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Merge pull request 'feature/calls-maket' (#5) from feature/calls-maket into xmpp-lib

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Reviewed-on: #5
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fmodf 2024-01-28 07:30:10 +00:00
parent b1604ab115 0f5019252b
commit ed82109382
898 changed files with 22239 additions and 61134 deletions
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3
.gitignore vendored
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/old/.idea
/Engine/Info.plist
/Snikket/Snikket.entitlements
/XMPPSwift/Client/VoIP/rickroll.mp4
/.nvim
/buildServer.json

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Lib/WebRTC.xcframework/Info.plist Normal file
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<?xml version="1.0" encoding="UTF-8"?>
<!DOCTYPE plist PUBLIC "-//Apple//DTD PLIST 1.0//EN" "http://www.apple.com/DTDs/PropertyList-1.0.dtd">
<plist version="1.0">
<dict>
<key>AvailableLibraries</key>
<array>
<dict>
<key>LibraryIdentifier</key>
<string>ios-arm64</string>
<key>LibraryPath</key>
<string>WebRTC.framework</string>
<key>SupportedArchitectures</key>
<array>
<string>arm64</string>
</array>
<key>SupportedPlatform</key>
<string>ios</string>
</dict>
<dict>
<key>LibraryIdentifier</key>
<string>ios-x86_64_arm64-simulator</string>
<key>LibraryPath</key>
<string>WebRTC.framework</string>
<key>SupportedArchitectures</key>
<array>
<string>arm64</string>
<string>x86_64</string>
</array>
<key>SupportedPlatform</key>
<string>ios</string>
<key>SupportedPlatformVariant</key>
<string>simulator</string>
</dict>
<dict>
<key>LibraryIdentifier</key>
<string>macos-x86_64_arm64</string>
<key>LibraryPath</key>
<string>WebRTC.framework</string>
<key>SupportedArchitectures</key>
<array>
<string>x86_64</string>
<string>arm64</string>
</array>
<key>SupportedPlatform</key>
<string>macos</string>
</dict>
<dict>
<key>LibraryIdentifier</key>
<string>ios-x86_64_arm64-maccatalyst</string>
<key>LibraryPath</key>
<string>WebRTC.framework</string>
<key>SupportedArchitectures</key>
<array>
<string>x86_64</string>
<string>arm64</string>
</array>
<key>SupportedPlatform</key>
<string>ios</string>
<key>SupportedPlatformVariant</key>
<string>maccatalyst</string>
</dict>
</array>
<key>CFBundlePackageType</key>
<string>XFWK</string>
<key>XCFrameworkFormatVersion</key>
<string>1.0</string>
</dict>
</plist>

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Lib/WebRTC.xcframework/LICENSE Normal file
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Copyright (c) 2011, The WebRTC project authors. All rights reserved.
Redistribution and use in source and binary forms, with or without
modification, are permitted provided that the following conditions are
met:
* Redistributions of source code must retain the above copyright
notice, this list of conditions and the following disclaimer.
* Redistributions in binary form must reproduce the above copyright
notice, this list of conditions and the following disclaimer in
the documentation and/or other materials provided with the
distribution.
* Neither the name of Google nor the names of its contributors may
be used to endorse or promote products derived from this software
without specific prior written permission.
THIS SOFTWARE IS PROVIDED BY THE COPYRIGHT HOLDERS AND CONTRIBUTORS
"AS IS" AND ANY EXPRESS OR IMPLIED WARRANTIES, INCLUDING, BUT NOT
LIMITED TO, THE IMPLIED WARRANTIES OF MERCHANTABILITY AND FITNESS FOR
A PARTICULAR PURPOSE ARE DISCLAIMED. IN NO EVENT SHALL THE COPYRIGHT
HOLDER OR CONTRIBUTORS BE LIABLE FOR ANY DIRECT, INDIRECT, INCIDENTAL,
SPECIAL, EXEMPLARY, OR CONSEQUENTIAL DAMAGES (INCLUDING, BUT NOT
LIMITED TO, PROCUREMENT OF SUBSTITUTE GOODS OR SERVICES; LOSS OF USE,
DATA, OR PROFITS; OR BUSINESS INTERRUPTION) HOWEVER CAUSED AND ON ANY
THEORY OF LIABILITY, WHETHER IN CONTRACT, STRICT LIABILITY, OR TORT
(INCLUDING NEGLIGENCE OR OTHERWISE) ARISING IN ANY WAY OUT OF THE USE
OF THIS SOFTWARE, EVEN IF ADVISED OF THE POSSIBILITY OF SUCH DAMAGE.

