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    Puis-je poster des contenus à partir d’une tablette Ipad ?
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  • Support audio et vidéo HTML5

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    MediaSPIP utilise les balises HTML5 video et audio pour la lecture de documents multimedia en profitant des dernières innovations du W3C supportées par les navigateurs modernes.
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    Le lecteur HTML5 utilisé a été spécifiquement créé pour MediaSPIP : il est complètement modifiable graphiquement pour correspondre à un thème choisi.
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  • HTML5 audio and video support

    13 avril 2011, par

    MediaSPIP uses HTML5 video and audio tags to play multimedia files, taking advantage of the latest W3C innovations supported by modern browsers.
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Sur d’autres sites (7205)

  • lavc : clarify meaning of avctx.level option

    26 août 2023, par Stefano Sabatini
    lavc : clarify meaning of avctx.level option
    
    • [DH] doc/codecs.texi
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  • Meaning of Timestamp Retrieved by AudioQueueGetCurrentTime() in AudioQueue Callback

    30 juillet 2024, par White0930

    I'm working on an audio project and have a question regarding the meaning of the timestamp retrieved by AudioQueueGetCurrentTime().

    


    According to the Apple Developer Documentation, the following calculation gives the audio time being played (since AudioQueueStart) :

    


    - (Float64) GetCurrentTime {
    AudioTimeStamp c;       
    AudioQueueGetCurrentTime(playState.queue, NULL, &c, NULL);  
    return c.mSampleTime / _av->audio.sample_rate;
}


    


    However, in a project I'm working on, I noticed the following code inside the fillAudioBuffer callback function of AudioQueue :

    


    
static void fillAudioBuffer(AudioQueueRef queue, AudioQueueBufferRef buffer){
    
    int lengthCopied = INT32_MAX;
    int dts= 0;
    int isDone = 0;

    buffer->mAudioDataByteSize = 0;
    buffer->mPacketDescriptionCount = 0;
    
    OSStatus err = 0;
    AudioTimeStamp bufferStartTime;

    AudioQueueGetCurrentTime(queue, NULL, &bufferStartTime, NULL);
    

    
    while(buffer->mPacketDescriptionCount < numPacketsToRead && lengthCopied > 0){
        if (buffer->mAudioDataByteSize) {
            break;
        }
        
        lengthCopied = getNextAudio(_av,buffer->mAudioDataBytesCapacity-buffer->mAudioDataByteSize, (uint8_t*)buffer->mAudioData+buffer->mAudioDataByteSize,&dts,&isDone);
        if(!lengthCopied || isDone) break;
      
        if(aqStartDts < 0) aqStartDts = dts;
        if (dts>0) currentDts = dts;
        if(buffer->mPacketDescriptionCount ==0){
            bufferStartTime.mFlags = kAudioTimeStampSampleTimeValid;
            bufferStartTime.mSampleTime = (Float64)(dts-aqStartDts) * _av->audio.frame_size;
            
            if (bufferStartTime.mSampleTime <0 ) bufferStartTime.mSampleTime = 0;
            PMSG2("AQHandler.m fillAudioBuffer: DTS for %x: %lf time base: %lf StartDTS: %d\n", (unsigned int)buffer, bufferStartTime.mSampleTime, _av->audio.time_base, aqStartDts);
            
        }
        buffer->mPacketDescriptions[buffer->mPacketDescriptionCount].mStartOffset = buffer->mAudioDataByteSize;
        buffer->mPacketDescriptions[buffer->mPacketDescriptionCount].mDataByteSize = lengthCopied;
        buffer->mPacketDescriptions[buffer->mPacketDescriptionCount].mVariableFramesInPacket = _av->audio.frame_size;
        
        buffer->mPacketDescriptionCount++;
        buffer->mAudioDataByteSize += lengthCopied;
        
    }
    
#ifdef DEBUG
    int audioBufferCount, audioBufferTotal,  videoBufferCount, videoBufferTotal;
    bufferCheck(_av,&videoBufferCount, &videoBufferTotal, &audioBufferCount, &audioBufferTotal);
    
    PMSG2("AQHandler.m fillAudioBuffer: Video Buffer: %d/%d Audio Buffer: %d/%d\n", videoBufferCount, videoBufferTotal, audioBufferCount, audioBufferTotal);
    
    PMSG2("AQHandler.m fillAudioBuffer: Bytes copied for buffer 0x%x: %d\n",(unsigned int)buffer, (int)buffer->mAudioDataByteSize );
#endif  
    if(buffer->mAudioDataByteSize){
        
        if(err=AudioQueueEnqueueBufferWithParameters(queue, buffer, 0, NULL, 0, 0, 0, NULL, &bufferStartTime, NULL))
        {
#ifdef DEBUG
            char sErr[10];

            PMSG2(@"AQHandler.m fillAudioBuffer: Could not enqueue buffer 0x%x: %d %s.", buffer, err, FormatError(sErr, err));
#endif
        }
    }

}


    


    Based on the documentation for AudioQueueEnqueueBufferWithParameters and the variable naming used by the author, bufferStartTime seems to represent the time when the newly filled audio buffer will start playing, i.e., the time when all current audio in the queue has finished playing and the new audio starts. This interpretation suggests bufferStartTime is not the same as the time of the audio currently being played.

    


    I have browsed through many related questions, but I still have some doubts.. I'm currently fixing an audio-video synchronization issue in my project, and there isn't much detailed information in the Apple Developer Documentation (or maybe my search skills are lacking).

    


    Can someone clarify the exact meaning of the timestamp returned by AudioQueueGetCurrentTime() in this context ? Is it the time when the current audio will finish playing, or is it the time when the new audio will start playing ? Any additional resources or documentation that explain this in detail would also be appreciated.

    


  • Add Language Name for each file, rename the language code according to the standard ISO 639 for Estonian, Georgian, Ukrainian and Chinese (http://en.wikipedia.org/wiki/List_of_ISO_639-1_codes)

    14 août 2012, par Coto

    m localization/messages_ar.js m localization/messages_bg.js m localization/messages_ca.js m localization/messages_cs.js m localization/messages_da.js m localization/messages_de.js m localization/messages_el.js m localization/messages_es.js m localization/messages_et.js m (...)