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https://github.com/Swiftgram/Telegram-iOS.git
synced 2025-12-24 07:05:35 +00:00
Roll back tgcalls
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@@ -28,9 +28,6 @@
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#include "platform/darwin/iOS/tgcalls_audio_device_module_ios.h"
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#include "platform/darwin/iOS/RTCAudioSession.h"
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#include "platform/darwin/iOS/RTCAudioSessionConfiguration.h"
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#endif
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#import "group/GroupInstanceImpl.h"
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@@ -629,7 +626,7 @@ tgcalls::VideoCaptureInterfaceObject *GetVideoCaptureAssumingSameThread(tgcalls:
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}
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std::shared_ptr<tgcalls::VideoCaptureInterface> interface = strongSelf->_interface;
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/*if (false && requestClone) {
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if (false && requestClone) {
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VideoSampleBufferView *remoteRenderer = [[VideoSampleBufferView alloc] initWithFrame:CGRectZero];
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remoteRenderer.videoContentMode = UIViewContentModeScaleAspectFill;
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@@ -646,7 +643,7 @@ tgcalls::VideoCaptureInterfaceObject *GetVideoCaptureAssumingSameThread(tgcalls:
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}
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completion(remoteRenderer, cloneRenderer);
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} else */if ([VideoMetalView isSupported]) {
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} else if ([VideoMetalView isSupported]) {
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VideoMetalView *remoteRenderer = [[VideoMetalView alloc] initWithFrame:CGRectZero];
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remoteRenderer.videoContentMode = UIViewContentModeScaleAspectFill;
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@@ -708,8 +705,6 @@ tgcalls::VideoCaptureInterfaceObject *GetVideoCaptureAssumingSameThread(tgcalls:
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id<OngoingCallThreadLocalContextQueueWebrtc> _queue;
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int32_t _contextId;
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bool _useManualAudioSessionControl;
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OngoingCallNetworkTypeWebrtc _networkType;
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NSTimeInterval _callReceiveTimeout;
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NSTimeInterval _callRingTimeout;
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@@ -848,22 +843,7 @@ static void (*InternalVoipLoggingFunction)(NSString *) = NULL;
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}
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}
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- (instancetype _Nonnull)initWithVersion:(NSString * _Nonnull)version queue:(id<OngoingCallThreadLocalContextQueueWebrtc> _Nonnull)queue
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proxy:(VoipProxyServerWebrtc * _Nullable)proxy
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networkType:(OngoingCallNetworkTypeWebrtc)networkType dataSaving:(OngoingCallDataSavingWebrtc)dataSaving
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derivedState:(NSData * _Nonnull)derivedState
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key:(NSData * _Nonnull)key
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isOutgoing:(bool)isOutgoing
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connections:(NSArray<OngoingCallConnectionDescriptionWebrtc *> * _Nonnull)connections maxLayer:(int32_t)maxLayer
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allowP2P:(BOOL)allowP2P
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allowTCP:(BOOL)allowTCP
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enableStunMarking:(BOOL)enableStunMarking
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logPath:(NSString * _Nonnull)logPath
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statsLogPath:(NSString * _Nonnull)statsLogPath
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sendSignalingData:(void (^ _Nonnull)(NSData * _Nonnull))sendSignalingData videoCapturer:(OngoingCallThreadLocalContextVideoCapturer * _Nullable)videoCapturer
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preferredVideoCodec:(NSString * _Nullable)preferredVideoCodec
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audioInputDeviceId:(NSString * _Nonnull)audioInputDeviceId
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useManualAudioSessionControl:(bool)useManualAudioSessionControl {
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- (instancetype _Nonnull)initWithVersion:(NSString * _Nonnull)version queue:(id<OngoingCallThreadLocalContextQueueWebrtc> _Nonnull)queue proxy:(VoipProxyServerWebrtc * _Nullable)proxy networkType:(OngoingCallNetworkTypeWebrtc)networkType dataSaving:(OngoingCallDataSavingWebrtc)dataSaving derivedState:(NSData * _Nonnull)derivedState key:(NSData * _Nonnull)key isOutgoing:(bool)isOutgoing connections:(NSArray<OngoingCallConnectionDescriptionWebrtc *> * _Nonnull)connections maxLayer:(int32_t)maxLayer allowP2P:(BOOL)allowP2P allowTCP:(BOOL)allowTCP enableStunMarking:(BOOL)enableStunMarking logPath:(NSString * _Nonnull)logPath statsLogPath:(NSString * _Nonnull)statsLogPath sendSignalingData:(void (^)(NSData * _Nonnull))sendSignalingData videoCapturer:(OngoingCallThreadLocalContextVideoCapturer * _Nullable)videoCapturer preferredVideoCodec:(NSString * _Nullable)preferredVideoCodec audioInputDeviceId: (NSString * _Nonnull)audioInputDeviceId {
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self = [super init];
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if (self != nil) {
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_version = version;
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@@ -872,25 +852,6 @@ static void (*InternalVoipLoggingFunction)(NSString *) = NULL;
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assert([[OngoingCallThreadLocalContextWebrtc versionsWithIncludeReference:true] containsObject:version]);
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_useManualAudioSessionControl = useManualAudioSessionControl;
