diff --git a/src/audio/audiolayer.cpp b/src/audio/audiolayer.cpp
index c5ba9570e5289fe37b6b150be333ea1264199bc3..b179db8002f26f7d6fff5c19b5911b7199e40828 100644
--- a/src/audio/audiolayer.cpp
+++ b/src/audio/audiolayer.cpp
@@ -28,11 +28,11 @@
#include "../global.h"
#include "../manager.h"
-AudioLayer::AudioLayer ()
+AudioLayer::AudioLayer(ManagerImpl& manager)
: _urgentRingBuffer(SIZEBUF)
, _mainSndRingBuffer(SIZEBUF)
, _micRingBuffer(SIZEBUF)
- , _stream(NULL)
+ , _stream(NULL), _manager(manager)
{
portaudio::System::initialize();
listDevices();
@@ -98,7 +98,7 @@ void
AudioLayer::startStream(void)
{
ost::MutexLock guard(_mutex);
- if (Manager::instance().isDriverLoaded()) {
+ if (_manager.isDriverLoaded()) {
if (_stream && !_stream->isActive()) {
_debug("Thread: start audiolayer stream\n");
_stream->start();
@@ -110,10 +110,11 @@ void
AudioLayer::stopStream(void)
{
ost::MutexLock guard(_mutex);
- if (Manager::instance().isDriverLoaded()) {
+ if (_manager.isDriverLoaded()) {
if (_stream && !_stream->isStopped()) {
_debug("Thread: stop audiolayer stream\n");
_stream->stop();
+ _mainSndRingBuffer.flush();
}
}
}
@@ -142,6 +143,14 @@ AudioLayer::putMain(void* buffer, int toCopy)
ost::MutexLock guard(_mutex);
_mainSndRingBuffer.Put(buffer, toCopy);
}
+
+void
+AudioLayer::flushMain()
+{
+ ost::MutexLock guard(_mutex);
+ _mainSndRingBuffer.flush();
+}
+
void
AudioLayer::putUrgent(void* buffer, int toCopy)
{
@@ -165,13 +174,14 @@ AudioLayer::audioCallback (const void *inputBuffer, void *outputBuffer,
unsigned long framesPerBuffer,
const PaStreamCallbackTimeInfo *timeInfo,
PaStreamCallbackFlags statusFlags) {
- (void) timeInfo;
- (void) statusFlags;
+
+ (void) timeInfo;
+ (void) statusFlags;
int16 *in = (int16 *) inputBuffer;
int16 *out = (int16 *) outputBuffer;
int toGet, toPut, urgentAvail, normalAvail, micAvailPut;
-
+
urgentAvail = _urgentRingBuffer.AvailForGet();
if (urgentAvail > 0) {
// Urgent data (dtmf, incoming call signal) come first.
@@ -180,11 +190,10 @@ AudioLayer::audioCallback (const void *inputBuffer, void *outputBuffer,
} else {
toGet = framesPerBuffer;
}
- _urgentRingBuffer.Get(out, SAMPLES_SIZE(toGet));
+ _urgentRingBuffer.Get(out, SAMPLES_SIZE(toGet), _manager.getSpkrVolume());
// Consume the regular one as well (same amount of bytes)
_mainSndRingBuffer.Discard(SAMPLES_SIZE(toGet));
-
}
else {
// If nothing urgent, play the regular sound samples
@@ -194,19 +203,18 @@ AudioLayer::audioCallback (const void *inputBuffer, void *outputBuffer,
//_debug("%d vs %d : %d\t", normalAvail, (int)framesPerBuffer, toGet);
if (toGet) {
- _mainSndRingBuffer.Get(out, SAMPLES_SIZE(toGet));
+ _mainSndRingBuffer.Get(out, SAMPLES_SIZE(toGet), _manager.getSpkrVolume());
} else {
toGet = SAMPLES_SIZE(framesPerBuffer);
_mainSndRingBuffer.PutZero(toGet);
- _mainSndRingBuffer.Get(out, toGet);
+ _mainSndRingBuffer.Get(out, toGet, 0);
}
}
// Additionally handle the mike's audio stream
micAvailPut = _micRingBuffer.AvailForPut();
- toPut = (micAvailPut <= (int)framesPerBuffer) ? micAvailPut :
- framesPerBuffer;
- _micRingBuffer.Put(in, SAMPLES_SIZE(toPut));
+ toPut = (micAvailPut <= (int)framesPerBuffer) ? micAvailPut : framesPerBuffer;
+ _micRingBuffer.Put(in, SAMPLES_SIZE(toPut), _manager.getMicVolume());
return paContinue;
}
diff --git a/src/audio/audiolayer.h b/src/audio/audiolayer.h
index 69a0d4db36fd64adeda82491c324e156a1e74452..9ed53488a7cbbbd7f06ee88155ea572dbf2962c7 100644
--- a/src/audio/audiolayer.h
+++ b/src/audio/audiolayer.h
@@ -29,6 +29,7 @@
#include "../global.h"
#include "ringbuffer.h"
#include <cc++/thread.h>
+#include "../manager.h"
#define FRAME_PER_BUFFER 160
#define MIC_CHANNELS 2 // 1=mono 2=stereo
@@ -38,10 +39,9 @@
class RingBuffer;
-
class AudioLayer {
public:
- AudioLayer();
+ AudioLayer(ManagerImpl& manager);
~AudioLayer (void);
void listDevices();
@@ -52,6 +52,7 @@ public:
bool isStreamActive (void);
bool isStreamStopped (void);
+ void flushMain();
void putMain(void* buffer, int toCopy);
void putUrgent(void* buffer, int toCopy);
@@ -59,7 +60,6 @@ public:
const PaStreamCallbackTimeInfo*, PaStreamCallbackFlags);
inline RingBuffer &urgentRingBuffer(void) { return _urgentRingBuffer; }
- inline RingBuffer &mainSndRingBuffer(void) { return _mainSndRingBuffer; }
inline RingBuffer &micRingBuffer(void) { return _micRingBuffer; }
private:
@@ -71,6 +71,7 @@ private:
portaudio::MemFunCallbackStream<AudioLayer> *_stream;
portaudio::AutoSystem autoSys;
ost::Mutex _mutex;
+ ManagerImpl& _manager;
};
#endif // _AUDIO_LAYER_H_
diff --git a/src/audio/audiortp.cpp b/src/audio/audiortp.cpp
index 03e1e7607afdbe9a2d17a960f5190c623574ab74..55f7f3bc270fe99e1785c438abae58ec1d224f01 100644
--- a/src/audio/audiortp.cpp
+++ b/src/audio/audiortp.cpp
@@ -29,7 +29,6 @@
#include "audiocodec.h"
#include "audiortp.h"
#include "audiolayer.h"
-#include "codecDescriptor.h"
#include "ringbuffer.h"
#include "../user_cfg.h"
#include "../sipcall.h"
@@ -73,29 +72,16 @@ AudioRtp::createNewSession (SipCall *ca) {
void
-AudioRtp::closeRtpSession (SipCall *ca) {
- // This will make RTP threads finish.
