search for: audiobuffers

Hi, I've been trying to encode a live audio input from the microphone on iOS device using opus. Uncompressed audio recording works fine with http://theamazingaudioengine.com/ Then, when I tried to do encoding, I'm stuck at figuring out why the buffer is empty: static int ec_write_byte(ec_enc *_this,unsigned _value){ if(_this->offs+_this->end_offs>=_this->storage)return

Hi all, i use speex preprocessor features in this way: =================================== #define NN 160 /* 20msec di audio */ ... int tbc=0,c,d,ret; spx_int16_t TEMP_Buffer[NN]; char DLECODE; /* Inizializza il preprocessore Speex se non inizializzato */ if(Modem->speex_pp_state == NULL) { Modem->speex_pp_state = speex_preprocess_state_init(NN,AUDIO_SAMPLERATE); }

...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. In...

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

Hi Folks, i implemented the required callback funktions to open an ogg file from memory. Now ov_pcm_total() returns always 0, strange thing about is that ov_pcm_total does not call any of the callback funktions. I also like to mention that ov_info (which uses the read and seek callback funktions) delivers correct values. Thats why i doubt that there's an error in (the rather simple) callback

...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)

Hi everyone, I'm trying to encode/decode with speex, when I do not, the audio is loud and clear, but when I encode/decode to test audio quality, I get a really poor audio quality and a robotic sound. Here's my init audio method : #define AUDIO_QUALITY 10 - (void) initAudio { try { //SPEEX CONFIG speex_bits_init(&bits_in);