From b626904724b907c309f5e96f6d21edc0bbdbebef Mon Sep 17 00:00:00 2001
From: Emmanuel Milou <emmanuel.milou@savoirfairelinux.com>
Date: Fri, 23 May 2008 17:16:41 -0400
Subject: [PATCH] Playback functional - Capture not
problem with capture: delay and audio quality
Dtmf, ringtones missing
---
src/audio/audiortp.cpp | 1 -
src/audio/audiostream.cpp | 8 +++-
src/audio/audiostream.h | 1 +
src/audio/pulselayer.cpp | 94 ++++++++++++++++++++++++++++++++-------
src/audio/pulselayer.h | 4 ++
src/managerimpl.cpp | 8 ++--
6 files changed, 93 insertions(+), 23 deletions(-)
diff --git a/src/audio/audiortp.cpp b/src/audio/audiortp.cpp
index 513b29db58..788feb6193 100644
--- a/src/audio/audiortp.cpp
+++ b/src/audio/audiortp.cpp
@@ -1,5 +1,4 @@
/*
- *
* Copyright (C) 2004-2008 Savoir-Faire Linux inc.
* Author: Emmanuel Milou <emmanuel.milou@savoirfairelinux.com>
* Author: Alexandre Bourget <alexandre.bourget@savoirfairelinux.com>
diff --git a/src/audio/audiostream.cpp b/src/audio/audiostream.cpp
index 97a5edda2d..c3164d06fe 100644
--- a/src/audio/audiostream.cpp
+++ b/src/audio/audiostream.cpp
@@ -29,7 +29,8 @@ AudioStream::AudioStream( pa_context* context, int type, std::string desc )
sample_spec.rate = 44100;
sample_spec.channels = 1;
channel_map.channels = 1;
- flag = PA_STREAM_AUTO_TIMING_UPDATE;
+ flag = PA_STREAM_AUTO_TIMING_UPDATE ;
+ volume = PA_VOLUME_NORM;
_audiostream = createStream( context );
}
@@ -64,6 +65,7 @@ AudioStream::createStream( pa_context* c )
{
_debug("Creating %s stream...\n" , _streamDescription.c_str());
pa_stream* s;
+ pa_cvolume cv;
assert(pa_sample_spec_valid(&sample_spec));
assert(pa_channel_map_valid(&channel_map));
@@ -74,7 +76,9 @@ AudioStream::createStream( pa_context* c )
assert( s );
if( _streamType == PLAYBACK_STREAM ){
- pa_stream_connect_playback( s , NULL , NULL , flag , NULL, NULL );
+ pa_stream_connect_playback( s , NULL , NULL ,
+ PA_STREAM_INTERPOLATE_TIMING,
+ pa_cvolume_set(&cv, sample_spec.channels , volume) , NULL );
}
else if( _streamType == CAPTURE_STREAM ){
pa_stream_connect_record( s , NULL , NULL , flag );
diff --git a/src/audio/audiostream.h b/src/audio/audiostream.h
index a4464500f5..c2411d4812 100644
--- a/src/audio/audiostream.h
+++ b/src/audio/audiostream.h
@@ -59,6 +59,7 @@ class AudioStream {
pa_stream_flags_t flag;
pa_sample_spec sample_spec ;
//pa_channel_map channel_map;
+ pa_volume_t volume;
};
diff --git a/src/audio/pulselayer.cpp b/src/audio/pulselayer.cpp
index 480dde04e1..e26bdb63b2 100644
--- a/src/audio/pulselayer.cpp
+++ b/src/audio/pulselayer.cpp
@@ -21,11 +21,13 @@
int framesPerBuffer = 4096;
+const pa_buffer_attr *a;
-PulseLayer::PulseLayer(ManagerImpl* manager)
+ PulseLayer::PulseLayer(ManagerImpl* manager)
: AudioLayer( manager , PULSEAUDIO )
, _urgentRingBuffer( SIZEBUF)
- ,_mainSndRingBuffer( SIZEBUF )
+ ,_mainSndRingBuffer( SIZEBUF )
+ ,_micRingBuffer( SIZEBUF )
