/**
* メッセージ
* @param[in] nMessage メッセージ
* @param[in] nParameter パラメータ
* @return 結果
*/
INTPTR CExternalPsg::Message(UINT nMessage, INTPTR nParameter)
{
switch (nMessage)
{
case kMute:
Mute(nParameter != 0);
break;
}
return 0;
}
GlobalShortcuts::GlobalShortcuts(QObject *parent)
: QObject(parent),
gnome_backend_(NULL),
system_backend_(NULL),
use_gnome_(false),
rating_signals_mapper_(new QSignalMapper(this))
{
settings_.beginGroup(kSettingsGroup);
// Create actions
AddShortcut("play", tr("Play"), SIGNAL(Play()));
AddShortcut("pause", tr("Pause"), SIGNAL(Pause()));
AddShortcut("play_pause", tr("Play/Pause"), SIGNAL(PlayPause()), QKeySequence(Qt::Key_MediaPlay));
AddShortcut("stop", tr("Stop"), SIGNAL(Stop()), QKeySequence(Qt::Key_MediaStop));
AddShortcut("stop_after", tr("Stop playing after current track"), SIGNAL(StopAfter()));
AddShortcut("next_track", tr("Next track"), SIGNAL(Next()), QKeySequence(Qt::Key_MediaNext));
AddShortcut("prev_track", tr("Previous track"), SIGNAL(Previous()), QKeySequence(Qt::Key_MediaPrevious));
AddShortcut("inc_volume", tr("Increase volume"), SIGNAL(IncVolume()));
AddShortcut("dec_volume", tr("Decrease volume"), SIGNAL(DecVolume()));
AddShortcut("mute", tr("Mute"), SIGNAL(Mute()));
AddShortcut("seek_forward", tr("Seek forward"), SIGNAL(SeekForward()));
AddShortcut("seek_backward", tr("Seek backward"), SIGNAL(SeekBackward()));
AddShortcut("show_hide", tr("Show/Hide"), SIGNAL(ShowHide()));
AddShortcut("show_osd", tr("Show OSD"), SIGNAL(ShowOSD()));
AddShortcut("toggle_pretty_osd", tr("Toggle Pretty OSD"), SIGNAL(TogglePrettyOSD())); // Toggling possible only for pretty OSD
AddShortcut("shuffle_mode", tr("Change shuffle mode"), SIGNAL(CycleShuffleMode()));
AddShortcut("repeat_mode", tr("Change repeat mode"), SIGNAL(CycleRepeatMode()));
AddShortcut("toggle_last_fm_scrobbling", tr("Enable/disable Last.fm scrobbling"), SIGNAL(ToggleScrobbling()));
AddShortcut("global_search_popup", tr("Show Global Search Popup"), SIGNAL(ShowGlobalSearch()));
AddRatingShortcut("rate_zero_star", tr("Rate the current song 0 stars"), rating_signals_mapper_, 0);
AddRatingShortcut("rate_one_star", tr("Rate the current song 1 star"), rating_signals_mapper_, 1);
AddRatingShortcut("rate_two_star", tr("Rate the current song 2 stars"), rating_signals_mapper_, 2);
AddRatingShortcut("rate_three_star", tr("Rate the current song 3 stars"), rating_signals_mapper_, 3);
AddRatingShortcut("rate_four_star", tr("Rate the current song 4 stars"), rating_signals_mapper_, 4);
AddRatingShortcut("rate_five_star", tr("Rate the current song 5 stars"), rating_signals_mapper_, 5);
connect(rating_signals_mapper_, SIGNAL(mapped(int)), SIGNAL(RateCurrentSong(int)));
// Create backends - these do the actual shortcut registration
gnome_backend_ = new GnomeGlobalShortcutBackend(this);
#ifndef Q_OS_DARWIN
system_backend_ = new QxtGlobalShortcutBackend(this);
#else
system_backend_ = new MacGlobalShortcutBackend(this);
#endif
ReloadSettings();
}
