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[WIP] Conference

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This commit is contained in:
Isaac
2025-03-19 18:55:24 +01:00
parent 40b19cfef2
commit 17dd059f13
33 changed files with 2034 additions and 283 deletions
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90
submodules/TgVoipWebrtc/Sources/OngoingCallThreadLocalContext.mm
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@@ -127,7 +127,7 @@ public:
public:
virtual rtc::scoped_refptr<tgcalls::WrappedAudioDeviceModule> audioDeviceModule() = 0;
virtual rtc::scoped_refptr<tgcalls::WrappedAudioDeviceModule> makeChildAudioDeviceModule() = 0;
virtual rtc::scoped_refptr<tgcalls::WrappedAudioDeviceModule> makeChildAudioDeviceModule(bool isActive) = 0;
virtual void start() = 0;
};
@@ -147,14 +147,14 @@ public:
return 0;
}
void UpdateAudioCallback(webrtc::AudioTransport *previousAudioCallback, webrtc::AudioTransport *audioCallback) {
void UpdateAudioCallback(webrtc::AudioTransport *previousAudioCallback, webrtc::AudioTransport *audioCallback, bool isActive) {
_mutex.Lock();
if (audioCallback) {
_audioTransports.push_back(audioCallback);
_audioTransports.push_back(std::make_pair(audioCallback, isActive));
} else if (previousAudioCallback) {
for (size_t i = 0; i < _audioTransports.size(); i++) {
if (_audioTransports[i] == previousAudioCallback) {
if (_audioTransports[i].first == previousAudioCallback) {
_audioTransports.erase(_audioTransports.begin() + i);
break;
}
@@ -163,6 +163,18 @@ public:
_mutex.Unlock();
}
void UpdateAudioCallbackIsActive(webrtc::AudioTransport *audioCallback, bool isActive) {
_mutex.Lock();
for (auto &it : _audioTransports) {
if (it.first == audioCallback) {
it.second = isActive;
}
}
_mutex.Unlock();
}
virtual int32_t RegisterAudioCallback(webrtc::AudioTransport *audioCallback) override {
return 0;
@@ -474,7 +486,7 @@ public:
_mutex.Lock();
if (!_audioTransports.empty()) {
for (size_t i = 0; i < _audioTransports.size(); i++) {
_audioTransports[i]->RecordedDataIsAvailable(
_audioTransports[i].first->RecordedDataIsAvailable(
audioSamples,
nSamples,
nBytesPerSample,
@@ -508,7 +520,7 @@ public:
_mutex.Lock();
if (!_audioTransports.empty()) {
for (size_t i = 0; i < _audioTransports.size(); i++) {
_audioTransports[i]->RecordedDataIsAvailable(
_audioTransports[i].first->RecordedDataIsAvailable(
audioSamples,
nSamples,
nBytesPerSample,
@@ -552,11 +564,14 @@ public:
int16_t *resultAudioSamples = (int16_t *)audioSamples;
for (size_t i = 0; i < _audioTransports.size(); i++) {
if (!_audioTransports[i].second) {
continue;
}
int64_t localElapsedTimeMs = 0;
int64_t localNtpTimeMs = 0;
size_t localNSamplesOut = 0;
_audioTransports[i]->NeedMorePlayData(
_audioTransports[i].first->NeedMorePlayData(
nSamples,
nBytesPerSample,
nChannels,
@@ -584,7 +599,7 @@ public:
}
nSamplesOut = nSamples;
} else {
result = _audioTransports[_audioTransports.size() - 1]->NeedMorePlayData(
result = _audioTransports[_audioTransports.size() - 1].first->NeedMorePlayData(
nSamples,
nBytesPerSample,
nChannels,
@@ -616,7 +631,7 @@ public:
_mutex.Lock();
if (!_audioTransports.empty()) {
_audioTransports[_audioTransports.size() - 1]->PullRenderData(
_audioTransports[_audioTransports.size() - 1].first->PullRenderData(
bits_per_sample,
sample_rate,
number_of_channels,
@@ -650,6 +665,9 @@ public:
virtual void Stop() override {
}
virtual void setIsActive(bool isActive) override {
}
virtual void ActualStop() {
if (_isStarted) {
_isStarted = false;
@@ -661,22 +679,23 @@ public:
private:
bool _isStarted = false;
std::vector<webrtc::AudioTransport *> _audioTransports;
std::vector<std::pair<webrtc::AudioTransport *, bool>> _audioTransports;
webrtc::Mutex _mutex;
std::vector<int16_t> _mixAudioSamples;
};
class WrappedChildAudioDeviceModule : public tgcalls::DefaultWrappedAudioDeviceModule {
public:
WrappedChildAudioDeviceModule(webrtc::scoped_refptr<WrappedAudioDeviceModuleIOS> impl) :
tgcalls::DefaultWrappedAudioDeviceModule(impl) {
WrappedChildAudioDeviceModule(webrtc::scoped_refptr<WrappedAudioDeviceModuleIOS> impl, bool isActive) :
tgcalls::DefaultWrappedAudioDeviceModule(impl),
_isActive(isActive) {