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Lib/WebRTC.xcframework/ios-arm64/WebRTC.framework/Headers/RTCAudioDevice.h Normal file
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/*
* Copyright 2022 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 <AudioUnit/AudioUnit.h>
#import <Foundation/Foundation.h>
#import <WebRTC/RTCMacros.h>
NS_ASSUME_NONNULL_BEGIN
typedef OSStatus (^RTC_OBJC_TYPE(RTCAudioDeviceGetPlayoutDataBlock))(
AudioUnitRenderActionFlags *_Nonnull actionFlags,
const AudioTimeStamp *_Nonnull timestamp,
NSInteger inputBusNumber,
UInt32 frameCount,
AudioBufferList *_Nonnull outputData);
typedef OSStatus (^RTC_OBJC_TYPE(RTCAudioDeviceRenderRecordedDataBlock))(
AudioUnitRenderActionFlags *_Nonnull actionFlags,
const AudioTimeStamp *_Nonnull timestamp,
NSInteger inputBusNumber,
UInt32 frameCount,
AudioBufferList *_Nonnull inputData,
void *_Nullable renderContext);
typedef OSStatus (^RTC_OBJC_TYPE(RTCAudioDeviceDeliverRecordedDataBlock))(
AudioUnitRenderActionFlags *_Nonnull actionFlags,
const AudioTimeStamp *_Nonnull timestamp,
NSInteger inputBusNumber,
UInt32 frameCount,
const AudioBufferList *_Nullable inputData,
void *_Nullable renderContext,
NS_NOESCAPE RTC_OBJC_TYPE(RTCAudioDeviceRenderRecordedDataBlock) _Nullable renderBlock);
/**
* Delegate object provided by native ADM during RTCAudioDevice initialization.
* Provides blocks to poll playback audio samples from native ADM and to feed
* recorded audio samples into native ADM.
*/
RTC_OBJC_EXPORT @protocol RTC_OBJC_TYPE
(RTCAudioDeviceDelegate)<NSObject>
/**
* Implementation of RTCAudioSource should call this block to feed recorded PCM (16-bit integer)
* into native ADM. Stereo data is expected to be interleaved starting with the left channel.
* Either `inputData` with pre-filled audio data must be provided during block
* call or `renderBlock` must be provided which must fill provided audio buffer with recorded
* samples.
*
* NOTE: Implementation of RTCAudioDevice is expected to call the block on the same thread until
* `notifyAudioInterrupted` is called. When `notifyAudioInterrupted` is called implementation
* can call the block on a different thread.
*/
@property(readonly, nonnull)
RTC_OBJC_TYPE(RTCAudioDeviceDeliverRecordedDataBlock) deliverRecordedData;
/**
* Provides input sample rate preference as it preferred by native ADM.
*/
@property(readonly) double preferredInputSampleRate;
/**
* Provides input IO buffer duration preference as it preferred by native ADM.
*/
@property(readonly) NSTimeInterval preferredInputIOBufferDuration;
/**
* Provides output sample rate preference as it preferred by native ADM.
*/
@property(readonly) double preferredOutputSampleRate;
/**
* Provides output IO buffer duration preference as it preferred by native ADM.
*/
@property(readonly) NSTimeInterval preferredOutputIOBufferDuration;
/**
* Implementation of RTCAudioDevice should call this block to request PCM (16-bit integer)
* from native ADM to play. Stereo data is interleaved starting with the left channel.
*
* NOTE: Implementation of RTCAudioDevice is expected to invoke of this block on the
* same thread until `notifyAudioInterrupted` is called. When `notifyAudioInterrupted` is called
* implementation can call the block from a different thread.
*/
@property(readonly, nonnull) RTC_OBJC_TYPE(RTCAudioDeviceGetPlayoutDataBlock) getPlayoutData;
/**
* Notifies native ADM that some of the audio input parameters of RTCAudioDevice like
* samle rate and/or IO buffer duration and/or IO latency had possibly changed.
* Native ADM will adjust its audio input buffer to match current parameters of audio device.
*
* NOTE: Must be called within block executed via `dispatchAsync` or `dispatchSync`.
*/
- (void)notifyAudioInputParametersChange;
/**
