Commit c592ae1d authored by Emmanuel Milou's avatar Emmanuel Milou
Browse files

[#2402] Code indentation

parent 2cff4a7f
......@@ -37,8 +37,8 @@ AlsaLayer::AlsaLayer (ManagerImpl* manager)
, _is_running_capture (false)
, _is_open_playback (false)
, _is_open_capture (false)
, _trigger_request (false)
, _audioThread(NULL)
, _trigger_request (false)
, _audioThread (NULL)
{
_debug (" Constructor of AlsaLayer called\n");
......@@ -56,12 +56,14 @@ AlsaLayer::~AlsaLayer (void)
_debug ("Destructor of AlsaLayer called\n");
closeLayer();
if(_converter) {
delete _converter; _converter = NULL;
if (_converter) {
delete _converter;
_converter = NULL;
}
if(dcblocker) {
delete dcblocker; dcblocker = NULL;
if (dcblocker) {
delete dcblocker;
dcblocker = NULL;
}
}
......@@ -73,7 +75,7 @@ AlsaLayer::closeLayer()
try {
/* Stop the audio thread first */
if (_audioThread) {
_debug("Stop Audio Thread\n");
_debug ("Stop Audio Thread\n");
delete _audioThread;
_audioThread=NULL;
}
......@@ -115,7 +117,7 @@ AlsaLayer::openDevice (int indexIn, int indexOut, int sampleRate, int frameSize,
_frameSize = frameSize;
_audioPlugin = std::string(plugin);
_audioPlugin = std::string (plugin);
_debugAlsa (" Setting AlsaLayer: device in=%2d, out=%2d\n", _indexIn, _indexOut);
......@@ -147,15 +149,16 @@ AlsaLayer::startStream (void)
std::string pcmp = buildDeviceTopo (_audioPlugin, _indexOut, 0);
std::string pcmc = buildDeviceTopo (_audioPlugin, _indexIn, 0);
if(!is_playback_open()){
open_device (pcmp, pcmc, SFL_PCM_PLAYBACK);
if (!is_playback_open()) {
open_device (pcmp, pcmc, SFL_PCM_PLAYBACK);
}
if(!is_capture_open()){
open_device (pcmp, pcmc, SFL_PCM_CAPTURE);
if (!is_capture_open()) {
open_device (pcmp, pcmc, SFL_PCM_CAPTURE);
}
prepareCaptureStream ();
preparePlaybackStream ();
startCaptureStream ();
startPlaybackStream ();
......@@ -166,14 +169,14 @@ AlsaLayer::startStream (void)
// getMainBuffer()->flushAllBuffers();
// getMainBuffer()->flushDefault();
if(_audioThread == NULL) {
try {
_debug("Start Audio Thread\n");
_audioThread = new AudioThread (this);
_audioThread->start();
} catch (...) {
_debugException ("Fail to start audio thread\n");
}
if (_audioThread == NULL) {
try {
_debug ("Start Audio Thread\n");
_audioThread = new AudioThread (this);
_audioThread->start();
} catch (...) {
_debugException ("Fail to start audio thread\n");
}
}
}
......@@ -186,7 +189,7 @@ AlsaLayer::stopStream (void)
try {
/* Stop the audio thread first */
if (_audioThread) {
_debug("Stop Audio Thread\n");
_debug ("Stop Audio Thread\n");
delete _audioThread;
_audioThread=NULL;
}
......@@ -196,6 +199,7 @@ AlsaLayer::stopStream (void)
}
closeCaptureStream ();
closePlaybackStream ();
/* Flush the ring buffers */
......@@ -224,11 +228,12 @@ void AlsaLayer::stopCaptureStream (void)
int err;
if (_CaptureHandle) {
_debug("AlsaLayer:: stop Alsa capture\n");
if((err = snd_pcm_drop (_CaptureHandle)) < 0)
_debug("AlsaLayer:: Error stopping ALSA capture: %s\n", snd_strerror(err));
else
stop_capture ();
_debug ("AlsaLayer:: stop Alsa capture\n");
if ( (err = snd_pcm_drop (_CaptureHandle)) < 0)
_debug ("AlsaLayer:: Error stopping ALSA capture: %s\n", snd_strerror (err));
else
stop_capture ();
}
}
......@@ -241,11 +246,12 @@ void AlsaLayer::closeCaptureStream (void)
stopCaptureStream ();
if (is_capture_open()) {
_debug("AlsaLayer:: close ALSA capture\n");
if ((err = snd_pcm_close (_CaptureHandle)) < 0)
_debug("Error closing ALSA capture: %s\n", snd_strerror(err));
