Repo created

TMessagesProj/jni/voip/tgcalls/platform/darwin/iOS/CallAudioTone.h

@@ -0,0 +1,48 @@
+#ifndef TGCALLS_CALL_AUDIO_TONE_H_
+#define TGCALLS_CALL_AUDIO_TONE_H_
+
+#include <vector>
+
+namespace tgcalls {
+
+class CallAudioTone {
+public:
+    CallAudioTone(std::vector<int16_t> &&samples, int sampleRate, int loopCount) :
+    _samples(std::move(samples)), _sampleRate(sampleRate), _loopCount(loopCount) {
+    }
+
+public:
+    std::vector<int16_t> const samples() const {
+        return _samples;
+    }
+
+    int sampleRate() const {
+        return _sampleRate;
+    }
+
+    int loopCount() const {
+        return _loopCount;
+    }
+    
+    void setLoopCount(int loopCount) {
+        _loopCount = loopCount;
+    }
+    
+    size_t offset() const {
+        return _offset;
+    }
+    
+    void setOffset(size_t offset) {
+        _offset = offset;
+    }
+
+private:
+    std::vector<int16_t> _samples;
+    int _sampleRate = 48000;
+    int _loopCount = 1;
+    size_t _offset = 0;
+};
+
+}
+
+#endif  // TGCALLS_CALL_AUDIO_TONE_H_

TMessagesProj/jni/voip/tgcalls/platform/darwin/iOS/RTCAudioSession+Configuration.mm

@@ -0,0 +1,180 @@
+/*
+ *  Copyright 2016 The WebRTC Project Authors. All rights reserved.
+ *
+ *  Use of this source code is governed by a BSD-style license
+ *  that can be found in the LICENSE file in the root of the source
+ *  tree. An additional intellectual property rights grant can be found
+ *  in the file PATENTS.  All contributing project authors may
+ *  be found in the AUTHORS file in the root of the source tree.
+ */
+
+#import "RTCAudioSession+Private.h"
+#import "RTCAudioSessionConfiguration.h"
+
+#import "base/RTCLogging.h"
+
+@implementation RTC_OBJC_TYPE (RTCAudioSession)
+(Configuration)
+
+    - (BOOL)setConfiguration : (RTC_OBJC_TYPE(RTCAudioSessionConfiguration) *)configuration error
+    : (NSError **)outError disableRecording:(BOOL)disableRecording {
+  return [self setConfiguration:configuration
+                         active:NO
+                shouldSetActive:NO
+                          error:outError
+               disableRecording:disableRecording];
+}
+
+- (BOOL)setConfiguration:(RTC_OBJC_TYPE(RTCAudioSessionConfiguration) *)configuration
+                  active:(BOOL)active
+                   error:(NSError **)outError disableRecording:(BOOL)disableRecording {
+  return [self setConfiguration:configuration
+                         active:active
+                shouldSetActive:YES
+                          error:outError
+               disableRecording:disableRecording];
+}
+
+#pragma mark - Private
+
+- (BOOL)setConfiguration:(RTC_OBJC_TYPE(RTCAudioSessionConfiguration) *)configuration
+                  active:(BOOL)active
+         shouldSetActive:(BOOL)shouldSetActive
+                   error:(NSError **)outError disableRecording:(BOOL)disableRecording {
+  NSParameterAssert(configuration);
+  if (outError) {
+    *outError = nil;
+  }
+
+  // Provide an error even if there isn't one so we can log it. We will not
+  // return immediately on error in this function and instead try to set
+  // everything we can.
+  NSError *error = nil;
+
+  if (!disableRecording) {
+  if (self.category != configuration.category ||
+      self.categoryOptions != configuration.categoryOptions) {
+    NSError *categoryError = nil;
+    if (![self setCategory:configuration.category
+               withOptions:configuration.categoryOptions
+                     error:&categoryError]) {
+      RTCLogError(@"Failed to set category: %@",
+                  categoryError.localizedDescription);
+      error = categoryError;
+    } else {
+      RTCLog(@"Set category to: %@", configuration.category);
+    }
+  }
+
+  if (self.mode != configuration.mode) {
+    NSError *modeError = nil;
+    if (![self setMode:configuration.mode error:&modeError]) {
+      RTCLogError(@"Failed to set mode: %@",
+                  modeError.localizedDescription);
+      error = modeError;
+    } else {
+      RTCLog(@"Set mode to: %@", configuration.mode);
+    }
+  }
+
+  // Sometimes category options don't stick after setting mode.
+  if (self.categoryOptions != configuration.categoryOptions) {
+    NSError *categoryError = nil;
+    if (![self setCategory:configuration.category
+               withOptions:configuration.categoryOptions
+                     error:&categoryError]) {
+      RTCLogError(@"Failed to set category options: %@",
+                  categoryError.localizedDescription);
+      error = categoryError;
+    } else {
+      RTCLog(@"Set category options to: %ld",
+             (long)configuration.categoryOptions);
+    }
+  }
+  }
+
+  if (self.preferredSampleRate != configuration.sampleRate) {
+    NSError *sampleRateError = nil;
+    if (![self setPreferredSampleRate:configuration.sampleRate
+                                error:&sampleRateError]) {
+      RTCLogError(@"Failed to set preferred sample rate: %@",
+                  sampleRateError.localizedDescription);
+      if (!self.ignoresPreferredAttributeConfigurationErrors) {
+        error = sampleRateError;
+      }
+    } else {
+      RTCLog(@"Set preferred sample rate to: %.2f",
+             configuration.sampleRate);
+    }
+  }
+
+  if (self.preferredIOBufferDuration != configuration.ioBufferDuration) {
+    NSError *bufferDurationError = nil;
+    if (![self setPreferredIOBufferDuration:configuration.ioBufferDuration
+                                      error:&bufferDurationError]) {
+      RTCLogError(@"Failed to set preferred IO buffer duration: %@",
+                  bufferDurationError.localizedDescription);
+      if (!self.ignoresPreferredAttributeConfigurationErrors) {
+        error = bufferDurationError;
+      }
+    } else {
+      RTCLog(@"Set preferred IO buffer duration to: %f",
+             configuration.ioBufferDuration);
+    }
+  }
+
+  if (shouldSetActive) {
+    NSError *activeError = nil;
+    if (![self setActive:active error:&activeError]) {
+      RTCLogError(@"Failed to setActive to %d: %@",
+                  active, activeError.localizedDescription);
+      error = activeError;
+    }
+  }
+
+  if (self.isActive &&
+      // TODO(tkchin): Figure out which category/mode numChannels is valid for.
+      [self.mode isEqualToString:AVAudioSessionModeVoiceChat]) {
+    // Try to set the preferred number of hardware audio channels. These calls
+    // must be done after setting the audio session’s category and mode and
+    // activating the session.
+    NSInteger inputNumberOfChannels = configuration.inputNumberOfChannels;
+    if (self.inputNumberOfChannels != inputNumberOfChannels) {
+      NSError *inputChannelsError = nil;
+      if (![self setPreferredInputNumberOfChannels:inputNumberOfChannels
+                                             error:&inputChannelsError]) {
+       RTCLogError(@"Failed to set preferred input number of channels: %@",
+                   inputChannelsError.localizedDescription);
+       if (!self.ignoresPreferredAttributeConfigurationErrors) {
+         error = inputChannelsError;
+       }
+      } else {
+        RTCLog(@"Set input number of channels to: %ld",
+               (long)inputNumberOfChannels);
+      }
+    }
+    NSInteger outputNumberOfChannels = configuration.outputNumberOfChannels;
+    if (self.outputNumberOfChannels != outputNumberOfChannels) {
+      NSError *outputChannelsError = nil;
+      if (![self setPreferredOutputNumberOfChannels:outputNumberOfChannels
+                                              error:&outputChannelsError]) {
+        RTCLogError(@"Failed to set preferred output number of channels: %@",
+                    outputChannelsError.localizedDescription);
+        if (!self.ignoresPreferredAttributeConfigurationErrors) {
+          error = outputChannelsError;
+        }
+      } else {
+        RTCLog(@"Set output number of channels to: %ld",
+               (long)outputNumberOfChannels);
+      }
+    }
+  }
+
+  if (outError) {
+    *outError = error;
+  }
+
+  return error == nil;
+}
+
+@end

TMessagesProj/jni/voip/tgcalls/platform/darwin/iOS/RTCAudioSession+Private.h

@@ -0,0 +1,95 @@
+/*
+ *  Copyright 2016 The WebRTC Project Authors. All rights reserved.
+ *
+ *  Use of this source code is governed by a BSD-style license
+ *  that can be found in the LICENSE file in the root of the source
+ *  tree. An additional intellectual property rights grant can be found
+ *  in the file PATENTS.  All contributing project authors may
+ *  be found in the AUTHORS file in the root of the source tree.
+ */
+
+#import "RTCAudioSession.h"
+
+NS_ASSUME_NONNULL_BEGIN
+
+@class RTC_OBJC_TYPE(RTCAudioSessionConfiguration);
+
+@interface RTC_OBJC_TYPE (RTCAudioSession)
+()
+
+    /** Number of times setActive:YES has succeeded without a balanced call to
+     *  setActive:NO.
+     */
+    @property(nonatomic, readonly) int activationCount;
+
+/** The number of times `beginWebRTCSession` was called without a balanced call
+ *  to `endWebRTCSession`.
+ */
+@property(nonatomic, readonly) int webRTCSessionCount;
+
+/** Convenience BOOL that checks useManualAudio and isAudioEnebled. */
+@property(readonly) BOOL canPlayOrRecord;
+
+/** Tracks whether we have been sent an interruption event that hasn't been matched by either an
+ *  interrupted end event or a foreground event.
+ */
+@property(nonatomic, assign) BOOL isInterrupted;
+
+/** Adds the delegate to the list of delegates, and places it at the front of
+ *  the list. This delegate will be notified before other delegates of
+ *  audio events.
+ */
+- (void)pushDelegate:(id<RTC_OBJC_TYPE(RTCAudioSessionDelegate)>)delegate;
+
+/** Signals RTCAudioSession that a WebRTC session is about to begin and
+ *  audio configuration is needed. Will configure the audio session for WebRTC
+ *  if not already configured and if configuration is not delayed.
+ *  Successful calls must be balanced by a call to endWebRTCSession.
+ */
+- (BOOL)beginWebRTCSession:(NSError **)outError;
+
+/** Signals RTCAudioSession that a WebRTC session is about to end and audio
+ *  unconfiguration is needed. Will unconfigure the audio session for WebRTC
+ *  if this is the last unmatched call and if configuration is not delayed.
+ */
+- (BOOL)endWebRTCSession:(NSError **)outError;
+
+/** Configure the audio session for WebRTC. This call will fail if the session
+ *  is already configured. On other failures, we will attempt to restore the
+ *  previously used audio session configuration.
+ *  `lockForConfiguration` must be called first.
+ *  Successful calls to configureWebRTCSession must be matched by calls to
+ *  `unconfigureWebRTCSession`.
+ */
+- (BOOL)configureWebRTCSession:(NSError **)outError disableRecording:(BOOL)disableRecording;
+
+/** Unconfigures the session for WebRTC. This will attempt to restore the
+ *  audio session to the settings used before `configureWebRTCSession` was
+ *  called.
+ *  `lockForConfiguration` must be called first.
+ */
+- (BOOL)unconfigureWebRTCSession:(NSError **)outError;
+
+/** Returns a configuration error with the given description. */
+- (NSError *)configurationErrorWithDescription:(NSString *)description;
+
+/** Notifies the receiver that a playout glitch was detected. */
+- (void)notifyDidDetectPlayoutGlitch:(int64_t)totalNumberOfGlitches;
+
+/** Notifies the receiver that there was an error when starting an audio unit. */
+- (void)notifyAudioUnitStartFailedWithError:(OSStatus)error;
+
+// Properties and methods for tests.
+- (void)notifyDidBeginInterruption;
+- (void)notifyDidEndInterruptionWithShouldResumeSession:(BOOL)shouldResumeSession;
+- (void)notifyDidChangeRouteWithReason:(AVAudioSessionRouteChangeReason)reason
+                         previousRoute:(AVAudioSessionRouteDescription *)previousRoute;
+- (void)notifyMediaServicesWereLost;
+- (void)notifyMediaServicesWereReset;
+- (void)notifyDidChangeCanPlayOrRecord:(BOOL)canPlayOrRecord;
+- (void)notifyDidStartPlayOrRecord;
+- (void)notifyDidStopPlayOrRecord;
+
+@end
+
+NS_ASSUME_NONNULL_END