{
_debug("Pulse audio constructor: Create context\n");
}
@@ -86,7 +88,7 @@ PulseLayer::context_state_callback( pa_context* c, void* user_data )
break;
case PA_CONTEXT_FAILED:
default:
- _debug(" Error : %s" , pa_strerror(pa_context_errno(c)));
+ _debug(" Error : %n" , pa_strerror(pa_context_errno(c)));
exit(1);
}
}
@@ -96,11 +98,14 @@ PulseLayer::createStreams( pa_context* c )
{
playback = new AudioStream(c, PLAYBACK_STREAM, "SFLphone out");
pa_stream_set_write_callback( playback->pulseStream() , audioCallback, this);
+ pa_stream_set_overflow_callback( playback->pulseStream() , overflow , this);
record = new AudioStream(c, CAPTURE_STREAM, "SFLphone in");
pa_stream_set_read_callback( record->pulseStream() , audioCallback, this);
+ pa_stream_set_underflow_callback( playback->pulseStream() , underflow , this);
cache = new AudioStream(c, UPLOAD_STREAM, "Cache samples");
pa_threaded_mainloop_signal(m , 0);
+
}
bool
@@ -162,6 +167,7 @@ PulseLayer::putMain(void* buffer, int toCopy)
void
PulseLayer::flushMain()
{
+ _mainSndRingBuffer.flush();
}
int
@@ -179,25 +185,37 @@ PulseLayer::putUrgent(void* buffer, int toCopy)
int
PulseLayer::canGetMic()
{
- return 882;
+ if( record )
+ {
+ int a = _micRingBuffer.AvailForGet();
+ _debug("available for get = %i\n" , a);
+ return a;
+ }
+ else
+ return 0;
}
int
PulseLayer::getMic(void *buffer, int toCopy)
{
- return 160;
+ if( record ){
+ return _micRingBuffer.Get(buffer, toCopy, 100);
+ }
+ else
+ return 0;
}
void
PulseLayer::flushMic()
{
+ _micRingBuffer.flush();
}
/* bool
-PulseLayer::isStreamStopped (void)
-{
-}
-*/
+ PulseLayer::isStreamStopped (void)
+ {
+ }
+ */
void
PulseLayer::startStream (void)
{
@@ -221,16 +239,35 @@ PulseLayer::audioCallback ( pa_stream* s, size_t bytes, void* userdata )
{
PulseLayer* pulse = (PulseLayer*) userdata;
assert( s && bytes );
+ assert( bytes > 0 );
pulse->write();
+ pulse->read();
+}
+
+ void
+PulseLayer::underflow ( pa_stream* s, void* userdata )
+{
+ _debug("Buffer Underflow\n");
+}
+
+
+ void
+PulseLayer::overflow ( pa_stream* s, void* userdata )
+{
+ _debug("Buffer Overflow\n");
}
void
PulseLayer::write( void )
{
+ // a = pa_stream_get_buffer_attr(record->pulseStream());
+ //if (a)
+ //_debug("Buffer attributes: maxlength=%u , fragsize=%u\n" , a->maxlength, a->fragsize );
+ //_debug("Device name = %s ; device index = %i\n" , pa_stream_get_device_name( playback->pulseStream()) , pa_stream_get_device_index( playback->pulseStream()));
int toGet;
int urgentAvail; // number of data right and data left
int normalAvail; // number of data right and data left
- SFLDataFormat* out = (SFLDataFormat*)malloc(framesPerBuffer * sizeof(SFLDataFormat));
+ SFLDataFormat* out = (SFLDataFormat*)pa_xmalloc(framesPerBuffer * sizeof(SFLDataFormat));
urgentAvail = _urgentRingBuffer.AvailForGet();
if (urgentAvail > 0) {
// Urgent data (dtmf, incoming call signal) come first.