else
close_capture ();
_debug ("AlsaLayer:: close ALSA capture\n");
if ( (err = snd_pcm_close (_CaptureHandle)) < 0)
_debug ("Error closing ALSA capture: %s\n", snd_strerror (err));
else
close_capture ();
}
}
......@@ -255,10 +261,11 @@ void AlsaLayer::startCaptureStream (void)
if (_CaptureHandle && !is_capture_running()) {
_debug ("AlsaLayer:: start ALSA capture\n");
if((err = snd_pcm_start (_CaptureHandle)) < 0)
_debug("Error starting ALSA capture: %s\n", snd_strerror(err));
else
start_capture();
if ( (err = snd_pcm_start (_CaptureHandle)) < 0)
_debug ("Error starting ALSA capture: %s\n", snd_strerror (err));
else
start_capture();
}
}
......@@ -267,9 +274,10 @@ void AlsaLayer::prepareCaptureStream (void)
int err;
if (is_capture_open() && !is_capture_prepared()) {
_debug("AlsaLayer:: prepare ALSA capture\n");
if ((err = snd_pcm_prepare (_CaptureHandle)) < 0)
_debug ("Error preparing ALSA capture: %s\n", snd_strerror(err));
_debug ("AlsaLayer:: prepare ALSA capture\n");
if ( (err = snd_pcm_prepare (_CaptureHandle)) < 0)
_debug ("Error preparing ALSA capture: %s\n", snd_strerror (err));
else
prepare_capture ();
}
......@@ -280,11 +288,12 @@ void AlsaLayer::stopPlaybackStream (void)
int err;
if (_PlaybackHandle && is_playback_running()) {
_debug("AlsaLayer:: stop ALSA playback\n");
if((err = snd_pcm_drop (_PlaybackHandle)) < 0)
_debug("Error stopping ALSA playback: %s\n", snd_strerror(err));
else
stop_playback ();
_debug ("AlsaLayer:: stop ALSA playback\n");
if ( (err = snd_pcm_drop (_PlaybackHandle)) < 0)
_debug ("Error stopping ALSA playback: %s\n", snd_strerror (err));
else
stop_playback ();
}
}
......@@ -297,11 +306,12 @@ void AlsaLayer::closePlaybackStream (void)
stopPlaybackStream ();
if (is_playback_open()) {
_debug("AlsaLayer:: close ALSA playback\n");
if ((err = snd_pcm_close (_PlaybackHandle)) < 0)
_debug("Error closing ALSA playback: %s\n", snd_strerror(err));
_debug ("AlsaLayer:: close ALSA playback\n");
if ( (err = snd_pcm_close (_PlaybackHandle)) < 0)
_debug ("Error closing ALSA playback: %s\n", snd_strerror (err));
else
close_playback ();
close_playback ();
}
}
......@@ -310,11 +320,12 @@ void AlsaLayer::startPlaybackStream (void)
int err;
if (_PlaybackHandle && !is_playback_running()) {
_debug ("AlsaLayer:: start ALSA playback\n");
if ((err = snd_pcm_start (_PlaybackHandle)) < 0)
_debug("Error starting ALSA playback: %s\n", snd_strerror(err));
else
start_playback();
_debug ("AlsaLayer:: start ALSA playback\n");
if ( (err = snd_pcm_start (_PlaybackHandle)) < 0)
_debug ("Error starting ALSA playback: %s\n", snd_strerror (err));
else
start_playback();
}
}
......@@ -323,11 +334,12 @@ void AlsaLayer::preparePlaybackStream (void)
int err;
if (is_playback_open() && !is_playback_prepared()) {
_debug("AlsaLayer:: prepare playback stream\n");
if ((err = snd_pcm_prepare (_PlaybackHandle)) < 0)
_debug ("Error preparing the device: %s\n", snd_strerror(err));
else
prepare_playback ();
_debug ("AlsaLayer:: prepare playback stream\n");
if ( (err = snd_pcm_prepare (_PlaybackHandle)) < 0)
_debug ("Error preparing the device: %s\n", snd_strerror (err));
else
prepare_playback ();
}
}
......@@ -491,8 +503,9 @@ AlsaLayer::open_device (std::string pcm_p, std::string pcm_c, int flag)
if (flag == SFL_PCM_BOTH || flag == SFL_PCM_PLAYBACK) {
_debug("AlsaLayer:: open playback device\n");
_debug ("AlsaLayer:: open playback device\n");
