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Autres articles (25)

  • 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 (...)

  • Support de tous types de médias

    10 avril 2011

    Contrairement à beaucoup de logiciels et autres plate-formes modernes de partage de documents, MediaSPIP a l’ambition de gérer un maximum de formats de documents différents qu’ils soient de type : images (png, gif, jpg, bmp et autres...) ; audio (MP3, Ogg, Wav et autres...) ; vidéo (Avi, MP4, Ogv, mpg, mov, wmv et autres...) ; contenu textuel, code ou autres (open office, microsoft office (tableur, présentation), web (html, css), LaTeX, Google Earth) (...)

  • List of compatible distributions

    26 avril 2011, par

    The table below is the list of Linux distributions compatible with the automated installation script of MediaSPIP. Distribution nameVersion nameVersion number Debian Squeeze 6.x.x Debian Weezy 7.x.x Debian Jessie 8.x.x Ubuntu The Precise Pangolin 12.04 LTS Ubuntu The Trusty Tahr 14.04
    If you want to help us improve this list, you can provide us access to a machine whose distribution is not mentioned above or send the necessary fixes to add (...)

Sur d’autres sites (5935)

  • mp4 generated by moviepy no sound with tweepy

    2 décembre 2020, par Will Rowe

    I'm using MoviePy to edit a mp4 file and then Tweepy to tweet out the video.

    


    Here's what I have currently :

    


    clip = VideoFileClip("c:/users/.../UNIQUENAME2.mp4").subclip(8) #cut off the first 8 seconds of the clip
clip.write_videofile("c:/users/.../UNIQUENAME2-finished.mp4")
clip.close()
highlightz = TweetMachine()
highlightz.makeAVidTweet('c:/users/.../UNIQUENAME2-finished.mp4','audio test')


    


    Here's my TweetMachine :

    


    def makeAVidTweet(self,fileLoc,text):
        upload_result = self.api.media_upload(fileLoc)
        time.sleep(120) #wait just in case things are still processing
        media_ids = [upload_result.media_id_string]
        self.api.update_status(status=text, media_ids=media_ids)


    


    Currently, the video is cut fine and UNIQUENAME2-finished.mp4 is constructed correctly such that if I open the file on my computer, the video is cut correctly and the sound works. However, the video posted on Twitter has no sound.

    


    I'm assuming it's some sort of issue with how MoviePy makes the mp4 file and Twitter not liking something with the sound settings, but I'm pretty unfamiliar with mp4 stuff and I couldn't find anything about it on the MoviePy or Tweepy docs.

    


    Any tips or thoughts would be greatly appreciated !

    


    Stream info from VLC :
Codec : MPEG Audio layer 1/2 (mpga),
Type : Audio,
Channels : Stereo,
Sample Rate : 44100 Hz,
Bits per sample : 32,
Bitrate : 128 kb/s

    


  • ffmpeg how to ignore initial empty audio frames when decoding to loop a sound

    1er décembre 2020, par cs guy

    I am trying to loop a ogg sound file. The goal is to make a loopable audio interface for my mobile app.

    


    I decode the given ogg file into a buffer and that buffer is sent to audio card for playing. All good until it the audio finishes (end of file). When it finishes I use av_seek_frame(avFormatContext, streamInfoIndex, 0, AVSEEK_FLAG_FRAME); to basically loop back to beginning. And continue decoding into writing to the same buffer. At first sight I thought this would give me perfect loops. One problem I had was, the decoder in the end gives me extra empty frames. So I ignored them by keeping track of how many samples are decoded :

    


    durationInMillis = avFormatContext->duration * 1000;
numOfTotalSamples =
                (uint64_t) avFormatContext->duration *
                (uint64_t) pLocalCodecParameters->sample_rate *
                (uint64_t) pLocalCodecParameters->channels /
                (uint64_t) AV_TIME_BASE;


    


    When the threshold is reached I ignore the frames sent by the codec. I thought this was it and ran some test. I recorded 5 minutes of my app and in the end I compared the results in FL studio by customly adding the same sound clip several times to match the length of my audio recording :

    


    Here it is after 5 minutes :

    


    enter image description here

    


    In the first loops the difference is very low I thought it was working and I used this for several days until I tested this on 5 minute recording. As the looping approached to 5 minutes mark the difference got very huge. My code is not looping the audio correctly. I suspect that the codec is adding 1 or 2 empty frames at the very beginning in each loop caused by av_seek_frame knowing that a frame can contain up several audio samples. These probably accumulate and cause the mismatch.

    


    My question is : how can I drop the empty frames that is sent by codec while decoding so that I can create a perfect loop of the audio ?

    


    My code is below here. Please be aware that I deleted lots of if checks that was inteded for safety to make it more readable in the code below, these removed checks are always false so it doesnt matter for the reader.

