10.  6.101 / 10.  6.101  libswscale      8. 12.100 /  8. 12.100  libswresample   5.  4.100 /  5.  4.100
  libpostproc    58.  4.100 / 58.  4.100

  configuration:    --prefix=/usr/local
    --extra-cflags=-I/home/ubuntu/src/ffmpeg_build/include
    --extra-ldflags=-L/home/ubuntu/src/ffmpeg_build/lib
    --bindir=/usr/local/bin
    --enable-gpl    --enable-nonfree    --enable-libfreetype    --enable-libx264    --enable-libvpx    --enable-libmp3lame    --enable-libopus    --enable-libass    --enable-libvorbis    --enable-libaom    --enable-libdrm    --enable-version3    --enable-shared    --enable-filter=drawtext


    


    However the filter drawtext is not enabled :

    


    $ ffmpeg -filters | grep drawtext
ffmpeg version N-117989-gcb27e478f7 Copyright (c) 2000-2024 the FFmpeg developers  built with gcc 11 (Ubuntu 11.4.0-1ubuntu1~22.04)  configuration: --prefix=/usr/local --extra-cflags=-I/home/ubuntu/src/ffmpeg_build/include --extra-ldflags=-L/home/ubuntu/src/ffmpeg_build/lib --bindir=/usr/local/bin --enable-gpl --enable-nonfree --enable-libfreetype --enable-libx264 --enable-libvpx --enable-libmp3lame --enable-libopus --enable-libass --enable-libvorbis --enable-libaom --enable-libdrm --enable-version3 --enable-shared --enable-filter=drawtext  libavutil      59. 47.101 / 59. 47.101  libavcodec     61. 26.100 / 61. 26.100  libavformat    61.  9.100 / 61.  9.100  libavdevice    61.  4.100 / 61.  4.100  libavfilter    10.  6.101 / 10.  6.101  libswscale      8. 12.100 /  8. 12.100  libswresample   5.  4.100 /  5.  4.100  libpostproc    58.  4.100 / 58.  4.100


    


  • Converting HLS Stream to stream supported by old radio

    29 novembre 2024, par Alberto Faenza

    I have an old internet radio that does not support HLS streams.
Therefore I cannot listen to my favourite radio at this url :
https://streamcdnf31-4c4b867c89244861ac216426883d1ad0.msvdn.net/radiodeejay/radiodeejay/master_ma.m3u8

    


    I found a solution using a paid software https://minimradio.com/ which is based on minimserver and minimstreamer.

    


    This solution works if I install mininmserver and minimstreamer on a local computer and use the internet radio to point to the converter stream but I will have to pay if I want to use this.

    


    Checking the documentation of minimradio and ministreamer I can see the following :

    


    *Some internet radios can play the previous AAC ADTS streams but can't play these new HLS streams
...

    


    If the network stream URL points to an HLS .m3u8 master playlist or media playlist file, MinimStreamer reads this file and uses the HLS protocol to read the stream audio data and send it to the music player as a conventional HTTP stream. This makes the stream playable on music players that don't support the HLS protocol. The audio data in the stream must be encoded in AAC format.*
and not a single destination receiver I should use a streaming (broadcasting) server. What can I use to do that ?

    


    My question is the following :
Is there a way to replicate what minimstreamer is doing using ffmpeg ?
I have tried this :

    


    


    ffmpeg -re -i https://streamcdnf31-4c4b867c89244861ac216426883d1ad0.msvdn.net/radiodeejay/radiodeejay/master_ma.m3u8 -c copy -listen 1 -f mpegts http://192.168.1.9:10000

    


    


    which is playing corrctly in local vlc on the same computer. But when I stop VLC is got this error in ffmpeg :

    


    [https @ 00000291de047400] Cannot reuse HTTP connection for different host: StreamCdnG20-4c4b867c89244861ac216426883d1ad0.msvdn.net:-1 != 4c4b867c89244861ac216426883d1ad0.msvdn.net:-1
[hls @ 00000291dd96d140] keepalive request failed for 'https://4c4b867c89244861ac216426883d1ad0.msvdn.net/radiodeejay/radiodeejay/play1.m3u8' with error: 'Invalid argument' when parsing playlist
[hls @ 00000291dd96d140] Opening 'https://4c4b867c89244861ac216426883d1ad0.msvdn.net/radiodeejay/radiodeejay/play1.m3u8' for reading
[hls @ 00000291dd96d140] Skip ('#EXT-X-DISCONTINUITY-SEQUENCE:0')
[hls @ 00000291dd96d140] Skip ('#EXT-X-PROGRAM-DATE-TIME:2024-11-29T06:36:56.926Z')
[hls @ 00000291dd96d140] Skip ('#EXT-X-PROGRAM-DATE-TIME:2024-11-29T06:37:07.314Z')
[hls @ 00000291dd96d140] Skip ('#EXT-X-PROGRAM-DATE-TIME:2024-11-29T06:37:17.571Z')
[https @ 00000291de4e00c0] Opening 'https://StreamCdnG20-4c4b867c89244861ac216426883d1ad0.msvdn.net/radiodeejay/radiodeejay/20240722T095729_p1s_001086632.ts' for reading
[aost#0:0/copy @ 00000291de1c4f40] Error submitting a packet to the muxer: Error number -10054 occurred
    Last message repeated 1 times
[out#0/mpegts @ 00000291deaa7e40] Error muxing a packet
[out#0/mpegts @ 00000291deaa7e40] Task finished with error code: -10054 (Error number -10054 occurred)
[out#0/mpegts @ 00000291deaa7e40] Terminating thread with return code -10054 (Error number -10054 occurred)
[out#0/mpegts @ 00000291deaa7e40] Error writing trailer: Error number -10054 occurred
[http @ 00000291de8870c0] URL read error: Error number -10054 occurred
[out#0/mpegts @ 00000291deaa7e40] Error closing file: Error number -10054 occurred
[out#0/mpegts @ 00000291deaa7e40] video:0KiB audio:797KiB subtitle:0KiB other streams:0KiB global headers:0KiB muxing overhead: 21.849292%
size=     971KiB time=00:00:50.98 bitrate= 156.0kbits/s speed=1.01x
Conversion failed!


    


    And if I try to connect from my internet radio I immediately got this error :

    


    [aost#0:0/copy @ 0000027081584a40] Error submitting a packet to the muxer: Error number -10053 occurred
    Last message repeated 1 times
[out#0/mpegts @ 0000027081e684c0] Error muxing a packet
[out#0/mpegts @ 0000027081e684c0] Task finished with error code: -10053 (Error number -10053 occurred)
[out#0/mpegts @ 0000027081e684c0] Terminating thread with return code -10053 (Error number -10053 occurred)
[out#0/mpegts @ 0000027081e684c0] Error writing trailer: Error number -10053 occurred
[http @ 0000027081c47680] URL read error: Error number -10053 occurred
[out#0/mpegts @ 0000027081e684c0] Error closing file: Error number -10053 occurred
[out#0/mpegts @ 0000027081e684c0] video:0KiB audio:46KiB subtitle:0KiB other streams:0KiB global headers:0KiB muxing overhead: 13.917515%
size=      52KiB time=00:00:02.94 bitrate= 145.1kbits/s speed= 1.2x
Conversion failed!


    


    What is the correct way to stream this one locally in order to be listened in my internet radio ?
Shall I use ffmpeg or can be done directly with ngnix ? Or shall I use both ?

    


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