// if((err = snd_pcm_open(&_PlaybackHandle, pcm_p.c_str(), SND_PCM_STREAM_PLAYBACK, 0 )) < 0){
if ( (err = snd_pcm_open (&_PlaybackHandle, pcm_p.c_str(), SND_PCM_STREAM_PLAYBACK, 0)) < 0) {
_debugAlsa ("Error while opening playback device %s\n", pcm_p.c_str());
setErrorMessage (ALSA_PLAYBACK_DEVICE);
......@@ -512,7 +525,8 @@ AlsaLayer::open_device (std::string pcm_p, std::string pcm_c, int flag)
if (flag == SFL_PCM_BOTH || flag == SFL_PCM_CAPTURE) {
_debug("AlsaLayer:: open capture device\n");
_debug ("AlsaLayer:: open capture device\n");
if ( (err = snd_pcm_open (&_CaptureHandle, pcm_c.c_str(), SND_PCM_STREAM_CAPTURE, 0)) < 0) {
_debugAlsa ("Error while opening capture device %s\n", pcm_c.c_str());
setErrorMessage (ALSA_CAPTURE_DEVICE);
......@@ -576,7 +590,8 @@ AlsaLayer::write (void* buffer, int length)
break;
default:
_debugAlsa ("Write error unknown - dropping frames **********************************: %s\n", snd_strerror(err));
_debugAlsa ("Write error unknown - dropping frames **********************************: %s\n", snd_strerror (err));
stopPlaybackStream ();
break;
......@@ -635,7 +650,7 @@ AlsaLayer::read (void* buffer, int toCopy)
void
AlsaLayer::handle_xrun_capture (void)
{
_debugAlsa("handle_xrun_capture\n");
_debugAlsa ("handle_xrun_capture\n");
snd_pcm_status_t* status;
snd_pcm_status_alloca (&status);
......@@ -655,7 +670,7 @@ AlsaLayer::handle_xrun_capture (void)
void
AlsaLayer::handle_xrun_playback (void)
{
_debugAlsa("AlsaLayer:: handle_xrun_playback\n");
_debugAlsa ("AlsaLayer:: handle_xrun_playback\n");
int state;
snd_pcm_status_t* status;
......@@ -669,7 +684,7 @@ AlsaLayer::handle_xrun_playback (void)
stopPlaybackStream ();
preparePlaybackStream ();
_trigger_request = true;
}
}
......@@ -807,7 +822,7 @@ void AlsaLayer::audioCallback (void)
spkrVolume = _manager->getSpkrVolume();
micVolume = _manager->getMicVolume();
/*
int writeableSize = snd_pcm_avail_update(_PlaybackHandle);
_debug("writeableSize %i\n", writeableSize);
......@@ -816,6 +831,7 @@ void AlsaLayer::audioCallback (void)
// AvailForGet tell the number of chars inside the buffer
// framePerBuffer are the number of data for one channel (left)
urgentAvailBytes = _urgentRingBuffer.AvailForGet();
if (urgentAvailBytes > 0) {
// Urgent data (dtmf, incoming call signal) come first.
......@@ -835,7 +851,7 @@ void AlsaLayer::audioCallback (void)
} else {
tone = _manager->getTelephoneTone();
file_tone = _manager->getTelephoneFile();
file_tone = _manager->getTelephoneFile();
toGet = framesPerBufferAlsa;
maxBytes = toGet * sizeof (SFLDataFormat);
......@@ -846,92 +862,93 @@ void AlsaLayer::audioCallback (void)
tone->getNext (out, toGet, spkrVolume);
write (out , maxBytes);
free(out);
out = 0;
free (out);
out = 0;
} else if (file_tone != 0) {
out = (SFLDataFormat*) malloc (maxBytes * sizeof (SFLDataFormat));
file_tone->getNext(out, toGet, spkrVolume);
file_tone->getNext (out, toGet, spkrVolume);
write (out , maxBytes);
free(out);
out = 0;
free (out);
out = 0;
} else {
// If nothing urgent, play the regular sound samples
int _mainBufferSampleRate = getMainBuffer()->getInternalSamplingRate();
int maxNbSamplesToGet = 0;
int maxNbBytesToGet = 0;
// If nothing urgent, play the regular sound samples
int _mainBufferSampleRate = getMainBuffer()->getInternalSamplingRate();
int maxNbSamplesToGet = 0;
int maxNbBytesToGet = 0;
// Compute maximal value to get into the ring buffer
if (_mainBufferSampleRate && ( (int) _audioSampleRate != _mainBufferSampleRate)) {