- if (ca) {
- if (ca->enable_audio > 0) {
- ca->enable_audio = -1;
-
- if (_RTXThread != NULL) {
- _debug("Thread: stop AudioRTP for sipcall: %d\n", ca->getId());
- delete _RTXThread; _RTXThread = NULL;
- }
- }
- } else {
- _debug("Thread: stop AudioRTP for no sipcall\n");
- delete _RTXThread; _RTXThread = NULL;
- }
+AudioRtp::closeRtpSession () {
+ // This will make RTP threads finish.
+ delete _RTXThread; _RTXThread = NULL;
}
////////////////////////////////////////////////////////////////////////////////
// AudioRtpRTX Class //
////////////////////////////////////////////////////////////////////////////////
-AudioRtpRTX::AudioRtpRTX (SipCall *sipcall,
- AudioLayer* driver,
- bool sym) {
+AudioRtpRTX::AudioRtpRTX (SipCall *sipcall, AudioLayer* driver,
+ bool sym) : _codecBuilder(0) {
time = new ost::Time();
_ca = sipcall;
_sym = sym;
@@ -123,6 +109,7 @@ AudioRtpRTX::~AudioRtpRTX () {
} catch (...) {
_debug("AudioRtpRTX: try to terminate, but catch an exception...\n");
}
+ _ca = NULL;
_debug("Thread: AudioRtpRTX stop session\n");
if (!_sym) {
delete _sessionRecv; _sessionRecv = NULL;
@@ -189,7 +176,8 @@ AudioRtpRTX::initAudioRtpSession (void)
}
void
-AudioRtpRTX::sendSessionFromMic (unsigned char* data_to_send, int16* data_from_mic, int16* data_from_mic_tmp, int timestamp, int micVolume)
+// AudioRtpRTX::sendSessionFromMic (unsigned char* data_to_send, int16* data_from_mic, int16* data_from_mic_tmp, int timestamp, int micVolume)
+AudioRtpRTX::sendSessionFromMic (unsigned char* data_to_send, int16* data_from_mic, int16* data_from_mic_tmp, int timestamp)
{
int k;
int compSize;
@@ -204,34 +192,34 @@ AudioRtpRTX::sendSessionFromMic (unsigned char* data_to_send, int16* data_from_m
}
// Get bytes from micRingBuffer to data_from_mic
- Manager::instance().getAudioDriver()->micRingBuffer().Get(data_from_mic,
- SAMPLES_SIZE(bytesAvail));
+ Manager::instance().getAudioDriver()->micRingBuffer().Get(data_from_mic, SAMPLES_SIZE(bytesAvail), 0);
// control volume and stereo->mono
for (int j = 0; j < RTP_FRAMES2SEND; j++) {
- k = j*2;
- data_from_mic_tmp[j] = (int16)(0.5f*(data_from_mic[k] +
- data_from_mic[k+1]) *
- micVolume/100);
+ k = j<<1;
+ data_from_mic_tmp[j] = (int16)(0.5f*(data_from_mic[k] + data_from_mic[k+1]));
+ //micVolume/100);
}
- // Encode acquired audio sample
- compSize = _ca->getAudioCodec()->codecEncode (data_to_send,
- data_from_mic_tmp,
- RTP_FRAMES2SEND*2);
- // Send encoded audio sample over the network
- if (!_sym) {
- _sessionSend->putData(timestamp, data_to_send, compSize);
- } else {
- _session->putData(timestamp, data_to_send, compSize);
- }
-
+ if ( _ca != NULL ) {
+ // Encode acquired audio sample
+ AudioCodec* ac = _ca->getAudioCodec();
+ if ( ac != NULL ) {
+ compSize = ac->codecEncode (data_to_send, data_from_mic_tmp, RTP_FRAMES2SEND*2);
+ // Send encoded audio sample over the network
+ if (!_sym) {
+ _sessionSend->putData(timestamp, data_to_send, compSize);
+ } else {
+ _session->putData(timestamp, data_to_send, compSize);
+ }
+ }
+ }
}
void
AudioRtpRTX::receiveSessionForSpkr (int16* data_for_speakers,
- int16* data_for_speakers_tmp, int spkrVolume, int& countTime)
+// int16* data_for_speakers_tmp, int spkrVolume, int& countTime)
+ int16* data_for_speakers_tmp, int& countTime)
{
- int expandedSize;
int k;
const ost::AppDataUnit* adu = NULL;
@@ -242,31 +230,33 @@ AudioRtpRTX::receiveSessionForSpkr (int16* data_for_speakers,
adu = _session->getData(_session->getFirstTimestamp());
}
if (adu == NULL) {
-// Manager::instance().getAudioDriver()->mainSndRingBuffer().flush();
-// Manager::instance().getAudioDriver()->stopStream();
+ Manager::instance().getAudioDriver()->flushMain();
return;
}
- // Decode data with relevant codec
- CodecDescriptor* cd = new CodecDescriptor (adu->getType());
-
- AudioCodec* ac = cd->alloc(adu->getType(), "");