    


    helper.cpp

    


    int32_t&#xA;outputAudioFrame(AVCodecContext *avCodecContext, AVFrame *avResampledDecFrame, int32_t &amp;ret,&#xA;                 LockFreeQueue<float> *&amp;buffer, int8_t *&amp;mediaLoadPointer,&#xA;                 AVFrame *avDecoderFrame, SwrContext *swrContext,&#xA;                 std::atomic_bool *&amp;signalExitFuture,&#xA;                 uint64_t &amp;currentNumSamples, uint64_t &amp;numOfTotalSamples) {&#xA;    // resampling is done here but its boiler code so I removed it.&#xA;    auto *floatArrPtr = (float *) (avResampledDecFrame->data[0]);&#xA;&#xA;    int32_t numOfSamples = avResampledDecFrame->nb_samples * avResampledDecFrame->channels;&#xA;&#xA;    for (int32_t i = 0; i &lt; numOfSamples; i&#x2B;&#x2B;) {&#xA;        if (currentNumSamples == numOfTotalSamples) {&#xA;            break;&#xA;        }&#xA;&#xA;        buffer->push(*floatArrPtr);&#xA;        currentNumSamples&#x2B;&#x2B;;&#xA;        floatArrPtr&#x2B;&#x2B;;&#xA;    }&#xA;&#xA;    return 0;&#xA;}&#xA;&#xA;&#xA;&#xA;int32_t decode(int32_t &amp;ret, AVCodecContext *avCodecContext, AVPacket *avPacket,&#xA;               LockFreeQueue<float> *&amp;buffer,&#xA;               AVFrame *avDecoderFrame,&#xA;               AVFrame *avResampledDecFrame,&#xA;               std::atomic_bool *&amp;signalExitFuture,&#xA;               int8_t *&amp;mediaLoadPointer, SwrContext *swrContext,&#xA;               uint64_t &amp;currentNumSamples, uint64_t &amp;numOfTotalSamples) {&#xA;   &#xA;    ret = avcodec_send_packet(avCodecContext, avPacket);&#xA;    if (ret &lt; 0) {&#xA;        LOGE("decode: Error submitting a packet for decoding %s", av_err2str(ret));&#xA;        return ret;&#xA;    }&#xA;&#xA;    // get all the available frames from the decoder&#xA;    while (ret >= 0) {&#xA;&#xA;        // submit the packet to the decoder&#xA;        ret = avcodec_receive_frame(avCodecContext, avDecoderFrame);&#xA;        if (ret &lt; 0) {&#xA;            // those two return values are special and mean there is no output&#xA;            // frame available, but there were no errors during decoding&#xA;            if (ret == AVERROR_EOF || ret == AVERROR(EAGAIN)) {&#xA;                //LOGD("avcodec_receive_frame returned special %s", av_err2str(ret));&#xA;                return 0;&#xA;            }&#xA;&#xA;            LOGE("avcodec_receive_frame Error during decoding %s", av_err2str(ret));&#xA;            return ret;&#xA;        }&#xA;&#xA;        ret = outputAudioFrame(avCodecContext, avResampledDecFrame, ret, buffer,&#xA;                               mediaLoadPointer, avDecoderFrame, swrContext, signalExitFuture,&#xA;                               currentNumSamples, numOfTotalSamples);&#xA;&#xA;        av_frame_unref(avDecoderFrame);&#xA;        av_frame_unref(avResampledDecFrame);&#xA;&#xA;        if (ret &lt; 0)&#xA;            return ret;&#xA;    }&#xA;&#xA;    return 0;&#xA;}&#xA;</float></float>

    &#xA;

    Main.cpp

    &#xA;

    while (!*signalExitFuture) {&#xA;            while ((ret = av_read_frame(avFormatContext, avPacket)) >= 0) {&#xA;&#xA;                ret = decode(ret, avCodecContext, avPacket, buffer, avDecoderFrame,&#xA;                             avResampledDecFrame, signalExitFuture,&#xA;                             mediaLoadPointer, swrContext,&#xA;                             currentNumSamples, numOfTotalSamples);&#xA;&#xA;                // The packet must be freed with av_packet_unref() when it is no longer needed.&#xA;                av_packet_unref(avPacket);&#xA;&#xA;                if (ret &lt; 0) {&#xA;                    LOGE("Error! %s", av_err2str(ret));&#xA;&#xA;                    goto cleanup;&#xA;                }&#xA;            }&#xA;&#xA;            if (ret == AVERROR_EOF) {&#xA;&#xA;                ret = av_seek_frame(avFormatContext, streamInfoIndex, 0, AVSEEK_FLAG_FRAME);&#xA;&#xA;                currentNumSamples = 0;&#xA;                avcodec_flush_buffers(avCodecContext);&#xA;            }&#xA;        }&#xA;

    &#xA;

  • Create background sound for video with ffmpeg

    23 novembre 2020, par Neret

    I have a video file without sound and a stereo audio file. Video is several times longer than audio. I'd like to create a background sound which starts from 2 second silence, then trim silence of the audio at both ends and duplicate trimmed audio several times to the end of the video.

    &#xA;

    enter image description here

    &#xA;

    I found how to trim audio :

    &#xA;

    ffmpeg -y -i audio.wav -af silenceremove=start_periods=1:start_threshold=-75dB,areverse,silenceremove=start_periods=1:start_threshold=-75dB,areverse trimmed_audio.wav&#xA;

    &#xA;

    And how to create silence :

    &#xA;

    ffmpeg -y -f lavfi -i anullsrc=channel_layout=stereo:sample_rate=48000:duration=2 silence.wav&#xA;

    &#xA;

    How can I duplicate the audio and combine it with the video ?

    &#xA;