double upsampleFactor = (double) _audioSampleRate / _mainBufferSampleRate;
// Compute maximal value to get into the ring buffer
if (_mainBufferSampleRate && ((int)_audioSampleRate != _mainBufferSampleRate)) {
double upsampleFactor = (double) _audioSampleRate / _mainBufferSampleRate;
maxNbSamplesToGet = (int) ( (double) framesPerBufferAlsa / upsampleFactor);
maxNbSamplesToGet = (int) ((double) framesPerBufferAlsa / upsampleFactor);
} else {
} else {
maxNbSamplesToGet = framesPerBufferAlsa;
maxNbSamplesToGet = framesPerBufferAlsa;
}
}
maxNbBytesToGet = maxNbSamplesToGet * sizeof (SFLDataFormat);
maxNbBytesToGet = maxNbSamplesToGet * sizeof(SFLDataFormat);
normalAvailBytes = getMainBuffer()->availForGet();
toGet = (normalAvailBytes < (int)maxNbBytesToGet) ? normalAvailBytes : maxNbBytesToGet;
toGet = (normalAvailBytes < (int) maxNbBytesToGet) ? normalAvailBytes : maxNbBytesToGet;
out = (SFLDataFormat*) malloc (maxNbBytesToGet);
if (normalAvailBytes) {
getMainBuffer()->getData(out, toGet, spkrVolume);
getMainBuffer()->getData (out, toGet, spkrVolume);
if (_mainBufferSampleRate && ( (int) _audioSampleRate != _mainBufferSampleRate)) {
if (_mainBufferSampleRate && ((int)_audioSampleRate != _mainBufferSampleRate)) {
rsmpl_out = (SFLDataFormat*) malloc (framesPerBufferAlsa * sizeof (SFLDataFormat));
rsmpl_out = (SFLDataFormat*) malloc (framesPerBufferAlsa * sizeof (SFLDataFormat));
// Do sample rate conversion
int nb_sample_down = toGet / sizeof(SFLDataFormat);
// Do sample rate conversion
int nb_sample_down = toGet / sizeof (SFLDataFormat);
int nbSample = _converter->upsampleData((SFLDataFormat*)out, rsmpl_out, _mainBufferSampleRate, _audioSampleRate, nb_sample_down);
int nbSample = _converter->upsampleData ( (SFLDataFormat*) out, rsmpl_out, _mainBufferSampleRate, _audioSampleRate, nb_sample_down);
write (rsmpl_out, nbSample*sizeof(SFLDataFormat));
free(rsmpl_out);
rsmpl_out = 0;
} else {
write (rsmpl_out, nbSample*sizeof (SFLDataFormat));
write (out, toGet);
free (rsmpl_out);
rsmpl_out = 0;
}
} else {
write (out, toGet);
}
} else {
if((tone == 0) && (file_tone == 0)) {
if ( (tone == 0) && (file_tone == 0)) {
SFLDataFormat* zeros = (SFLDataFormat*)malloc (framesPerBufferAlsa * sizeof (SFLDataFormat));
SFLDataFormat* zeros = (SFLDataFormat*) malloc (framesPerBufferAlsa * sizeof (SFLDataFormat));
bzero (zeros, framesPerBufferAlsa * sizeof (SFLDataFormat));
write (zeros, framesPerBufferAlsa * sizeof (SFLDataFormat));
bzero (zeros, framesPerBufferAlsa * sizeof (SFLDataFormat));
write (zeros, framesPerBufferAlsa * sizeof (SFLDataFormat));
free (zeros);
}
free (zeros);
}
}
_urgentRingBuffer.Discard (toGet);
_urgentRingBuffer.Discard (toGet);
free (out);
out = 0;
free (out);
out = 0;
}
......@@ -939,59 +956,62 @@ void AlsaLayer::audioCallback (void)
// Additionally handle the mic's audio stream
int micAvailBytes;
int micAvailPut;
int toPut;
SFLDataFormat* in;
// snd_pcm_sframes_t micAvailAlsa;
in = 0;
if(is_capture_running())
{
micAvailBytes = snd_pcm_avail_update(_CaptureHandle);
// _debug("micAvailBytes %i\n", micAvailBytes);
if(micAvailBytes > 0)
{
if (is_capture_running()) {
micAvailBytes = snd_pcm_avail_update (_CaptureHandle);
// _debug("micAvailBytes %i\n", micAvailBytes);
if (micAvailBytes > 0) {
micAvailPut = getMainBuffer()->availForPut();
toPut = (micAvailBytes <= framesPerBufferAlsa) ? micAvailBytes : framesPerBufferAlsa;
in = (SFLDataFormat*)malloc(toPut * sizeof(SFLDataFormat));