@@ -242,14 +279,14 @@ PulseLayer::write( void )
}
else
{
- AudioLoop* tone = _manager->getTelephoneTone();
+ AudioLoop* tone = 0;//_manager->getTelephoneTone();
if ( tone != 0) {
- tone->getNext(out, framesPerBuffer, 100);
+ //tone->getNext(out, framesPerBuffer, 100);
toGet = framesPerBuffer;
- } else if ( (tone=_manager->getTelephoneFile()) != 0 ) {
+ } /*else if ( (tone=_manager->getTelephoneFile()) != 0 ) {
tone->getNext(out, framesPerBuffer, 100);
toGet = framesPerBuffer;
- } else {
+ } */else {
normalAvail = _mainSndRingBuffer.AvailForGet();
toGet = (normalAvail < (int)(framesPerBuffer * sizeof(SFLDataFormat))) ? normalAvail : framesPerBuffer * sizeof(SFLDataFormat);
if (toGet) {
@@ -264,7 +301,34 @@ PulseLayer::write( void )
pa_stream_write( playback->pulseStream() , out , toGet , pa_xfree, 0 , PA_SEEK_RELATIVE);
}
-int
+ void
+PulseLayer::read( void )
+{
+ //assert( record->pulseStream() );
+ int micAvailPut;
+ size_t toPut;
+ const void* data;
+ // Handle the mic's audio stream
+ micAvailPut = _micRingBuffer.AvailForPut();
+ toPut = 2048;
+ //toPut = (micAvailPut <= (size_t)(framesPerBuffer * sizeof(SFLDataFormat))) ? micAvailPut : framesPerBuffer * sizeof(SFLDataFormat);
+ //_debug("micAvailPut: %i - - toPut : %i\n", micAvailPut, toPut);
+
+ if( pa_stream_peek( record->pulseStream() , &data , &toPut ) < 0 ){
+ _debug("pa_stream_peek() failed: %s\n" , pa_strerror( pa_context_errno( context) ));
+ return;
+ }
+
+ _debug("- toPut : %i\n", toPut);
+ if( data != NULL )
+ _micRingBuffer.Put( (void*)data , toPut, 100);
+
+ if( pa_stream_drop( record->pulseStream()) < 0 )
+ _debug("pa_stream_drop() failed: %s\n" , pa_strerror( pa_context_errno( context) ));
+
+}
+
+ int
PulseLayer::putInCache( char code, void *buffer, int toCopy )
{
_debug("Put the DTMF in cache\n");
diff --git a/src/audio/pulselayer.h b/src/audio/pulselayer.h
index 947df7fca1..9e864caf37 100644
--- a/src/audio/pulselayer.h
+++ b/src/audio/pulselayer.h
@@ -76,6 +76,8 @@ class PulseLayer : public AudioLayer {
int playSamples(void* buffer, int toCopy, bool isTalking) ;
static void audioCallback ( pa_stream* s, size_t bytes, void* userdata );
+ static void overflow ( pa_stream* s, void* userdata );
+ static void underflow ( pa_stream* s, void* userdata );
static void context_state_callback( pa_context* c, void* user_data );
@@ -125,6 +127,7 @@ class PulseLayer : public AudioLayer {
void closeCaptureStream( void );
void write( void );
+ void read( void );
void createStreams( pa_context* c );
/**
* Drop the pending frames and close the playback device
@@ -136,6 +139,7 @@ class PulseLayer : public AudioLayer {
/** Ringbuffers for data */
RingBuffer _mainSndRingBuffer;
RingBuffer _urgentRingBuffer;
+ RingBuffer _micRingBuffer;
/** PulseAudio streams and context */
pa_context* context;
diff --git a/src/managerimpl.cpp b/src/managerimpl.cpp
index 90f17c965d..8053c79da3 100644
--- a/src/managerimpl.cpp
+++ b/src/managerimpl.cpp
@@ -497,11 +497,10 @@ ManagerImpl::playDtmf(char code, bool isTalking)
// put the size in bytes...
// so size * 1 channel (mono) * sizeof (bytes for the data)
#if CHECK_INTERFACE( layer , ALSA )
- _debug("%i No good\n", layer);
audiolayer->playSamples(_buf, size * sizeof(SFLDataFormat), isTalking);
#else
- _debug("%i Good\n" , layer);
- audiolayer->putUrgent( _buf, size * sizeof(SFLDataFormat) );
+ _debug("DTMF disabled\n");
+ //audiolayer->putUrgent( _buf, size * sizeof(SFLDataFormat) );
#endif
}
returnValue = true;
@@ -832,14 +831,13 @@ ManagerImpl::ringtone()
_audiofile.start();
_toneMutex.leaveMutex();
#if CHECK_INTERFACE( layer, ALSA )
- _debug()
int size = _audiofile.getSize();
SFLDataFormat output[ size ];
_audiofile.getNext(output, size , 100);
audiolayer->putUrgent( output , size );
#else
// pulseaudio code
- audiolayer->startStream();
+ //audiolayer->startStream();
#endif
} else {
ringback();
--
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