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    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

  • Contribute to translation

    13 avril 2011

    You can help us to improve the language used in the software interface to make MediaSPIP more accessible and user-friendly. You can also translate the interface into any language that allows it to spread to new linguistic communities.
    To do this, we use the translation interface of SPIP where the all the language modules of MediaSPIP are available. Just subscribe to the mailing list and request further informantion on translation.
    MediaSPIP is currently available in French and English (...)

  • Encoding and processing into web-friendly formats

    13 avril 2011, par

    MediaSPIP automatically converts uploaded files to internet-compatible formats.
    Video files are encoded in MP4, Ogv and WebM (supported by HTML5) and MP4 (supported by Flash).
    Audio files are encoded in MP3 and Ogg (supported by HTML5) and MP3 (supported by Flash).
    Where possible, text is analyzed in order to retrieve the data needed for search engine detection, and then exported as a series of image files.
    All uploaded files are stored online in their original format, so you can (...)

Sur d’autres sites (13972)

  • Script doesnt recognize bars / length right for cutting audio , ffmpeg terminal

    14 avril 2024, par totzillarbeats

    This terminal script doesn't recognize bars / length right for cutting audio, maybe somebody knows what's wrong with the calculation ...

    


    Would be happy about any help the cutting already works !

    


    #!/bin/bash

# Function to extract BPM from filename

get_bpm() {
    local filename="$1"
    local bpm=$(echo "$filename" | grep -oE '[0-9]{1,3}' | head -n1)
    echo "$bpm"
}

# Function to cut audio based on BPM
cut_audio() {
    local input_file="$1"
    local bpm="$2"
    local output_file="${input_file%.*}_cut.${input_file##*.}" # Appends "_cut" to original filename

    # Define the number of beats per bar (assuming 4 beats per bar)
    beats_per_bar=4

    # Calculate the duration of each bar in seconds
    bar_duration=$((60 * beats_per_bar / bpm))

    # Define start and end times for each bar range
    start_times=(0 21 33 45 57 69 81 93 105 117 129 141)
    end_times=(20 29 41 53 65 77 89 101 113 125 137 149)

    # Iterate through each bar range
    for ((i = 0; i < ${#start_times[@]}; i++)); do
        start_time=${start_times[$i]}
        end_time=${end_times[$i]}
        echo "Cutting audio file $input_file at $bpm BPM for bar $((i + 1)) ($start_time-$end_time) for $bar_duration seconds..."

        # Cut audio for current bar range using ffmpeg
        ffmpeg -i "$input_file" -ss "$start_time" -to "$end_time" -c copy "$output_file"_"$((i + 1)).${input_file##*.}" -y
    done

    # Check if the output files are empty and delete them if so
    for output_file in "${output_file}"_*; do
        if [ ! -s "$output_file" ]; then
            echo "Output file $output_file is empty. Deleting..."
            rm "$output_file"
        fi
    done

    echo "Audio cut and saved as $output_file"
}


# Main script
if [ "$#" -eq 0 ]; then
    echo "Usage: $0 [audio_file1] [audio_file2] ..."
    exit 1
fi

for file in "$@"; do
    bpm=$(get_bpm "$file")
    if [ -z "$bpm" ]; then
        echo "Error: No BPM found in filename $file"
    else
        cut_audio "$file" "$bpm"
    fi
done


    


    Maybe its only the math calc in the beginning but idk :)

    


    If you need more details just lmk

    


  • avutil/rational : Check that av_reduce() returns values within the requested max

    29 décembre 2014, par Michael Niedermayer
    avutil/rational : Check that av_reduce() returns values within the requested max
    

    Signed-off-by : Michael Niedermayer <michaelni@gmx.at>

    • [DH] libavutil/rational.c
  • ffmpeg : avcodec_open2 returns invalid argument

    12 mai 2016, par roari

    i’m reusing the sample code from the developer 64bit release of ffmpeg in my application to encode a video :

    AVCodec* pCodec_{nullptr};
    AVCodecContext* pContext_{nullptr};

    avcodec_register_all();
    pCodec_ = avcodec_find_encoder(AV_CODEC_ID_MPEG2VIDEO);
    if (!pCodec_) {}

    pContext_ = avcodec_alloc_context3(pCodec_);
    if (!pContext_) {}

    pContext_->bit_rate = 400000;
    pContext_->width = size.width();
    pContext_->height = size.height();

    pContext_->time_base.den = 1;
    pContext_->time_base.num = fps;

    pContext_->gop_size = 10;
    pContext_->max_b_frames = 1;
    pContext_->pix_fmt = AV_PIX_FMT_BGR0;

    if (codec_id == AV_CODEC_ID_H264) {
       av_opt_set(pContext_->priv_data, "preset", "slow", 0);
    }

    int err = avcodec_open2(pContext_, pCodec_, nullptr);
    if (err &lt; 0) {}

    AVCodec* and AVCodecContext* look like they are allocated correctly. avcodec_open2 then returns invalid argument (-22).

    I use : Windows 10 64, VS2013 Compiler, Qt Creator IDE, ffmpeg(2016-05-12) 64bit.

    The sample i took the code from is "decoding_encoding.c".

    Any ideas ?