+ int payload = adu->getType();
+ unsigned char* data = (unsigned char*)adu->getData();
+ unsigned int size = adu->getSize();
- expandedSize = ac->codecDecode (data_for_speakers,
- (unsigned char*) adu->getData(),
- adu->getSize());
-
- // control volume for speakers and mono->stereo
- for (int j = 0; j < expandedSize; j++) {
- k = j*2;
- data_for_speakers_tmp[k] = data_for_speakers_tmp[k+1]=
- data_for_speakers[j] * spkrVolume/100;
+ // Decode data with relevant codec
+ int expandedSize = 0;
+ AudioCodec* ac = _codecBuilder.alloc(payload, "");
+ if (ac != NULL) {
+ expandedSize = ac->codecDecode (data_for_speakers, data, size);
}
+ ac = NULL;
+
+ // control volume for speakers and mono->stereo
+ for (int j = 0; j < expandedSize; j++) {
+ k = j<<1; // fast multiply by two
+ data_for_speakers_tmp[k] = data_for_speakers_tmp[k+1] = data_for_speakers[j];
+ // * spkrVolume/100;
+ }
- // If the current call is the call which is answered
- // Set decoded data to sound device
- Manager::instance().getAudioDriver()->putMain(data_for_speakers_tmp, SAMPLES_SIZE(RTP_FRAMES2SEND));
- //}
+ // If the current call is the call which is answered
+ // Set decoded data to sound device
+ Manager::instance().getAudioDriver()->putMain(data_for_speakers_tmp, SAMPLES_SIZE(RTP_FRAMES2SEND));
+ //}
// Notify (with a beep) an incoming call when there is already a call
countTime += time->getSecond();
@@ -279,15 +269,13 @@ AudioRtpRTX::receiveSessionForSpkr (int16* data_for_speakers,
Manager::instance().getAudioDriver()->startStream();
- delete ac; ac = NULL;
- delete cd; cd = NULL;
delete adu; adu = NULL;
}
void
AudioRtpRTX::run (void) {
- int micVolume;
- int spkrVolume;
+// int micVolume;
+// int spkrVolume;
unsigned char *data_to_send;
int16 *data_from_mic;
int16 *data_from_mic_tmp;
@@ -314,7 +302,6 @@ AudioRtpRTX::run (void) {
// flush stream:
ManagerImpl& manager = Manager::instance();
AudioLayer *audiolayer = manager.getAudioDriver();
- audiolayer->mainSndRingBuffer().flush();
audiolayer->urgentRingBuffer().flush();
// start running the packet queue scheduler.
@@ -326,23 +313,23 @@ AudioRtpRTX::run (void) {
_session->startRunning();
}
- while (!testCancel() && _ca->enable_audio != -1) {
- micVolume = manager.getMicVolume();
- spkrVolume = manager.getSpkrVolume();
+ while (!testCancel() && _ca != NULL && _ca->enable_audio != -1) {
+ //micVolume = manager.getMicVolume();
+ //spkrVolume = manager.getSpkrVolume();
////////////////////////////
// Send session
////////////////////////////
- sendSessionFromMic(data_to_send, data_from_mic, data_from_mic_tmp,
- timestamp, micVolume);
+ //sendSessionFromMic(data_to_send, data_from_mic, data_from_mic_tmp, timestamp, micVolume);
+ sendSessionFromMic(data_to_send, data_from_mic, data_from_mic_tmp, timestamp);
timestamp += RTP_FRAMES2SEND;
////////////////////////////
// Recv session
////////////////////////////
- receiveSessionForSpkr(data_for_speakers, data_for_speakers_tmp,
- spkrVolume, countTime);
+ //receiveSessionForSpkr(data_for_speakers, data_for_speakers_tmp, spkrVolume, countTime);
+ receiveSessionForSpkr(data_for_speakers, data_for_speakers_tmp, countTime);
// Let's wait for the next transmit cycle
Thread::sleep(TimerPort::getTimer());
@@ -356,8 +343,6 @@ AudioRtpRTX::run (void) {
delete [] data_from_mic; data_from_mic = 0;
audiolayer->stopStream();
- audiolayer->mainSndRingBuffer().flush();
- //audiolayer->urgentRingBuffer().flush();
}
diff --git a/src/audio/audiortp.h b/src/audio/audiortp.h
index 273136cdd6ae72ed4e0297f79bf7492632c6e1c5..716296f5eb1b9a8be0f1c649d91bb1f62c683559 100644
--- a/src/audio/audiortp.h
+++ b/src/audio/audiortp.h
@@ -25,6 +25,7 @@
#include <ccrtp/rtp.h>
#include <cc++/numbers.h>
+#include "codecDescriptor.h"
#define RTP_FRAMES2SEND 160
@@ -52,9 +53,14 @@ private:
ost::SymmetricRTPSession *_session;
bool _sym;
+ // build codec...
