diff --git a/src/audio/audiostream.cpp b/src/audio/audiostream.cpp
index b1c0fcfd3d781f38be74ef2885042addc7d21ea3..196a17e1086fb714ca9e2d511f6f2bcfa08703a0 100644
--- a/src/audio/audiostream.cpp
+++ b/src/audio/audiostream.cpp
@@ -87,7 +87,12 @@ AudioStream::createStream( pa_context* c )
assert( s );
if( _streamType == PLAYBACK_STREAM ){
- pa_stream_connect_playback( s , NULL , NULL ,
+ pa_buffer_attr* attributes;
+ attributes->maxlength = 66500;
+ attributes->tlength = 44100;
+ attributes->prebuf = 10000;
+ attributes->minreq = 882;
+ pa_stream_connect_playback( s , NULL , attributes ,
PA_STREAM_INTERPOLATE_TIMING,
&_volume, NULL);
//pa_cvolume_set(&cv, sample_spec.channels , _volume) , NULL );
diff --git a/src/audio/pulselayer.cpp b/src/audio/pulselayer.cpp
index 734a2dec9531ad6b49fa4f308373d2f59726e07f..5515a350aed8d0596e40230225b1a73249d8cf4b 100644
--- a/src/audio/pulselayer.cpp
+++ b/src/audio/pulselayer.cpp
@@ -115,10 +115,10 @@ PulseLayer::createStreams( pa_context* c )
{
playback = new AudioStream(c, PLAYBACK_STREAM, "SFLphone out", _manager->getSpkrVolume());
pa_stream_set_write_callback( playback->pulseStream() , audioCallback, this);
- pa_stream_set_overflow_callback( playback->pulseStream() , overflow , this);
+ //pa_stream_set_overflow_callback( playback->pulseStream() , overflow , this);
record = new AudioStream(c, CAPTURE_STREAM, "SFLphone in", _manager->getMicVolume());
pa_stream_set_read_callback( record->pulseStream() , audioCallback, this);
- pa_stream_set_underflow_callback( record->pulseStream() , underflow , this);
+ //pa_stream_set_underflow_callback( record->pulseStream() , underflow , this);
cache = new AudioStream(c, UPLOAD_STREAM, "Cache samples", _manager->getSpkrVolume());
pa_threaded_mainloop_signal(m , 0);
@@ -286,82 +286,94 @@ PulseLayer::overflow ( pa_stream* s, void* userdata )
void
PulseLayer::processData( void )
{
- const char* data;
- size_t r;
- int toGet;
- int urgentAvail; // number of data right and data left
- int normalAvail; // number of data right and data left
- size_t maxsize;
- int toPlay;
-
// Handle the mic
- if( (record->pulseStream()) && pa_stream_get_state( record->pulseStream()) == PA_STREAM_READY) {
+ // We check if the stream is ready
+ if( (record->pulseStream()) && pa_stream_get_state( record->pulseStream()) == PA_STREAM_READY)
+ readFromMic();
- if( pa_stream_peek( record->pulseStream() , (const void**)&data , &r ) < 0 || !data ){
- _debug("pa_stream_peek() failed: %s\n" , pa_strerror( pa_context_errno( context) ));
- //return;
- }
-
- if( data != 0 ){
- _micRingBuffer.Put( (void*)data , r, 100);
- }
+ // Handle the data for the speakers
+ if( (playback->pulseStream()) && pa_stream_get_state( playback->pulseStream()) == PA_STREAM_READY){
- if( pa_stream_drop( record->pulseStream() ) < 0 ) {
- _debug("pa_stream_drop() failed: %s\n" , pa_strerror( pa_context_errno( context) ));
- //return;
- }
- }
- _debug("writable size = %d\n" , pa_stream_writable_size(playback->pulseStream()));
- if( pa_stream_writable_size(playback->pulseStream()) == 0 ){
- _debug("flush it\n");
+ // If the playback buffer is full, we don't overflow it; wait for it to have free space
+ if( pa_stream_writable_size(playback->pulseStream()) == 0 )
return;
- }
- if( (playback->pulseStream()) && pa_stream_get_state( playback->pulseStream()) == PA_STREAM_READY || !(maxsize=pa_stream_writable_size(playback->pulseStream()))) {
- maxsize = 2048;
- SFLDataFormat* out;
- urgentAvail = _urgentRingBuffer.AvailForGet();
- if (urgentAvail > 0) {
- // Urgent data (dtmf, incoming call signal) come first.