TMessagesProj/jni/voip/tgcalls/platform/darwin/iOS/RTCAudioSession.h

@@ -0,0 +1,266 @@
+/*
+ *  Copyright 2016 The WebRTC Project Authors. All rights reserved.
+ *
+ *  Use of this source code is governed by a BSD-style license
+ *  that can be found in the LICENSE file in the root of the source
+ *  tree. An additional intellectual property rights grant can be found
+ *  in the file PATENTS.  All contributing project authors may
+ *  be found in the AUTHORS file in the root of the source tree.
+ */
+
+#import <AVFoundation/AVFoundation.h>
+#import <Foundation/Foundation.h>
+
+#import "RTCMacros.h"
+
+NS_ASSUME_NONNULL_BEGIN
+
+extern NSString *const kRTCAudioSessionErrorDomain;
+/** Method that requires lock was called without lock. */
+extern NSInteger const kRTCAudioSessionErrorLockRequired;
+/** Unknown configuration error occurred. */
+extern NSInteger const kRTCAudioSessionErrorConfiguration;
+
+@class RTC_OBJC_TYPE(RTCAudioSession);
+@class RTC_OBJC_TYPE(RTCAudioSessionConfiguration);
+
+// Surfaces AVAudioSession events. WebRTC will listen directly for notifications
+// from AVAudioSession and handle them before calling these delegate methods,
+// at which point applications can perform additional processing if required.
+RTC_OBJC_EXPORT
+@protocol RTC_OBJC_TYPE
+(RTCAudioSessionDelegate)<NSObject>
+
+    @optional
+/** Called on a system notification thread when AVAudioSession starts an
+ *  interruption event.
+ */
+- (void)audioSessionDidBeginInterruption:(RTC_OBJC_TYPE(RTCAudioSession) *)session;
+
+/** Called on a system notification thread when AVAudioSession ends an
+ *  interruption event.
+ */
+- (void)audioSessionDidEndInterruption:(RTC_OBJC_TYPE(RTCAudioSession) *)session
+                   shouldResumeSession:(BOOL)shouldResumeSession;
+
+/** Called on a system notification thread when AVAudioSession changes the
+ *  route.
+ */
+- (void)audioSessionDidChangeRoute:(RTC_OBJC_TYPE(RTCAudioSession) *)session
+                            reason:(AVAudioSessionRouteChangeReason)reason
+                     previousRoute:(AVAudioSessionRouteDescription *)previousRoute;
+
+/** Called on a system notification thread when AVAudioSession media server
+ *  terminates.
+ */
+- (void)audioSessionMediaServerTerminated:(RTC_OBJC_TYPE(RTCAudioSession) *)session;
+
+/** Called on a system notification thread when AVAudioSession media server
+ *  restarts.
+ */
+- (void)audioSessionMediaServerReset:(RTC_OBJC_TYPE(RTCAudioSession) *)session;
+
+// TODO(tkchin): Maybe handle SilenceSecondaryAudioHintNotification.
+
+- (void)audioSession:(RTC_OBJC_TYPE(RTCAudioSession) *)session
+    didChangeCanPlayOrRecord:(BOOL)canPlayOrRecord;
+
+/** Called on a WebRTC thread when the audio device is notified to begin
+ *  playback or recording.
+ */
+- (void)audioSessionDidStartPlayOrRecord:(RTC_OBJC_TYPE(RTCAudioSession) *)session;
+
+/** Called on a WebRTC thread when the audio device is notified to stop
+ *  playback or recording.
+ */
+- (void)audioSessionDidStopPlayOrRecord:(RTC_OBJC_TYPE(RTCAudioSession) *)session;
+
+/** Called when the AVAudioSession output volume value changes. */
+- (void)audioSession:(RTC_OBJC_TYPE(RTCAudioSession) *)audioSession
+    didChangeOutputVolume:(float)outputVolume;
+
+/** Called when the audio device detects a playout glitch. The argument is the
+ *  number of glitches detected so far in the current audio playout session.
+ */
+- (void)audioSession:(RTC_OBJC_TYPE(RTCAudioSession) *)audioSession
+    didDetectPlayoutGlitch:(int64_t)totalNumberOfGlitches;
+
+/** Called when the audio session is about to change the active state.
+ */
+- (void)audioSession:(RTC_OBJC_TYPE(RTCAudioSession) *)audioSession willSetActive:(BOOL)active;
+
+/** Called after the audio session sucessfully changed the active state.
+ */
+- (void)audioSession:(RTC_OBJC_TYPE(RTCAudioSession) *)audioSession didSetActive:(BOOL)active;
+
+/** Called after the audio session failed to change the active state.
+ */
+- (void)audioSession:(RTC_OBJC_TYPE(RTCAudioSession) *)audioSession
+    failedToSetActive:(BOOL)active
+                error:(NSError *)error;
+
+- (void)audioSession:(RTC_OBJC_TYPE(RTCAudioSession) *)audioSession
+    audioUnitStartFailedWithError:(NSError *)error;
+
+@end
+
+/** This is a protocol used to inform RTCAudioSession when the audio session
+ *  activation state has changed outside of RTCAudioSession. The current known use
+ *  case of this is when CallKit activates the audio session for the application
+ */
+RTC_OBJC_EXPORT
+@protocol RTC_OBJC_TYPE
+(RTCAudioSessionActivationDelegate)<NSObject>
+
+    /** Called when the audio session is activated outside of the app by iOS. */
+    - (void)audioSessionDidActivate : (AVAudioSession *)session;
+
+/** Called when the audio session is deactivated outside of the app by iOS. */
+- (void)audioSessionDidDeactivate:(AVAudioSession *)session;
+
+@end
+
+/** Proxy class for AVAudioSession that adds a locking mechanism similar to
+ *  AVCaptureDevice. This is used to that interleaving configurations between
+ *  WebRTC and the application layer are avoided.
+ *
+ *  RTCAudioSession also coordinates activation so that the audio session is
+ *  activated only once. See `setActive:error:`.
+ */
+RTC_OBJC_EXPORT
+@interface RTC_OBJC_TYPE (RTCAudioSession) : NSObject <RTC_OBJC_TYPE(RTCAudioSessionActivationDelegate)>
+
+/** Convenience property to access the AVAudioSession singleton. Callers should
+ *  not call setters on AVAudioSession directly, but other method invocations
+ *  are fine.
+ */
+@property(nonatomic, readonly) AVAudioSession *session;
+
+/** Our best guess at whether the session is active based on results of calls to
+ *  AVAudioSession.
+ */
+@property(nonatomic, readonly) BOOL isActive;
+
+/** If YES, WebRTC will not initialize the audio unit automatically when an
+ *  audio track is ready for playout or recording. Instead, applications should
+ *  call setIsAudioEnabled. If NO, WebRTC will initialize the audio unit
+ *  as soon as an audio track is ready for playout or recording.
+ */
+@property(nonatomic, assign) BOOL useManualAudio;
+
+/** This property is only effective if useManualAudio is YES.
+ *  Represents permission for WebRTC to initialize the VoIP audio unit.
+ *  When set to NO, if the VoIP audio unit used by WebRTC is active, it will be
+ *  stopped and uninitialized. This will stop incoming and outgoing audio.
+ *  When set to YES, WebRTC will initialize and start the audio unit when it is
+ *  needed (e.g. due to establishing an audio connection).
+ *  This property was introduced to work around an issue where if an AVPlayer is
+ *  playing audio while the VoIP audio unit is initialized, its audio would be
+ *  either cut off completely or played at a reduced volume. By preventing
+ *  the audio unit from being initialized until after the audio has completed,
+ *  we are able to prevent the abrupt cutoff.
+ */
+@property(nonatomic, assign) BOOL isAudioEnabled;
+
+// Proxy properties.
+@property(readonly) NSString *category;
+@property(readonly) AVAudioSessionCategoryOptions categoryOptions;
+@property(readonly) NSString *mode;
+@property(readonly) BOOL secondaryAudioShouldBeSilencedHint;
+@property(readonly) AVAudioSessionRouteDescription *currentRoute;
+@property(readonly) NSInteger maximumInputNumberOfChannels;
+@property(readonly) NSInteger maximumOutputNumberOfChannels;
+@property(readonly) float inputGain;
+@property(readonly) BOOL inputGainSettable;
+@property(readonly) BOOL inputAvailable;
+@property(readonly, nullable) NSArray<AVAudioSessionDataSourceDescription *> *inputDataSources;
+@property(readonly, nullable) AVAudioSessionDataSourceDescription *inputDataSource;
+@property(readonly, nullable) NSArray<AVAudioSessionDataSourceDescription *> *outputDataSources;
+@property(readonly, nullable) AVAudioSessionDataSourceDescription *outputDataSource;
+@property(readonly) double sampleRate;
+@property(readonly) double preferredSampleRate;
+@property(readonly) NSInteger inputNumberOfChannels;
+@property(readonly) NSInteger outputNumberOfChannels;
+@property(readonly) float outputVolume;
+@property(readonly) NSTimeInterval inputLatency;
+@property(readonly) NSTimeInterval outputLatency;
+@property(readonly) NSTimeInterval IOBufferDuration;
+@property(readonly) NSTimeInterval preferredIOBufferDuration;
+
+/**
+ When YES, calls to -setConfiguration:error: and -setConfiguration:active:error: ignore errors in
+ configuring the audio session's "preferred" attributes (e.g. preferredInputNumberOfChannels).
+ Typically, configurations to preferred attributes are optimizations, and ignoring this type of
+ configuration error allows code flow to continue along the happy path when these optimization are
+ not available. The default value of this property is NO.
+ */
+@property(nonatomic) BOOL ignoresPreferredAttributeConfigurationErrors;
+
+/** Default constructor. */
++ (instancetype)sharedInstance;
+- (instancetype)init NS_UNAVAILABLE;
+
+/** Adds a delegate, which is held weakly. */
+- (void)addDelegate:(id<RTC_OBJC_TYPE(RTCAudioSessionDelegate)>)delegate;
+/** Removes an added delegate. */
+- (void)removeDelegate:(id<RTC_OBJC_TYPE(RTCAudioSessionDelegate)>)delegate;
+
+/** Request exclusive access to the audio session for configuration. This call
+ *  will block if the lock is held by another object.
+ */
+- (void)lockForConfiguration;
+/** Relinquishes exclusive access to the audio session. */
+- (void)unlockForConfiguration;
+
+/** If `active`, activates the audio session if it isn't already active.
+ *  Successful calls must be balanced with a setActive:NO when activation is no
+ *  longer required. If not `active`, deactivates the audio session if one is
+ *  active and this is the last balanced call. When deactivating, the
+ *  AVAudioSessionSetActiveOptionNotifyOthersOnDeactivation option is passed to
+ *  AVAudioSession.
+ */
+- (BOOL)setActive:(BOOL)active error:(NSError **)outError;
+
+// The following methods are proxies for the associated methods on
+// AVAudioSession. `lockForConfiguration` must be called before using them
+// otherwise they will fail with kRTCAudioSessionErrorLockRequired.
+
+- (BOOL)setCategory:(NSString *)category
+        withOptions:(AVAudioSessionCategoryOptions)options
+              error:(NSError **)outError;
+- (BOOL)setMode:(NSString *)mode error:(NSError **)outError;
+- (BOOL)setInputGain:(float)gain error:(NSError **)outError;
+- (BOOL)setPreferredSampleRate:(double)sampleRate error:(NSError **)outError;
+- (BOOL)setPreferredIOBufferDuration:(NSTimeInterval)duration error:(NSError **)outError;
+- (BOOL)setPreferredInputNumberOfChannels:(NSInteger)count error:(NSError **)outError;
+- (BOOL)setPreferredOutputNumberOfChannels:(NSInteger)count error:(NSError **)outError;
+- (BOOL)overrideOutputAudioPort:(AVAudioSessionPortOverride)portOverride error:(NSError **)outError;
+- (BOOL)setPreferredInput:(AVAudioSessionPortDescription *)inPort error:(NSError **)outError;
+- (BOOL)setInputDataSource:(AVAudioSessionDataSourceDescription *)dataSource
+                     error:(NSError **)outError;
+- (BOOL)setOutputDataSource:(AVAudioSessionDataSourceDescription *)dataSource
+                      error:(NSError **)outError;
+@end
+
+@interface RTC_OBJC_TYPE (RTCAudioSession)
+(Configuration)
+
+    /** Applies the configuration to the current session. Attempts to set all
+     *  properties even if previous ones fail. Only the last error will be
+     *  returned.
+     *  `lockForConfiguration` must be called first.
+     */
+    - (BOOL)setConfiguration : (RTC_OBJC_TYPE(RTCAudioSessionConfiguration) *)configuration error
+    : (NSError **)outError;
+
+/** Convenience method that calls both setConfiguration and setActive.
+ *  `lockForConfiguration` must be called first.
+ */
+- (BOOL)setConfiguration:(RTC_OBJC_TYPE(RTCAudioSessionConfiguration) *)configuration
+                  active:(BOOL)active
+                   error:(NSError **)outError
+        disableRecording:(BOOL)disableRecording;
+
+@end
+
+NS_ASSUME_NONNULL_END

TMessagesProj/jni/voip/tgcalls/platform/darwin/iOS/RTCAudioSessionConfiguration.h

@@ -0,0 +1,47 @@
+/*
+ *  Copyright 2016 The WebRTC Project Authors. All rights reserved.
+ *
+ *  Use of this source code is governed by a BSD-style license
+ *  that can be found in the LICENSE file in the root of the source
+ *  tree. An additional intellectual property rights grant can be found
+ *  in the file PATENTS.  All contributing project authors may
+ *  be found in the AUTHORS file in the root of the source tree.
+ */
+
+#import <AVFoundation/AVFoundation.h>
+#import <Foundation/Foundation.h>
+
+#import "RTCMacros.h"
+
+NS_ASSUME_NONNULL_BEGIN
+
+RTC_EXTERN const double kRTCAudioSessionHighPerformanceSampleRate;
+RTC_EXTERN const double kRTCAudioSessionLowComplexitySampleRate;
+RTC_EXTERN const double kRTCAudioSessionHighPerformanceIOBufferDuration;
+RTC_EXTERN const double kRTCAudioSessionLowComplexityIOBufferDuration;
+
+// Struct to hold configuration values.
+RTC_OBJC_EXPORT
+@interface RTC_OBJC_TYPE (RTCAudioSessionConfiguration) : NSObject
+
+@property(nonatomic, strong) NSString *category;
+@property(nonatomic, assign) AVAudioSessionCategoryOptions categoryOptions;
+@property(nonatomic, strong) NSString *mode;
+@property(nonatomic, assign) double sampleRate;
+@property(nonatomic, assign) NSTimeInterval ioBufferDuration;
+@property(nonatomic, assign) NSInteger inputNumberOfChannels;
+@property(nonatomic, assign) NSInteger outputNumberOfChannels;
+
+/** Initializes configuration to defaults. */
+- (instancetype)init NS_DESIGNATED_INITIALIZER;
+
+/** Returns the current configuration of the audio session. */
++ (instancetype)currentConfiguration;
+/** Returns the configuration that WebRTC needs. */
++ (instancetype)webRTCConfiguration;
+/** Provide a way to override the default configuration. */
++ (void)setWebRTCConfiguration:(RTC_OBJC_TYPE(RTCAudioSessionConfiguration) *)configuration;
+
+@end
+
+NS_ASSUME_NONNULL_END

TMessagesProj/jni/voip/tgcalls/platform/darwin/iOS/RTCAudioSessionConfiguration.mm

