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Commit b7286e03 authored by Julien Bonjean's avatar Julien Bonjean
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[#2402] Code indent

parent 3298edc0
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astylerc
View file @ b7286e03
......@@ -5,7 +5,7 @@
# Savoir-faire Linux Inc
# http://www.sflphone.org
style=kr # Kernighan & Ritchie style formatting/indenting uses linux bracket
style=k&r # Kernighan & Ritchie style formatting/indenting uses linux bracket
indent=spaces=4 # Use spaces instead of tabs for indentation
indent-classes # Indent 'class' and 'struct' blocks so that the blocks 'public:', 'protected:' and 'private:' are indented
indent-switches # Indent 'switch' blocks so that the 'case X:' statements are indented in the switch block
......
sflphone-common/src/account.cpp
View file @ b7286e03
......@@ -34,81 +34,87 @@
#include "manager.h"
Account::Account (const AccountID& accountID, std::string type) :
_accountID (accountID)
, _link (NULL)
, _enabled (true)
, _type (type)
, _codecOrder ()
, _codecStr("")
, _ringtonePath("/usr/share/sflphone/ringtones/konga.ul")
, _ringtoneEnabled(true)
, _displayName("")
, _useragent("SFLphone")
_accountID (accountID)
, _link (NULL)
, _enabled (true)
, _type (type)
, _codecOrder ()
, _codecStr ("")
, _ringtonePath ("/usr/share/sflphone/ringtones/konga.ul")
, _ringtoneEnabled (true)
, _displayName ("")
, _useragent ("SFLphone")
{
setRegistrationState (Unregistered);
setRegistrationState (Unregistered);
}
Account::~Account()
{
}
void Account::loadConfig() {
void Account::loadConfig()
{
// If IAX is not supported, do not register this account
// If IAX is not supported, do not register this account
#ifndef USE_IAX
if (_type == "IAX")
_enabled = false;
if (_type == "IAX")
_enabled = false;
#endif
loadAudioCodecs ();
loadAudioCodecs ();
}
void Account::setRegistrationState (RegistrationState state) {
void Account::setRegistrationState (RegistrationState state)
{
if (state != _registrationState) {
_debug ("Account: set registration state");
_registrationState = state;
if (state != _registrationState) {
_debug ("Account: set registration state");
_registrationState = state;
// Notify the client
Manager::instance().connectionStatusNotification();
}
// Notify the client
Manager::instance().connectionStatusNotification();
}
}
void Account::loadAudioCodecs (void) {
// if the user never set the codec list, use the default configuration for this account
if(_codecStr == "") {
_info ("Account: use the default order");
Manager::instance ().getCodecDescriptorMap ().setDefaultOrder();
}
// else retrieve the one set in the user config file
else {
std::vector<std::string> active_list = Manager::instance ().retrieveActiveCodecs();
// This property is now set per account basis
// std::string s = Manager::instance ().getConfigString (_accountID, "ActiveCodecs");
setActiveCodecs (Manager::instance ().unserialize (_codecStr));
}
void Account::loadAudioCodecs (void)
{
// if the user never set the codec list, use the default configuration for this account
if (_codecStr == "") {
_info ("Account: use the default order");
Manager::instance ().getCodecDescriptorMap ().setDefaultOrder();
}
// else retrieve the one set in the user config file
else {
std::vector<std::string> active_list = Manager::instance ().retrieveActiveCodecs();
// This property is now set per account basis
// std::string s = Manager::instance ().getConfigString (_accountID, "ActiveCodecs");
setActiveCodecs (Manager::instance ().unserialize (_codecStr));
}
}
void Account::setActiveCodecs (const std::vector <std::string> &list) {
_codecOrder.clear();
// list contains the ordered payload of active codecs picked by the user for this account
// we used the CodecOrder vector to save the order.