GlobalShortcuts::GlobalShortcuts(QObject *parent)
: QObject(parent),
backend(nullptr) {
// Create actions
AddShortcut("play", tr("Play"), SIGNAL(Play()));
AddShortcut("pause", tr("Pause"), SIGNAL(Pause()));
AddShortcut("play_pause", tr("Play/Pause"), SIGNAL(PlayPause()), QKeySequence(Qt::Key_MediaPlay));
AddShortcut("stop", tr("Stop"), SIGNAL(Stop()), QKeySequence(Qt::Key_MediaStop));
AddShortcut("stop_after", tr("Stop playing after current track"), SIGNAL(StopAfter()));
AddShortcut("next_track", tr("Next track"), SIGNAL(Next()), QKeySequence(Qt::Key_MediaNext));
AddShortcut("prev_track", tr("Previous track"), SIGNAL(Previous()), QKeySequence(Qt::Key_MediaPrevious));
AddShortcut("inc_volume", tr("Increase volume"), SIGNAL(IncVolume()));
AddShortcut("dec_volume", tr("Decrease volume"), SIGNAL(DecVolume()));
AddShortcut("mute", tr("Mute"), SIGNAL(Mute()));
AddShortcut("seek_forward", tr("Seek forward"), SIGNAL(SeekForward()));
AddShortcut("seek_backward", tr("Seek backward"), SIGNAL(SeekBackward()));
}
int FG::qt_metacall(QMetaObject::Call _c, int _id, void **_a)
{
_id = QMainWindow::qt_metacall(_c, _id, _a);
if (_id < 0)
return _id;
if (_c == QMetaObject::InvokeMetaMethod) {
switch (_id) {
case 0: receiveValue((*reinterpret_cast< double(*)>(_a[1]))); break;
case 1: receiveValue((*reinterpret_cast< QString(*)>(_a[1]))); break;
case 2: receiveValue((*reinterpret_cast< uint(*)>(_a[1]))); break;
case 3: callLineEditInput((*reinterpret_cast< int(*)>(_a[1]))); break;
case 4: Beep(); break;
case 5: Mute(); break;
case 6: on_AWGBox_clicked(); break;
case 7: on_modeBox_currentIndexChanged((*reinterpret_cast< int(*)>(_a[1]))); break;
case 8: on_calibrate_clicked(); break;
case 9: on_appBckPSoC_clicked(); break;
case 10: on_fineCoarse_clicked(); break;
case 11: on_leftBut_clicked(); break;
case 12: on_rightBut_clicked(); break;
case 13: on_exit_clicked(); break;
case 14: on_upBut_clicked(); break;
case 15: on_downBut_clicked(); break;
case 16: on_GateBut_clicked(); break;
case 17: on_EXTBut_clicked(); break;
case 18: on_INTBut_clicked(); break;
case 19: on_MANBut_clicked(); break;
case 20: on_RUNBut_clicked(); break;
case 21: on_hiZBut_clicked(); break;
case 22: on_triangleBut_clicked(); break;
case 23: on_rampBut_clicked(); break;
case 24: on_squareBut_clicked(); break;
case 25: on_sineBut_clicked(); break;
case 26: on_singleBut_clicked(); break;
case 27: on_burstBut_clicked(); break;
case 28: on_continuousBut_clicked(); break;
default: ;
}
_id -= 29;
}
return _id;
}
void
AudioContext::Shutdown()
{
mIsShutDown = true;
// We mute rather than suspending, because the delay between the ::Shutdown
// call and the CC would make us overbuffer in the MediaStreamGraph.
// See bug 936784 for details.
if (!mIsOffline) {
Mute();
}
// Release references to active nodes.
// Active AudioNodes don't unregister in destructors, at which point the
// Node is already unregistered.
mActiveNodes.Clear();
// For offline contexts, we can destroy the MediaStreamGraph at this point.