+ CodecDescriptor _codecBuilder;
+
void initAudioRtpSession (void);
- void sendSessionFromMic (unsigned char*, int16*, int16*, int, int);
- void receiveSessionForSpkr (int16*, int16*, int, int&);
+// void sendSessionFromMic (unsigned char*, int16*, int16*, int, int);
+ void sendSessionFromMic (unsigned char*, int16*, int16*, int);
+// void receiveSessionForSpkr (int16*, int16*, int, int&);
+ void receiveSessionForSpkr (int16*, int16*, int&);
};
///////////////////////////////////////////////////////////////////////////////
@@ -66,7 +72,7 @@ public:
~AudioRtp();
int createNewSession (SipCall *);
- void closeRtpSession (SipCall *);
+ void closeRtpSession ();
private:
AudioRtpRTX* _RTXThread;
diff --git a/src/audio/codecDescriptor.cpp b/src/audio/codecDescriptor.cpp
index 7c64f397a793ca191011af361b13e62ec35efd5e..9e5feae22bb35eebff797bf9934da20a62119a47 100644
--- a/src/audio/codecDescriptor.cpp
+++ b/src/audio/codecDescriptor.cpp
@@ -46,36 +46,63 @@ CodecDescriptor::CodecDescriptor (int payload)
{
_payload = payload;
_codecName = rtpmapPayload(_payload);
+ initCache();
}
CodecDescriptor::CodecDescriptor (const std::string& name)
{
_codecName = name;
_payload = matchPayloadCodec(name);
+ initCache();
}
CodecDescriptor::CodecDescriptor (int payload, const std::string& name)
{
_payload = payload;
_codecName = name;
+ initCache();
}
-CodecDescriptor::~CodecDescriptor (void)
+void CodecDescriptor::initCache()
{
+ _ac1 = NULL;
+ _ac2 = NULL;
+ _ac3 = NULL;
+}
+
+
+CodecDescriptor::~CodecDescriptor (void)
+{
+ delete _ac1;
+ delete _ac2;
+ delete _ac3;
}
AudioCodec*
CodecDescriptor::alloc (int payload, const std::string& name)
{
+ // _ac1, _ac2 and _ac3 are caching...
switch(payload) {
case PAYLOAD_CODEC_ULAW:
- return new Ulaw(payload, name);
+ if ( _ac1 == NULL ) {
+ _ac1 = new Ulaw(payload, name);
+ //return new Ulaw(payload, name);
+ }
+ return _ac1;
break;
case PAYLOAD_CODEC_ALAW:
- return new Alaw(payload, name);
+ if ( _ac2 == NULL ) {
+ _ac2 = new Alaw(payload, name);
+ //return new Alaw(payload, name);
+ }
+ return _ac2;
break;
case PAYLOAD_CODEC_GSM:
- return new Gsm(payload, name);
+ if ( _ac3 == NULL ) {
+ _ac3 = new Gsm(payload, name);
+ //return new Gsm(payload, name);
+ }
+ return _ac3;
break;
default:
return NULL;
diff --git a/src/audio/codecDescriptor.h b/src/audio/codecDescriptor.h
index 4ca94aea42402656a83343789ac7829d654ead1b..db8e6a96dd3710f3f667bea7c891ecde70815d51 100644
--- a/src/audio/codecDescriptor.h
+++ b/src/audio/codecDescriptor.h
@@ -77,8 +77,13 @@ public:
std::string rtpmapPayload (int);
private:
+ void initCache();
int _payload;
std::string _codecName;
+
+ AudioCodec* _ac1;
+ AudioCodec* _ac2;
+ AudioCodec* _ac3;
};
#endif // __CODEC_DESCRIPTOR_H__
diff --git a/src/audio/gsmcodec.cpp b/src/audio/gsmcodec.cpp
index 36767d1c410c18f8e7fb6bd99c8ad1273e20481b..60d54eda887faf917524db2bd74780476aaec9ae 100644
--- a/src/audio/gsmcodec.cpp
+++ b/src/audio/gsmcodec.cpp
@@ -36,7 +36,6 @@ Gsm::~Gsm (void)
{
gsm_destroy(_decode_gsmhandle);
gsm_destroy(_encode_gsmhandle);
-
}
int
diff --git a/src/audio/ringbuffer.cpp b/src/audio/ringbuffer.cpp
index 54168f31dce3783f25e9ced2738a75f4d596a225..c7825b35ff7271c41cd16b63f0f18960b5859d17 100644
--- a/src/audio/ringbuffer.cpp
+++ b/src/audio/ringbuffer.cpp
@@ -50,10 +50,8 @@ RingBuffer::~RingBuffer() {
void
RingBuffer::flush (void) {
- mMutex.enterMutex();
mStart = 0;
mEnd = 0;
- mMutex.leaveMutex();
}
int
@@ -78,19 +76,19 @@ RingBuffer::PutZero(int toZero)
}
// This one puts some data inside the ring buffer.
+// Change the volume if it's not 0
int
-RingBuffer::Put(void* buffer, int toCopy) {
+RingBuffer::Put(void* buffer, int toCopy, unsigned short volume) {
samplePtr src;
int block;
int copied;
int pos;
int len = Len();
- mMutex.enterMutex();
if (toCopy > (mBufferSize-4) - len)
toCopy = (mBufferSize-4) - len;
- src = (samplePtr) buffer;
+ src = (samplePtr) buffer;
copied = 0;
pos = mEnd;
@@ -98,9 +96,15 @@ RingBuffer::Put(void* buffer, int toCopy) {
while(toCopy) {
block = toCopy;
if (block > mBufferSize - pos) // from current pos. to end of buffer
- block = mBufferSize - pos;
+ block = mBufferSize - pos;
// put the data inside the buffer.