int i=0;
int payload;
size_t size = list.size();
while ( (unsigned int) i < size) {
payload = std::atoi (list[i].data());
_info ("Account: Adding codec with RTP payload=%i", payload);
//if (Manager::instance ().getCodecDescriptorMap ().isCodecLoaded (payload)) {
_codecOrder.push_back ( (AudioCodecType) payload);
//}
i++;
}
// setConfig
_codecStr = Manager::instance ().serialize (list);
void Account::setActiveCodecs (const std::vector <std::string> &list)
{
_codecOrder.clear();
// list contains the ordered payload of active codecs picked by the user for this account
// we used the CodecOrder vector to save the order.
int i=0;
int payload;
size_t size = list.size();
while ( (unsigned int) i < size) {
payload = std::atoi (list[i].data());
_info ("Account: Adding codec with RTP payload=%i", payload);
//if (Manager::instance ().getCodecDescriptorMap ().isCodecLoaded (payload)) {
_codecOrder.push_back ( (AudioCodecType) payload);
//}
i++;
}
// setConfig
_codecStr = Manager::instance ().serialize (list);
}
sflphone-common/src/accountcreator.cpp
View file @ b7286e03
......@@ -50,26 +50,26 @@ AccountCreator::createAccount (AccountType type, AccountID accountID)
{
switch (type) {
case SIP_ACCOUNT: {
_debug("AccountCreator: create account %s", accountID.c_str());
return new SIPAccount (accountID);
break;
}
case SIP_DIRECT_IP_ACCOUNT: {
_debug("AccountCreator: create account IP2IP_PROFILE");
return new SIPAccount (IP2IP_PROFILE);
break;
}
case SIP_ACCOUNT: {
_debug ("AccountCreator: create account %s", accountID.c_str());
return new SIPAccount (accountID);
break;
}
case SIP_DIRECT_IP_ACCOUNT: {
_debug ("AccountCreator: create account IP2IP_PROFILE");
return new SIPAccount (IP2IP_PROFILE);
break;
}
#ifdef USE_IAX
case IAX_ACCOUNT: {
_debug("AccountCreator: create account %s", accountID.c_str());
return new IAXAccount (accountID);
break;
}
case IAX_ACCOUNT: {
_debug ("AccountCreator: create account %s", accountID.c_str());
return new IAXAccount (accountID);
break;
}
#endif
default:
_error("AccountCreator: Error: unknown account type");
default:
_error ("AccountCreator: Error: unknown account type");
}
return 0;
......
sflphone-common/src/audio/alsa/alsalayer.cpp
View file @ b7286e03
......@@ -39,7 +39,7 @@ int framesPerBufferAlsa = 2048;
AlsaLayer::AlsaLayer (ManagerImpl* manager)
: AudioLayer (manager , ALSA)
, _PlaybackHandle (NULL)
, _RingtoneHandle (NULL)
, _RingtoneHandle (NULL)
, _CaptureHandle (NULL)
, _periodSize()
, _audioPlugin()
......@@ -59,7 +59,7 @@ AlsaLayer::AlsaLayer (ManagerImpl* manager)
// _audioThread = new AudioThread (this);
// _audioThread = NULL;
_urgentRingBuffer.createReadPointer();
AudioLayer::_echocancelstate = true;
AudioLayer::_noisesuppressstate = true;
}
......@@ -111,9 +111,9 @@ AlsaLayer::openDevice (int indexIn, int indexOut, int indexRing, int sampleRate,
if (stream == SFL_PCM_BOTH && is_capture_open() == true && is_playback_open() == true) {
closeCaptureStream();
closePlaybackStream();
} else if((stream == SFL_PCM_CAPTURE || stream == SFL_PCM_BOTH) && is_capture_open() == true)
