From d82299d385915b6e492ce951bffef1d23aa2a6dd Mon Sep 17 00:00:00 2001
From: yanmorin <yanmorin>
Date: Tue, 18 Oct 2005 19:36:02 +0000
Subject: [PATCH] I tried to fix the sound, but nah, it's noisy...
---
src/audio/audiolayer.cpp | 42 +++++++++++++++++++------------------
src/audio/ringbuffer.cpp | 2 ++
src/audio/tonegenerator.cpp | 11 ++++------
src/global.h | 3 ---
src/managerimpl.cpp | 38 ++++++++++++++++++++-------------
5 files changed, 51 insertions(+), 45 deletions(-)
diff --git a/src/audio/audiolayer.cpp b/src/audio/audiolayer.cpp
index 7cb08167c5..76aaa634fd 100644
--- a/src/audio/audiolayer.cpp
+++ b/src/audio/audiolayer.cpp
@@ -35,7 +35,6 @@ AudioLayer::AudioLayer(ManagerImpl& manager)
, _stream(NULL), _manager(manager)
{
portaudio::System::initialize();
- listDevices();
}
// Destructor
@@ -113,7 +112,7 @@ AudioLayer::stopStream(void)
if (_manager.isDriverLoaded()) {
if (_stream && !_stream->isStopped()) {
_debug("Thread: stop audiolayer stream\n");
- _stream->stop();
+ _stream->stop();
_mainSndRingBuffer.flush();
}
}
@@ -180,41 +179,44 @@ AudioLayer::audioCallback (const void *inputBuffer, void *outputBuffer,
int16 *in = (int16 *) inputBuffer;
int16 *out = (int16 *) outputBuffer;
- int toGet, toPut, urgentAvail, normalAvail, micAvailPut;
-
- urgentAvail = _urgentRingBuffer.AvailForGet();
+ int toGet, toPut;
+ int urgentAvail, // number of int16 right and int16 left
+ normalAvail, // number of int16 right and int16 left
+ micAvailPut;
+
+ // AvailForGet tell the number of chars inside the buffer
+ // framePerBuffer are the number of int16 for one channel (left)
+ // so we divise by short/char * 2 channels
+ int NBCHARFORTWOINT16 = sizeof(int16)/sizeof(char) * CHANNELS;
+ urgentAvail = _urgentRingBuffer.AvailForGet() / NBCHARFORTWOINT16;
if (urgentAvail > 0) {
// Urgent data (dtmf, incoming call signal) come first.
- if (urgentAvail < (int)framesPerBuffer) {
- toGet = urgentAvail;
- } else {
- toGet = framesPerBuffer;
- }
- _urgentRingBuffer.Get(out, SAMPLES_SIZE(toGet), _manager.getSpkrVolume());
+ toGet = (urgentAvail < (int)framesPerBuffer) ? urgentAvail : framesPerBuffer;
+ _urgentRingBuffer.Get(out, toGet * NBCHARFORTWOINT16, _manager.getSpkrVolume());
// Consume the regular one as well (same amount of bytes)
- _mainSndRingBuffer.Discard(SAMPLES_SIZE(toGet));
+ _mainSndRingBuffer.Discard(toGet * NBCHARFORTWOINT16);
}
else {
// If nothing urgent, play the regular sound samples
- normalAvail = _mainSndRingBuffer.AvailForGet() / (MIC_CHANNELS * SAMPLE_BYTES);
+ normalAvail = _mainSndRingBuffer.AvailForGet() / NBCHARFORTWOINT16;
toGet = (normalAvail < (int)framesPerBuffer) ? normalAvail : framesPerBuffer;
- // MIC_CHANNELS * SAMPLE_BYTES
//_debug("mainsndringbuffer.get: %d vs %d : %d\n", normalAvail, (int)framesPerBuffer, toGet);
if (toGet) {
- _mainSndRingBuffer.Get(out, SAMPLES_SIZE(toGet), _manager.getSpkrVolume());
+ _mainSndRingBuffer.Get(out, toGet*NBCHARFORTWOINT16, _manager.getSpkrVolume());
} else {
- toGet = SAMPLES_SIZE(framesPerBuffer);
+ toGet = framesPerBuffer * NBCHARFORTWOINT16;
_mainSndRingBuffer.PutZero(toGet);
_mainSndRingBuffer.Get(out, toGet, 100);
}
}
- // Additionally handle the mike's audio stream
- micAvailPut = _micRingBuffer.AvailForPut();
- toPut = (micAvailPut <= (int)framesPerBuffer) ? micAvailPut : framesPerBuffer;
- _micRingBuffer.Put(in, SAMPLES_SIZE(toPut), _manager.getMicVolume());
+ // Additionally handle the mic's audio stream
+ short micVolume = _manager.getMicVolume();
+ micAvailPut = _micRingBuffer.AvailForPut();
+ toPut = (micAvailPut <= (int)framesPerBuffer) ? micAvailPut : framesPerBuffer;
+ _micRingBuffer.Put(in, SAMPLES_SIZE(toPut), micVolume );
return paContinue;
}
diff --git a/src/audio/ringbuffer.cpp b/src/audio/ringbuffer.cpp
index aa193b73b3..662f784c6a 100644
--- a/src/audio/ringbuffer.cpp
+++ b/src/audio/ringbuffer.cpp
@@ -103,6 +103,7 @@ RingBuffer::Put(void* buffer, int toCopy, unsigned short volume) {
for (int i=0; i < int16len; i++) { src16[i] = src16[i] * volume / 100; }
}
// bcopy(src, dest, len)
+ //fprintf(stderr, "has %d put %d\t", len, block);
bcopy (src, mBuffer + pos, block);
src += block;
pos = (pos + block) % mBufferSize;
@@ -153,6 +154,7 @@ RingBuffer::Get(void *buffer, int toCopy, unsigned short volume) {
}
// bcopy(src, dest, len)
bcopy (mBuffer + mStart, dest, block);
+ //fprintf(stderr, "has %d get %d\t", len, block);
//_debug("get %d chars at address %ld, mBufferSize=%d, toCopy=%d\n", block, mBuffer+mStart, mBufferSize, toCopy);
dest += block;
mStart = (mStart + block) % mBufferSize;
diff --git a/src/audio/tonegenerator.cpp b/src/audio/tonegenerator.cpp
index 1c116b9946..4864a7ea18 100644
--- a/src/audio/tonegenerator.cpp
+++ b/src/audio/tonegenerator.cpp
@@ -60,29 +60,26 @@ ToneThread::run (void) {
bool started = false;
// How long do 'size' samples play ?
