search for: audiobuff

...ction. I set the max data bytes for output payload to 2048, application mode as VoIP, and complexity to 5. The following is a snippet in ObjC where I did the encoding: self.inputReceiver = [AEBlockAudioReceiver audioReceiverWithBlock:^(void *source, const AudioTimeStamp *time, UInt32 frames, AudioBufferList *bufferList) { UInt32 numberOfBuffers = bufferList->mNumberBuffers; for (int bufferIndex = 0; bufferIndex < numberOfBuffers ; ++bufferIndex) { AudioBuffer audioBuffer = bufferList->mBuffers[bufferIndex]; int frameSize = (samplingRate/1000) *...

...Modem->trigger++; } else if(Modem->trigger > 0) { Modem->trigger--; } if(Modem->is_onair) { Modem->trigger = Modem->trigger_high; /* Se siamo ON AIR, colleziona i dati audio VALIDI in un buffer separato */ for(d=0;d<tbc;d++) { Modem->audiobuffer[Modem->audio_count] = TEMP_Buffer[d]; Modem->audio_count++; } } tbc = 0; bzero(&TEMP_Buffer,NN); } =============================================== As you can image i use speex to clean and elaborate audio from a voice modem. All work correctly BUT i've a ugly...

...because the first goal is a good raw performance of the whole loop. If there is no frame dropping mechanism, I would expect that the audio/video playback goes out of sync... but this is not the case. Instead the video playback slows down and the audio playback has gaps (even with precached decoded audiobuffers). Not only the Theora decoding and the YUV2RGB stuff takes time and those seem to be not the main reason why audio gets stalled (the vorbis decoder is in the same loop and it is called very frequently, also when heavy work load on theora). The problem occures when ogg_stream_pagein() is called....

Hi, I was wondering if there's any sense in increasing audiobuffer above the minimal '2' in meetme, if every channel is already dejittered before (Local/.../nj - as described at: http://www.russellbryant.net/blog/2007/10/09/asterisk-jitterbuffer-support-for-applications/) Will it help in anything, or just increase delay? Thanks, Stan

...uses the read and seek callback funktions) delivers correct values. Thats why i doubt that there's an error in (the rather simple) callback funktions... Is it valid to call ov_pcm_total() on an ogg file that was opened with ov_open_callbacks()? If it is not: How can i calculate the size of the audiobuffer? I have realy no idea what i've done wrong, so please let me know if you have any ideas. Thank's in advance, Mike <p>... by the way, how can i find out with which version of the librarys i am working with? <p>--- >8 ---- List archives: http://www.xiph.org/archives/ Ogg...

...udio bytes: 640 Buffer size: 320 This always happens, even if there's only one participant in the conference. All phones are Supura/Linksys SPA-941 phones. Everything is working fine (users can talk to each other, voicemail is working, etc), exept for meetme. In meetme.conf I've got audiobuffers=32, which doesn't help. Any clue? -- Erik Hensema (erik@hensema.net)

...buffer, encodedSize * FRAME_SIZE * *2); speex_decode_int(dec_state, &bits_out, outTemp); return outTemp; } And at last, the function that call speex encode & decode and copy it to the buffer that will be playback by the specific callback : - (void) processAudio: (AudioBufferList*) bufferList { AudioBuffer sourceBuffer = bufferList->mBuffers[0]; NSLog(@"Origin size: %lu", sourceBuffer.mDataByteSize); int size = 0; char *encodedAudio = encodeSpeex((spx_int16_t*) sourceBuffer.mData, sourceBuffer.mDataByteSize, &...