}
virtual ~WrappedChildAudioDeviceModule() {
if (_audioCallback) {
auto previousAudioCallback = _audioCallback;
_audioCallback = nullptr;
((WrappedAudioDeviceModuleIOS *)WrappedInstance().get())->UpdateAudioCallback(previousAudioCallback, nullptr);
((WrappedAudioDeviceModuleIOS *)WrappedInstance().get())->UpdateAudioCallback(previousAudioCallback, nullptr, false);
}
}
@@ -685,21 +704,20 @@ public:
_audioCallback = audioCallback;
if (_isActive) {
((WrappedAudioDeviceModuleIOS *)WrappedInstance().get())->UpdateAudioCallback(previousAudioCallback, audioCallback);
((WrappedAudioDeviceModuleIOS *)WrappedInstance().get())->UpdateAudioCallback(previousAudioCallback, audioCallback, _isActive);
}
return 0;
}
public:
void setIsActive() {
if (_isActive) {
void setIsActive(bool isActive) override {
if (_isActive == isActive) {
return;
}
_isActive = true;
_isActive = isActive;
if (_audioCallback) {
((WrappedAudioDeviceModuleIOS *)WrappedInstance().get())->UpdateAudioCallback(nullptr, _audioCallback);
((WrappedAudioDeviceModuleIOS *)WrappedInstance().get())->UpdateAudioCallbackIsActive(_audioCallback, isActive);
}
}
@@ -733,8 +751,8 @@ public:
return _audioDeviceModule;
}
rtc::scoped_refptr<tgcalls::WrappedAudioDeviceModule> makeChildAudioDeviceModule() override {
return rtc::make_ref_counted<WrappedChildAudioDeviceModule>(_audioDeviceModule);
rtc::scoped_refptr<tgcalls::WrappedAudioDeviceModule> makeChildAudioDeviceModule(bool isActive) override {
return rtc::make_ref_counted<WrappedChildAudioDeviceModule>(_audioDeviceModule, isActive);
}
virtual void start() override {
@@ -1928,8 +1946,7 @@ static void (*InternalVoipLoggingFunction)(NSString *) = NULL;
},
.createWrappedAudioDeviceModule = [audioDeviceModule](webrtc::TaskQueueFactory *taskQueueFactory) -> rtc::scoped_refptr<tgcalls::WrappedAudioDeviceModule> {
if (audioDeviceModule) {
auto result = audioDeviceModule->getSyncAssumingSameThread()->makeChildAudioDeviceModule();
((WrappedChildAudioDeviceModule *)result.get())->setIsActive();
auto result = audioDeviceModule->getSyncAssumingSameThread()->makeChildAudioDeviceModule(true);
return result;
} else {
return nullptr;
@@ -2260,6 +2277,17 @@ static void (*InternalVoipLoggingFunction)(NSString *) = NULL;
}
}
- (void)deactivateIncomingAudio {
if (_currentAudioDeviceModuleThread) {
auto currentAudioDeviceModule = _currentAudioDeviceModule;
if (currentAudioDeviceModule) {
_currentAudioDeviceModuleThread->PostTask([currentAudioDeviceModule]() {
((tgcalls::WrappedAudioDeviceModule *)currentAudioDeviceModule.get())->setIsActive(false);
});
}
}
}
@end
namespace {
@@ -2347,7 +2375,8 @@ statsLogPath:(NSString * _Nonnull)statsLogPath
onMutedSpeechActivityDetected:(void (^ _Nullable)(bool))onMutedSpeechActivityDetected
audioDevice:(SharedCallAudioDevice * _Nullable)audioDevice
encryptionKey:(NSData * _Nullable)encryptionKey
isConference:(bool)isConference {
isConference:(bool)isConference
isActiveByDefault:(bool)isActiveByDefault {
self = [super init];
if (self != nil) {
_queue = queue;
@@ -2636,10 +2665,9 @@ isConference:(bool)isConference {
return resultModule;
}
},
.createWrappedAudioDeviceModule = [audioDeviceModule](webrtc::TaskQueueFactory *taskQueueFactory) -> rtc::scoped_refptr<tgcalls::WrappedAudioDeviceModule> {
.createWrappedAudioDeviceModule = [audioDeviceModule, isActiveByDefault](webrtc::TaskQueueFactory *taskQueueFactory) -> rtc::scoped_refptr<tgcalls::WrappedAudioDeviceModule> {
if (audioDeviceModule) {
auto result = audioDeviceModule->getSyncAssumingSameThread()->makeChildAudioDeviceModule();
((WrappedChildAudioDeviceModule *)result.get())->setIsActive();
auto result = audioDeviceModule->getSyncAssumingSameThread()->makeChildAudioDeviceModule(isActiveByDefault);
return result;
} else {
return nullptr;
@@ -2983,6 +3011,14 @@ isConference:(bool)isConference {
}
- (void)activateIncomingAudio {
if (_currentAudioDeviceModuleThread) {
auto currentAudioDeviceModule = _currentAudioDeviceModule;
if (currentAudioDeviceModule) {
_currentAudioDeviceModuleThread->PostTask([currentAudioDeviceModule]() {
((tgcalls::WrappedAudioDeviceModule *)currentAudioDeviceModule.get())->setIsActive(true);
});
}
}
}
@end