@@ -0,0 +1,133 @@
+/*
+ *  Copyright 2016 The WebRTC Project Authors. All rights reserved.
+ *
+ *  Use of this source code is governed by a BSD-style license
+ *  that can be found in the LICENSE file in the root of the source
+ *  tree. An additional intellectual property rights grant can be found
+ *  in the file PATENTS.  All contributing project authors may
+ *  be found in the AUTHORS file in the root of the source tree.
+ */
+
+#import "RTCAudioSessionConfiguration.h"
+#import "RTCAudioSession.h"
+
+#import "helpers/RTCDispatcher.h"
+#import "helpers/UIDevice+RTCDevice.h"
+
+// Try to use mono to save resources. Also avoids channel format conversion
+// in the I/O audio unit. Initial tests have shown that it is possible to use
+// mono natively for built-in microphones and for BT headsets but not for
+// wired headsets. Wired headsets only support stereo as native channel format
+// but it is a low cost operation to do a format conversion to mono in the
+// audio unit. Hence, we will not hit a RTC_CHECK in
+// VerifyAudioParametersForActiveAudioSession() for a mismatch between the
+// preferred number of channels and the actual number of channels.
+const int kRTCAudioSessionPreferredNumberOfChannels = 1;
+
+// Preferred hardware sample rate (unit is in Hertz). The client sample rate
+// will be set to this value as well to avoid resampling the the audio unit's
+// format converter. Note that, some devices, e.g. BT headsets, only supports
+// 8000Hz as native sample rate.
+const double kRTCAudioSessionHighPerformanceSampleRate = 48000.0;
+
+// A lower sample rate will be used for devices with only one core
+// (e.g. iPhone 4). The goal is to reduce the CPU load of the application.
+const double kRTCAudioSessionLowComplexitySampleRate = 16000.0;
+
+// Use a hardware I/O buffer size (unit is in seconds) that matches the 10ms
+// size used by WebRTC. The exact actual size will differ between devices.
+// Example: using 48kHz on iPhone 6 results in a native buffer size of
+// ~10.6667ms or 512 audio frames per buffer. The FineAudioBuffer instance will
+// take care of any buffering required to convert between native buffers and
+// buffers used by WebRTC. It is beneficial for the performance if the native
+// size is as an even multiple of 10ms as possible since it results in "clean"
+// callback sequence without bursts of callbacks back to back.
+const double kRTCAudioSessionHighPerformanceIOBufferDuration = 0.02;
+
+// Use a larger buffer size on devices with only one core (e.g. iPhone 4).
+// It will result in a lower CPU consumption at the cost of a larger latency.
+// The size of 60ms is based on instrumentation that shows a significant
+// reduction in CPU load compared with 10ms on low-end devices.
+// TODO(henrika): monitor this size and determine if it should be modified.
+const double kRTCAudioSessionLowComplexityIOBufferDuration = 0.06;
+
+static RTC_OBJC_TYPE(RTCAudioSessionConfiguration) *gWebRTCConfiguration = nil;
+
+@implementation RTC_OBJC_TYPE (RTCAudioSessionConfiguration)
+
+@synthesize category = _category;
+@synthesize categoryOptions = _categoryOptions;
+@synthesize mode = _mode;
+@synthesize sampleRate = _sampleRate;
+@synthesize ioBufferDuration = _ioBufferDuration;
+@synthesize inputNumberOfChannels = _inputNumberOfChannels;
+@synthesize outputNumberOfChannels = _outputNumberOfChannels;
+
+- (instancetype)init {
+  if (self = [super init]) {
+    // Use a category which supports simultaneous recording and playback.
+    // By default, using this category implies that our app’s audio is
+    // nonmixable, hence activating the session will interrupt any other
+    // audio sessions which are also nonmixable.
+    _category = AVAudioSessionCategoryPlayAndRecord;
+    _categoryOptions = AVAudioSessionCategoryOptionAllowBluetooth;
+
+    // Specify mode for two-way voice communication (e.g. VoIP).
+    _mode = AVAudioSessionModeVoiceChat;
+
+    // Set the session's sample rate or the hardware sample rate.
+    // It is essential that we use the same sample rate as stream format
+    // to ensure that the I/O unit does not have to do sample rate conversion.
+    // Set the preferred audio I/O buffer duration, in seconds.
+    NSUInteger processorCount = [NSProcessInfo processInfo].processorCount;
+    // Use best sample rate and buffer duration if the CPU has more than one
+    // core.
+    if (processorCount > 1) {
+      _sampleRate = kRTCAudioSessionHighPerformanceSampleRate;
+      _ioBufferDuration = kRTCAudioSessionHighPerformanceIOBufferDuration;
+    } else {
+      _sampleRate = kRTCAudioSessionLowComplexitySampleRate;
+      _ioBufferDuration = kRTCAudioSessionLowComplexityIOBufferDuration;
+    }
+
+    // We try to use mono in both directions to save resources and format
+    // conversions in the audio unit. Some devices does only support stereo;
+    // e.g. wired headset on iPhone 6.
+    // TODO(henrika): add support for stereo if needed.
+    _inputNumberOfChannels = kRTCAudioSessionPreferredNumberOfChannels;
+    _outputNumberOfChannels = kRTCAudioSessionPreferredNumberOfChannels;
+  }
+  return self;
+}
+
++ (void)initialize {
+  gWebRTCConfiguration = [[self alloc] init];
+}
+
++ (instancetype)currentConfiguration {
+  RTC_OBJC_TYPE(RTCAudioSession) *session = [RTC_OBJC_TYPE(RTCAudioSession) sharedInstance];
+  RTC_OBJC_TYPE(RTCAudioSessionConfiguration) *config =
+      [[RTC_OBJC_TYPE(RTCAudioSessionConfiguration) alloc] init];
+  config.category = session.category;
+  config.categoryOptions = session.categoryOptions;
+  config.mode = session.mode;
+  config.sampleRate = session.sampleRate;
+  config.ioBufferDuration = session.IOBufferDuration;
+  config.inputNumberOfChannels = session.inputNumberOfChannels;
+  config.outputNumberOfChannels = session.outputNumberOfChannels;
+  return config;
+}
+
++ (instancetype)webRTCConfiguration {
+  @synchronized(self) {
+    return (RTC_OBJC_TYPE(RTCAudioSessionConfiguration) *)gWebRTCConfiguration;
+  }
+}
+
++ (void)setWebRTCConfiguration:(RTC_OBJC_TYPE(RTCAudioSessionConfiguration) *)configuration {
+  @synchronized(self) {
+    gWebRTCConfiguration = configuration;
+  }
+}
+
+@end

TMessagesProj/jni/voip/tgcalls/platform/darwin/iOS/RTCNativeAudioSessionDelegateAdapter.h

@@ -0,0 +1,33 @@
+/*
+ *  Copyright 2018 The WebRTC Project Authors. All rights reserved.
+ *
+ *  Use of this source code is governed by a BSD-style license
+ *  that can be found in the LICENSE file in the root of the source
+ *  tree. An additional intellectual property rights grant can be found
+ *  in the file PATENTS.  All contributing project authors may
+ *  be found in the AUTHORS file in the root of the source tree.
+ */
+
+#import "RTCAudioSession.h"
+
+NS_ASSUME_NONNULL_BEGIN
+
+namespace webrtc {
+class AudioSessionObserver;
+}
+
+/** Adapter that forwards RTCAudioSessionDelegate calls to the appropriate
+ *  methods on the AudioSessionObserver.
+ */
+@interface RTCNativeAudioSessionDelegateAdapter : NSObject <RTC_OBJC_TYPE (RTCAudioSessionDelegate)>
+
+- (instancetype)init NS_UNAVAILABLE;
+
+/** `observer` is a raw pointer and should be kept alive
+ *  for this object's lifetime.
+ */
+- (instancetype)initWithObserver:(webrtc::AudioSessionObserver *)observer NS_DESIGNATED_INITIALIZER;
+
+@end
+
+NS_ASSUME_NONNULL_END

TMessagesProj/jni/voip/tgcalls/platform/darwin/iOS/RTCNativeAudioSessionDelegateAdapter.mm

@@ -0,0 +1,89 @@
+/*
+ *  Copyright 2018 The WebRTC Project Authors. All rights reserved.
+ *
+ *  Use of this source code is governed by a BSD-style license
+ *  that can be found in the LICENSE file in the root of the source
+ *  tree. An additional intellectual property rights grant can be found
+ *  in the file PATENTS.  All contributing project authors may
+ *  be found in the AUTHORS file in the root of the source tree.
+ */
+
+#import "RTCNativeAudioSessionDelegateAdapter.h"
+
+#include "sdk/objc/native/src/audio/audio_session_observer.h"
+
+#import "base/RTCLogging.h"
+
+@implementation RTCNativeAudioSessionDelegateAdapter {
+  webrtc::AudioSessionObserver *_observer;
+}
+
+- (instancetype)initWithObserver:(webrtc::AudioSessionObserver *)observer {
+  RTC_DCHECK(observer);
+  if (self = [super init]) {
+    _observer = observer;
+  }
+  return self;
+}
+
+#pragma mark - RTC_OBJC_TYPE(RTCAudioSessionDelegate)
+
+- (void)audioSessionDidBeginInterruption:(RTC_OBJC_TYPE(RTCAudioSession) *)session {
+  _observer->OnInterruptionBegin();
+}
+
+- (void)audioSessionDidEndInterruption:(RTC_OBJC_TYPE(RTCAudioSession) *)session
+                   shouldResumeSession:(BOOL)shouldResumeSession {
+  _observer->OnInterruptionEnd();
+}
+
+- (void)audioSessionDidChangeRoute:(RTC_OBJC_TYPE(RTCAudioSession) *)session
+                            reason:(AVAudioSessionRouteChangeReason)reason
+                     previousRoute:(AVAudioSessionRouteDescription *)previousRoute {
+  switch (reason) {
+    case AVAudioSessionRouteChangeReasonUnknown:
+    case AVAudioSessionRouteChangeReasonNewDeviceAvailable:
+    case AVAudioSessionRouteChangeReasonOldDeviceUnavailable:
+    case AVAudioSessionRouteChangeReasonCategoryChange:
+      // It turns out that we see a category change (at least in iOS 9.2)
+      // when making a switch from a BT device to e.g. Speaker using the
+      // iOS Control Center and that we therefore must check if the sample
+      // rate has changed. And if so is the case, restart the audio unit.
+    case AVAudioSessionRouteChangeReasonOverride:
+    case AVAudioSessionRouteChangeReasonWakeFromSleep:
+    case AVAudioSessionRouteChangeReasonNoSuitableRouteForCategory:
+      _observer->OnValidRouteChange();
+      break;
+    case AVAudioSessionRouteChangeReasonRouteConfigurationChange:
+      // The set of input and output ports has not changed, but their
+      // configuration has, e.g., a port’s selected data source has
+      // changed. Ignore this type of route change since we are focusing
+      // on detecting headset changes.
+      RTCLog(@"Ignoring RouteConfigurationChange");
+      break;
+  }
+}
+
+- (void)audioSessionMediaServerTerminated:(RTC_OBJC_TYPE(RTCAudioSession) *)session {
+}
+
+- (void)audioSessionMediaServerReset:(RTC_OBJC_TYPE(RTCAudioSession) *)session {
+}
+
+- (void)audioSession:(RTC_OBJC_TYPE(RTCAudioSession) *)session
+    didChangeCanPlayOrRecord:(BOOL)canPlayOrRecord {
+  _observer->OnCanPlayOrRecordChange(canPlayOrRecord);
+}
+
+- (void)audioSessionDidStartPlayOrRecord:(RTC_OBJC_TYPE(RTCAudioSession) *)session {
+}
+
+- (void)audioSessionDidStopPlayOrRecord:(RTC_OBJC_TYPE(RTCAudioSession) *)session {
+}
+
+- (void)audioSession:(RTC_OBJC_TYPE(RTCAudioSession) *)audioSession
+    didChangeOutputVolume:(float)outputVolume {
+  _observer->OnChangedOutputVolume();
+}
+
+@end