+ if (volume) {
+ int16* src16 = (int16*)src;
+ int int16len = (block >> 1);
+ for (int i=0; i< int16len; i++) { src16[i] = src16[i] * volume / 100; }
+ }
+ // bcopy(src, dest, len)
bcopy (src, mBuffer + pos, block);
src += block;
@@ -110,7 +114,6 @@ RingBuffer::Put(void* buffer, int toCopy) {
}
mEnd = pos;
- mMutex.leaveMutex();
// How many items copied.
return copied;
@@ -128,13 +131,14 @@ RingBuffer::AvailForGet() const {
// Get will move 'toCopy' bytes from the internal FIFO to 'buffer'
int
-RingBuffer::Get(void *buffer, int toCopy) {
+RingBuffer::Get(void *buffer, int toCopy, unsigned short volume) {
samplePtr dest;
int block;
int copied;
int len = Len();
-
- mMutex.enterMutex();
+ int int16len;
+ int16* start;
+
if (toCopy > len)
toCopy = len;
@@ -146,14 +150,17 @@ RingBuffer::Get(void *buffer, int toCopy) {
if (block > (mBufferSize - mStart))
block = mBufferSize - mStart;
- bcopy (mBuffer + mStart, dest, block);
+ start = (int16*)(mBuffer + mStart);
+ int16len = (block >> 1);
+ for (int i=0; i<int16len; i++) { start[i] = start[i] * volume / 100; }
+ // bcopy(src, dest, len)
+ bcopy (start, dest, block);
dest += block;
mStart = (mStart + block) % mBufferSize;
toCopy -= block;
copied += block;
}
- mMutex.leaveMutex();
return copied;
}
@@ -169,10 +176,3 @@ RingBuffer::Discard(int toDiscard) {
return toDiscard;
}
-
-/*
-T
-RingBuffer::getNextItem (void) {
- return (T) 0;
-}
-*/
diff --git a/src/audio/ringbuffer.h b/src/audio/ringbuffer.h
index c18f82b08f614608ecc41018bee594c44ae0bfeb..9e1ef6bef4177f45a0f92e7ef5e9d59982fcc411 100644
--- a/src/audio/ringbuffer.h
+++ b/src/audio/ringbuffer.h
@@ -42,14 +42,14 @@ class RingBuffer {
// For the writer only:
//
int AvailForPut (void) const;
- int Put (void*, int);
+ int Put (void*, int, unsigned short = 0);
void PutZero(int);
//
// For the reader only:
//
int AvailForGet (void) const;
- int Get (void *, int);
+ int Get (void *, int, unsigned short);
int Discard(int);
int Len() const;
@@ -57,7 +57,6 @@ class RingBuffer {
private:
// T getNextSample(void);
- ost::Mutex mMutex;
int mStart;
int mEnd;
int mBufferSize;
diff --git a/src/audio/tonegenerator.cpp b/src/audio/tonegenerator.cpp
index c7b1488e4a2833b6e139fb3bb837941d4154d8ff..7a430f38179c90e1ec9f3355a75af7a76ff5061c 100644
--- a/src/audio/tonegenerator.cpp
+++ b/src/audio/tonegenerator.cpp
@@ -56,7 +56,7 @@ ToneThread::~ToneThread (void) {
void
ToneThread::run (void) {
int k;
- int spkrVolume;
+ //int spkrVolume;
bool started = false;
// How long do 'size' samples play ?
@@ -64,14 +64,16 @@ ToneThread::run (void) {
unsigned int pause = (size * 1000) / SAMPLING_RATE - 100;
ManagerImpl& manager = Manager::instance();
- manager.getAudioDriver()->mainSndRingBuffer().flush();
+ manager.getAudioDriver()->flushMain();
while (!testCancel()) {
// Create a new stereo buffer with the volume adjusted
- spkrVolume = manager.getSpkrVolume();
+ //spkrVolume = manager.getSpkrVolume();
for (int j = 0; j < size; j++) {
- k = j*2; // channels is 2 (global.h)
- buf_ctrl_vol[k] = buf_ctrl_vol[k+1] = buffer[j] * spkrVolume/100;
+ k = j<<1; // channels is 2 (global.h)
+ // split in two
+ buf_ctrl_vol[k] = buf_ctrl_vol[k+1] = buffer[j];
+ // * spkrVolume/100;
}
// Push the tone to the audio FIFO
diff --git a/src/call.cpp b/src/call.cpp
index 3804e160d325db2c497c8c13d0f714c192bc4dfc..e1b74afc2c8419b589d48bf72f831590fdf37e51 100644
--- a/src/call.cpp
+++ b/src/call.cpp
@@ -185,7 +185,7 @@ Call::isAnswered (void)
bool
Call::isNotAnswered (void)
{
- return (_state == Error || _state == NotExist ) ? true : false;
+ return (_state == Error || _state == NotExist || _state == Busy) ? true : false;
}
bool
diff --git a/src/gui/server/tcpstreampool.h b/src/gui/server/tcpstreampool.h
index 46f8e90af0d61e9389f48879652d0c59d8fe3d98..6bec3bd783f1fec6ae55c9071bbace7356084457 100644
--- a/src/gui/server/tcpstreampool.h
+++ b/src/gui/server/tcpstreampool.h
@@ -43,6 +43,7 @@ public:
{
setCancel(cancelDeferred);
}
+ TCPStreamPool::~TCPStreamPool() {terminate();}
void run();
void send(const std::string& response);
diff --git a/src/managerimpl.cpp b/src/managerimpl.cpp
index 3656c65cd71578dbea9ab7958fb76a9a4764d098..7716a8e8e894de15f0ec1c75899867e0351087dc 100644
--- a/src/managerimpl.cpp
+++ b/src/managerimpl.cpp
@@ -336,7 +336,7 @@ ManagerImpl::onHoldCall (CALLID id)
if (call == NULL) {
return -1;
}
- if ( call->getState() == Call::OnHold) {
+ if ( call->getState() == Call::OnHold || call->isNotAnswered()) {
return 1;
}
return call->onHold();
@@ -349,6 +349,7 @@ ManagerImpl::onHoldCall (CALLID id)
int