} else if ( (stream == SFL_PCM_CAPTURE || stream == SFL_PCM_BOTH) && is_capture_open() == true)
closeCaptureStream ();
else if((stream == SFL_PCM_PLAYBACK || stream == SFL_PCM_BOTH) && is_playback_open () == true)
else if ( (stream == SFL_PCM_PLAYBACK || stream == SFL_PCM_BOTH) && is_playback_open () == true)
closePlaybackStream ();
_indexIn = indexIn;
......@@ -136,16 +136,16 @@ AlsaLayer::openDevice (int indexIn, int indexOut, int indexRing, int sampleRate,
_converter = new SamplerateConverter (_audioSampleRate, 1000);
AudioLayer::_echoCancel = new EchoCancel();
AudioLayer::_echoCanceller = new AudioProcessing(static_cast<Algorithm *>(_echoCancel));
AudioLayer::_echoCanceller = new AudioProcessing (static_cast<Algorithm *> (_echoCancel));
AudioLayer::_echoCancel->setEchoCancelState(AudioLayer::_echocancelstate);
AudioLayer::_echoCancel->setNoiseSuppressState(AudioLayer::_noisesuppressstate);
AudioLayer::_echoCancel->setEchoCancelState (AudioLayer::_echocancelstate);
AudioLayer::_echoCancel->setNoiseSuppressState (AudioLayer::_noisesuppressstate);
AudioLayer::_dcblocker = new DcBlocker();
AudioLayer::_audiofilter = new AudioProcessing(static_cast<Algorithm *>(_dcblocker));
AudioLayer::_audiofilter = new AudioProcessing (static_cast<Algorithm *> (_dcblocker));
return true;
return true;
}
void
......@@ -153,33 +153,32 @@ AlsaLayer::startStream (void)
{
_debug ("Audio: Start stream");
if(_audiofilter)
_audiofilter->resetAlgorithm();
if (_audiofilter)
_audiofilter->resetAlgorithm();
if(_echoCanceller)
_echoCanceller->resetAlgorithm();
if (_echoCanceller)
_echoCanceller->resetAlgorithm();
if(is_playback_running() && is_capture_running() )
if (is_playback_running() && is_capture_running())
return;
std::string pcmp;
std::string pcmr;
std::string pcmc;
if(_audioPlugin == PCM_DMIX_DSNOOP) {
pcmp = buildDeviceTopo (PCM_DMIX, _indexOut, 0);
pcmr = buildDeviceTopo (PCM_DMIX, _indexRing, 0);
pcmc = buildDeviceTopo(PCM_DSNOOP, _indexIn, 0);
}
else {
pcmp = buildDeviceTopo (_audioPlugin, _indexOut, 0);
pcmr = buildDeviceTopo (_audioPlugin, _indexRing, 0);
pcmc = buildDeviceTopo(_audioPlugin, _indexIn, 0);
if (_audioPlugin == PCM_DMIX_DSNOOP) {
pcmp = buildDeviceTopo (PCM_DMIX, _indexOut, 0);
pcmr = buildDeviceTopo (PCM_DMIX, _indexRing, 0);
pcmc = buildDeviceTopo (PCM_DSNOOP, _indexIn, 0);
} else {
pcmp = buildDeviceTopo (_audioPlugin, _indexOut, 0);
pcmr = buildDeviceTopo (_audioPlugin, _indexRing, 0);
pcmc = buildDeviceTopo (_audioPlugin, _indexIn, 0);
}
_debug("pcmp: %s, index %d", pcmp.c_str(), _indexOut);
_debug("pcmr: %s, index %d", pcmr.c_str(), _indexRing);
_debug("pcmc: %s, index %d", pcmc.c_str(), _indexIn);
_debug ("pcmp: %s, index %d", pcmp.c_str(), _indexOut);
_debug ("pcmr: %s, index %d", pcmr.c_str(), _indexRing);
_debug ("pcmc: %s, index %d", pcmc.c_str(), _indexIn);
if (!is_playback_open()) {
open_device (pcmp, pcmc, pcmr, SFL_PCM_PLAYBACK);
......@@ -255,22 +254,22 @@ bool AlsaLayer::isCaptureActive (void)
}
void AlsaLayer::setEchoCancelState(bool state)