if (mIsOffline && mDestination) {
mDestination->OfflineShutdown();
}
}
void MpvHandler::SetProperties()
{
Volume(volume);
Speed(speed);
Mute(mute);
}
//*****************************************************************************
//receiving loop: grab 48KHz baseband samples from Line,
//demodulate, decrypt, decode, play 8KHz voice over Speaker
int rx(int typing)
{
//input: -1 for no typing chars, 1 - exist some chars in input buffer
//output: 0 - no any jobs doing, 1 - some jobs were doing
int i;
float f;
int job=0; //flag of any job were doing
char lag_flag=0; //block lag is locked (modems synchronization complete)
//char lock_flag=0; //phase of carrier (1333Hz, 6 samples per period) is locked
//char sync_flag=0; //the difference of frequency transmitter-to-receiver sampling rate is locked
//char current_lag=0; //block lag (0-90, aligned to last bit position, the 6 samples for bit)
char info[8]={0}; //call info
//regularly play Speaker's buffer
job=playjit(); //the first try to play a tail of samples in buffer
//check for we have enough samples for demodulation
if(cnt<180*6) //check we haven't enough of unprocessed samples
{
//move tail to start of receiving buffer
if(samples>speech) //check for tail
{
for(i=0; i<cnt; i++) speech[i]=samples[i]; //move tail to start of buffer
samples=speech; //set pointer to start of buffer
}
//record
i=_soundgrab((char*)(samples+cnt), 180*6); //try to grab new 48KHZ samples from Line
if((i>0)&&(i<=(180*6))) //some samples grabbed
{
cnt+=i; //add grabbed samples to account
job+=4; //set job
}
}
else //we have enough samples for processing
{
i=Demodulate(samples, buf); //process samples: 36*6 (35-37)*6 samples
samples+=i; //move pointer to next samples (with frequency adjusting)
cnt-=i; //decrease the number of unprocessed samples
if(0x80&buf[11]) //checks flag for output data block is ready
{
//check for synck and averages BER
lag_flag=!(!(buf[11]&0x40)); //block lag is locked (synchronization compleet)
//lock_flag=!(!(buf[11]&0x20)); //phaze of carrier (1333Hz, 6 samples per period) is locked
//sync_flag=!(!(buf[11]&0x10)); //the differency of frequency transmitter-to-receiver sampling rate is locked
//current_lag=buf[10]>>1; //block lag (0-90, aligned to last bit position, the 6 samples for bit)
if(lag_flag) //check modem sync
{
//averages BER
i=(0x0F&buf[11]); //count symbols errors (only 1 error per 9-bit symbol can be detected)
fber*=0.99; //fber in range 0-900
fber+=i; //in range 0-9 errored bits per 90 bits treceived
}
//output statistics
if(typing<0) //output call's info if no characters were typed by user
{
f=Mute(0); //get packets counter value
i=State(0); //get current connection step * vad flag
//notification of state and voice output
if(!i) strcpy(info, (char*)"IDLE");
else if(abs(i)<8) strcpy(info, (char*)"CALL");
else if(f<=0) strcpy(info, (char*)"MUTE");
else if(i<0) strcpy(info, (char*)"PAUS");
else strcpy(info, (char*)"TALK");
if(f<0) f=-f; //absolute value
i=f*0.0675; //computes total time of the call in sec: each packet 67,5 ms
f=fau/4-100; //computes authentification level in %
if(f<0) f=0; //only positive results have reason
//current state notification
if(lag_flag) printf("%s %dmin %dsec BER:%0.02f AU:%d%%\r", info, i/60, i%60, fber/90, (int)f);
else printf("%s %dmin %dsec BER:---- AU:%d%%\r", info, i/60, i%60, (int)f); //lost of sync in modem
}
//process received packet detects voice/silence type
buf[11]=0xFE; //set flag default as for silence descriptor
if(lag_flag) //check modem sync
{
i=ProcessPkt(buf); //decode received packet
if(i>=0) //received packet is a control type
{
fau*=0.99; //fau in range 0-800 (400 for random data)
fau+=i; //averages authentication level
}
else if(i==-3)
{
buf[11]=0xFF; //set flag for voice data received
}
} //end of sync ok, packets processing
} //end of data block received
} //end of a portion of sampless processing
//check we have received data and output buffer is empty for decoding
if((0x0E&buf[11])&&(l_jit_buf<=180))
{
//decode voice data or set silency
job+=16; //set job
if(1&buf[11]) //this is a voice frame, decode it
{
melpe_s(sp, buf); //decode 81 bits in 11 bytes to 540 8KHz samples
}
else memset(sp, 0, 1080); //or output 67.5 mS of silence
buf[11]=0; //clears flag: data buffer is processed
//computes average playing delay
i=getdelay()+l_jit_buf; //total number of unplayed samples in buffers
fdelay*=0.9; //averages
fdelay+=i;
//computes optimal resapling ratio for the optimum delay
f=fabs(fdelay/10-720)/10000000; //correction rate due inconsistency
if(i<360) qff-=f; //adjust current ratio
else if(i>1080) qff+=f;
if(qff<0.888) qff=0.888; //restrictions
else if(qff>1.142) qff=1.142;
//resample and play to Headset
if(l_jit_buf>180) l_jit_buf=0; //prevent overflow
l_jit_buf+=resample(sp, jit_buf+l_jit_buf, qff); //resample buffer for playing
playjit(); //immediately try to play buffer
}
return job;
}
~GimicIf() {
Log("YMF288::Reset()\n");
Mute();
}
//-----------------------------------------------------------------------------
//! Create all the menu actions.