TMessagesProj/jni/voip/tgcalls/platform/darwin/iOS/tgcalls_audio_device_ios.h

@@ -0,0 +1,329 @@
+/*
+ *  Copyright (c) 2012 The WebRTC project authors. All Rights Reserved.
+ *
+ *  Use of this source code is governed by a BSD-style license
+ *  that can be found in the LICENSE file in the root of the source
+ *  tree. An additional intellectual property rights grant can be found
+ *  in the file PATENTS.  All contributing project authors may
+ *  be found in the AUTHORS file in the root of the source tree.
+ */
+
+#ifndef TGCALLS_AUDIO_AUDIO_DEVICE_IOS_H_
+#define TGCALLS_AUDIO_AUDIO_DEVICE_IOS_H_
+
+#include <atomic>
+#include <memory>
+
+#include "api/scoped_refptr.h"
+#include "api/sequence_checker.h"
+#include "sdk/objc/native/src/audio/audio_session_observer.h"
+#include "api/task_queue/pending_task_safety_flag.h"
+#include "modules/audio_device/audio_device_generic.h"
+#include "rtc_base/buffer.h"
+#include "rtc_base/thread.h"
+#include "rtc_base/thread_annotations.h"
+#include "sdk/objc/base/RTCMacros.h"
+#include "tgcalls_voice_processing_audio_unit.h"
+
+#include "platform/darwin/iOS/CallAudioTone.h"
+
+RTC_FWD_DECL_OBJC_CLASS(RTCNativeAudioSessionDelegateAdapter);
+
+namespace webrtc {
+
+class FineAudioBuffer;
+
+namespace tgcalls_ios_adm {
+
+// Implements full duplex 16-bit mono PCM audio support for iOS using a
+// Voice-Processing (VP) I/O audio unit in Core Audio. The VP I/O audio unit
+// supports audio echo cancellation. It also adds automatic gain control,
+// adjustment of voice-processing quality and muting.
+//
+// An instance must be created and destroyed on one and the same thread.
+// All supported public methods must also be called on the same thread.
+// A thread checker will RTC_DCHECK if any supported method is called on an
+// invalid thread.
+//
+// Recorded audio will be delivered on a real-time internal I/O thread in the
+// audio unit. The audio unit will also ask for audio data to play out on this
+// same thread.
+class AudioDeviceIOS : public AudioDeviceGeneric,
+                       public AudioSessionObserver,
+                       public VoiceProcessingAudioUnitObserver {
+ public:
+  explicit AudioDeviceIOS(bool bypass_voice_processing, bool disable_recording, bool enableSystemMute, int numChannels);
+  ~AudioDeviceIOS() override;
+
+  void AttachAudioBuffer(AudioDeviceBuffer* audioBuffer) override;
+
+  InitStatus Init() override;
+  int32_t Terminate() override;
+  bool Initialized() const override;
+
+  int32_t InitPlayout() override;
+  bool PlayoutIsInitialized() const override;
+
+  int32_t InitRecording() override;
+  bool RecordingIsInitialized() const override;
+
+  int32_t StartPlayout() override;
+  int32_t StopPlayout() override;
+  bool Playing() const override;
+
+  int32_t StartRecording() override;
+  int32_t StopRecording() override;
+  bool Recording() const override;
+                           
+  void setIsBufferPlaying(bool isBufferPlaying);
+  void setIsBufferRecording(bool isBufferRecording);
+
+  // These methods returns hard-coded delay values and not dynamic delay
+  // estimates. The reason is that iOS supports a built-in AEC and the WebRTC
+  // AEC will always be disabled in the Libjingle layer to avoid running two
+  // AEC implementations at the same time. And, it saves resources to avoid
+  // updating these delay values continuously.
+  // TODO(henrika): it would be possible to mark these two methods as not
+  // implemented since they are only called for A/V-sync purposes today and
+  // A/V-sync is not supported on iOS. However, we avoid adding error messages
+  // the log by using these dummy implementations instead.
+  int32_t PlayoutDelay(uint16_t& delayMS) const override;
+
+  // No implementation for playout underrun on iOS. We override it to avoid a
+  // periodic log that it isn't available from the base class.
+  int32_t GetPlayoutUnderrunCount() const override { return -1; }
+
+  // Native audio parameters stored during construction.
+  // These methods are unique for the iOS implementation.
+  int GetPlayoutAudioParameters(AudioParameters* params) const override;
+  int GetRecordAudioParameters(AudioParameters* params) const override;
+
+  // These methods are currently not fully implemented on iOS:
+
+  // See audio_device_not_implemented.cc for trivial implementations.
+  int32_t ActiveAudioLayer(
+      AudioDeviceModule::AudioLayer& audioLayer) const override;
+  int32_t PlayoutIsAvailable(bool& available) override;
+  int32_t RecordingIsAvailable(bool& available) override;
+  int16_t PlayoutDevices() override;
+  int16_t RecordingDevices() override;
+  int32_t PlayoutDeviceName(uint16_t index,
+                            char name[kAdmMaxDeviceNameSize],
+                            char guid[kAdmMaxGuidSize]) override;
+  int32_t RecordingDeviceName(uint16_t index,
+                              char name[kAdmMaxDeviceNameSize],
+                              char guid[kAdmMaxGuidSize]) override;
+  int32_t SetPlayoutDevice(uint16_t index) override;
+  int32_t SetPlayoutDevice(
+      AudioDeviceModule::WindowsDeviceType device) override;
+  int32_t SetRecordingDevice(uint16_t index) override;
+  int32_t SetRecordingDevice(
+      AudioDeviceModule::WindowsDeviceType device) override;
+  int32_t InitSpeaker() override;
+  bool SpeakerIsInitialized() const override;
+  int32_t InitMicrophone() override;
+  bool MicrophoneIsInitialized() const override;
+  int32_t SpeakerVolumeIsAvailable(bool& available) override;
+  int32_t SetSpeakerVolume(uint32_t volume) override;
+  int32_t SpeakerVolume(uint32_t& volume) const override;
+  int32_t MaxSpeakerVolume(uint32_t& maxVolume) const override;
+  int32_t MinSpeakerVolume(uint32_t& minVolume) const override;
+  int32_t MicrophoneVolumeIsAvailable(bool& available) override;
+  int32_t SetMicrophoneVolume(uint32_t volume) override;
+  int32_t MicrophoneVolume(uint32_t& volume) const override;
+  int32_t MaxMicrophoneVolume(uint32_t& maxVolume) const override;
+  int32_t MinMicrophoneVolume(uint32_t& minVolume) const override;
+  int32_t MicrophoneMuteIsAvailable(bool& available) override;
+  int32_t SetMicrophoneMute(bool enable) override;
+  int32_t MicrophoneMute(bool& enabled) const override;
+  int32_t SpeakerMuteIsAvailable(bool& available) override;
+  int32_t SetSpeakerMute(bool enable) override;
+  int32_t SpeakerMute(bool& enabled) const override;
+  int32_t StereoPlayoutIsAvailable(bool& available) override;
+  int32_t SetStereoPlayout(bool enable) override;
+  int32_t StereoPlayout(bool& enabled) const override;
+  int32_t StereoRecordingIsAvailable(bool& available) override;
+  int32_t SetStereoRecording(bool enable) override;
+  int32_t StereoRecording(bool& enabled) const override;
+
+  // AudioSessionObserver methods. May be called from any thread.
+  void OnInterruptionBegin() override;
+  void OnInterruptionEnd() override;
+  void OnValidRouteChange() override;
+  void OnCanPlayOrRecordChange(bool can_play_or_record) override;
+  void OnChangedOutputVolume() override;
+                           
+  void setTone(std::shared_ptr<tgcalls::CallAudioTone> tone);
+
+  // VoiceProcessingAudioUnitObserver methods.
+  OSStatus OnDeliverRecordedData(AudioUnitRenderActionFlags* flags,
+                                 const AudioTimeStamp* time_stamp,
+                                 UInt32 bus_number,
+                                 UInt32 num_frames,
+                                 AudioBufferList* io_data) override;
+  OSStatus OnGetPlayoutData(AudioUnitRenderActionFlags* flags,
+                            const AudioTimeStamp* time_stamp,
+                            UInt32 bus_number,
+                            UInt32 num_frames,
+                            AudioBufferList* io_data) override;
+  void OnMutedSpeechStatusChanged(bool isDetectingSpeech) override;
+
+  bool IsInterrupted();
+                           
+ public:
+  void (^mutedSpeechDetectionChanged)(bool);
+
+ private:
+  // Called by the relevant AudioSessionObserver methods on `thread_`.
+  void HandleInterruptionBegin();
+  void HandleInterruptionEnd();
+  void HandleValidRouteChange();
+  void HandleCanPlayOrRecordChange(bool can_play_or_record);
+  void HandleSampleRateChange();
+  void HandlePlayoutGlitchDetected();
+  void HandleOutputVolumeChange();
+
+  // Uses current `playout_parameters_` and `record_parameters_` to inform the
+  // audio device buffer (ADB) about our internal audio parameters.
+  void UpdateAudioDeviceBuffer();
+
+  // Since the preferred audio parameters are only hints to the OS, the actual
+  // values may be different once the AVAudioSession has been activated.
+  // This method asks for the current hardware parameters and takes actions
+  // if they should differ from what we have asked for initially. It also
+  // defines `playout_parameters_` and `record_parameters_`.
+  void SetupAudioBuffersForActiveAudioSession();
+
+  // Creates the audio unit.
+  bool CreateAudioUnit();
+
+  // Updates the audio unit state based on current state.
+  void UpdateAudioUnit(bool can_play_or_record);
+
+  // Configures the audio session for WebRTC.
+  bool ConfigureAudioSession();
+
+  // Like above, but requires caller to already hold session lock.
+  bool ConfigureAudioSessionLocked();
+
+  // Unconfigures the audio session.
+  void UnconfigureAudioSession();
+
+  // Activates our audio session, creates and initializes the voice-processing
+  // audio unit and verifies that we got the preferred native audio parameters.
+  bool InitPlayOrRecord();
+
+  // Closes and deletes the voice-processing I/O unit.
+  void ShutdownPlayOrRecord();
+
+  // Resets thread-checkers before a call is restarted.
+  void PrepareForNewStart();
+
+  // Determines whether voice processing should be enabled or disabled.
+  const bool bypass_voice_processing_;
+                           
+  const bool disable_recording_;
+  const bool enableSystemMute_ = false;
+  const int numChannels_;
+
+  // Native I/O audio thread checker.
+  SequenceChecker io_thread_checker_;
+
+  // Thread that this object is created on.
+  rtc::Thread* thread_;
+
+  // Raw pointer handle provided to us in AttachAudioBuffer(). Owned by the
+  // AudioDeviceModuleImpl class and called by AudioDeviceModule::Create().
+  // The AudioDeviceBuffer is a member of the AudioDeviceModuleImpl instance
+  // and therefore outlives this object.
+  AudioDeviceBuffer* audio_device_buffer_;
+
+  // Contains audio parameters (sample rate, #channels, buffer size etc.) for
+  // the playout and recording sides. These structure is set in two steps:
+  // first, native sample rate and #channels are defined in Init(). Next, the
+  // audio session is activated and we verify that the preferred parameters
+  // were granted by the OS. At this stage it is also possible to add a third
+  // component to the parameters; the native I/O buffer duration.
+  // A RTC_CHECK will be hit if we for some reason fail to open an audio session
+  // using the specified parameters.
+  AudioParameters playout_parameters_;
+  AudioParameters record_parameters_;
+
+  // The AudioUnit used to play and record audio.
+  std::unique_ptr<VoiceProcessingAudioUnit> audio_unit_;
+
+  // FineAudioBuffer takes an AudioDeviceBuffer which delivers audio data
+  // in chunks of 10ms. It then allows for this data to be pulled in
+  // a finer or coarser granularity. I.e. interacting with this class instead
+  // of directly with the AudioDeviceBuffer one can ask for any number of
+  // audio data samples. Is also supports a similar scheme for the recording
+  // side.
+  // Example: native buffer size can be 128 audio frames at 16kHz sample rate.
+  // WebRTC will provide 480 audio frames per 10ms but iOS asks for 128
+  // in each callback (one every 8ms). This class can then ask for 128 and the
+  // FineAudioBuffer will ask WebRTC for new data only when needed and also
+  // cache non-utilized audio between callbacks. On the recording side, iOS
+  // can provide audio data frames of size 128 and these are accumulated until
+  // enough data to supply one 10ms call exists. This 10ms chunk is then sent
+  // to WebRTC and the remaining part is stored.
+  std::unique_ptr<FineAudioBuffer> fine_audio_buffer_;
+
+  // Temporary storage for recorded data. AudioUnitRender() renders into this
+  // array as soon as a frame of the desired buffer size has been recorded.
+  // On real iOS devices, the size will be fixed and set once. For iOS
+  // simulators, the size can vary from callback to callback and the size
+  // will be changed dynamically to account for this behavior.
+  rtc::BufferT<int16_t> record_audio_buffer_;
+
+  // Set to 1 when recording is active and 0 otherwise.
+  std::atomic<int> recording_;
+
+  // Set to 1 when playout is active and 0 otherwise.
+  std::atomic<int> playing_;
+
+  // Set to true after successful call to Init(), false otherwise.
+  bool initialized_ RTC_GUARDED_BY(thread_);
+
+  // Set to true after successful call to InitRecording() or InitPlayout(),
+  // false otherwise.
+  bool audio_is_initialized_;
+
+  // Set to true if audio session is interrupted, false otherwise.
+  bool is_interrupted_;
+
+  // Audio interruption observer instance.
+  RTCNativeAudioSessionDelegateAdapter* audio_session_observer_
+      RTC_GUARDED_BY(thread_);
+
+  // Set to true if we've activated the audio session.
+  bool has_configured_session_ RTC_GUARDED_BY(thread_);
+
+  // Counts number of detected audio glitches on the playout side.
+  int64_t num_detected_playout_glitches_ RTC_GUARDED_BY(thread_);
+  int64_t last_playout_time_ RTC_GUARDED_BY(io_thread_checker_);
+
+  // Counts number of playout callbacks per call.
+  // The value is updated on the native I/O thread and later read on the
+  // creating `thread_` but at this stage no audio is active.
+  // Hence, it is a "thread safe" design and no lock is needed.
+  int64_t num_playout_callbacks_;
+
+  // Contains the time for when the last output volume change was detected.
+  int64_t last_output_volume_change_time_ RTC_GUARDED_BY(thread_);
+
+  // Avoids running pending task after `this` is Terminated.
+  webrtc::scoped_refptr<PendingTaskSafetyFlag> safety_ =
+      PendingTaskSafetyFlag::Create();
+                           
+  std::atomic<bool> _hasTone;
+  std::shared_ptr<tgcalls::CallAudioTone> _tone;
+                           
+  bool isBufferPlaying_ = false;
+  bool isBufferRecording_ = false;
+                           
+  bool isMicrophoneMuted_ = false;
+};
+}  // namespace tgcalls_ios_adm
+}  // namespace webrtc
+
+#endif  // SDK_OBJC_NATIVE_SRC_AUDIO_AUDIO_DEVICE_IOS_H_

TMessagesProj/jni/voip/tgcalls/platform/darwin/iOS/tgcalls_audio_device_module_ios.h

@@ -0,0 +1,161 @@
+/*
+ *  Copyright (c) 2012 The WebRTC project authors. All Rights Reserved.
+ *
+ *  Use of this source code is governed by a BSD-style license
+ *  that can be found in the LICENSE file in the root of the source
+ *  tree. An additional intellectual property rights grant can be found
+ *  in the file PATENTS.  All contributing project authors may
+ *  be found in the AUTHORS file in the root of the source tree.
+ */
+
+#ifndef TGCALLS_AUDIO_AUDIO_DEVICE_MODULE_IOS_H_
+#define TGCALLS_AUDIO_AUDIO_DEVICE_MODULE_IOS_H_
+
+#include <memory>
+
+#include "tgcalls_audio_device_ios.h"
+
+#include "api/task_queue/task_queue_factory.h"
+#include "modules/audio_device/audio_device_buffer.h"
+#include "modules/audio_device/include/audio_device.h"
+#include "rtc_base/checks.h"
+
+#include "platform/darwin/iOS/CallAudioTone.h"
+
+namespace webrtc {
+
+class AudioDeviceGeneric;
+
+namespace tgcalls_ios_adm {
+
+class AudioDeviceModuleIOS : public AudioDeviceModule {
+ public:
+  int32_t AttachAudioBuffer();
+
+  explicit AudioDeviceModuleIOS(bool bypass_voice_processing, bool disable_recording, bool enableSystemMute, int numChannels);
+  ~AudioDeviceModuleIOS() override;
+
+  // Retrieve the currently utilized audio layer
+  int32_t ActiveAudioLayer(AudioLayer* audioLayer) const override;
+
+  // Full-duplex transportation of PCM audio
+  int32_t RegisterAudioCallback(AudioTransport* audioCallback) override;
+
+  // Main initializaton and termination
+  int32_t Init() override;
+  int32_t Terminate() override;
+  bool Initialized() const override;
+
+  // Device enumeration
+  int16_t PlayoutDevices() override;
+  int16_t RecordingDevices() override;
+  int32_t PlayoutDeviceName(uint16_t index,
+                            char name[kAdmMaxDeviceNameSize],
+                            char guid[kAdmMaxGuidSize]) override;
+  int32_t RecordingDeviceName(uint16_t index,
+                              char name[kAdmMaxDeviceNameSize],
+                              char guid[kAdmMaxGuidSize]) override;
+
+  // Device selection
+  int32_t SetPlayoutDevice(uint16_t index) override;
+  int32_t SetPlayoutDevice(WindowsDeviceType device) override;
+  int32_t SetRecordingDevice(uint16_t index) override;
+  int32_t SetRecordingDevice(WindowsDeviceType device) override;
+
+  // Audio transport initialization
+  int32_t PlayoutIsAvailable(bool* available) override;
+  int32_t InitPlayout() override;
+  bool PlayoutIsInitialized() const override;
+  int32_t RecordingIsAvailable(bool* available) override;
+  int32_t InitRecording() override;
+  bool RecordingIsInitialized() const override;
+
+  // Audio transport control
+  int32_t InternalStartPlayout();
+  int32_t StartPlayout() override;
+  int32_t StopPlayout() override;
+  bool Playing() const override;
+  int32_t InternalStartRecording();
+  int32_t StartRecording() override;
+  int32_t StopRecording() override;
+  bool Recording() const override;
+
+  // Audio mixer initialization
+  int32_t InitSpeaker() override;
+  bool SpeakerIsInitialized() const override;
+  int32_t InitMicrophone() override;
+  bool MicrophoneIsInitialized() const override;
+
+  // Speaker volume controls
+  int32_t SpeakerVolumeIsAvailable(bool* available) override;
+  int32_t SetSpeakerVolume(uint32_t volume) override;
+  int32_t SpeakerVolume(uint32_t* volume) const override;
+  int32_t MaxSpeakerVolume(uint32_t* maxVolume) const override;
+  int32_t MinSpeakerVolume(uint32_t* minVolume) const override;
+
+  // Microphone volume controls
+  int32_t MicrophoneVolumeIsAvailable(bool* available) override;
+  int32_t SetMicrophoneVolume(uint32_t volume) override;
+  int32_t MicrophoneVolume(uint32_t* volume) const override;
+  int32_t MaxMicrophoneVolume(uint32_t* maxVolume) const override;
+  int32_t MinMicrophoneVolume(uint32_t* minVolume) const override;
+
+  // Speaker mute control
+  int32_t SpeakerMuteIsAvailable(bool* available) override;
+  int32_t SetSpeakerMute(bool enable) override;
+  int32_t SpeakerMute(bool* enabled) const override;
+
+  // Microphone mute control
+  int32_t MicrophoneMuteIsAvailable(bool* available) override;
+  int32_t SetMicrophoneMute(bool enable) override;
+  int32_t MicrophoneMute(bool* enabled) const override;
+
+  // Stereo support
+  int32_t StereoPlayoutIsAvailable(bool* available) const override;
+  int32_t SetStereoPlayout(bool enable) override;
+  int32_t StereoPlayout(bool* enabled) const override;
+  int32_t StereoRecordingIsAvailable(bool* available) const override;
+  int32_t SetStereoRecording(bool enable) override;
+  int32_t StereoRecording(bool* enabled) const override;
+
+  // Delay information and control
+  int32_t PlayoutDelay(uint16_t* delayMS) const override;
+
+  bool BuiltInAECIsAvailable() const override;
+  int32_t EnableBuiltInAEC(bool enable) override;
+  bool BuiltInAGCIsAvailable() const override;
+  int32_t EnableBuiltInAGC(bool enable) override;
+  bool BuiltInNSIsAvailable() const override;
+  int32_t EnableBuiltInNS(bool enable) override;
+
+  int32_t GetPlayoutUnderrunCount() const override;
+    
+  void setTone(std::shared_ptr<tgcalls::CallAudioTone> tone);
+
+#if defined(WEBRTC_IOS)
+  int GetPlayoutAudioParameters(AudioParameters* params) const override;
+  int GetRecordAudioParameters(AudioParameters* params) const override;
+#endif  // WEBRTC_IOS
+    
+ public:
+    void (^mutedSpeechDetectionChanged)(bool);
+
+ private:
+  const bool bypass_voice_processing_;
+  const bool disable_recording_;
+  const bool enableSystemMute_;
+  const int numChannels_;
+  bool initialized_ = false;
+  bool internalIsPlaying_ = false;
+  bool audioBufferPlayoutStarted_ = false;
+  bool audioBufferRecordingStarted_ = false;
+  const std::unique_ptr<TaskQueueFactory> task_queue_factory_;
+  std::unique_ptr<AudioDeviceIOS> audio_device_;
+  std::unique_ptr<AudioDeviceBuffer> audio_device_buffer_;
+    
+  std::shared_ptr<tgcalls::CallAudioTone> pendingAudioTone_;
+};
+}  // namespace tgcalls_ios_adm
+}  // namespace webrtc
+
+#endif  // SDK_OBJC_NATIVE_SRC_AUDIO_AUDIO_DEVICE_MODULE_IOS_H_

