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Commit a4d28f2c authored by Emmanuel Milou's avatar Emmanuel Milou
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Clean up pulseaudio code + DTMK ok

parent 46d4f9ab
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src/audio/audiostream.cpp
+ 6
− 1
View file @ a4d28f2c
......@@ -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 );
......
src/audio/pulselayer.cpp
+ 82
− 70
View file @ a4d28f2c
......@@ -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
......
src/audio/pulselayer.h
+ 2
− 0
View file @ a4d28f2c
......@@ -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
......
src/managerimpl.cpp
+ 1
− 2
View file @ a4d28f2c
......@@ -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;
......
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