void AlsaLayer::setEchoCancelState (bool state)
{
// if a stream already running
if(AudioLayer::_echoCancel)
_echoCancel->setEchoCancelState(state);
// if a stream already running
if (AudioLayer::_echoCancel)
_echoCancel->setEchoCancelState (state);
AudioLayer::_echocancelstate = state;
AudioLayer::_echocancelstate = state;
}
void AlsaLayer::setNoiseSuppressState(bool state)
void AlsaLayer::setNoiseSuppressState (bool state)
{
// if a stream already opened
if(AudioLayer::_echoCancel)
_echoCancel->setNoiseSuppressState(state);
// if a stream already opened
if (AudioLayer::_echoCancel)
_echoCancel->setNoiseSuppressState (state);
AudioLayer::_noisesuppressstate = state;
AudioLayer::_noisesuppressstate = state;
}
......@@ -343,12 +342,12 @@ void AlsaLayer::stopPlaybackStream (void)
{
int err;
if(_RingtoneHandle && is_playback_running()) {
_debug("Audio: Stop ALSA ringtone");
if (_RingtoneHandle && is_playback_running()) {
_debug ("Audio: Stop ALSA ringtone");
if( (err = snd_pcm_drop(_RingtoneHandle)) < 0) {
_debug("Audio: Error: Stop ALSA ringtone: %s", snd_strerror(err));
}
if ( (err = snd_pcm_drop (_RingtoneHandle)) < 0) {
_debug ("Audio: Error: Stop ALSA ringtone: %s", snd_strerror (err));
}
}
if (_PlaybackHandle && is_playback_running()) {
......@@ -372,16 +371,16 @@ void AlsaLayer::closePlaybackStream (void)
if (is_playback_open()) {
_debug("Audio: Close ALSA playback");
_debug ("Audio: Close ALSA playback");
if(_RingtoneHandle) {
if((err = snd_pcm_close(_RingtoneHandle)) < 0) {
_warn("Audio: Error: Closing ALSA ringtone: %s", snd_strerror(err));
}
if (_RingtoneHandle) {
if ( (err = snd_pcm_close (_RingtoneHandle)) < 0) {
_warn ("Audio: Error: Closing ALSA ringtone: %s", snd_strerror (err));
}
}
if ( (err = snd_pcm_close (_PlaybackHandle)) < 0)
_warn("Audio: Error: Closing ALSA playback: %s", snd_strerror (err));
_warn ("Audio: Error: Closing ALSA playback: %s", snd_strerror (err));
else
close_playback ();
}
......@@ -493,9 +492,8 @@ bool AlsaLayer::alsa_set_params (snd_pcm_t *pcm_handle, int type, int rate)
if ( (err = snd_pcm_hw_params_set_rate_near (pcm_handle, hwparams, &exact_ivalue, &dir) < 0)) {
_debugAlsa ("Audio: Error: Cannot set sample rate (%s)", snd_strerror (err));
return false;
}
else
_debug("Audio: Set audio rate to %d", rate);
} else
_debug ("Audio: Set audio rate to %d", rate);
if (dir!= 0) {
_debugAlsa ("Audio: Error: (%i) The choosen rate %d Hz is not supported by your hardware.Using %d Hz instead. ",type ,rate, exact_ivalue);
......@@ -581,8 +579,8 @@ AlsaLayer::open_device (std::string pcm_p, std::string pcm_c, std::string pcm_r,
_debug ("Audio: Open playback device (and ringtone)");
if ((err = snd_pcm_open(&_PlaybackHandle, pcm_p.c_str(), SND_PCM_STREAM_PLAYBACK, 0)) < 0) {
_warn("Audio: Error while opening playback device %s", pcm_p.c_str());
if ( (err = snd_pcm_open (&_PlaybackHandle, pcm_p.c_str(), SND_PCM_STREAM_PLAYBACK, 0)) < 0) {
_warn ("Audio: Error while opening playback device %s", pcm_p.c_str());
setErrorMessage (ALSA_PLAYBACK_DEVICE);
close_playback ();
return false;