TMessagesProj/jni/voip/tgcalls/platform/darwin/iOS/tgcalls_audio_device_module_ios.mm

@@ -0,0 +1,756 @@
+/*
+ *  Copyright (c) 2012 The WebRTC project authors. All Rights Reserved.
+ *
+ *  Use of this source code is governed by a BSD-style license
+ *  that can be found in the LICENSE file in the root of the source
+ *  tree. An additional intellectual property rights grant can be found
+ *  in the file PATENTS.  All contributing project authors may
+ *  be found in the AUTHORS file in the root of the source tree.
+ */
+
+#include "tgcalls_audio_device_module_ios.h"
+
+#include "api/task_queue/default_task_queue_factory.h"
+#include "modules/audio_device/audio_device_config.h"
+#include "modules/audio_device/audio_device_generic.h"
+#include "rtc_base/checks.h"
+#include "rtc_base/logging.h"
+#include "rtc_base/ref_count.h"
+#include "system_wrappers/include/metrics.h"
+#include "api/make_ref_counted.h"
+
+#if defined(WEBRTC_IOS)
+#include "tgcalls_audio_device_ios.h"
+#endif
+
+#define CHECKinitialized_() \
+  {                         \
+    if (!initialized_) {    \
+      return -1;            \
+    };                      \
+  }
+
+#define CHECKinitialized__BOOL() \
+  {                              \
+    if (!initialized_) {         \
+      return false;              \
+    };                           \
+  }
+
+
+namespace webrtc {
+
+webrtc::scoped_refptr<AudioDeviceModule> CreateAudioDeviceModule(bool bypass_voice_processing) {
+    return rtc::make_ref_counted<webrtc::tgcalls_ios_adm::AudioDeviceModuleIOS>(bypass_voice_processing, false, false, 1);
+}
+
+webrtc::scoped_refptr<AudioDeviceModule> AudioDeviceModule::Create(
+    AudioLayer audio_layer,
+    TaskQueueFactory* task_queue_factory) {
+  RTC_DLOG(LS_INFO) << __FUNCTION__;
+    return CreateAudioDeviceModule(false);
+}
+
+}
+
+namespace webrtc {
+namespace tgcalls_ios_adm {
+
+AudioDeviceModuleIOS::AudioDeviceModuleIOS(bool bypass_voice_processing, bool disable_recording, bool enableSystemMute, int numChannels)
+    : bypass_voice_processing_(bypass_voice_processing),
+      disable_recording_(disable_recording),
+      enableSystemMute_(enableSystemMute),
+      numChannels_(numChannels),
+      task_queue_factory_(CreateDefaultTaskQueueFactory()) {
+  RTC_LOG(LS_INFO) << "current platform is IOS";
+  RTC_LOG(LS_INFO) << "iPhone Audio APIs will be utilized.";
+}
+
+  int32_t AudioDeviceModuleIOS::AttachAudioBuffer() {
+    RTC_DLOG(LS_INFO) << __FUNCTION__;
+    audio_device_->AttachAudioBuffer(audio_device_buffer_.get());
+    return 0;
+  }
+
+  AudioDeviceModuleIOS::~AudioDeviceModuleIOS() {
+    RTC_DLOG(LS_INFO) << __FUNCTION__;
+  }
+
+  int32_t AudioDeviceModuleIOS::ActiveAudioLayer(AudioLayer* audioLayer) const {
+    RTC_DLOG(LS_INFO) << __FUNCTION__;
+    AudioLayer activeAudio;
+    if (audio_device_->ActiveAudioLayer(activeAudio) == -1) {
+      return -1;
+    }
+    *audioLayer = activeAudio;
+    return 0;
+  }
+
+  int32_t AudioDeviceModuleIOS::Init() {
+    RTC_DLOG(LS_INFO) << __FUNCTION__;
+    if (initialized_)
+      return 0;
+
+    audio_device_buffer_.reset(new webrtc::AudioDeviceBuffer(task_queue_factory_.get()));
+    audio_device_.reset(new tgcalls_ios_adm::AudioDeviceIOS(bypass_voice_processing_, disable_recording_, enableSystemMute_, numChannels_));
+    RTC_CHECK(audio_device_);
+      
+      if (audio_device_) {
+          audio_device_->mutedSpeechDetectionChanged = [mutedSpeechDetectionChanged copy];
+      }
+
+    this->AttachAudioBuffer();
+
+    AudioDeviceGeneric::InitStatus status = audio_device_->Init();
+    RTC_HISTOGRAM_ENUMERATION(
+        "WebRTC.Audio.InitializationResult", static_cast<int>(status),
+        static_cast<int>(AudioDeviceGeneric::InitStatus::NUM_STATUSES));
+    if (status != AudioDeviceGeneric::InitStatus::OK) {
+      RTC_LOG(LS_ERROR) << "Audio device initialization failed.";
+      return -1;
+    }
+    initialized_ = true;
+      
+    if (pendingAudioTone_) {
+      audio_device_->setTone(pendingAudioTone_);
+      pendingAudioTone_ = nullptr;
+    }
+    return 0;
+  }
+
+  int32_t AudioDeviceModuleIOS::Terminate() {
+    RTC_DLOG(LS_INFO) << __FUNCTION__;
+    if (!initialized_)
+      return 0;
+    if (audio_device_->Terminate() == -1) {
+      return -1;
+    }
+    initialized_ = false;
+    return 0;
+  }
+
+  bool AudioDeviceModuleIOS::Initialized() const {
+    RTC_DLOG(LS_INFO) << __FUNCTION__ << ": " << initialized_;
+    return initialized_;
+  }
+
+  int32_t AudioDeviceModuleIOS::InitSpeaker() {
+    RTC_DLOG(LS_INFO) << __FUNCTION__;
+    CHECKinitialized_();
+    return audio_device_->InitSpeaker();
+  }
+
+  int32_t AudioDeviceModuleIOS::InitMicrophone() {
+    RTC_DLOG(LS_INFO) << __FUNCTION__;
+    CHECKinitialized_();
+    return audio_device_->InitMicrophone();
+  }
+
+  int32_t AudioDeviceModuleIOS::SpeakerVolumeIsAvailable(bool* available) {
+    RTC_DLOG(LS_INFO) << __FUNCTION__;
+    CHECKinitialized_();
+    bool isAvailable = false;
+    if (audio_device_->SpeakerVolumeIsAvailable(isAvailable) == -1) {
+      return -1;
+    }
+    *available = isAvailable;
+    RTC_DLOG(LS_INFO) << "output: " << isAvailable;
+    return 0;
+  }
+
+  int32_t AudioDeviceModuleIOS::SetSpeakerVolume(uint32_t volume) {
+    RTC_DLOG(LS_INFO) << __FUNCTION__ << "(" << volume << ")";
+    CHECKinitialized_();
+    return audio_device_->SetSpeakerVolume(volume);
+  }
+
+  int32_t AudioDeviceModuleIOS::SpeakerVolume(uint32_t* volume) const {
+    RTC_DLOG(LS_INFO) << __FUNCTION__;
+    CHECKinitialized_();
+    uint32_t level = 0;
+    if (audio_device_->SpeakerVolume(level) == -1) {
+      return -1;
+    }
+    *volume = level;
+    RTC_DLOG(LS_INFO) << "output: " << *volume;
+    return 0;
+  }
+
+  bool AudioDeviceModuleIOS::SpeakerIsInitialized() const {
+    RTC_DLOG(LS_INFO) << __FUNCTION__;
+    CHECKinitialized__BOOL();
+    bool isInitialized = audio_device_->SpeakerIsInitialized();
+    RTC_DLOG(LS_INFO) << "output: " << isInitialized;
+    return isInitialized;
+  }
+
+  bool AudioDeviceModuleIOS::MicrophoneIsInitialized() const {
+    RTC_DLOG(LS_INFO) << __FUNCTION__;
+    CHECKinitialized__BOOL();
+    bool isInitialized = audio_device_->MicrophoneIsInitialized();
+    RTC_DLOG(LS_INFO) << "output: " << isInitialized;
+    return isInitialized;
+  }
+
+  int32_t AudioDeviceModuleIOS::MaxSpeakerVolume(uint32_t* maxVolume) const {
+    CHECKinitialized_();
+    uint32_t maxVol = 0;
+    if (audio_device_->MaxSpeakerVolume(maxVol) == -1) {
+      return -1;
+    }
+    *maxVolume = maxVol;
+    return 0;
+  }
+
+  int32_t AudioDeviceModuleIOS::MinSpeakerVolume(uint32_t* minVolume) const {
+    CHECKinitialized_();
+    uint32_t minVol = 0;
+    if (audio_device_->MinSpeakerVolume(minVol) == -1) {
+      return -1;
+    }
+    *minVolume = minVol;
+    return 0;
+  }
+
+  int32_t AudioDeviceModuleIOS::SpeakerMuteIsAvailable(bool* available) {
+    RTC_DLOG(LS_INFO) << __FUNCTION__;
+    CHECKinitialized_();
+    bool isAvailable = false;
+    if (audio_device_->SpeakerMuteIsAvailable(isAvailable) == -1) {
+      return -1;
+    }
+    *available = isAvailable;
+    RTC_DLOG(LS_INFO) << "output: " << isAvailable;
+    return 0;
+  }
+
+  int32_t AudioDeviceModuleIOS::SetSpeakerMute(bool enable) {
+    RTC_DLOG(LS_INFO) << __FUNCTION__ << "(" << enable << ")";
+    CHECKinitialized_();
+    return audio_device_->SetSpeakerMute(enable);
+  }
+
+  int32_t AudioDeviceModuleIOS::SpeakerMute(bool* enabled) const {
+    RTC_DLOG(LS_INFO) << __FUNCTION__;
+    CHECKinitialized_();
+    bool muted = false;
+    if (audio_device_->SpeakerMute(muted) == -1) {
+      return -1;
+    }
+    *enabled = muted;
+    RTC_DLOG(LS_INFO) << "output: " << muted;
+    return 0;
+  }
+
+  int32_t AudioDeviceModuleIOS::MicrophoneMuteIsAvailable(bool* available) {
+    RTC_DLOG(LS_INFO) << __FUNCTION__;
+    CHECKinitialized_();
+    bool isAvailable = false;
+    if (audio_device_->MicrophoneMuteIsAvailable(isAvailable) == -1) {
+      return -1;
+    }
+    *available = isAvailable;
+    RTC_DLOG(LS_INFO) << "output: " << isAvailable;
+    return 0;
+  }
+
+  int32_t AudioDeviceModuleIOS::SetMicrophoneMute(bool enable) {
+    RTC_DLOG(LS_INFO) << __FUNCTION__ << "(" << enable << ")";
+    CHECKinitialized_();
+    return (audio_device_->SetMicrophoneMute(enable));
+  }
+
+  int32_t AudioDeviceModuleIOS::MicrophoneMute(bool* enabled) const {
+    RTC_DLOG(LS_INFO) << __FUNCTION__;
+    CHECKinitialized_();
+    bool muted = false;
+    if (audio_device_->MicrophoneMute(muted) == -1) {
+      return -1;
+    }
+    *enabled = muted;
+    RTC_DLOG(LS_INFO) << "output: " << muted;
+    return 0;
+  }
+
+  int32_t AudioDeviceModuleIOS::MicrophoneVolumeIsAvailable(bool* available) {
+    RTC_DLOG(LS_INFO) << __FUNCTION__;
+    CHECKinitialized_();
+    bool isAvailable = false;
+    if (audio_device_->MicrophoneVolumeIsAvailable(isAvailable) == -1) {
+      return -1;
+    }
+    *available = isAvailable;
+    RTC_DLOG(LS_INFO) << "output: " << isAvailable;
+    return 0;
+  }
+
+  int32_t AudioDeviceModuleIOS::SetMicrophoneVolume(uint32_t volume) {
+    RTC_DLOG(LS_INFO) << __FUNCTION__ << "(" << volume << ")";
+    CHECKinitialized_();
+    return (audio_device_->SetMicrophoneVolume(volume));
+  }
+
+  int32_t AudioDeviceModuleIOS::MicrophoneVolume(uint32_t* volume) const {
+    RTC_DLOG(LS_INFO) << __FUNCTION__;
+    CHECKinitialized_();
+    uint32_t level = 0;
+    if (audio_device_->MicrophoneVolume(level) == -1) {
+      return -1;
+    }
+    *volume = level;
+    RTC_DLOG(LS_INFO) << "output: " << *volume;
+    return 0;
+  }
+
+  int32_t AudioDeviceModuleIOS::StereoRecordingIsAvailable(
+      bool* available) const {
+    RTC_DLOG(LS_INFO) << __FUNCTION__;
+    CHECKinitialized_();
+    bool isAvailable = false;
+    if (audio_device_->StereoRecordingIsAvailable(isAvailable) == -1) {
+      return -1;
+    }
+    *available = isAvailable;
+    RTC_DLOG(LS_INFO) << "output: " << isAvailable;
+    return 0;
+  }
+
+  int32_t AudioDeviceModuleIOS::SetStereoRecording(bool enable) {
+    RTC_DLOG(LS_INFO) << __FUNCTION__ << "(" << enable << ")";
+    CHECKinitialized_();
+    if (enable) {
+      RTC_LOG(LS_WARNING) << "recording in stereo is not supported";
+    }
+    return -1;
+  }
+
+  int32_t AudioDeviceModuleIOS::StereoRecording(bool* enabled) const {
+    RTC_DLOG(LS_INFO) << __FUNCTION__;
+    CHECKinitialized_();
+    bool stereo = false;
+    if (audio_device_->StereoRecording(stereo) == -1) {
+      return -1;
+    }
+    *enabled = stereo;
+    RTC_DLOG(LS_INFO) << "output: " << stereo;
+    return 0;
+  }
+
+  int32_t AudioDeviceModuleIOS::StereoPlayoutIsAvailable(bool* available) const {
+    RTC_DLOG(LS_INFO) << __FUNCTION__;
+    CHECKinitialized_();
+    bool isAvailable = false;
+    if (audio_device_->StereoPlayoutIsAvailable(isAvailable) == -1) {
+      return -1;
+    }
+    *available = isAvailable;
+    RTC_DLOG(LS_INFO) << "output: " << isAvailable;
+    return 0;
+  }
+
+  int32_t AudioDeviceModuleIOS::SetStereoPlayout(bool enable) {
+    RTC_DLOG(LS_INFO) << __FUNCTION__ << "(" << enable << ")";
+    CHECKinitialized_();
+    if (audio_device_->PlayoutIsInitialized()) {
+      RTC_LOG(LS_ERROR) << "unable to set stereo mode while playing side is initialized";
+      return -1;
+    }
+    if (audio_device_->SetStereoPlayout(enable)) {
+      RTC_LOG(LS_WARNING) << "stereo playout is not supported";
+      return -1;
+    }
+    int8_t nChannels(1);
+    if (enable) {
+      nChannels = 2;
+    }
+    audio_device_buffer_.get()->SetPlayoutChannels(nChannels);
+    return 0;
+  }
+
+  int32_t AudioDeviceModuleIOS::StereoPlayout(bool* enabled) const {
+    RTC_DLOG(LS_INFO) << __FUNCTION__;
+    CHECKinitialized_();
+    bool stereo = false;
+    if (audio_device_->StereoPlayout(stereo) == -1) {
+      return -1;
+    }
+    *enabled = stereo;
+    RTC_DLOG(LS_INFO) << "output: " << stereo;
+    return 0;
+  }
+
+  int32_t AudioDeviceModuleIOS::PlayoutIsAvailable(bool* available) {
+    RTC_DLOG(LS_INFO) << __FUNCTION__;
+    CHECKinitialized_();
