Commit 073bfc05 authored by Rafaël Carré's avatar Rafaël Carré

* #6629 : remove unused and unconfigurable frameSize from audiolayer

parent 3ddc5dbd
......@@ -133,7 +133,7 @@ AlsaLayer::closeLayer()
}
void
AlsaLayer::openDevice (int indexIn, int indexOut, int indexRing, int sampleRate, int frameSize, int stream , const std::string &plugin)
AlsaLayer::openDevice (int indexIn, int indexOut, int indexRing, int sampleRate, int stream , const std::string &plugin)
{
/* Close the devices before open it */
if (stream == SFL_PCM_BOTH and is_capture_open_ and is_playback_open_) {
......@@ -149,13 +149,11 @@ AlsaLayer::openDevice (int indexIn, int indexOut, int indexRing, int sampleRate,
indexRing_ = indexRing;
audioSampleRate_ = sampleRate;
frameSize_ = frameSize;
audioPlugin_ = plugin;
_debug (" Setting AlsaLayer: device in=%2d, out=%2d, ring=%2d", indexIn_, indexOut_, indexRing_);
_debug (" : alsa plugin=%s", audioPlugin_.c_str());
_debug (" : nb channel in=%2d, out=%2d", inChannel_, outChannel_);
_debug (" : sample rate=%5d, format=%s", audioSampleRate_, SFLDataFormatString);
audioThread_ = NULL;
......
......@@ -64,14 +64,13 @@ class AlsaLayer : public AudioLayer
* @param indexIn The number of the card chosen for capture
* @param indexOut The number of the card chosen for playback
* @param sampleRate The sample rate
* @param frameSize The frame size
* @param stream To indicate which kind of stream you want to open
* SFL_PCM_CAPTURE
* SFL_PCM_PLAYBACK
* SFL_PCM_BOTH
* @param plugin The alsa plugin ( dmix , default , front , surround , ...)
*/
void openDevice (int indexIn, int indexOut, int indexRing, int sampleRate, int frameSize, int stream, const std::string &plugin);
void openDevice (int indexIn, int indexOut, int indexRing, int sampleRate, int stream, const std::string &plugin);
/**
* Start the capture stream and prepare the playback stream.
......
......@@ -42,9 +42,6 @@ AudioLayer::AudioLayer (int type)
, indexOut_ (0)
, indexRing_ (0)
, audioSampleRate_ (0)
, frameSize_ (0)
, inChannel_ (1)
, outChannel_ (1)
, mutex_ ()
, audioPref(Manager::instance().audioPreference)
, layerType_ (type)
......
......@@ -78,14 +78,13 @@ class AudioLayer
* @param indexIn The number of the card chosen for capture
* @param indexOut The number of the card chosen for playback
* @param sampleRate The sample rate
* @param frameSize The frame size
* @param stream To indicate which kind of stream you want to open
* SFL_PCM_CAPTURE
* SFL_PCM_PLAYBACK
* SFL_PCM_BOTH
* @param plugin The alsa plugin ( dmix , default , front , surround , ...)
*/
virtual void openDevice (int indexIn, int indexOut, int indexRing, int sampleRate, int frameSize, int stream , const std::string &plugin) = 0;
virtual void openDevice (int indexIn, int indexOut, int indexRing, int sampleRate, int stream , const std::string &plugin) = 0;
/**
* Start the capture stream and prepare the playback stream.
......@@ -151,15 +150,6 @@ class AudioLayer
return audioSampleRate_;
}
/**
* Get the frame size of the audio layer
* @return unsigned int The frame size
* default: 20 ms
*/
unsigned int getFrameSize() const {
return frameSize_;
}
/**
* Get the layer type for this instance (either Alsa or PulseAudio)
* @return unsigned int The layer type
......@@ -252,22 +242,6 @@ class AudioLayer
*/
unsigned int audioSampleRate_;
/**
* Length of the sound frame we capture or read in ms
* The value can be set in the user config file - now: 20ms
*/
unsigned int frameSize_;
/**
* Input channel (mic) should be 1 mono
*/
unsigned int inChannel_;
/**
* Output channel (stereo) should be 1 mono
*/
unsigned int outChannel_;
/**
* Lock for the entire audio layer
*/
......