toPut = read (in, toPut* sizeof(SFLDataFormat));
in = (SFLDataFormat*) malloc (toPut * sizeof (SFLDataFormat));
toPut = read (in, toPut* sizeof (SFLDataFormat));
adjustVolume (in, toPut, SFL_PCM_CAPTURE);
if (in != 0) {
int _mainBufferSampleRate = getMainBuffer()->getInternalSamplingRate();
adjustVolume (in, toPut, SFL_PCM_CAPTURE);
if (_mainBufferSampleRate && ( (int) _audioSampleRate != _mainBufferSampleRate)) {
if (in != 0)
{
int _mainBufferSampleRate = getMainBuffer()->getInternalSamplingRate();
SFLDataFormat* rsmpl_out = (SFLDataFormat*) malloc (framesPerBufferAlsa * sizeof (SFLDataFormat));
if (_mainBufferSampleRate && ((int)_audioSampleRate != _mainBufferSampleRate)) {
int nbSample = toPut / sizeof (SFLDataFormat);
int nb_sample_up = nbSample;
SFLDataFormat* rsmpl_out = (SFLDataFormat*) malloc (framesPerBufferAlsa * sizeof (SFLDataFormat));
// _debug("nb_sample_up %i\n", nb_sample_up);
nbSample = _converter->downsampleData ( (SFLDataFormat*) in, rsmpl_out, _mainBufferSampleRate, _audioSampleRate, nb_sample_up);
int nbSample = toPut / sizeof(SFLDataFormat);
int nb_sample_up = nbSample;
dcblocker->filter_signal (rsmpl_out, nbSample);
// _debug("nb_sample_up %i\n", nb_sample_up);
nbSample = _converter->downsampleData ((SFLDataFormat*)in, rsmpl_out, _mainBufferSampleRate, _audioSampleRate, nb_sample_up);
getMainBuffer()->putData (rsmpl_out, nbSample * sizeof (SFLDataFormat), 100);
dcblocker->filter_signal(rsmpl_out, nbSample);
free (rsmpl_out);
rsmpl_out = 0;
getMainBuffer()->putData(rsmpl_out, nbSample * sizeof (SFLDataFormat), 100);
} else {
getMainBuffer()->putData (in, toPut, 100);
}
}
free(rsmpl_out);
rsmpl_out = 0;
} else {
free (in);
getMainBuffer()->putData(in, toPut, 100);
}
}
free(in); in=0;
in=0;
} else if (micAvailBytes < 0) {
_debug ("AlsaLayer::audioCallback (mic): error: %s\n", snd_strerror (micAvailBytes));
}
else if(micAvailBytes < 0)
{
_debug("AlsaLayer::audioCallback (mic): error: %s\n", snd_strerror(micAvailBytes));
}
}
}
......
......@@ -25,7 +25,8 @@ void AudioLayer::flushMain (void)
// should pass call id
MainBuffer* mainbuffer = getMainBuffer();
if(mainbuffer)
if (mainbuffer)
mainbuffer->flushAllBuffers();
}
......@@ -39,7 +40,7 @@ void AudioLayer::flushUrgent (void)
int AudioLayer::putUrgent (void* buffer, int toCopy)
{
_debug("------------------- AudioLayer::putUrgent --------------------\n");
_debug ("------------------- AudioLayer::putUrgent --------------------\n");
int a;
ost::MutexLock guard (_mutex);
......@@ -59,7 +60,7 @@ int AudioLayer::putMain (void *buffer, int toCopy, CallID call_id)
int a;
ost::MutexLock guard (_mutex);
a = getMainBuffer()->availForPut(call_id);
a = getMainBuffer()->availForPut (call_id);
if (a >= toCopy) {
return getMainBuffer()->putData (buffer, toCopy, _defaultVolume, call_id);
......
......@@ -55,8 +55,7 @@ AudioRtp::~AudioRtp (void)
{
ost::MutexLock m (_rtpMutex);
if (_RTXThread != _RTXThread)
{
if (_RTXThread != _RTXThread) {
delete _RTXThread;
_RTXThread = 0;
}
......@@ -122,10 +121,10 @@ AudioRtp::closeRtpSession ()
_debug ("AudioRtp::Stopping rtp session\n");
try {
if (_RTXThread != 0) {
if (_RTXThread != 0) {
delete _RTXThread;
_RTXThread = 0;
}
}
} catch (...) {
_debugException ("! ARTP Exception: when stopping audiortp\n");
throw;
......@@ -190,18 +189,18 @@ AudioRtpRTX::AudioRtpRTX (SIPCall *sipcall, bool sym) : time (new ost::Time()),
// convert count into string
std::stringstream out;