......@@ -595,19 +593,19 @@ AlsaLayer::open_device (std::string pcm_p, std::string pcm_c, std::string pcm_r,
return false;
}
if (getIndexOut() != getIndexRing()) {
if (getIndexOut() != getIndexRing()) {
if((err = snd_pcm_open(&_RingtoneHandle, pcm_r.c_str(), SND_PCM_STREAM_PLAYBACK, 0)) < 0) {
_warn("Audio: Error: Opening ringtone device %s", pcm_r.c_str());
// setErrorMessage(ALSA_RINGTONE_DEVICE);
}
if ( (err = snd_pcm_open (&_RingtoneHandle, pcm_r.c_str(), SND_PCM_STREAM_PLAYBACK, 0)) < 0) {
_warn ("Audio: Error: Opening ringtone device %s", pcm_r.c_str());
// setErrorMessage(ALSA_RINGTONE_DEVICE);
}
if(!alsa_set_params(_RingtoneHandle, 1, getSampleRate())) {
_warn("Audio: Error: Ringtone failed");
snd_pcm_close(_RingtoneHandle);
}
}
if (!alsa_set_params (_RingtoneHandle, 1, getSampleRate())) {
_warn ("Audio: Error: Ringtone failed");
snd_pcm_close (_RingtoneHandle);
}
}
open_playback ();
}
......@@ -616,8 +614,8 @@ AlsaLayer::open_device (std::string pcm_p, std::string pcm_c, std::string pcm_r,
_debug ("Audio: Open capture device");
if ((err = snd_pcm_open(&_CaptureHandle, pcm_c.c_str(), SND_PCM_STREAM_CAPTURE, 0)) < 0){
_warn("Audio: Error: Opening capture device %s", pcm_c.c_str());
if ( (err = snd_pcm_open (&_CaptureHandle, pcm_c.c_str(), SND_PCM_STREAM_CAPTURE, 0)) < 0) {
_warn ("Audio: Error: Opening capture device %s", pcm_c.c_str());
setErrorMessage (ALSA_CAPTURE_DEVICE);
close_capture ();
......@@ -625,7 +623,7 @@ AlsaLayer::open_device (std::string pcm_p, std::string pcm_c, std::string pcm_r,
}
if (!alsa_set_params (_CaptureHandle, 0, 8000)) {
_warn("Audio: Error: Capture failed");
_warn ("Audio: Error: Capture failed");
snd_pcm_close (_CaptureHandle);
close_capture ();
return false;
......@@ -636,7 +634,7 @@ AlsaLayer::open_device (std::string pcm_p, std::string pcm_c, std::string pcm_r,
// prepare_capture ();
}
return true;
}
......@@ -663,7 +661,7 @@ AlsaLayer::write (void* buffer, int length, snd_pcm_t * handle)
case -EIO:
//_debugAlsa(" XRUN playback ignored (%s)", snd_strerror(err));
handle_xrun_playback(handle);
handle_xrun_playback (handle);
if (snd_pcm_writei (handle, buffer , frames) <0)
_debugAlsa ("Audio: XRUN handling failed");
......@@ -759,13 +757,13 @@ AlsaLayer::handle_xrun_playback (snd_pcm_t *handle)
snd_pcm_status_t* status;
snd_pcm_status_alloca (&status);
if ( (state = snd_pcm_status (handle, status)) < 0)
_debugAlsa ("Audio: Error: Cannot get playback handle status (%s)" , snd_strerror (state));
if ( (state = snd_pcm_status (handle, status)) < 0)
_debugAlsa ("Audio: Error: Cannot get playback handle status (%s)" , snd_strerror (state));
else {
state = snd_pcm_status_get_state (status);
if (state == SND_PCM_STATE_XRUN) {
_debug("Audio: audio device in state SND_PCM_STATE_XRUN, restart device");
_debug ("Audio: audio device in state SND_PCM_STATE_XRUN, restart device");
stopPlaybackStream ();
preparePlaybackStream ();
......@@ -796,7 +794,7 @@ AlsaLayer::buildDeviceTopo (std::string plugin, int card, int subdevice)
pcm.append (ss1.str());
}