- unsigned int play_time = (_size * 1000) / SAMPLING_RATE - 10;
+ unsigned int play_time = (_size * 1000) / SAMPLING_RATE - 20;
ManagerImpl& manager = Manager::instance();
manager.getAudioDriver()->flushMain();
// this loop can be outside the stream, since we put the volume inside the ringbuffer
+ // Create a new stereo buffer
+ // Push the tone to the audio FIFO
for (int j = 0; j < _size; j++) {
k = j<<1; // channels is 2 (global.h)
// split in two
buf_ctrl_vol[k] = buf_ctrl_vol[k+1] = buffer[j];
- // * spkrVolume/100;
}
- // Create a new stereo buffer with the volume adjusted
- // spkrVolume = manager.getSpkrVolume();
- // Push the tone to the audio FIFO
// size = number of int16 * 2 (two channels) *
// int16 are the buf_ctrl_vol
// unsigned char are the sample_ptr inside ringbuffer
-
int size_in_char = _size * 2 * (sizeof(int16)/sizeof(unsigned char));
- _debug(" size : %d\t size_in_char : %d\n", _size, size_in_char);
+ //_debug(" size : %d\t size_in_char : %d\n", _size, size_in_char);
while (!testCancel()) {
manager.getAudioDriver()->putMain(buf_ctrl_vol, size_in_char);
diff --git a/src/global.h b/src/global.h
index 021f6cd1a6..57cd515b87 100644
--- a/src/global.h
+++ b/src/global.h
@@ -48,7 +48,4 @@ typedef short int16;
#define SAMPLING_RATE 8000
#define SIZEBUF 1024*1024
#define FORMAT 4
-#define OCTETS SAMPLING_RATE * FORMAT // Number of writen
- // bytes in buffer
-
#endif // __GLOBAL_H__
diff --git a/src/managerimpl.cpp b/src/managerimpl.cpp
index 18bf93f100..e7f78f945b 100644
--- a/src/managerimpl.cpp
+++ b/src/managerimpl.cpp
@@ -510,36 +510,44 @@ ManagerImpl::playDtmf(char code)
{
stopTone();
- int16* _buf = new int16[SIZEBUF];
+ // length in milliseconds
+ int pulselen = getConfigInt(SIGNALISATION, PULSE_LENGTH);
+ if (!pulselen) { return false; }
+
+ // numbers of int = length in milliseconds / 1000 (number of seconds)
+ // = number of seconds * SAMPLING_RATE by SECONDS
+ int size = pulselen * (SAMPLING_RATE/1000);
+
+ // this buffer is for mono
+ int16* _buf = new int16[size];
bool returnValue = false;
// Handle dtmf
_key.startTone(code);
- if ( _key.generateDTMF(_buf, SAMPLING_RATE) ) {
+ // copy the sound...
+ if ( _key.generateDTMF(_buf, size) ) {
int k;
- //int spkrVolume;
- int16* buf_ctrl_vol;
- // Determine dtmf pulse length
- int pulselen = getConfigInt(SIGNALISATION, PULSE_LENGTH);
- int size = pulselen * (OCTETS /1000);
-
- buf_ctrl_vol = new int16[size*CHANNELS];
- //spkrVolume = getSpkrVolume();
+ // allocation of more space, for stereo conversion
+ int16* buf_ctrl_vol = new int16[size*CHANNELS];
// Control volume and format mono->stereo
for (int j = 0; j < size; j++) {
- k = j<<1; // fast multiply by two
+ k = j<<1; // fast multiplication by two
buf_ctrl_vol[k] = buf_ctrl_vol[k+1] = _buf[j];
- // * spkrVolume/100;
}
- _toneMutex.enterMutex();
+
AudioLayer *audiolayer = getAudioDriver();
+
+ _toneMutex.enterMutex();
audiolayer->urgentRingBuffer().flush();
// Put buffer to urgentRingBuffer
- audiolayer->urgentRingBuffer().Put(buf_ctrl_vol, size * CHANNELS);
+ // put the size in bytes...
+ // so size * CHANNELS * 2 (bytes for the int16)
+ int nbInt16InChar = sizeof(int16)/sizeof(char);
+ audiolayer->urgentRingBuffer().Put(buf_ctrl_vol, size * CHANNELS * nbInt16InChar);
// We activate the stream if it's not active yet.
if (!audiolayer->isStreamActive()) {
@@ -548,7 +556,7 @@ ManagerImpl::playDtmf(char code)
audiolayer->urgentRingBuffer().flush();
audiolayer->stopStream();
} else {
- audiolayer->sleep(pulselen);
+ audiolayer->sleep(pulselen); // in milliseconds
}
_toneMutex.leaveMutex();
//setZonetone(false);
--
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