* Notifies native ADM that some of the audio output parameters of RTCAudioDevice like
* samle rate and/or IO buffer duration and/or IO latency had possibly changed.
* Native ADM will adjust its audio output buffer to match current parameters of audio device.
*
* NOTE: Must be called within block executed via `dispatchAsync` or `dispatchSync`.
*/
- (void)notifyAudioOutputParametersChange;
/**
* Notifies native ADM that audio input is interrupted and further audio playout
* and recording might happen on a different thread.
*
* NOTE: Must be called within block executed via `dispatchAsync` or `dispatchSync`.
*/
- (void)notifyAudioInputInterrupted;
/**
* Notifies native ADM that audio output is interrupted and further audio playout
* and recording might happen on a different thread.
*
* NOTE: Must be called within block executed via `dispatchAsync` or `dispatchSync`.
*/
- (void)notifyAudioOutputInterrupted;
/**
* Asynchronously execute block of code within the context of
* thread which owns native ADM.
*
* NOTE: Intended to be used to invoke `notifyAudioInputParametersChange`,
* `notifyAudioOutputParametersChange`, `notifyAudioInputInterrupted`,
* `notifyAudioOutputInterrupted` on native ADM thread.
* Also could be used by `RTCAudioDevice` implementation to tie
* mutations of underlying audio objects (AVAudioEngine, AudioUnit, etc)
* to the native ADM thread. Could be useful to handle events like audio route change, which
* could lead to audio parameters change.
*/
- (void)dispatchAsync:(dispatch_block_t)block;
/**
* Synchronously execute block of code within the context of
* thread which owns native ADM. Allows reentrancy.
*
* NOTE: Intended to be used to invoke `notifyAudioInputParametersChange`,
* `notifyAudioOutputParametersChange`, `notifyAudioInputInterrupted`,
* `notifyAudioOutputInterrupted` on native ADM thread and make sure
* aforementioned is completed before `dispatchSync` returns. Could be useful
* when implementation of `RTCAudioDevice` tie mutation to underlying audio objects (AVAudioEngine,
* AudioUnit, etc) to own thread to satisfy requirement that native ADM audio parameters
* must be kept in sync with current audio parameters before audio is actually played or recorded.
*/
- (void)dispatchSync:(dispatch_block_t)block;
@end
/**
* Protocol to abstract platform specific ways to implement playback and recording.
*
* NOTE: All the members of protocol are called by native ADM from the same thread
* between calls to `initializeWithDelegate` and `terminate`.
* NOTE: Implementation is fully responsible for configuring application's AVAudioSession.
* An example implementation of RTCAudioDevice: https://github.com/mstyura/RTCAudioDevice
* TODO(yura.yaroshevich): Implement custom RTCAudioDevice for AppRTCMobile demo app.
*/
RTC_OBJC_EXPORT @protocol RTC_OBJC_TYPE
(RTCAudioDevice)<NSObject>
/**
* Indicates current sample rate of audio recording. Changes to this property
* must be notified back to native ADM via `-[RTCAudioDeviceDelegate
* notifyAudioParametersChange]`.
*/
@property(readonly) double deviceInputSampleRate;
/**
* Indicates current size of record buffer. Changes to this property
* must be notified back to native ADM via `-[RTCAudioDeviceDelegate notifyAudioParametersChange]`.
*/
@property(readonly) NSTimeInterval inputIOBufferDuration;
/**
* Indicates current number of recorded audio channels. Changes to this property
* must be notified back to native ADM via `-[RTCAudioDeviceDelegate notifyAudioParametersChange]`.
*/
@property(readonly) NSInteger inputNumberOfChannels;
/**
* Indicates current input latency
*/
@property(readonly) NSTimeInterval inputLatency;
/**
* Indicates current sample rate of audio playback. Changes to this property