_debug("Audio: Device topo: %s", pcm.c_str());
_debug ("Audio: Device topo: %s", pcm.c_str());
return pcm;
}
......@@ -807,7 +805,7 @@ AlsaLayer::getSoundCardsInfo (int stream)
std::vector<std::string> cards_id;
HwIDPair p;
_debug("Audio: Get sound cards info: ");
_debug ("Audio: Get sound cards info: ");
snd_ctl_t* handle;
snd_ctl_card_info_t *info;
......@@ -899,7 +897,7 @@ AlsaLayer::soundCardGetIndex (std::string description)
return 0;
}
void AlsaLayer::audioCallback(void)
void AlsaLayer::audioCallback (void)
{
int toGet, urgentAvailBytes, normalAvailBytes;
......@@ -920,13 +918,13 @@ void AlsaLayer::audioCallback(void)
// framePerBuffer are the number of data for one channel (left)
urgentAvailBytes = _urgentRingBuffer.AvailForGet();
if(!_PlaybackHandle || !_CaptureHandle)
return;
if (!_PlaybackHandle || !_CaptureHandle)
return;
snd_pcm_wait(_PlaybackHandle, 20);
snd_pcm_wait (_PlaybackHandle, 20);
int playbackAvailSmpl = snd_pcm_avail_update(_PlaybackHandle);
int playbackAvailBytes = playbackAvailSmpl*sizeof(SFLDataFormat);
int playbackAvailSmpl = snd_pcm_avail_update (_PlaybackHandle);
int playbackAvailBytes = playbackAvailSmpl*sizeof (SFLDataFormat);
// _debug("PLAYBACK: %d", playbackAvailSmpl);
if (urgentAvailBytes > 0) {
......@@ -957,18 +955,17 @@ void AlsaLayer::audioCallback(void)
free (out);
out = 0;
}
else if (file_tone && !_RingtoneHandle && (normalAvailBytes <= 0)) {
out = (SFLDataFormat *) malloc (playbackAvailBytes);
file_tone->getNext (out, playbackAvailSmpl, spkrVolume);
write (out, playbackAvailBytes, _PlaybackHandle);
} else if (file_tone && !_RingtoneHandle && (normalAvailBytes <= 0)) {
free (out);
out = NULL;
out = (SFLDataFormat *) malloc (playbackAvailBytes);
file_tone->getNext (out, playbackAvailSmpl, spkrVolume);
write (out, playbackAvailBytes, _PlaybackHandle);
free (out);
out = NULL;
} else {
} else {
// If nothing urgent, play the regular sound samples
......@@ -983,9 +980,9 @@ void AlsaLayer::audioCallback(void)
double upsampleFactor = (double) _audioSampleRate / _mainBufferSampleRate;
maxNbSamplesToGet = (int) ( (double) playbackAvailSmpl / upsampleFactor);
maxNbBytesToGet = maxNbSamplesToGet * sizeof (SFLDataFormat);
maxNbBytesToGet = maxNbSamplesToGet * sizeof (SFLDataFormat);
}
}
toGet = (normalAvailBytes < (int) maxNbBytesToGet) ? normalAvailBytes : maxNbBytesToGet;
......@@ -995,9 +992,9 @@ void AlsaLayer::audioCallback(void)
getMainBuffer()->getData (out, toGet, spkrVolume);
// TODO: Audio processing should be performed inside mainbuffer
// to avoid such problem
AudioLayer::_echoCancel->setSamplingRate(_mainBufferSampleRate);
// TODO: Audio processing should be performed inside mainbuffer
// to avoid such problem
AudioLayer::_echoCancel->setSamplingRate (_mainBufferSampleRate);
if (_mainBufferSampleRate && ( (int) _audioSampleRate != _mainBufferSampleRate)) {
......@@ -1018,18 +1015,18 @@ void AlsaLayer::audioCallback(void)
} else {
write (out, toGet, _PlaybackHandle);
write (out, toGet, _PlaybackHandle);
}
// Copy far-end signal in echo canceller to adapt filter coefficient