- _debug("Play urgent!: %i\n" , urgentAvail);
- toGet = (urgentAvail < (int)(framesPerBuffer * sizeof(SFLDataFormat))) ? urgentAvail : framesPerBuffer * sizeof(SFLDataFormat);
- out = (SFLDataFormat*)pa_xmalloc(toGet * sizeof(SFLDataFormat) );
- _urgentRingBuffer.Get(out, toGet, 100);
- // Consume the regular one as well (same amount of bytes)
- _mainSndRingBuffer.Discard(toGet);
- pa_stream_write( playback->pulseStream() , out , toGet , pa_xfree, 0 , PA_SEEK_RELATIVE);
- }
- else
- {
- AudioLoop* tone = _manager->getTelephoneTone();
- if ( tone != 0) {
- toGet = framesPerBuffer;
- out = (SFLDataFormat*)pa_xmalloc(toGet * sizeof(SFLDataFormat) * sizeof(SFLDataFormat));
- tone->getNext(out, toGet * sizeof(SFLDataFormat) , 100);
- pa_stream_write( playback->pulseStream() , out , toGet * sizeof(SFLDataFormat) * sizeof(SFLDataFormat) , pa_xfree, 0 , PA_SEEK_RELATIVE);
- } else if ( (tone=_manager->getTelephoneFile()) != 0 ) {
- toGet = framesPerBuffer;
- toPlay = ( toGet * sizeof(SFLDataFormat) * sizeof(SFLDataFormat) > maxsize )? maxsize : toGet* sizeof(SFLDataFormat) * sizeof(SFLDataFormat);
- out = (SFLDataFormat*)pa_xmalloc(toPlay);
- tone->getNext(out, toPlay/2 , 100);
- if( pa_stream_write( playback->pulseStream() , out , toPlay , pa_xfree, 0 , PA_SEEK_RELATIVE) < 0 )
- _debug("aie aie aie\n");
- } else {
- out = (SFLDataFormat*)pa_xmalloc(framesPerBuffer * sizeof(SFLDataFormat));
- normalAvail = _mainSndRingBuffer.AvailForGet();
- toGet = (normalAvail < (int)(framesPerBuffer * sizeof(SFLDataFormat))) ? normalAvail : framesPerBuffer * sizeof(SFLDataFormat);
- if (toGet) {
- _mainSndRingBuffer.Get(out, toGet, 100);
- //_debug("Write %i bytes\n" , toGet);
- _mainSndRingBuffer.Discard(toGet);
- } else {
+ writeToSpeaker();
+ }
+}
+
+void PulseLayer::writeToSpeaker( void )
+{
+ /** Bytes available in the urgent ringbuffer ( reserved for DTMF ) */
+ int urgentAvail;
+ /** Bytes available in the regular ringbuffer ( reserved for voice ) */
+ int normalAvail;
+ int toGet;
+ int toPlay;
+
+ SFLDataFormat* out;
+ urgentAvail = _urgentRingBuffer.AvailForGet();
+ if (urgentAvail > 0) {
+ // Urgent data (dtmf, incoming call signal) come first.