ManagerImpl::offHoldCall (CALLID id)
{
+ stopTone();
ost::MutexLock m(_mutex);
Call* call = getCall(id);
if (call == NULL) {
@@ -480,6 +481,7 @@ ManagerImpl::sendDtmf (CALLID id, char code)
int returnValue = false;
switch (sendType) {
case 0: // SIP INFO
+ playDtmf(code);
_voIPLinkVector.at(DFT_VOIP_LINK)->carryingDTMFdigits(id, code);
returnValue = true;
break;
@@ -509,7 +511,8 @@ ManagerImpl::playDtmf(char code)
_key.startTone(code);
if ( _key.generateDTMF(_buf, SAMPLING_RATE) ) {
- int k, spkrVolume;
+ int k;
+ //int spkrVolume;
int16* buf_ctrl_vol;
// Determine dtmf pulse length
@@ -517,12 +520,13 @@ ManagerImpl::playDtmf(char code)
int size = pulselen * (OCTETS /1000);
buf_ctrl_vol = new int16[size*CHANNELS];
- spkrVolume = getSpkrVolume();
+ //spkrVolume = getSpkrVolume();
// Control volume and format mono->stereo
for (int j = 0; j < size; j++) {
- k = j*2;
- buf_ctrl_vol[k] = buf_ctrl_vol[k+1] = _buf[j] * spkrVolume/100;
+ k = j<<1; // fast multiply by two
+ buf_ctrl_vol[k] = buf_ctrl_vol[k+1] = _buf[j];
+ // * spkrVolume/100;
}
AudioLayer *audiolayer = getAudioDriver();
@@ -815,7 +819,6 @@ ManagerImpl::stopTone() {
_tone->stopTone();
}
_toneMutex.leaveMutex();
- getAudioDriver()->mainSndRingBuffer().flush();
getAudioDriver()->stopStream();
}
}
@@ -899,16 +902,18 @@ ManagerImpl::notificationIncomingCall (void) {
int16* buf_ctrl_vol;
int16* buffer = new int16[SAMPLING_RATE];
int size = SAMPLES_SIZE(FRAME_PER_BUFFER);//SAMPLING_RATE/2;
- int k, spkrVolume;
+ int k;
+ //int spkrVolume;
_tone->generateSin(440, 0, buffer);
// Volume Control
buf_ctrl_vol = new int16[size*CHANNELS];
- spkrVolume = getSpkrVolume();
+ // spkrVolume = getSpkrVolume();
for (int j = 0; j < size; j++) {
- k = j*2;
- buf_ctrl_vol[k] = buf_ctrl_vol[k+1] = buffer[j] * spkrVolume/100;
+ k = j<<1; // fast multiply by two
+ buf_ctrl_vol[k] = buf_ctrl_vol[k+1] = buffer[j];
+ // * spkrVolume/100;
}
getAudioDriver()->putUrgent(buf_ctrl_vol, SAMPLES_SIZE(FRAME_PER_BUFFER));
@@ -1083,7 +1088,7 @@ ManagerImpl::selectAudioDriver (void)
{
#if defined(AUDIO_PORTAUDIO)
try {
- _audiodriverPA = new AudioLayer();
+ _audiodriverPA = new AudioLayer(*this);
_audiodriverPA->openDevice(getConfigInt(AUDIO, DRIVER_NAME));
} catch(...) {
throw;
diff --git a/src/managerimpl.h b/src/managerimpl.h
index 3a631a9b24e7cf824e9869af74085817dddba93d..cc5050bbf1b7d0fa52007c37c26d1e02591f97e5 100644
--- a/src/managerimpl.h
+++ b/src/managerimpl.h
@@ -222,13 +222,27 @@ name);
void getStunInfo (StunAddress4& stunSvrAddr);
bool useStun (void);
- /*
- * Inline functions to manage volume control
- */
- inline int getSpkrVolume (void) { ost::MutexLock m(_mutex); return _spkr_volume; }
- inline void setSpkrVolume (int spkr_vol) { ost::MutexLock m(_mutex); _spkr_volume = spkr_vol; }
- inline int getMicVolume (void) { ost::MutexLock m(_mutex); return _mic_volume; }
- inline void setMicVolume (int mic_vol) { ost::MutexLock m(_mutex); _mic_volume = _mic_volume_before_mute = mic_vol; }
+ /*
+ * Inline functions to manage volume control
+ * Read by main thread and AudioLayer thread
+ * Write by main thread only
+ */
+ unsigned short getSpkrVolume(void) {
+ //ost::MutexLock m(_mutex);
+ return _spkr_volume;
+ }
+ void setSpkrVolume(unsigned short spkr_vol) {
+ //ost::MutexLock m(_mutex);
+ _spkr_volume = spkr_vol;
+ }
+ unsigned short getMicVolume(void) {
+ //ost::MutexLock m(_mutex);
+ return _mic_volume;
+ }
+ void setMicVolume(unsigned short mic_vol) {
+ //ost::MutexLock m(_mutex);
+ _mic_volume = _mic_volume_before_mute = mic_vol;
+ }
bool hasLoadedSetup() { return _setupLoaded; }
@@ -378,10 +392,10 @@ private:
unsigned int _nCodecs;
- // To handle volume control
- int _spkr_volume;
- int _mic_volume;
- int _mic_volume_before_mute;
+ // To handle volume control
+ short _spkr_volume;
+ short _mic_volume;
+ short _mic_volume_before_mute;
// To handle firewall
int _firewallPort;
diff --git a/src/sipcall.cpp b/src/sipcall.cpp
index 2028c1481fb2a02a3445ba30b1cda76e87ee30f0..02ce78d86b61823d1dd78e30dc7a7a35e278d71b 100644
--- a/src/sipcall.cpp
+++ b/src/sipcall.cpp
@@ -44,7 +44,7 @@ SipCall::SipCall (CALLID id, CodecDescriptorVector* cdv) : _localIp("127.0.0.1")
alloc(); // char* allocation
_cdv = cdv;
_audiocodec = NULL;
- enable_audio = -1;
+ enable_audio = false;
_state = 0;
_local_audio_port = 0;
@@ -57,7 +57,8 @@ SipCall::SipCall (CALLID id, CodecDescriptorVector* cdv) : _localIp("127.0.0.1")
SipCall::~SipCall (void)
{
dealloc();
- delete _audiocodec; _audiocodec = NULL;
+ //delete _audiocodec; don't delete it, the Manager will do it...