//-----------------------------------------------------------------------------
void tNASBar::CreateActions()
{
TRACE_FUNCTION;
if ( m_FusionClientAgent.IsPowerButtonAllowed() )
{
// Some audio servers support a power button
m_pPowerAction = new tAction( tr("Power"), this );
m_pPowerAction->setCheckable(true);
m_pPowerAction->setChecked( m_FusionClientAgent.GetPowerState() != tFusionClient::ePS_Off );
Connect( m_pPowerAction , SIGNAL( triggered() ), this, SLOT( PowerClicked() ));
}
else
{
if (m_pPowerAction != NULL)
{
delete m_pPowerAction;
m_pPowerAction = 0;
}
}
if ( m_FusionClientAgent.IsOnAndInitialized() )
{
// Connect firmware progress message
Connect( &m_FusionClientAgent, SIGNAL(UpdateFirmwareProgress( quint8 )), this, SLOT(OnUpdateFirmwareProgress( quint8 )));
// Volume action
m_pVolumeAction = new tSliderAction(
tr( "Volume", "sound level for SonicHub/FUSION-Link" ),
m_FusionClientAgent.MinVolumeLevel(),
m_FusionClientAgent.MaxVolumeLevel(),
0, // current volume is set when the Audio zone or Sonic hub Fader actions are created
true,
"%v",
this
);
// Mute action
m_pMuteAction = new tAction( tr("Mute"), this );
m_pMuteAction->setCheckable( true );
m_pMuteAction->setChecked( m_FusionClientAgent.IsMuted() );
Connect( m_pMuteAction, SIGNAL( toggled( bool ) ), this, SLOT( SetMute( bool ) ) );
Connect( &m_FusionClientAgent, SIGNAL( Mute( tFusionClient::eState ) ), this, SLOT( OnMute( tFusionClient::eState ) ) );
Connect( &m_FusionClientAgent, SIGNAL( Mute( tFusionClient::eState ) ), m_pVolumeWidget, SLOT( OnMute( tFusionClient::eState ) ) );
// Audio sources.
m_pAudioSourceAction = new tAudioSourceAction( m_FusionClientAgent, m_MySourceId, this );
// Virtual head - giving an empty string as the text, the device bars themselves will fill it it.
// if you don't give it an empty string, it comes up as a dummy action and doesn't draw correctly on the softkey bar
// it is also disabled in the menus. Less work by just setting as empty string now.