* must be notified back to native ADM via `-[RTCAudioDeviceDelegate notifyAudioParametersChange]`.
*/
@property(readonly) double deviceOutputSampleRate;
/**
* Indicates current size of playback buffer. Changes to this property
* must be notified back to native ADM via `-[RTCAudioDeviceDelegate notifyAudioParametersChange]`.
*/
@property(readonly) NSTimeInterval outputIOBufferDuration;
/**
* Indicates current number of playback audio channels. Changes to this property
* must be notified back to WebRTC via `[RTCAudioDeviceDelegate notifyAudioParametersChange]`.
*/
@property(readonly) NSInteger outputNumberOfChannels;
/**
* Indicates current output latency
*/
@property(readonly) NSTimeInterval outputLatency;
/**
* Indicates if invocation of `initializeWithDelegate` required before usage of RTCAudioDevice.
* YES indicates that `initializeWithDelegate` was called earlier without subsequent call to
* `terminate`. NO indicates that either `initializeWithDelegate` not called or `terminate` called.
*/
@property(readonly) BOOL isInitialized;
/**
* Initializes RTCAudioDevice with RTCAudioDeviceDelegate.
* Implementation must return YES if RTCAudioDevice initialized successfully and NO otherwise.
*/
- (BOOL)initializeWithDelegate:(id<RTC_OBJC_TYPE(RTCAudioDeviceDelegate)>)delegate;
/**
* De-initializes RTCAudioDevice. Implementation should forget about `delegate` provided in
* `initializeWithDelegate`.
*/
- (BOOL)terminateDevice;
/**
* Property to indicate if `initializePlayout` call required before invocation of `startPlayout`.
* YES indicates that `initializePlayout` was successfully invoked earlier or not necessary,
* NO indicates that `initializePlayout` invocation required.
*/
@property(readonly) BOOL isPlayoutInitialized;
/**
* Prepares RTCAudioDevice to play audio.
* Called by native ADM before invocation of `startPlayout`.
* Implementation is expected to return YES in case of successful playout initialization and NO
* otherwise.
*/
- (BOOL)initializePlayout;
/**
* Property to indicate if RTCAudioDevice should be playing according to
* earlier calls of `startPlayout` and `stopPlayout`.
*/
@property(readonly) BOOL isPlaying;
/**
* Method is called when native ADM wants to play audio.
* Implementation is expected to return YES if playback start request
* successfully handled and NO otherwise.
*/
- (BOOL)startPlayout;
/**
* Method is called when native ADM no longer needs to play audio.
* Implementation is expected to return YES if playback stop request
* successfully handled and NO otherwise.
*/
- (BOOL)stopPlayout;
/**
* Property to indicate if `initializeRecording` call required before usage of `startRecording`.
* YES indicates that `initializeRecording` was successfully invoked earlier or not necessary,
* NO indicates that `initializeRecording` invocation required.
*/
@property(readonly) BOOL isRecordingInitialized;
/**
* Prepares RTCAudioDevice to record audio.
* Called by native ADM before invocation of `startRecording`.
* Implementation may use this method to prepare resources required to record audio.
* Implementation is expected to return YES in case of successful record initialization and NO
* otherwise.
*/
- (BOOL)initializeRecording;
/**
* Property to indicate if RTCAudioDevice should record audio according to
* earlier calls to `startRecording` and `stopRecording`.
*/
@property(readonly) BOOL isRecording;
/**
* Method is called when native ADM wants to record audio.
* Implementation is expected to return YES if recording start request
* successfully handled and NO otherwise.
*/
- (BOOL)startRecording;
/**
* Method is called when native ADM no longer needs to record audio.
* Implementation is expected to return YES if recording stop request
* successfully handled and NO otherwise.
*/
- (BOOL)stopRecording;
@end
NS_ASSUME_NONNULL_END

