From 7f71f3ac890512ecae2d21c6ca8022ec644ebe21 Mon Sep 17 00:00:00 2001 From: Emmanuel Milou <emmanuel.milou@savoirfairelinux.com> Date: Fri, 24 Oct 2008 17:02:54 -0400 Subject: [PATCH] warning removed --- src/audio/alsalayer.cpp | 28 +++------------------------- src/iaxvoiplink.cpp | 17 +++++++++-------- 2 files changed, 12 insertions(+), 33 deletions(-) diff --git a/src/audio/alsalayer.cpp b/src/audio/alsalayer.cpp index 4fe02554f3..cf1df47127 100644 --- a/src/audio/alsalayer.cpp +++ b/src/audio/alsalayer.cpp @@ -134,28 +134,7 @@ void AlsaLayer::AlsaCallBack( snd_async_handler_t* pcm_callback ) void AlsaLayer::fillHWBuffer( void) { - unsigned char* data; - int pcmreturn, l1, l2; - short s1, s2; - int periodSize = 128 ; - int frames = periodSize >> 2 ; - _debug("frames = %d\n", frames); - - data = (unsigned char*)malloc(periodSize); - for(l1 = 0; l1 < 100; l1++) { - for(l2 = 0; l2 < frames; l2++) { - s1 = 0; - s2 = 0; - data[4*l2] = (unsigned char)s1; - data[4*l2+1] = s1 >> 8; - data[4*l2+2] = (unsigned char)s2; - data[4*l2+3] = s2 >> 8; - } - while ((pcmreturn = snd_pcm_writei(_PlaybackHandle, data, frames)) < 0) { - snd_pcm_prepare(_PlaybackHandle); - //_debugAlsa("< Buffer Underrun >\n"); - } - } + } bool @@ -165,7 +144,6 @@ AlsaLayer::isStreamActive (void) return (isPlaybackActive() && isCaptureActive()); } - int AlsaLayer::playSamples(void* buffer, int toCopy, bool isTalking) { @@ -199,7 +177,7 @@ AlsaLayer::canGetMic() int avail; if ( _CaptureHandle ) { avail = snd_pcm_avail_update( _CaptureHandle ); - //printf("%d\n", avail ); + printf("%d\n", avail ); if(avail > 0) return avail; else @@ -339,7 +317,7 @@ bool AlsaLayer::alsa_set_params( snd_pcm_t *pcm_handle, int type, int rate ){ return false; } if(dir!=0) { - _debugAlsa("(%i) The choosen period size %d bytes is not supported by your hardware.\nUsing %d instead.\n ", type, periodsize, exact_lvalue); + _debugAlsa("(%i) The choosen period size %d bytes is not supported by your hardware.\nUsing %d instead.\n ", type, (int)periodsize, (int)exact_lvalue); } periodsize=exact_lvalue; /* Set the number of fragments */ diff --git a/src/iaxvoiplink.cpp b/src/iaxvoiplink.cpp index f8bf510607..1bbdd62889 100644 --- a/src/iaxvoiplink.cpp +++ b/src/iaxvoiplink.cpp @@ -219,6 +219,10 @@ IAXVoIPLink::sendAudioFromMic(void) int maxBytesToGet, availBytesFromMic, bytesAvail, nbSample, compSize; AudioCodec *ac; + // We have to update the audio layer type in case we switched + // TODO Find out a better way to do it + updateAudiolayer(); + IAXCall* currentCall = getIAXCall(Manager::instance().getCurrentCallId()); if (!currentCall) { @@ -231,15 +235,15 @@ IAXVoIPLink::sendAudioFromMic(void) // Just make sure the currentCall is in state to receive audio right now. //_debug("Here we get: connectionState: %d state: %d \n", - // currentCall->getConnectionState(), - // currentCall->getState()); + //currentCall->getConnectionState(), + //currentCall->getState()); if (currentCall->getConnectionState() != Call::Connected || currentCall->getState() != Call::Active) { return; } - ac = currentCall -> getCodecMap().getCodec( currentCall -> getAudioCodec() ); + ac = currentCall->getCodecMap().getCodec( currentCall -> getAudioCodec() ); if (!ac) { // Audio codec still not determined. if (audiolayer) { @@ -255,14 +259,11 @@ IAXVoIPLink::sendAudioFromMic(void) // we have to get 20ms of data from the mic *20/1000 = /50 // rate/50 shall be lower than IAX__20S_48KHZ_MAX maxBytesToGet = audiolayer->getSampleRate()* audiolayer->getFrameSize() / 1000 * sizeof(SFLDataFormat); - - // We have to update the audio layer type in case we switched - // TODO Find out a better way to do it - updateAudiolayer(); - + // available bytes inside ringbuffer availBytesFromMic = audiolayer->canGetMic(); + _debug("max bytes=%i - avail = %i\n", maxBytesToGet, availBytesFromMic); if (availBytesFromMic < maxBytesToGet) { // We need packets full! -- GitLab