m_pVirtualHeadAction = new tAction( "", this );
Connect( m_pVirtualHeadAction , SIGNAL( triggered() ), this, SLOT( ActivateVirtualHead() ), Qt::QueuedConnection );
m_pBluetoothDevicesAction = new tAction( tr( "Bluetooth devices" ), this );
Connect( m_pBluetoothDevicesAction , SIGNAL( triggered() ), this, SLOT( ShowBluetoothDialog() ), Qt::QueuedConnection );
m_pMixerActions = tMixerActions::Create( m_FusionClientAgent, this );
m_pZoneSetupAction = m_pMixerActions->MixerAction();
Connect( m_pMixerActions, SIGNAL( MasterVolumeChanged( int ) ), m_pVolumeWidget, SLOT( SetValue( int ) ) );
Connect( m_pMixerActions, SIGNAL( MasterVolumeChanged( int ) ), m_pVolumeAction, SLOT( SetValue( int ) ) );
Connect( m_pVolumeAction, SIGNAL( ValueChanged( int ) ), m_pMixerActions, SLOT( RequestVolumeLevel( int ) ) );
m_pMixerActions->UpdateMasterVolume();
}
//*****************************************************************************
//transmition loop: grab 8KHz speech samples from Mike,
//resample, collect frame (540 in 67.5 mS), encode
//encrypt, modulate, play 48KHz baseband signal into Line
int tx(int job)
{
int i,j;
//loop 1: try to play unplayed samples
job+=_playjit(); //the first try to play a tail of samples in buffer
//loop 2: try to grab next 180 samples
//check for number of grabbed samples
if(spcnt<540) //we haven't enought samples for melpe encoder
{
i=soundgrab((char*)spraw, 180); //grab up to 180 samples
if((i>0)&&(i<=180)) //if some samles was grabbed
{
//Since we are using different audio devices
//on headset and line sides, the sampling rates of grabbing
// and playing devices can slightly differ then 48/8 depends HW
//so we must adjusts one of rates for synchronizing grabbing and playing processes
//The line side is more sensitive (requirements for baseband is more hard)
//That why we resamples grabbed stream (slave) for matching rate with playing stream as a master
//The adjusting process doing approximation in iterative way
//and requires several seconds for adaptation during possible loss of some speech 67.5mS frames
//computes estimated rate depends recording delay obtained in moment of last block was modulated
j=8000-(_fdelay-27000)/50; //computes samplerate using optimal delay and adjusting sensitivity
if(j>9000) j=9000; //restrict resulting samplerate
if(j<7000) j=7000;
//change rate of grabbed samples for synchronizing grabbing and playing loops
i=_resample(spraw, spbuf+spcnt, i, j); //resample and collect speech samples
spcnt+=i; //the number of samples in buffer for processing
tgrab+=i; //the total difference between grabbed speech and played baseband samples
//this is actually recording delay and must be near 270 sample in average
//for jitter protecting (due PC multi threading etc.)
job+=32; //set job
}
}
//check for we have enough grabbed samples for processing
if(spcnt>=540) //we have enough samples for melpe encoder
{
if(Mute(0)>0)
{
i=vad2(spbuf+10, &vad); //check frame is speech (by VAD)
i+=vad2(spbuf+100,&vad);
i+=vad2(spbuf+190,&vad);
i+=vad2(spbuf+280,&vad);
i+=vad2(spbuf+370,&vad);
i+=vad2(spbuf+460,&vad);
}
else i=0;
txbuf[11]=0xFF; //set defaults flag for voiced frame
if(i) //frame is voices: compress it
{
melpe_a(txbuf, spbuf); //encode the speech frame
i=State(1); //set VAD flag
}
else //unvoiced frame: sync packet will be send
{
txbuf[11]=0xFE; //or set silence flag for control blocks
i=State(-1); //clears VAD flag
}
spcnt-=540; //samples rest
if(spcnt) memcpy((char*)spbuf, (char*)(spbuf+540), 2*spcnt); //move tail to start of buffer
job+=64;
}
//Loop 3: playing
//get number of unplayed samples in buffer
i=_getdelay();
//preventing of freezing audio output after underrun or overrun
if(i>540*3*6)
{
_soundflush1();
i=_getdelay();
}
//check for delay is acceptable for playing next portion of samples
if(i<720*6)
{
if(l__jit_buf) return job; //we have some unplayed samples in local buffer, not play now.
MakePkt(txbuf); //encrypt voice or get actual control packet
l__jit_buf=Modulate(txbuf, _jit_buf); //modulate block
txbuf[11]=0; //clear tx buffer (processed)
_playjit(); //immediately play baseband into Line
//estimate rate changing for grabbed samples for synchronizing grabbing and playing
_fdelay*=0.99; //smooth coefficient
_fdelay+=tgrab; //averages recording delay
tgrab-=540; //decrease counter of grabbed samples
job+=128;
}
return job;
}