+ _audiocodec = NULL;
}
void
@@ -142,7 +143,8 @@ void
SipCall::setAudioCodec (AudioCodec* ac)
{
// it use a new!
- delete _audiocodec; _audiocodec = ac;
+ // delete _audiocodec;
+ _audiocodec = ac;
}
// newIncomingCall is called when the IP-Phone user receives a new call.
@@ -253,10 +255,10 @@ SipCall::newIncomingCall (eXosip_event_t *event) {
eXosip_call_send_answer (_tid, 415, NULL);
} else {
/* start sending audio */
- if (enable_audio > 0) {
- enable_audio = -1;
+ if (enable_audio == true) {
+ enable_audio = false;
}
- if (enable_audio != 1) /* audio is started */ {
+ if (enable_audio == false) /* audio is started */ {
sdp_message_t *local_sdp;
local_sdp = eXosip_get_sdp_info (answer);
if (remote_sdp != NULL && local_sdp != NULL) {
@@ -284,12 +286,12 @@ SipCall::newIncomingCall (eXosip_event_t *event) {
&& local_med != NULL) {
tmp = (char *) osip_list_get (local_med->m_payloads, 0);
}
- if (tmp != NULL) {
- payload = atoi (tmp);
- _debug("SipCall::newIncomingCall: For incoming payload = %d\n", payload);
- setAudioCodec(_cdv->at(0)->alloc(payload, ""));
- }
- if (tmp != NULL
+ if (tmp != NULL) {
+ payload = atoi (tmp);
+ _debug("SipCall::newIncomingCall: For incoming payload = %d\n", payload);
+ setAudioCodec(_cdv->at(0)->alloc(payload, "")); // codec builder for the mic
+ }
+ if (tmp != NULL
&& audio_port > 0
&& _remote_sdp_audio_port > 0
&& _remote_sdp_audio_ip[0] != '\0') {
@@ -353,64 +355,6 @@ SipCall::ringingCall (eXosip_event_t *event) {
osip_free (tmp);
}
}
-/*
- sdp_message_t *remote_sdp;
- sdp_message_t *local_sdp;
-
- local_sdp = eXosip_get_sdp_info (event->request);
- remote_sdp = eXosip_get_sdp_info (event->response);
- if (remote_sdp == NULL) {
- _debug("SipCall::ringingCall: No remote SDP body found for call\n");
- // TODO: remote_sdp = retreive from ack above
- }
- if (local_sdp == NULL) {
- _debug("SipCall::ringingCall: SDP body was probably in the ACK (TODO)\n");
- }
- if (remote_sdp != NULL && local_sdp != NULL) {
- sdp_connection_t *conn;
- sdp_media_t *local_med;
- sdp_media_t *remote_med;
- char *tmp = NULL;
- int audio_port = 0;
-
- conn = eXosip_get_audio_connection (remote_sdp);
- if (conn != NULL && conn->c_addr != NULL) {
- snprintf (_remote_sdp_audio_ip, 49, "%s", conn->c_addr);
- }
- remote_med = eXosip_get_audio_media (remote_sdp);
- if (remote_med != NULL && remote_med->m_port != NULL) {
- _remote_sdp_audio_port = atoi (remote_med->m_port);
- }
- local_med = eXosip_get_audio_media (local_sdp);
- if (local_med != NULL && local_med->m_port != NULL) {
- audio_port = atoi (local_med->m_port);
- }
- if (_remote_sdp_audio_port > 0 && _remote_sdp_audio_ip[0] != '\0'
- && remote_med != NULL) {
- tmp = (char *) osip_list_get (remote_med->m_payloads, 0);
- }
- if (tmp != NULL) {
- ca->payload = atoi (tmp);
- }
- if (tmp != NULL
- && audio_port > 0
- && _remote_sdp_audio_port > 0
- && _remote_sdp_audio_ip[0] != '\0') {
- // search if stream is sendonly or recvonly
- _remote_sendrecv =
- sdp_analyse_attribute (remote_sdp, remote_med);
- _local_sendrecv = sdp_analyse_attribute (local_sdp, local_med);
- if (_local_sendrecv == _SENDRECV) {
- if (_remote_sendrecv == _SENDONLY)
- _local_sendrecv = _RECVONLY;
- else if (_remote_sendrecv == _RECVONLY)
- _local_sendrecv = _SENDONLY;
- }
- }
- }
- sdp_message_free (local_sdp);
- sdp_message_free (remote_sdp);
-*/
this->_state = event->type;;
return 0;
}
@@ -420,74 +364,6 @@ SipCall::receivedAck (eXosip_event_t *event)
{
_cid = event->cid;
_did = event->did;
-/*
- if (event->ack != NULL) {
- sdp_message_t *remote_sdp;
- remote_sdp = eXosip_get_sdp_info (event->ack);
- if (remote_sdp != NULL) {
- _debug("SipCall::receivedAck: SDP detected in ACK!\n");
- } else {
- _debug("SipCall::receivedAck: no SDP detected in ACK!\n");
- }
- }
-
- if (enable_audio != 1) { // audio is started
- sdp_message_t *remote_sdp;
- sdp_message_t *local_sdp;
-
- remote_sdp = eXosip_get_remote_sdp (_did);
- local_sdp = eXosip_get_local_sdp (_did);
- if (remote_sdp == NULL) {
- _debug("SipCall::receivedAck: No remote SDP body found for call\n");
- }
- if (remote_sdp != NULL && local_sdp != NULL) {
- sdp_connection_t *conn;
- sdp_media_t *local_med;
- sdp_media_t *remote_med;
- char *tmp = NULL;
- int audio_port = 0;
-
- conn = eXosip_get_audio_connection (remote_sdp);
- if (conn != NULL && conn->c_addr != NULL) {
- snprintf (_remote_sdp_audio_ip, 49, "%s", conn->c_addr);
- }