+    bool isAvailable = false;
+    if (audio_device_->PlayoutIsAvailable(isAvailable) == -1) {
+      return -1;
+    }
+    *available = isAvailable;
+    RTC_DLOG(LS_INFO) << "output: " << isAvailable;
+    return 0;
+  }
+
+  int32_t AudioDeviceModuleIOS::RecordingIsAvailable(bool* available) {
+    RTC_DLOG(LS_INFO) << __FUNCTION__;
+    CHECKinitialized_();
+    bool isAvailable = false;
+    if (audio_device_->RecordingIsAvailable(isAvailable) == -1) {
+      return -1;
+    }
+    *available = isAvailable;
+    RTC_DLOG(LS_INFO) << "output: " << isAvailable;
+    return 0;
+  }
+
+  int32_t AudioDeviceModuleIOS::MaxMicrophoneVolume(uint32_t* maxVolume) const {
+    CHECKinitialized_();
+    uint32_t maxVol(0);
+    if (audio_device_->MaxMicrophoneVolume(maxVol) == -1) {
+      return -1;
+    }
+    *maxVolume = maxVol;
+    return 0;
+  }
+
+  int32_t AudioDeviceModuleIOS::MinMicrophoneVolume(uint32_t* minVolume) const {
+    CHECKinitialized_();
+    uint32_t minVol(0);
+    if (audio_device_->MinMicrophoneVolume(minVol) == -1) {
+      return -1;
+    }
+    *minVolume = minVol;
+    return 0;
+  }
+
+  int16_t AudioDeviceModuleIOS::PlayoutDevices() {
+    RTC_DLOG(LS_INFO) << __FUNCTION__;
+    CHECKinitialized_();
+    uint16_t nPlayoutDevices = audio_device_->PlayoutDevices();
+    RTC_DLOG(LS_INFO) << "output: " << nPlayoutDevices;
+    return (int16_t)(nPlayoutDevices);
+  }
+
+  int32_t AudioDeviceModuleIOS::SetPlayoutDevice(uint16_t index) {
+    RTC_DLOG(LS_INFO) << __FUNCTION__ << "(" << index << ")";
+    CHECKinitialized_();
+    return audio_device_->SetPlayoutDevice(index);
+  }
+
+  int32_t AudioDeviceModuleIOS::SetPlayoutDevice(WindowsDeviceType device) {
+    RTC_DLOG(LS_INFO) << __FUNCTION__;
+    CHECKinitialized_();
+    return audio_device_->SetPlayoutDevice(device);
+  }
+
+  int32_t AudioDeviceModuleIOS::PlayoutDeviceName(
+      uint16_t index,
+      char name[kAdmMaxDeviceNameSize],
+      char guid[kAdmMaxGuidSize]) {
+    RTC_DLOG(LS_INFO) << __FUNCTION__ << "(" << index << ", ...)";
+    CHECKinitialized_();
+    if (name == NULL) {
+      return -1;
+    }
+    if (audio_device_->PlayoutDeviceName(index, name, guid) == -1) {
+      return -1;
+    }
+    if (name != NULL) {
+      RTC_DLOG(LS_INFO) << "output: name = " << name;
+    }
+    if (guid != NULL) {
+      RTC_DLOG(LS_INFO) << "output: guid = " << guid;
+    }
+    return 0;
+  }
+
+  int32_t AudioDeviceModuleIOS::RecordingDeviceName(
+      uint16_t index,
+      char name[kAdmMaxDeviceNameSize],
+      char guid[kAdmMaxGuidSize]) {
+    RTC_DLOG(LS_INFO) << __FUNCTION__ << "(" << index << ", ...)";
+    CHECKinitialized_();
+    if (name == NULL) {
+      return -1;
+    }
+    if (audio_device_->RecordingDeviceName(index, name, guid) == -1) {
+      return -1;
+    }
+    if (name != NULL) {
+      RTC_DLOG(LS_INFO) << "output: name = " << name;
+    }
+    if (guid != NULL) {
+      RTC_DLOG(LS_INFO) << "output: guid = " << guid;
+    }
+    return 0;
+  }
+
+  int16_t AudioDeviceModuleIOS::RecordingDevices() {
+    RTC_DLOG(LS_INFO) << __FUNCTION__;
+    CHECKinitialized_();
+    uint16_t nRecordingDevices = audio_device_->RecordingDevices();
+    RTC_DLOG(LS_INFO) << "output: " << nRecordingDevices;
+    return (int16_t)nRecordingDevices;
+  }
+
+  int32_t AudioDeviceModuleIOS::SetRecordingDevice(uint16_t index) {
+    RTC_DLOG(LS_INFO) << __FUNCTION__ << "(" << index << ")";
+    CHECKinitialized_();
+    return audio_device_->SetRecordingDevice(index);
+  }
+
+  int32_t AudioDeviceModuleIOS::SetRecordingDevice(WindowsDeviceType device) {
+    RTC_DLOG(LS_INFO) << __FUNCTION__;
+    CHECKinitialized_();
+    return audio_device_->SetRecordingDevice(device);
+  }
+
+  int32_t AudioDeviceModuleIOS::InitPlayout() {
+    RTC_DLOG(LS_INFO) << __FUNCTION__;
+    CHECKinitialized_();
+    if (PlayoutIsInitialized()) {
+      return 0;
+    }
+    int32_t result = audio_device_->InitPlayout();
+    RTC_DLOG(LS_INFO) << "output: " << result;
+    RTC_HISTOGRAM_BOOLEAN("WebRTC.Audio.InitPlayoutSuccess",
+                          static_cast<int>(result == 0));
+    return result;
+  }
+
+  int32_t AudioDeviceModuleIOS::InitRecording() {
+    RTC_DLOG(LS_INFO) << __FUNCTION__;
+    CHECKinitialized_();
+    if (RecordingIsInitialized()) {
+      return 0;
+    }
+    int32_t result = audio_device_->InitRecording();
+    RTC_DLOG(LS_INFO) << "output: " << result;
+    RTC_HISTOGRAM_BOOLEAN("WebRTC.Audio.InitRecordingSuccess",
+                          static_cast<int>(result == 0));
+    return result;
+  }
+
+  bool AudioDeviceModuleIOS::PlayoutIsInitialized() const {
+    RTC_DLOG(LS_INFO) << __FUNCTION__;
+    CHECKinitialized__BOOL();
+    return audio_device_->PlayoutIsInitialized();
+  }
+
+  bool AudioDeviceModuleIOS::RecordingIsInitialized() const {
+    RTC_DLOG(LS_INFO) << __FUNCTION__;
+    CHECKinitialized__BOOL();
+    return audio_device_->RecordingIsInitialized();
+  }
+
+  int32_t AudioDeviceModuleIOS::InternalStartPlayout() {
+    RTC_DLOG(LS_INFO) << __FUNCTION__;
+    CHECKinitialized_();
+    if (audio_device_->Playing()) {
+      return 0;
+    }
+    int32_t result = audio_device_->StartPlayout();
+    RTC_DLOG(LS_INFO) << "output: " << result;
+    RTC_HISTOGRAM_BOOLEAN("WebRTC.Audio.StartPlayoutSuccess",
+                          static_cast<int>(result == 0));
+    return result;
+  }
+
+  int32_t AudioDeviceModuleIOS::StartPlayout() {
+    RTC_DLOG(LS_INFO) << __FUNCTION__;
+    CHECKinitialized_();
+    if (audio_device_->Playing()) {
+      if (!audioBufferPlayoutStarted_) {
+        audioBufferPlayoutStarted_ = true;
+        audio_device_buffer_.get()->StartPlayout();
+        audio_device_->setIsBufferPlaying(true);
+      }
+      internalIsPlaying_ = true;
+      return 0;
+    }
+    audio_device_buffer_.get()->StartPlayout();
+    audio_device_->setIsBufferPlaying(true);
+    audioBufferPlayoutStarted_ = true;
+    int32_t result = audio_device_->StartPlayout();
+    RTC_DLOG(LS_INFO) << "output: " << result;
+    RTC_HISTOGRAM_BOOLEAN("WebRTC.Audio.StartPlayoutSuccess",
+                          static_cast<int>(result == 0));
+    internalIsPlaying_ = true;
+    return result;
+  }
+
+  int32_t AudioDeviceModuleIOS::StopPlayout() {
+    RTC_DLOG(LS_INFO) << __FUNCTION__;
+    CHECKinitialized_();
+    int32_t result = audio_device_->StopPlayout();
+    if (audioBufferPlayoutStarted_) {
+      audio_device_buffer_.get()->StopPlayout();
+      audioBufferPlayoutStarted_ = false;
+      audio_device_->setIsBufferPlaying(false);
+    }
+    RTC_DLOG(LS_INFO) << "output: " << result;
+    RTC_HISTOGRAM_BOOLEAN("WebRTC.Audio.StopPlayoutSuccess",
+                          static_cast<int>(result == 0));
+    return result;
+  }
+
+  bool AudioDeviceModuleIOS::Playing() const {
+    RTC_DLOG(LS_INFO) << __FUNCTION__;
+    CHECKinitialized__BOOL();
+    return internalIsPlaying_;
+  }
+
+  int32_t AudioDeviceModuleIOS::InternalStartRecording() {
+    RTC_DLOG(LS_INFO) << __FUNCTION__;
+    CHECKinitialized_();
+    if (Recording()) {
+      return 0;
+    }
+    int32_t result = audio_device_->StartRecording();
+    RTC_DLOG(LS_INFO) << "output: " << result;
+    RTC_HISTOGRAM_BOOLEAN("WebRTC.Audio.StartRecordingSuccess",
+                          static_cast<int>(result == 0));
+    return result;
+}
+
+  int32_t AudioDeviceModuleIOS::StartRecording() {
+    RTC_DLOG(LS_INFO) << __FUNCTION__;
+    CHECKinitialized_();
+    if (Recording()) {
+      if (!audioBufferRecordingStarted_) {
+        audioBufferRecordingStarted_ = true;
+        audio_device_buffer_.get()->StartRecording();
+        audio_device_->setIsBufferRecording(true);
+      }
+      return 0;
+    }
+    audio_device_buffer_.get()->StartRecording();
+    audioBufferRecordingStarted_ = true;
+    audio_device_->setIsBufferRecording(true);
+    int32_t result = audio_device_->StartRecording();
+    RTC_DLOG(LS_INFO) << "output: " << result;
+    RTC_HISTOGRAM_BOOLEAN("WebRTC.Audio.StartRecordingSuccess",
+                          static_cast<int>(result == 0));
+    return result;
+  }
+
+  int32_t AudioDeviceModuleIOS::StopRecording() {
+    RTC_DLOG(LS_INFO) << __FUNCTION__;
+    CHECKinitialized_();
+    int32_t result = audio_device_->StopRecording();
+    if (audioBufferRecordingStarted_) {
+      audioBufferRecordingStarted_ = false;
+      audio_device_buffer_.get()->StopRecording();
+        audio_device_->setIsBufferRecording(false);
+    }
+    RTC_DLOG(LS_INFO) << "output: " << result;
+    RTC_HISTOGRAM_BOOLEAN("WebRTC.Audio.StopRecordingSuccess",
+                          static_cast<int>(result == 0));
+    return result;
+  }
+
+  bool AudioDeviceModuleIOS::Recording() const {
+    RTC_DLOG(LS_INFO) << __FUNCTION__;
+    CHECKinitialized__BOOL();
+    return audio_device_->Recording();
+  }
+
+  int32_t AudioDeviceModuleIOS::RegisterAudioCallback(
+      AudioTransport* audioCallback) {
+    RTC_DLOG(LS_INFO) << __FUNCTION__;
+    return audio_device_buffer_.get()->RegisterAudioCallback(audioCallback);
+  }
+
+  int32_t AudioDeviceModuleIOS::PlayoutDelay(uint16_t* delayMS) const {
+    CHECKinitialized_();
+    uint16_t delay = 0;
+    if (audio_device_->PlayoutDelay(delay) == -1) {
+      RTC_LOG(LS_ERROR) << "failed to retrieve the playout delay";
+      return -1;
+    }
+    *delayMS = delay;
+    return 0;
+  }
+
+  bool AudioDeviceModuleIOS::BuiltInAECIsAvailable() const {
+    RTC_DLOG(LS_INFO) << __FUNCTION__;
+    CHECKinitialized__BOOL();
+    bool isAvailable = audio_device_->BuiltInAECIsAvailable();
+    RTC_DLOG(LS_INFO) << "output: " << isAvailable;
+    return isAvailable;
+  }
+
+  int32_t AudioDeviceModuleIOS::EnableBuiltInAEC(bool enable) {
+    RTC_DLOG(LS_INFO) << __FUNCTION__ << "(" << enable << ")";
+    CHECKinitialized_();
+    int32_t ok = audio_device_->EnableBuiltInAEC(enable);
+    RTC_DLOG(LS_INFO) << "output: " << ok;
+    return ok;
+  }
+
+  bool AudioDeviceModuleIOS::BuiltInAGCIsAvailable() const {
+    RTC_DLOG(LS_INFO) << __FUNCTION__;
+    CHECKinitialized__BOOL();
+    bool isAvailable = audio_device_->BuiltInAGCIsAvailable();
+    RTC_DLOG(LS_INFO) << "output: " << isAvailable;
+    return isAvailable;
+  }
+
+  int32_t AudioDeviceModuleIOS::EnableBuiltInAGC(bool enable) {
+    RTC_DLOG(LS_INFO) << __FUNCTION__ << "(" << enable << ")";
+    CHECKinitialized_();
+    int32_t ok = audio_device_->EnableBuiltInAGC(enable);
+    RTC_DLOG(LS_INFO) << "output: " << ok;
+    return ok;
+  }
+
+  bool AudioDeviceModuleIOS::BuiltInNSIsAvailable() const {
+    RTC_DLOG(LS_INFO) << __FUNCTION__;
+    CHECKinitialized__BOOL();
+    bool isAvailable = audio_device_->BuiltInNSIsAvailable();
+    RTC_DLOG(LS_INFO) << "output: " << isAvailable;
+    return isAvailable;
+  }
+
+  int32_t AudioDeviceModuleIOS::EnableBuiltInNS(bool enable) {
+    RTC_DLOG(LS_INFO) << __FUNCTION__ << "(" << enable << ")";
+    CHECKinitialized_();
+    int32_t ok = audio_device_->EnableBuiltInNS(enable);
+    RTC_DLOG(LS_INFO) << "output: " << ok;
+    return ok;
+  }
+
+  int32_t AudioDeviceModuleIOS::GetPlayoutUnderrunCount() const {
+    // Don't log here, as this method can be called very often.
+    CHECKinitialized_();
+    int32_t ok = audio_device_->GetPlayoutUnderrunCount();
+    return ok;
+  }
+
+  void AudioDeviceModuleIOS::setTone(std::shared_ptr<tgcalls::CallAudioTone> tone) {
+    if (audio_device_) {
+      audio_device_->setTone(tone);
+    } else {
+      pendingAudioTone_ = tone;
+    }
+  }
+
+#if defined(WEBRTC_IOS)
+  int AudioDeviceModuleIOS::GetPlayoutAudioParameters(
+      AudioParameters* params) const {
+    RTC_DLOG(LS_INFO) << __FUNCTION__;
+    int r = audio_device_->GetPlayoutAudioParameters(params);
+    RTC_DLOG(LS_INFO) << "output: " << r;
+    return r;
+  }
+
+  int AudioDeviceModuleIOS::GetRecordAudioParameters(
+      AudioParameters* params) const {
+    RTC_DLOG(LS_INFO) << __FUNCTION__;
+    int r = audio_device_->GetRecordAudioParameters(params);
+    RTC_DLOG(LS_INFO) << "output: " << r;
+    return r;
+  }
+#endif  // WEBRTC_IOS
+}
+}