+ _debug("Play urgent!: %i\n" , urgentAvail);
+ toGet = (urgentAvail < (int)(framesPerBuffer * sizeof(SFLDataFormat))) ? urgentAvail : framesPerBuffer * sizeof(SFLDataFormat);
+ out = (SFLDataFormat*)pa_xmalloc(toGet * sizeof(SFLDataFormat) );
+ _urgentRingBuffer.Get(out, toGet, 100);
+ pa_stream_write( playback->pulseStream() , out , toGet , pa_xfree, 0 , PA_SEEK_RELATIVE);
+ // Consume the regular one as well (same amount of bytes)
+ _mainSndRingBuffer.Discard(toGet);
+ }
+ else
+ {
+ AudioLoop* tone = _manager->getTelephoneTone();
+ if ( tone != 0) {
+ toGet = framesPerBuffer;
+ out = (SFLDataFormat*)pa_xmalloc(toGet * sizeof(SFLDataFormat) * sizeof(SFLDataFormat));
+ tone->getNext(out, toGet * sizeof(SFLDataFormat) , 100);
+ pa_stream_write( playback->pulseStream() , out , toGet * sizeof(SFLDataFormat) * sizeof(SFLDataFormat) , pa_xfree, 0 , PA_SEEK_RELATIVE);
+ }
+ if ( (tone=_manager->getTelephoneFile()) != 0 ) {
+ toGet = framesPerBuffer;
+ toPlay = ( toGet * sizeof(SFLDataFormat) * sizeof(SFLDataFormat)> framesPerBuffer )? framesPerBuffer : toGet* sizeof(SFLDataFormat)* sizeof(SFLDataFormat) ;
+ out = (SFLDataFormat*)pa_xmalloc(toPlay);
+ tone->getNext(out, toPlay/2 , 100);
+ pa_stream_write( playback->pulseStream() , out , toPlay , pa_xfree, 0 , PA_SEEK_RELATIVE) ;
+ //out = (SFLDataFormat*)pa_xmalloc(toGet * sizeof(SFLDataFormat));
+ //tone->getNext(out, toGet , 100);
+ //pa_stream_write( playback->pulseStream() , out , toGet , pa_xfree, 0 , PA_SEEK_RELATIVE) < 0;
+ }
+ else {
+ out = (SFLDataFormat*)pa_xmalloc(framesPerBuffer * sizeof(SFLDataFormat));
+ normalAvail = _mainSndRingBuffer.AvailForGet();
+ toGet = (normalAvail < (int)(framesPerBuffer * sizeof(SFLDataFormat))) ? normalAvail : framesPerBuffer * sizeof(SFLDataFormat);
+ if (toGet) {
+ _mainSndRingBuffer.Get(out, toGet, 100);
+ _mainSndRingBuffer.Discard(toGet);
+ }
+ else {
bzero(out, framesPerBuffer * sizeof(SFLDataFormat));
- }
- pa_stream_write( playback->pulseStream() , out , toGet , pa_xfree, 0 , PA_SEEK_RELATIVE);
}
+ pa_stream_write( playback->pulseStream() , out , toGet , pa_xfree, 0 , PA_SEEK_RELATIVE);
}
- }
- //}
+ }
+}
+
+void PulseLayer::readFromMic( void )
+{
+ const char* data;
+ size_t r;
+
+ if( pa_stream_peek( record->pulseStream() , (const void**)&data , &r ) < 0 || !data ){
+ _debug("pa_stream_peek() failed: %s\n" , pa_strerror( pa_context_errno( context) ));
+ }
+
+ if( data != 0 ){
+ _micRingBuffer.Put( (void*)data , r, 100);
+ }
+
+ if( pa_stream_drop( record->pulseStream() ) < 0 ) {
+ _debug("pa_stream_drop() failed: %s\n" , pa_strerror( pa_context_errno( context) ));
+ }
}
int
diff --git a/src/audio/pulselayer.h b/src/audio/pulselayer.h
index 5156591269151329a8c0d5c1c3b8919d3e47a293..d7fbd6cbf745cbd044e3167e408a8d9789cbdb4a 100644
--- a/src/audio/pulselayer.h
+++ b/src/audio/pulselayer.h
@@ -181,6 +181,8 @@ class PulseLayer : public AudioLayer {
* Write data from the ring buffer to the harware and read data from the hardware
*/
void processData( void );
+ void readFromMic( void );
+ void writeToSpeaker( void );
/**
* Create the audio streams into the given context
diff --git a/src/managerimpl.cpp b/src/managerimpl.cpp
index 817dfa51a12286ff0b809b3a169af9b96080d8f9..af9c12f6f6cd435f90193f6f505d9571885d86f3 100644
--- a/src/managerimpl.cpp
+++ b/src/managerimpl.cpp
@@ -503,7 +503,7 @@ ManagerImpl::playDtmf(char code, bool isTalking)
audiolayer->playSamples(_buf, size * sizeof(SFLDataFormat), isTalking);
else
_debug("DTMF disabled\n");
- //audiolayer->putUrgent( _buf, size * sizeof(SFLDataFormat) );
+ audiolayer->putUrgent( _buf, size * sizeof(SFLDataFormat) );
}
returnValue = true;
@@ -742,7 +742,6 @@ ManagerImpl::playATone(Tone::TONEID toneId) {
else{
audiolayer->startStream();
}
- // Pulseaudio code
}
else
return false;