- remote_med = eXosip_get_audio_media (remote_sdp);
- if (remote_med != NULL && remote_med->m_port != NULL) {
- _remote_sdp_audio_port = atoi (remote_med->m_port);
- }
- local_med = eXosip_get_audio_media (local_sdp);
- if (local_med != NULL && local_med->m_port != NULL) {
- audio_port = atoi (local_med->m_port);
- }
-
- if (_remote_sdp_audio_port > 0
- && _remote_sdp_audio_ip[0] != '\0' && local_med != NULL) {
- tmp = (char *) osip_list_get (local_med->m_payloads, 0);
- }
- if (tmp != NULL) {
- ca->payload = atoi (tmp);
- }
- if (tmp != NULL
- && audio_port > 0
- && _remote_sdp_audio_port > 0
- && _remote_sdp_audio_ip[0] != '\0') {
-
- // search if stream is sendonly or recvonly
- _remote_sendrecv =
- sdp_analyse_attribute (remote_sdp, remote_med);
- _local_sendrecv = sdp_analyse_attribute (local_sdp, local_med);
- if (_local_sendrecv == _SENDRECV) {
- if (_remote_sendrecv == _SENDONLY)
- _local_sendrecv = _RECVONLY;
- else if (_remote_sendrecv == _RECVONLY)
- _local_sendrecv = _SENDONLY;
- }
- }
- }
- sdp_message_free (local_sdp);
- sdp_message_free (remote_sdp);
- }
-*/
_state = event->type;
return 0;
}
@@ -557,16 +433,16 @@ SipCall::answeredCall(eXosip_event_t *event) {
void
SipCall::answeredCall_without_hold (eXosip_event_t *event)
{
- if (enable_audio == 1 && event->response != NULL) {
+ if (enable_audio == true && event->response != NULL) {
sdp_message_t *sdp = eXosip_get_sdp_info (event->response);
if (sdp != NULL) {
/* audio is started and session has just been modified */
- enable_audio = -1;
+ enable_audio = false;
sdp_message_free (sdp);
}
}
- if (enable_audio != 1 &&
+ if (enable_audio == false &&
event->request != NULL &&
event->response != NULL) { /* audio is started */
diff --git a/src/sipcall.h b/src/sipcall.h
index 18cb3864ae54fbde343acc1ec2f2bf92089a903a..95626bc9636b5dfd5c0d8d666229247f599028b8 100644
--- a/src/sipcall.h
+++ b/src/sipcall.h
@@ -44,7 +44,7 @@ public:
~SipCall (void);
int payload;
- int enable_audio; /* 1 started, -1 stopped */
+ bool enable_audio; /* true = started, false = stopped */
/*
* Store information about incoming call and negociate payload
diff --git a/src/sipvoiplink.cpp b/src/sipvoiplink.cpp
index e187dea7728c22e32c9e7b1a3164317fba070bab..b74a707a4ace39f86bee46d20828ce55a599b60d 100644
--- a/src/sipvoiplink.cpp
+++ b/src/sipvoiplink.cpp
@@ -379,11 +379,16 @@ SipVoIPLink::answer (CALLID id)
return i;
}
+
+/**
+ * @return > 0 is good, -1 is bad
+ */
int
SipVoIPLink::hangup (CALLID id)
{
int i = 0;
SipCall* sipcall = getSipCall(id);
+ if (sipcall == NULL) { return -1; }
_debug("Hang up call [id = %d, cid = %d, did = %d]\n",
id, sipcall->getCid(), sipcall->getDid());
// Release SIP stack.
@@ -392,7 +397,8 @@ SipVoIPLink::hangup (CALLID id)
eXosip_unlock();
// Release RTP channels
- _audiortp.closeRtpSession(sipcall);
+ sipcall->enable_audio = false;
+ _audiortp.closeRtpSession();
deleteSipCall(id);
return i;
@@ -413,6 +419,9 @@ SipVoIPLink::cancel (CALLID id)
return i;
}
+/*
+ * @return -1 = sipcall not present
+ */
int
SipVoIPLink::onhold (CALLID id)
{
@@ -422,12 +431,15 @@ SipVoIPLink::onhold (CALLID id)
sdp_message_t *local_sdp = NULL;
- did = getSipCall(id)->getDid();
-
+ SipCall *sipcall = getSipCall(id);
+ if ( sipcall == NULL ) { return -1; }
+
+ did = sipcall->getDid();
+
eXosip_lock ();
local_sdp = eXosip_get_local_sdp (did);
eXosip_unlock ();
-
+
if (local_sdp == NULL) {
return -1;
}
@@ -445,7 +457,7 @@ SipVoIPLink::onhold (CALLID id)
/* add sdp body */
{
char *tmp = NULL;
-
+
i = sdp_hold_call (local_sdp);
if (i != 0) {
sdp_message_free (local_sdp);
@@ -471,7 +483,8 @@ SipVoIPLink::onhold (CALLID id)
eXosip_unlock ();
// Disable audio
- _audiortp.closeRtpSession(getSipCall(id));
+ sipcall->enable_audio = false;
+ _audiortp.closeRtpSession();
return i;
}
@@ -740,7 +753,9 @@ SipVoIPLink::getEvent (void)
_debug("Call is closed [id = %d, cid = %d, did = %d]\n", id, event->cid, event->did);
if (id != 0) {
if (Manager::instance().callCanBeClosed(id)) {
- _audiortp.closeRtpSession(getSipCall(id));
+ SipCall* sipcall = getSipCall(id);
+ if ( sipcall != NULL ) { sipcall->enable_audio = false; }
+ _audiortp.closeRtpSession();
}
Manager::instance().peerHungupCall(id);
deleteSipCall(id);
@@ -1005,7 +1020,7 @@ SipVoIPLink::endSipCalls()
eXosip_unlock();
// Release RTP channels
- _audiortp.closeRtpSession(*iter);
+ _audiortp.closeRtpSession();
delete *iter; *iter = NULL;
}
iter++;