TMessagesProj/jni/voip/tgcalls/platform/darwin/iOS/tgcalls_voice_processing_audio_unit.h

@@ -0,0 +1,149 @@
+/*
+ *  Copyright 2016 The WebRTC Project Authors. All rights reserved.
+ *
+ *  Use of this source code is governed by a BSD-style license
+ *  that can be found in the LICENSE file in the root of the source
+ *  tree. An additional intellectual property rights grant can be found
+ *  in the file PATENTS.  All contributing project authors may
+ *  be found in the AUTHORS file in the root of the source tree.
+ */
+
+#ifndef TGCALLS_AUDIO_VOICE_PROCESSING_AUDIO_UNIT_H_
+#define TGCALLS_AUDIO_VOICE_PROCESSING_AUDIO_UNIT_H_
+
+#include <AudioUnit/AudioUnit.h>
+
+namespace webrtc {
+namespace tgcalls_ios_adm {
+
+class VoiceProcessingAudioUnitObserver {
+ public:
+  // Callback function called on a real-time priority I/O thread from the audio
+  // unit. This method is used to signal that recorded audio is available.
+  virtual OSStatus OnDeliverRecordedData(AudioUnitRenderActionFlags* flags,
+                                         const AudioTimeStamp* time_stamp,
+                                         UInt32 bus_number,
+                                         UInt32 num_frames,
+                                         AudioBufferList* io_data) = 0;
+
+  // Callback function called on a real-time priority I/O thread from the audio
+  // unit. This method is used to provide audio samples to the audio unit.
+  virtual OSStatus OnGetPlayoutData(AudioUnitRenderActionFlags* io_action_flags,
+                                    const AudioTimeStamp* time_stamp,
+                                    UInt32 bus_number,
+                                    UInt32 num_frames,
+                                    AudioBufferList* io_data) = 0;
+    
+  virtual void OnMutedSpeechStatusChanged(bool isDetectingSpeech) {}
+
+ protected:
+  ~VoiceProcessingAudioUnitObserver() {}
+};
+
+// Convenience class to abstract away the management of a Voice Processing
+// I/O Audio Unit. The Voice Processing I/O unit has the same characteristics
+// as the Remote I/O unit (supports full duplex low-latency audio input and
+// output) and adds AEC for for two-way duplex communication. It also adds AGC,
+// adjustment of voice-processing quality, and muting. Hence, ideal for
+// VoIP applications.
+class VoiceProcessingAudioUnit {
+ public:
+  VoiceProcessingAudioUnit(bool bypass_voice_processing, bool disable_recording, bool enableSystemMute, int numChannels,
+                           VoiceProcessingAudioUnitObserver* observer);
+  ~VoiceProcessingAudioUnit();
+
+  // TODO(tkchin): enum for state and state checking.
+  enum State : int32_t {
+    // Init() should be called.
+    kInitRequired,
+    // Audio unit created but not initialized.
+    kUninitialized,
+    // Initialized but not started. Equivalent to stopped.
+    kInitialized,
+    // Initialized and started.
+    kStarted,
+  };
+
+  // Number of bytes per audio sample for 16-bit signed integer representation.
+  static const UInt32 kBytesPerSample;
+
+  // Initializes this class by creating the underlying audio unit instance.
+  // Creates a Voice-Processing I/O unit and configures it for full-duplex
+  // audio. The selected stream format is selected to avoid internal resampling
+  // and to match the 10ms callback rate for WebRTC as well as possible.
+  // Does not intialize the audio unit.
+  bool Init();
+
+  VoiceProcessingAudioUnit::State GetState() const;
+
+  // Initializes the underlying audio unit with the given sample rate.
+  bool Initialize(Float64 sample_rate);
+
+  // Starts the underlying audio unit.
+  OSStatus Start();
+
+  // Stops the underlying audio unit.
+  bool Stop();
+
+  // Uninitializes the underlying audio unit.
+  bool Uninitialize();
+    
+  void setIsMicrophoneMuted(bool isMuted);
+
+  // Calls render on the underlying audio unit.
+  OSStatus Render(AudioUnitRenderActionFlags* flags,
+                  const AudioTimeStamp* time_stamp,
+                  UInt32 output_bus_number,
+                  UInt32 num_frames,
+                  AudioBufferList* io_data);
+
+ private:
+  // The C API used to set callbacks requires static functions. When these are
+  // called, they will invoke the relevant instance method by casting
+  // in_ref_con to VoiceProcessingAudioUnit*.
+  static OSStatus OnGetPlayoutData(void* in_ref_con,
+                                   AudioUnitRenderActionFlags* flags,
+                                   const AudioTimeStamp* time_stamp,
+                                   UInt32 bus_number,
+                                   UInt32 num_frames,
+                                   AudioBufferList* io_data);
+  static OSStatus OnDeliverRecordedData(void* in_ref_con,
+                                        AudioUnitRenderActionFlags* flags,
+                                        const AudioTimeStamp* time_stamp,
+                                        UInt32 bus_number,
+                                        UInt32 num_frames,
+                                        AudioBufferList* io_data);
+
+  // Notifies observer that samples are needed for playback.
+  OSStatus NotifyGetPlayoutData(AudioUnitRenderActionFlags* flags,
+                                const AudioTimeStamp* time_stamp,
+                                UInt32 bus_number,
+                                UInt32 num_frames,
+                                AudioBufferList* io_data);
+  // Notifies observer that recorded samples are available for render.
+  OSStatus NotifyDeliverRecordedData(AudioUnitRenderActionFlags* flags,
+                                     const AudioTimeStamp* time_stamp,
+                                     UInt32 bus_number,
+                                     UInt32 num_frames,
+                                     AudioBufferList* io_data);
+
+  // Returns the predetermined format with a specific sample rate. See
+  // implementation file for details on format.
+  AudioStreamBasicDescription GetFormat(Float64 sample_rate, int numChannels) const;
+
+  // Deletes the underlying audio unit.
+  void DisposeAudioUnit();
+
+  const bool bypass_voice_processing_;
+  const bool disable_recording_;
+  const int numChannels_;
+  bool enableSystemMute_ = false;
+  bool isMicrophoneMuted_ = true;
+  VoiceProcessingAudioUnitObserver* observer_;
+  AudioUnit vpio_unit_;
+  VoiceProcessingAudioUnit::State state_;
+};
+}  // namespace tgcalls_ios_adm
+}  // namespace webrtc
+
+#endif  // SDK_OBJC_NATIVE_SRC_AUDIO_VOICE_PROCESSING_AUDIO_UNIT_H_

TMessagesProj/jni/voip/tgcalls/platform/darwin/iOS/tgcalls_voice_processing_audio_unit.mm