......@@ -370,12 +370,9 @@ void PulseLayer::context_state_callback (pa_context* c, void* user_data)
}
}
void PulseLayer::openDevice (int indexIn UNUSED, int indexOut UNUSED, int indexRing UNUSED, int sampleRate, int frameSize , int stream UNUSED, const std::string &plugin UNUSED)
void PulseLayer::openDevice (int indexIn UNUSED, int indexOut UNUSED, int indexRing UNUSED, int sampleRate, int stream UNUSED, const std::string &plugin UNUSED)
{
_debug ("Audio: Open device sampling rate %d, frame size %d", audioSampleRate_, frameSize_);
audioSampleRate_ = sampleRate;
frameSize_ = frameSize;
flushUrgent();
......
......@@ -62,14 +62,13 @@ class PulseLayer : public AudioLayer
* @param indexIn The number of the card chosen for capture
* @param indexOut The number of the card chosen for playback
* @param sampleRate The sample rate
* @param frameSize The frame size
* @param stream To indicate which kind of stream you want to open
* SFL_PCM_CAPTURE
* SFL_PCM_PLAYBACK
* SFL_PCM_BOTH
* @param plugin The alsa plugin ( dmix , default , front , surround , ...)
*/
void openDevice (int indexIn, int indexOut, int indexRing, int sampleRate, int frameSize , int stream, const std::string &plugin) ;
void openDevice (int indexIn, int indexOut, int indexRing, int sampleRate, int stream, const std::string &plugin) ;
DeviceList* getSinkList (void) {
return &sinkList_;
......
......@@ -713,7 +713,7 @@ IAXVoIPLink::iaxHandleVoiceEvent (iax_event* event, IAXCall* call)
unsigned int size = event->datalen;
// Decode data with relevant codec
unsigned int max = audioCodec->getClockRate() * audiolayer_->getFrameSize() / 1000;
unsigned int max = audioCodec->getClockRate() * 20 / 1000;
if (size > max) {
_debug ("The size %d is bigger than expected %d. Packet cropped. Ouch!", size, max);
......
......@@ -2056,7 +2056,7 @@ void ManagerImpl::setAudioPlugin (const std::string& audioPlugin)
if (_audiodriver -> getLayerType() == ALSA) {
_audiodriver -> openDevice (_audiodriver->getIndexIn(), _audiodriver->getIndexOut(),
_audiodriver->getIndexRing(), _audiodriver -> getSampleRate(),
_audiodriver -> getFrameSize(), SFL_PCM_BOTH, audioPlugin);
SFL_PCM_BOTH, audioPlugin);
}
audioLayerMutexUnlock();
}
......@@ -2089,22 +2089,19 @@ void ManagerImpl::setAudioDevice (const int index, int streamType)
case SFL_PCM_PLAYBACK:
_debug ("Manager: Set output device");
_audiodriver->openDevice (_audiodriver->getIndexIn(), index, _audiodriver->getIndexRing(),
_audiodriver->getSampleRate(), _audiodriver->getFrameSize(),
SFL_PCM_PLAYBACK, alsaplugin);
_audiodriver->getSampleRate(), SFL_PCM_PLAYBACK, alsaplugin);
audioPreference.setCardout (index);
break;
case SFL_PCM_CAPTURE:
_debug ("Manager: Set input device");
_audiodriver->openDevice (index, _audiodriver->getIndexOut(), _audiodriver->getIndexRing(),
_audiodriver->getSampleRate(), _audiodriver->getFrameSize(),
SFL_PCM_CAPTURE, alsaplugin);
_audiodriver->getSampleRate(), SFL_PCM_CAPTURE, alsaplugin);
audioPreference.setCardin (index);
break;
case SFL_PCM_RINGTONE:
_debug ("Manager: Set ringtone device");
_audiodriver->openDevice (_audiodriver->getIndexOut(), _audiodriver->getIndexOut(), index,
_audiodriver->getSampleRate(), _audiodriver->getFrameSize(),
SFL_PCM_RINGTONE, alsaplugin);
_audiodriver->getSampleRate(), SFL_PCM_RINGTONE, alsaplugin);
audioPreference.setCardring (index);
break;
default:
......@@ -2491,7 +2488,6 @@ void ManagerImpl::selectAudioDriver (void)
int numCardRing = audioPreference.getCardring();
int sampleRate = getMainBuffer()->getInternalSamplingRate();
int frameSize = audioPreference.getFramesize();
/* Only for the ALSA layer, we check the sound card information */
......@@ -2518,8 +2514,7 @@ void ManagerImpl::selectAudioDriver (void)
}
/* Open the audio devices */
_audiodriver->openDevice (numCardIn, numCardOut, numCardRing, sampleRate, frameSize,