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[RTCAudioSession sharedInstance].useManualAudio = true;
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#ifdef WEBRTC_IOS
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RTCAudioSessionConfiguration *sharedConfiguration = [RTCAudioSessionConfiguration webRTCConfiguration];
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if (useManualAudioSessionControl) {
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sharedConfiguration.mode = AVAudioSessionModeVoiceChat;
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} else {
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sharedConfiguration.mode = AVAudioSessionModeVoiceChat;
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}
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sharedConfiguration.categoryOptions |= AVAudioSessionCategoryOptionMixWithOthers;
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sharedConfiguration.outputNumberOfChannels = 1;
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[RTCAudioSessionConfiguration setWebRTCConfiguration:sharedConfiguration];
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/*[RTCAudioSession sharedInstance].useManualAudio = true;
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[[RTCAudioSession sharedInstance] audioSessionDidActivate:[AVAudioSession sharedInstance]];
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[RTCAudioSession sharedInstance].isAudioEnabled = true;*/
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#endif
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_callReceiveTimeout = 20.0;
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_callRingTimeout = 90.0;
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_callConnectTimeout = 30.0;
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@@ -1133,17 +1094,6 @@ static void (*InternalVoipLoggingFunction)(NSString *) = NULL;
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- (void)beginTermination {
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}
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- (void)setManualAudioSessionIsActive:(bool)isAudioSessionActive {
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if (_useManualAudioSessionControl) {
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if (isAudioSessionActive) {
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[[RTCAudioSession sharedInstance] audioSessionDidActivate:[AVAudioSession sharedInstance]];
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} else {
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[[RTCAudioSession sharedInstance] audioSessionDidDeactivate:[AVAudioSession sharedInstance]];
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}
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[RTCAudioSession sharedInstance].isAudioEnabled = isAudioSessionActive;
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}
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}
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+ (void)stopWithTerminationResult:(OngoingCallThreadLocalContextWebrtcTerminationResult *)terminationResult completion:(void (^)(NSString *, int64_t, int64_t, int64_t, int64_t))completion {
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if (completion) {
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if (terminationResult) {
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@@ -1479,22 +1429,6 @@ private:
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}
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}
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#ifdef WEBRTC_IOS
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RTCAudioSessionConfiguration *sharedConfiguration = [RTCAudioSessionConfiguration webRTCConfiguration];
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sharedConfiguration.mode = AVAudioSessionModeVoiceChat;
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sharedConfiguration.categoryOptions |= AVAudioSessionCategoryOptionMixWithOthers;
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if (disableAudioInput) {
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sharedConfiguration.outputNumberOfChannels = 2;
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} else {
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sharedConfiguration.outputNumberOfChannels = 1;
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}
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[RTCAudioSessionConfiguration setWebRTCConfiguration:sharedConfiguration];
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/*[RTCAudioSession sharedInstance].useManualAudio = true;
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[[RTCAudioSession sharedInstance] audioSessionDidActivate:[AVAudioSession sharedInstance]];
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[RTCAudioSession sharedInstance].isAudioEnabled = true;*/
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#endif
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std::vector<tgcalls::VideoCodecName> videoCodecPreferences;
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int minOutgoingVideoBitrateKbit = 500;
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@@ -1678,15 +1612,6 @@ private:
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}
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}
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- (void)setManualAudioSessionIsActive:(bool)isAudioSessionActive {
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if (isAudioSessionActive) {
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[[RTCAudioSession sharedInstance] audioSessionDidActivate:[AVAudioSession sharedInstance]];
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} else {
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[[RTCAudioSession sharedInstance] audioSessionDidDeactivate:[AVAudioSession sharedInstance]];
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}
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[RTCAudioSession sharedInstance].isAudioEnabled = isAudioSessionActive;
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}
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- (void)setConnectionMode:(OngoingCallConnectionMode)connectionMode keepBroadcastConnectedIfWasEnabled:(bool)keepBroadcastConnectedIfWasEnabled isUnifiedBroadcast:(bool)isUnifiedBroadcast {
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if (_instance) {
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tgcalls::GroupConnectionMode mappedConnectionMode;
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