@@ -0,0 +1,544 @@
+/*
+ *  Copyright 2016 The WebRTC Project Authors. All rights reserved.
+ *
+ *  Use of this source code is governed by a BSD-style license
+ *  that can be found in the LICENSE file in the root of the source
+ *  tree. An additional intellectual property rights grant can be found
+ *  in the file PATENTS.  All contributing project authors may
+ *  be found in the AUTHORS file in the root of the source tree.
+ */
+
+#import "tgcalls_voice_processing_audio_unit.h"
+
+#include "rtc_base/checks.h"
+#include "system_wrappers/include/metrics.h"
+
+#import "base/RTCLogging.h"
+#import "platform/darwin/iOS/RTCAudioSessionConfiguration.h"
+
+#if !defined(NDEBUG)
+static void LogStreamDescription(AudioStreamBasicDescription description) {
+  char formatIdString[5];
+  UInt32 formatId = CFSwapInt32HostToBig(description.mFormatID);
+  bcopy(&formatId, formatIdString, 4);
+  formatIdString[4] = '\0';
+  RTCLog(@"AudioStreamBasicDescription: {\n"
+          "  mSampleRate: %.2f\n"
+          "  formatIDString: %s\n"
+          "  mFormatFlags: 0x%X\n"
+          "  mBytesPerPacket: %u\n"
+          "  mFramesPerPacket: %u\n"
+          "  mBytesPerFrame: %u\n"
+          "  mChannelsPerFrame: %u\n"
+          "  mBitsPerChannel: %u\n"
+          "  mReserved: %u\n}",
+         description.mSampleRate, formatIdString,
+         static_cast<unsigned int>(description.mFormatFlags),
+         static_cast<unsigned int>(description.mBytesPerPacket),
+         static_cast<unsigned int>(description.mFramesPerPacket),
+         static_cast<unsigned int>(description.mBytesPerFrame),
+         static_cast<unsigned int>(description.mChannelsPerFrame),
+         static_cast<unsigned int>(description.mBitsPerChannel),
+         static_cast<unsigned int>(description.mReserved));
+}
+#endif
+
+namespace webrtc {
+namespace tgcalls_ios_adm {
+
+// Calls to AudioUnitInitialize() can fail if called back-to-back on different
+// ADM instances. A fall-back solution is to allow multiple sequential calls
+// with as small delay between each. This factor sets the max number of allowed
+// initialization attempts.
+static const int kMaxNumberOfAudioUnitInitializeAttempts = 5;
+// A VP I/O unit's bus 1 connects to input hardware (microphone).
+static const AudioUnitElement kInputBus = 1;
+// A VP I/O unit's bus 0 connects to output hardware (speaker).
+static const AudioUnitElement kOutputBus = 0;
+
+// Returns the automatic gain control (AGC) state on the processed microphone
+// signal. Should be on by default for Voice Processing audio units.
+static OSStatus GetAGCState(AudioUnit audio_unit, UInt32* enabled) {
+  RTC_DCHECK(audio_unit);
+  UInt32 size = sizeof(*enabled);
+  OSStatus result = AudioUnitGetProperty(audio_unit,
+                                         kAUVoiceIOProperty_VoiceProcessingEnableAGC,
+                                         kAudioUnitScope_Global,
+                                         kInputBus,
+                                         enabled,
+                                         &size);
+  RTCLog(@"VPIO unit AGC: %u", static_cast<unsigned int>(*enabled));
+  return result;
+}
+
+VoiceProcessingAudioUnit::VoiceProcessingAudioUnit(bool bypass_voice_processing, bool disable_recording, bool enableSystemMute, int numChannels,
+                                                   VoiceProcessingAudioUnitObserver* observer)
+    : bypass_voice_processing_(bypass_voice_processing),
+      disable_recording_(disable_recording),
+      numChannels_(numChannels),
+      enableSystemMute_(enableSystemMute),
+      observer_(observer),
+      vpio_unit_(nullptr),
+      state_(kInitRequired) {
+  RTC_DCHECK(observer);
+}
+
+VoiceProcessingAudioUnit::~VoiceProcessingAudioUnit() {
+  DisposeAudioUnit();
+}
+
+const UInt32 VoiceProcessingAudioUnit::kBytesPerSample = 2;
+
+bool VoiceProcessingAudioUnit::Init() {
+  RTC_DCHECK_EQ(state_, kInitRequired);
+
+  // Create an audio component description to identify the Voice Processing
+  // I/O audio unit.
+  AudioComponentDescription vpio_unit_description;
+  vpio_unit_description.componentType = kAudioUnitType_Output;
+    
+  if (disable_recording_) {
+    vpio_unit_description.componentSubType = kAudioUnitSubType_RemoteIO;
+  } else {
+    vpio_unit_description.componentSubType = kAudioUnitSubType_VoiceProcessingIO;
+  }
+  vpio_unit_description.componentManufacturer = kAudioUnitManufacturer_Apple;
+  vpio_unit_description.componentFlags = 0;
+  vpio_unit_description.componentFlagsMask = 0;
+
+  // Obtain an audio unit instance given the description.
+  AudioComponent found_vpio_unit_ref =
+      AudioComponentFindNext(nullptr, &vpio_unit_description);
+
+  // Create a Voice Processing IO audio unit.
+  OSStatus result = noErr;
+  result = AudioComponentInstanceNew(found_vpio_unit_ref, &vpio_unit_);
+  if (result != noErr) {
+    vpio_unit_ = nullptr;
+    RTCLogError(@"AudioComponentInstanceNew failed. Error=%ld.", (long)result);
+    return false;
+  }
+
+    if (!disable_recording_) {
+        // Enable input on the input scope of the input element.
+        UInt32 enable_input = 1;
+        result = AudioUnitSetProperty(vpio_unit_, kAudioOutputUnitProperty_EnableIO,
+                                      kAudioUnitScope_Input, kInputBus, &enable_input,
+                                      sizeof(enable_input));
+        if (result != noErr) {
+            DisposeAudioUnit();
+            RTCLogError(@"Failed to enable input on input scope of input element. "
+                        "Error=%ld.",
+                        (long)result);
+            return false;
+        }
+    }
+
+  // Enable output on the output scope of the output element.
+  UInt32 enable_output = 1;
+  result = AudioUnitSetProperty(vpio_unit_, kAudioOutputUnitProperty_EnableIO,
+                                kAudioUnitScope_Output, kOutputBus,
+                                &enable_output, sizeof(enable_output));
+  if (result != noErr) {
+    DisposeAudioUnit();
+    RTCLogError(@"Failed to enable output on output scope of output element. "
+                 "Error=%ld.",
+                (long)result);
+    return false;
+  }
+
+  // Specify the callback function that provides audio samples to the audio
+  // unit.
+  AURenderCallbackStruct render_callback;
+  render_callback.inputProc = OnGetPlayoutData;
+  render_callback.inputProcRefCon = this;
+  result = AudioUnitSetProperty(
+      vpio_unit_, kAudioUnitProperty_SetRenderCallback, kAudioUnitScope_Input,
+      kOutputBus, &render_callback, sizeof(render_callback));
+  if (result != noErr) {
+    DisposeAudioUnit();
+    RTCLogError(@"Failed to specify the render callback on the output bus. "
+                 "Error=%ld.",
+                (long)result);
+    return false;
+  }
+
+  if (!disable_recording_) {
+      // Disable AU buffer allocation for the recorder, we allocate our own.
+      // TODO(henrika): not sure that it actually saves resource to make this call.
+      UInt32 flag = 0;
+      result = AudioUnitSetProperty(
+                                    vpio_unit_, kAudioUnitProperty_ShouldAllocateBuffer,
+                                    kAudioUnitScope_Output, kInputBus, &flag, sizeof(flag));
+      if (result != noErr) {
+          DisposeAudioUnit();
+          RTCLogError(@"Failed to disable buffer allocation on the input bus. "
+                      "Error=%ld.",
+                      (long)result);
+          return false;
+      }
+      
+      
+      // Specify the callback to be called by the I/O thread to us when input audio
+      // is available. The recorded samples can then be obtained by calling the
+      // AudioUnitRender() method.
+      AURenderCallbackStruct input_callback;
+      input_callback.inputProc = OnDeliverRecordedData;
+      input_callback.inputProcRefCon = this;
+      result = AudioUnitSetProperty(vpio_unit_,
+                                    kAudioOutputUnitProperty_SetInputCallback,
+                                    kAudioUnitScope_Global, kInputBus,
+                                    &input_callback, sizeof(input_callback));
+      if (result != noErr) {
+          DisposeAudioUnit();
+          RTCLogError(@"Failed to specify the input callback on the input bus. "
+                      "Error=%ld.",
+                      (long)result);
+          return false;
+      }
+      
+      if (@available(iOS 15.0, macOS 14.0, *)) {
+          VoiceProcessingAudioUnit *audio_unit = this;
+          AUVoiceIOMutedSpeechActivityEventListener listener = ^(AUVoiceIOSpeechActivityEvent event) {
+              if (event == kAUVoiceIOSpeechActivityHasStarted) {
+                  if (audio_unit->observer_) {
+                      audio_unit->observer_->OnMutedSpeechStatusChanged(true);
+                  }
+              } else if (event == kAUVoiceIOSpeechActivityHasEnded) {
+                  if (audio_unit->observer_) {
+                      audio_unit->observer_->OnMutedSpeechStatusChanged(false);
+                  }
+              }
+          };
+          result = AudioUnitSetProperty(vpio_unit_, kAUVoiceIOProperty_MutedSpeechActivityEventListener, kAudioUnitScope_Global, kInputBus, &listener, sizeof(AUVoiceIOMutedSpeechActivityEventListener));
+      }
+  }
+    
+    if (enableSystemMute_) {
+        UInt32 muteUplinkOutput = isMicrophoneMuted_ ? 1 : 0;
+        OSStatus result = AudioUnitSetProperty(vpio_unit_, kAUVoiceIOProperty_MuteOutput, kAudioUnitScope_Global, kInputBus, &muteUplinkOutput, sizeof(muteUplinkOutput));
+        if (result != noErr) {
+            RTCLogError(@"Failed to set kAUVoiceIOProperty_MuteOutput. Error=%ld", (long)result);
+        }
+    }
+
+  state_ = kUninitialized;
+  return true;
+}
+
+VoiceProcessingAudioUnit::State VoiceProcessingAudioUnit::GetState() const {
+  return state_;
+}
+
+bool VoiceProcessingAudioUnit::Initialize(Float64 sample_rate) {
+  RTC_DCHECK_GE(state_, kUninitialized);
+  RTCLog(@"Initializing audio unit with sample rate: %f", sample_rate);
+
+  OSStatus result = noErr;
+  AudioStreamBasicDescription outputFormat = GetFormat(sample_rate, numChannels_);
+  AudioStreamBasicDescription inputFormat = GetFormat(sample_rate, 1);
+  UInt32 size = sizeof(outputFormat);
+#if !defined(NDEBUG)
+  LogStreamDescription(outputFormat);
+#endif
+
+    if (!disable_recording_) {
+        // Set the format on the output scope of the input element/bus.
+        result =
+        AudioUnitSetProperty(vpio_unit_, kAudioUnitProperty_StreamFormat,
+                             kAudioUnitScope_Output, kInputBus, &inputFormat, size);
+        if (result != noErr) {
+            RTCLogError(@"Failed to set format on output scope of input bus. "
+                        "Error=%ld.",
+                        (long)result);
+            return false;
+        }
+    }
+
+  // Set the format on the input scope of the output element/bus.
+  result =
+      AudioUnitSetProperty(vpio_unit_, kAudioUnitProperty_StreamFormat,
+                           kAudioUnitScope_Input, kOutputBus, &outputFormat, size);
+  if (result != noErr) {
+    RTCLogError(@"Failed to set format on input scope of output bus. "
+                 "Error=%ld.",
+                (long)result);
+    return false;
+  }
+
+  // Initialize the Voice Processing I/O unit instance.
+  // Calls to AudioUnitInitialize() can fail if called back-to-back on
+  // different ADM instances. The error message in this case is -66635 which is
+  // undocumented. Tests have shown that calling AudioUnitInitialize a second
+  // time, after a short sleep, avoids this issue.
+  // See webrtc:5166 for details.
+  int failed_initalize_attempts = 0;
+  result = AudioUnitInitialize(vpio_unit_);
+  while (result != noErr) {
+    RTCLogError(@"Failed to initialize the Voice Processing I/O unit. "
+                 "Error=%ld.",
+                (long)result);
+    ++failed_initalize_attempts;
+    if (failed_initalize_attempts == kMaxNumberOfAudioUnitInitializeAttempts) {
+      // Max number of initialization attempts exceeded, hence abort.
+      RTCLogError(@"Too many initialization attempts.");
+      return false;
+    }
+    RTCLog(@"Pause 100ms and try audio unit initialization again...");
+    [NSThread sleepForTimeInterval:0.1f];
+    result = AudioUnitInitialize(vpio_unit_);
+  }
+  if (result == noErr) {
+    RTCLog(@"Voice Processing I/O unit is now initialized.");
+  }
+
+  if (bypass_voice_processing_) {
+    // Attempt to disable builtin voice processing.
+    UInt32 toggle = 1;
+    result = AudioUnitSetProperty(vpio_unit_,
+                                  kAUVoiceIOProperty_BypassVoiceProcessing,
+                                  kAudioUnitScope_Global,
+                                  kInputBus,
+                                  &toggle,
+                                  sizeof(toggle));
+    if (result == noErr) {
+      RTCLog(@"Successfully bypassed voice processing.");
+    } else {
+      RTCLogError(@"Failed to bypass voice processing. Error=%ld.", (long)result);
+    }
+    state_ = kInitialized;
+    return true;
+  }
+
+  if (!disable_recording_) {
+  // AGC should be enabled by default for Voice Processing I/O units but it is
+  // checked below and enabled explicitly if needed. This scheme is used
+  // to be absolutely sure that the AGC is enabled since we have seen cases
+  // where only zeros are recorded and a disabled AGC could be one of the
+  // reasons why it happens.
+  int agc_was_enabled_by_default = 0;
+  UInt32 agc_is_enabled = 0;
+  result = GetAGCState(vpio_unit_, &agc_is_enabled);
+  if (result != noErr) {
+    RTCLogError(@"Failed to get AGC state (1st attempt). "
+                 "Error=%ld.",
+                (long)result);
+    // Example of error code: kAudioUnitErr_NoConnection (-10876).
+    // All error codes related to audio units are negative and are therefore
+    // converted into a postive value to match the UMA APIs.
+    RTC_HISTOGRAM_COUNTS_SPARSE_100000(
+        "WebRTC.Audio.GetAGCStateErrorCode1", (-1) * result);
+  }
+      
+  if (agc_is_enabled) {
+    // Remember that the AGC was enabled by default. Will be used in UMA.
+    agc_was_enabled_by_default = 1;
+  } else {
+    // AGC was initially disabled => try to enable it explicitly.
+    UInt32 enable_agc = 1;
+    result =
+        AudioUnitSetProperty(vpio_unit_,
+                             kAUVoiceIOProperty_VoiceProcessingEnableAGC,
+                             kAudioUnitScope_Global, kInputBus, &enable_agc,
+                             sizeof(enable_agc));
+    if (result != noErr) {
+      RTCLogError(@"Failed to enable the built-in AGC. "
+                   "Error=%ld.",
+                  (long)result);
+      RTC_HISTOGRAM_COUNTS_SPARSE_100000(
+          "WebRTC.Audio.SetAGCStateErrorCode", (-1) * result);
+    }
+    result = GetAGCState(vpio_unit_, &agc_is_enabled);
+    if (result != noErr) {
+      RTCLogError(@"Failed to get AGC state (2nd attempt). "
+                   "Error=%ld.",
+                  (long)result);
+      RTC_HISTOGRAM_COUNTS_SPARSE_100000(
+          "WebRTC.Audio.GetAGCStateErrorCode2", (-1) * result);
+    }
+  }
+
+  // Track if the built-in AGC was enabled by default (as it should) or not.
+  RTC_HISTOGRAM_BOOLEAN("WebRTC.Audio.BuiltInAGCWasEnabledByDefault",
+                        agc_was_enabled_by_default);
+  RTCLog(@"WebRTC.Audio.BuiltInAGCWasEnabledByDefault: %d",
+         agc_was_enabled_by_default);
+  // As a final step, add an UMA histogram for tracking the AGC state.
+  // At this stage, the AGC should be enabled, and if it is not, more work is
+  // needed to find out the root cause.
+  RTC_HISTOGRAM_BOOLEAN("WebRTC.Audio.BuiltInAGCIsEnabled", agc_is_enabled);
+  RTCLog(@"WebRTC.Audio.BuiltInAGCIsEnabled: %u",
+         static_cast<unsigned int>(agc_is_enabled));
+  }
+
+  state_ = kInitialized;
+  return true;
+}
+
+OSStatus VoiceProcessingAudioUnit::Start() {
+  RTC_DCHECK_GE(state_, kUninitialized);
+  RTCLog(@"Starting audio unit.");
+
+  OSStatus result = AudioOutputUnitStart(vpio_unit_);
+  if (result != noErr) {
+    RTCLogError(@"Failed to start audio unit. Error=%ld", (long)result);
+    return result;
+  } else {
+    RTCLog(@"Started audio unit");
+  }
+  state_ = kStarted;
+  return noErr;
+}
+
+bool VoiceProcessingAudioUnit::Stop() {
+  RTC_DCHECK_GE(state_, kUninitialized);
+  RTCLog(@"Stopping audio unit.");
+
+  OSStatus result = AudioOutputUnitStop(vpio_unit_);
+  if (result != noErr) {
+    RTCLogError(@"Failed to stop audio unit. Error=%ld", (long)result);
+    return false;
+  } else {
+    RTCLog(@"Stopped audio unit");
+  }
+
+  state_ = kInitialized;
+  return true;
+}
+
+bool VoiceProcessingAudioUnit::Uninitialize() {
+  RTC_DCHECK_GE(state_, kUninitialized);
+  RTCLog(@"Unintializing audio unit.");
+
+  OSStatus result = AudioUnitUninitialize(vpio_unit_);
+  if (result != noErr) {
+    RTCLogError(@"Failed to uninitialize audio unit. Error=%ld", (long)result);
+    return false;
+  } else {
+    RTCLog(@"Uninitialized audio unit.");
+  }
+
+  state_ = kUninitialized;
+  return true;
+}
+
+void VoiceProcessingAudioUnit::setIsMicrophoneMuted(bool isMuted) {
+    isMicrophoneMuted_ = isMuted;
+    if (enableSystemMute_) {
+        if (vpio_unit_ && state_ == kStarted) {
+            UInt32 muteUplinkOutput = isMuted ? 1 : 0;
+            OSStatus result = AudioUnitSetProperty(vpio_unit_, kAUVoiceIOProperty_MuteOutput, kAudioUnitScope_Global, kInputBus, &muteUplinkOutput, sizeof(muteUplinkOutput));
+            if (result != noErr) {
+                RTCLogError(@"Failed to set kAUVoiceIOProperty_MuteOutput. Error=%ld", (long)result);
+            }
+        }
+    }
+}
+
+OSStatus VoiceProcessingAudioUnit::Render(AudioUnitRenderActionFlags* flags,
+                                          const AudioTimeStamp* time_stamp,
+                                          UInt32 output_bus_number,
+                                          UInt32 num_frames,
+                                          AudioBufferList* io_data) {
+  RTC_DCHECK(vpio_unit_) << "Init() not called.";
+
+  OSStatus result = AudioUnitRender(vpio_unit_, flags, time_stamp,
+                                    output_bus_number, num_frames, io_data);
+  if (result != noErr) {
+    RTCLogError(@"Failed to render audio unit. Error=%ld", (long)result);
+  }
+  return result;
+}
+
+OSStatus VoiceProcessingAudioUnit::OnGetPlayoutData(
+    void* in_ref_con,
+    AudioUnitRenderActionFlags* flags,
+    const AudioTimeStamp* time_stamp,
+    UInt32 bus_number,
+    UInt32 num_frames,
+    AudioBufferList* io_data) {
+  VoiceProcessingAudioUnit* audio_unit =
+      static_cast<VoiceProcessingAudioUnit*>(in_ref_con);
+  return audio_unit->NotifyGetPlayoutData(flags, time_stamp, bus_number,
+                                          num_frames, io_data);
+}
+
+OSStatus VoiceProcessingAudioUnit::OnDeliverRecordedData(
+    void* in_ref_con,
+    AudioUnitRenderActionFlags* flags,
+    const AudioTimeStamp* time_stamp,
+    UInt32 bus_number,
+    UInt32 num_frames,
+    AudioBufferList* io_data) {
+  VoiceProcessingAudioUnit* audio_unit =
+      static_cast<VoiceProcessingAudioUnit*>(in_ref_con);
+  return audio_unit->NotifyDeliverRecordedData(flags, time_stamp, bus_number,
+                                               num_frames, io_data);
+}
+
+OSStatus VoiceProcessingAudioUnit::NotifyGetPlayoutData(
+    AudioUnitRenderActionFlags* flags,
+    const AudioTimeStamp* time_stamp,
+    UInt32 bus_number,
+    UInt32 num_frames,
+    AudioBufferList* io_data) {
+  return observer_->OnGetPlayoutData(flags, time_stamp, bus_number, num_frames,
+                                     io_data);
+}
+
+OSStatus VoiceProcessingAudioUnit::NotifyDeliverRecordedData(
+    AudioUnitRenderActionFlags* flags,
+    const AudioTimeStamp* time_stamp,
+    UInt32 bus_number,
+    UInt32 num_frames,
+    AudioBufferList* io_data) {
+  return observer_->OnDeliverRecordedData(flags, time_stamp, bus_number,
+                                          num_frames, io_data);
+}
+
+AudioStreamBasicDescription VoiceProcessingAudioUnit::GetFormat(
+    Float64 sample_rate, int numChannels) const {
+  // Set the application formats for input and output:
+  // - use same format in both directions
+  // - avoid resampling in the I/O unit by using the hardware sample rate
+  // - linear PCM => noncompressed audio data format with one frame per packet
+  // - no need to specify interleaving since only mono is supported
+  AudioStreamBasicDescription format;
+  format.mSampleRate = sample_rate;
+  format.mFormatID = kAudioFormatLinearPCM;
+  format.mFormatFlags =
+      kLinearPCMFormatFlagIsSignedInteger | kLinearPCMFormatFlagIsPacked | kAudioFormatFlagsNativeEndian;
+  format.mBytesPerPacket = kBytesPerSample * numChannels;
+  format.mFramesPerPacket = 1;  // uncompressed.
+  format.mBytesPerFrame = kBytesPerSample * numChannels;
+  format.mChannelsPerFrame = numChannels;
+  format.mBitsPerChannel = 8 * kBytesPerSample;
+  return format;
+}
+
+void VoiceProcessingAudioUnit::DisposeAudioUnit() {
+  if (vpio_unit_) {
+    switch (state_) {
+      case kStarted:
+        Stop();
+        [[fallthrough]];
+      case kInitialized:
+        Uninitialize();
+        break;
+      case kUninitialized:
+      case kInitRequired:
+        break;
+    }
+
+    RTCLog(@"Disposing audio unit.");
+    OSStatus result = AudioComponentInstanceDispose(vpio_unit_);
+    if (result != noErr) {
+      RTCLogError(@"AudioComponentInstanceDispose failed. Error=%ld.",
+                  (long)result);
+    }
+    vpio_unit_ = nullptr;
+  }
+}
+
+}  // namespace tgcalls_ios_adm
+}  // namespace webrtc