SFL_PCM_BOTH, alsaPlugin);
_audiodriver->openDevice (numCardIn, numCardOut, numCardRing, sampleRate, SFL_PCM_BOTH, alsaPlugin);
audioLayerMutexUnlock();
}
......@@ -2540,9 +2535,6 @@ void ManagerImpl::switchAudioManager (void)
int type = _audiodriver->getLayerType();
int samplerate = _mainBuffer.getInternalSamplingRate();
int framesize = audioPreference.getFramesize();
_debug ("Manager: samplerate: %d, framesize %d", samplerate, framesize);
std::string alsaPlugin(audioPreference.getPlugin());
......@@ -2558,8 +2550,7 @@ void ManagerImpl::switchAudioManager (void)
else
_audiodriver = new AlsaLayer();
_audiodriver->openDevice (numCardIn, numCardOut, numCardRing, samplerate, framesize,
SFL_PCM_BOTH, alsaPlugin);
_audiodriver->openDevice (numCardIn, numCardOut, numCardRing, samplerate, SFL_PCM_BOTH, alsaPlugin);
_debug ("Manager: Current device: %d ", type);
......@@ -2591,9 +2582,8 @@ void ManagerImpl::audioSamplingRateChanged (int samplerate)
_debug ("Manager: Audio sampling rate changed");
int type = _audiodriver->getLayerType();
int framesize = audioPreference.getFramesize();
_debug ("Manager: New samplerate: %d, New framesize %d", samplerate, framesize);
_debug ("Manager: New samplerate: %d", samplerate);
std::string alsaPlugin(audioPreference.getPlugin());
......@@ -2611,8 +2601,7 @@ void ManagerImpl::audioSamplingRateChanged (int samplerate)
else
_audiodriver = new PulseLayer;
_audiodriver->openDevice (numCardIn, numCardOut, numCardRing, samplerate, framesize,
SFL_PCM_BOTH, alsaPlugin);
_audiodriver->openDevice (numCardIn, numCardOut, numCardRing, samplerate, SFL_PCM_BOTH, alsaPlugin);
_debug ("Manager: Current device: %d ", type);
......
......@@ -39,7 +39,6 @@ namespace {
static const char * const DFT_PULSE_LENGTH_STR ="250"; /** Default DTMF lenght */
static const char * const ZRTP_ZIDFILE = "zidFile"; /** The filename used for storing ZIDs */
static const char * const ALSA_DFT_CARD = "0"; /** Default sound card index */
static const char * const DFT_FRAME_SIZE = "20"; /** Default frame size in millisecond */
static const char * const DFT_VOL_SPKR_STR = "100"; /** Default speaker volume */
static const char * const DFT_VOL_MICRO_STR = "100"; /** Default mic volume */
} // end anonymous namespace
......@@ -282,7 +281,6 @@ void HookPreference::unserialize (Conf::MappingNode *map)
AudioPreference::AudioPreference() : _cardin (atoi (ALSA_DFT_CARD)) // ALSA_DFT_CARD
, _cardout (atoi (ALSA_DFT_CARD)) // ALSA_DFT_CARD
, _cardring (atoi (ALSA_DFT_CARD)) // ALSA_DFT_CARD
, _framesize (atoi (DFT_FRAME_SIZE)) // DFT_FRAME_SIZE
, _plugin ("default") // PCM_DEFAULT
, _smplrate (44100) // DFT_SAMPLE_RATE
, _devicePlayback ("")
......@@ -318,9 +316,6 @@ void AudioPreference::serialize (Conf::YamlEmitter *emitter)
std::stringstream ringstr;
ringstr << _cardring;
Conf::ScalarNode cardring (ringstr.str());// 0
std::stringstream framestr;
framestr << _framesize;
Conf::ScalarNode framesize (framestr.str()); // 20
Conf::ScalarNode plugin (_plugin); // default
std::stringstream ratestr;
......@@ -361,7 +356,6 @@ void AudioPreference::serialize (Conf::YamlEmitter *emitter)
alsapreferencemap.setKeyValue (cardinKey, &cardin);
alsapreferencemap.setKeyValue (cardoutKey, &cardout);
alsapreferencemap.setKeyValue (cardringKey, &cardring);
alsapreferencemap.setKeyValue (framesizeKey, &framesize);
alsapreferencemap.setKeyValue (pluginKey, &plugin);
alsapreferencemap.setKeyValue (smplrateKey, &smplrate);
......@@ -395,7 +389,6 @@ void AudioPreference::unserialize (Conf::MappingNode *map)
alsamap->getValue (cardinKey, &_cardin);
alsamap->getValue (cardoutKey, &_cardout);
alsamap->getValue (cardringKey, &_cardring);
alsamap->getValue (framesizeKey, &_framesize);
alsamap->getValue (smplrateKey, &_smplrate);
alsamap->getValue (pluginKey, &_plugin);
}
......
......@@ -76,7 +76,6 @@ const std::string pulsemapKey ("pulse");
const std::string cardinKey ("cardIn");// : 0
const std::string cardoutKey ("cardOut");// 0
const std::string cardringKey ("cardRing");// : 0
const std::string framesizeKey ("frameSize");// : 20
const std::string pluginKey ("plugin"); //: default
const std::string smplrateKey ("smplRate");//: 44100
const std::string devicePlaybackKey ("devicePlayback");//:
......@@ -444,13 +443,6 @@ class AudioPreference : public Serializable
_cardring = c;
}
int getFramesize (void) const {
return _framesize;
}
void setFramesize (int f) {
_framesize = f;
}
std::string getPlugin (void) const {
return _plugin;
}
......@@ -555,7 +547,6 @@ class AudioPreference : public Serializable
int _cardin; // 0
int _cardout; // 0
int _cardring;// 0
int _framesize; // 20
std::string _plugin; // default
int _smplrate;// 44100
......
......@@ -44,7 +44,6 @@ void AudioLayerTest::testAudioLayerConfig()
_debug ("-------------------- AudioLayerTest::testAudioLayerConfig --------------------\n");
CPPUNIT_ASSERT( Manager::instance().audioPreference.getSmplrate() == 44100);
CPPUNIT_ASSERT( Manager::instance().audioPreference.getFramesize() == 20);
// alsa preferences
CPPUNIT_ASSERT( Manager::instance().audioPreference.getCardin() == 0);
......@@ -62,8 +61,6 @@ void AudioLayerTest::testAudioLayerConfig()
// TODO: Fix tests
//CPPUNIT_ASSERT ( (int) Manager::instance().getAudioDriver()->getSampleRate() == sampling_rate);
//CPPPUNIT_ASSERT ( (int) Manager::instance().getAudioDriver()->getFrameSize() == frame_size);
}
void AudioLayerTest::testAudioLayerSwitch()
......@@ -104,21 +101,15 @@ void AudioLayerTest::testPulseConnect()
CPPUNIT_ASSERT (_pulselayer->getLayerType() == PULSEAUDIO);
std::string alsaPlugin;
int numCardIn, numCardOut, numCardRing, sampleRate, frameSize;
int sampleRate;
alsaPlugin = manager->audioPreference.getPlugin();
numCardIn = manager->audioPreference.getCardin();
numCardOut = manager->audioPreference.getCardout();
numCardRing = manager->audioPreference.getCardring();
sampleRate = manager->audioPreference.getSmplrate();
frameSize = manager->audioPreference.getFramesize();
CPPUNIT_ASSERT (_pulselayer->getPlaybackStream() == NULL);
CPPUNIT_ASSERT (_pulselayer->getRecordStream() == NULL);
try {
_pulselayer->openDevice (numCardIn, numCardOut, numCardRing, sampleRate, frameSize, SFL_PCM_BOTH, alsaPlugin);
_pulselayer->openDevice (sampleRate);
} catch (...) {
_debug ("Exception occured wile opening device! ");
}
......
......@@ -44,7 +44,6 @@ void ConfigurationTest::testDefaultValueAudio()
CPPUNIT_ASSERT (Manager::instance().audioPreference.getCardin() == 0); // ALSA_DFT_CARD);
CPPUNIT_ASSERT (Manager::instance().audioPreference.getCardout() == 0); // ALSA_DFT_CARD);
CPPUNIT_ASSERT (Manager::instance().audioPreference.getSmplrate() == 44100); // DFT_SAMPLE_RATE);
CPPUNIT_ASSERT (Manager::instance().audioPreference.getFramesize() == 20); // DFT_FRAME_SIZE);
CPPUNIT_ASSERT (Manager::instance().audioPreference.getPlugin() == PCM_DEFAULT);
CPPUNIT_ASSERT (Manager::instance().audioPreference.getVolumespkr() == 100);
CPPUNIT_ASSERT (Manager::instance().audioPreference.getVolumemic() == 100);
......
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