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Lib/WebRTC.xcframework/ios-arm64/WebRTC.framework/Headers/RTCAudioSession.h Normal file
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/*
* 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 <WebRTC/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:(AVAudioSessionCategory)category
mode:(AVAudioSessionMode)mode
options:(AVAudioSessionCategoryOptions)options
error:(NSError **)outError;
- (BOOL)setCategory:(AVAudioSessionCategory)category
withOptions:(AVAudioSessionCategoryOptions)options
error:(NSError **)outError;
- (BOOL)setMode:(AVAudioSessionMode)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;
@end
NS_ASSUME_NONNULL_END

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/*
* 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 <WebRTC/RTCMacros.h>
NS_ASSUME_NONNULL_BEGIN
RTC_EXTERN const int kRTCAudioSessionPreferredNumberOfChannels;
RTC_EXTERN const double kRTCAudioSessionHighPerformanceSampleRate;
RTC_EXTERN const double kRTCAudioSessionHighPerformanceIOBufferDuration;
// 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

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/*
* 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 <Foundation/Foundation.h>
#import <WebRTC/RTCMacros.h>
#import <WebRTC/RTCMediaSource.h>
NS_ASSUME_NONNULL_BEGIN
RTC_OBJC_EXPORT
@interface RTC_OBJC_TYPE (RTCAudioSource) : RTC_OBJC_TYPE(RTCMediaSource)
- (instancetype)init NS_UNAVAILABLE;
// Sets the volume for the RTCMediaSource. `volume` is a gain value in the range
// [0, 10].
// Temporary fix to be able to modify volume of remote audio tracks.
// TODO(kthelgason): Property stays here temporarily until a proper volume-api
// is available on the surface exposed by webrtc.
@property(nonatomic, assign) double volume;
@end
NS_ASSUME_NONNULL_END

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Lib/WebRTC.xcframework/ios-arm64/WebRTC.framework/Headers/RTCAudioTrack.h Normal file
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/*
* Copyright 2015 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 <WebRTC/RTCMacros.h>
#import <WebRTC/RTCMediaStreamTrack.h>
NS_ASSUME_NONNULL_BEGIN
@class RTC_OBJC_TYPE(RTCAudioSource);
RTC_OBJC_EXPORT
@interface RTC_OBJC_TYPE (RTCAudioTrack) : RTC_OBJC_TYPE(RTCMediaStreamTrack)
- (instancetype)init NS_UNAVAILABLE;
/** The audio source for this audio track. */
@property(nonatomic, readonly) RTC_OBJC_TYPE(RTCAudioSource) * source;
@end
NS_ASSUME_NONNULL_END

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/*
* 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 <AVFoundation/AVFoundation.h>
#import <WebRTC/RTCMacros.h>
#import <WebRTC/RTCVideoFrameBuffer.h>
NS_ASSUME_NONNULL_BEGIN
/** RTCVideoFrameBuffer containing a CVPixelBufferRef */
RTC_OBJC_EXPORT
@interface RTC_OBJC_TYPE (RTCCVPixelBuffer) : NSObject <RTC_OBJC_TYPE(RTCVideoFrameBuffer)>
@property(nonatomic, readonly) CVPixelBufferRef pixelBuffer;
@property(nonatomic, readonly) int cropX;
@property(nonatomic, readonly) int cropY;
@property(nonatomic, readonly) int cropWidth;
@property(nonatomic, readonly) int cropHeight;
+ (NSSet<NSNumber *> *)supportedPixelFormats;
- (instancetype)initWithPixelBuffer:(CVPixelBufferRef)pixelBuffer;
- (instancetype)initWithPixelBuffer:(CVPixelBufferRef)pixelBuffer
adaptedWidth:(int)adaptedWidth
adaptedHeight:(int)adaptedHeight
cropWidth:(int)cropWidth
cropHeight:(int)cropHeight
cropX:(int)cropX
cropY:(int)cropY;
- (BOOL)requiresCropping;
- (BOOL)requiresScalingToWidth:(int)width height:(int)height;
- (int)bufferSizeForCroppingAndScalingToWidth:(int)width height:(int)height;
/** The minimum size of the `tmpBuffer` must be the number of bytes returned from the
* bufferSizeForCroppingAndScalingToWidth:height: method.
* If that size is 0, the `tmpBuffer` may be nil.
*/
- (BOOL)cropAndScaleTo:(CVPixelBufferRef)outputPixelBuffer
withTempBuffer:(nullable uint8_t *)tmpBuffer;
@end
NS_ASSUME_NONNULL_END

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/*
* 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 <Foundation/Foundation.h>
#import <WebRTC/RTCLogging.h>
#import <WebRTC/RTCMacros.h>
NS_ASSUME_NONNULL_BEGIN
typedef void (^RTCCallbackLoggerMessageHandler)(NSString *message);
typedef void (^RTCCallbackLoggerMessageAndSeverityHandler)(NSString *message,
RTCLoggingSeverity severity);
// This class intercepts WebRTC logs and forwards them to a registered block.
// This class is not threadsafe.
RTC_OBJC_EXPORT
@interface RTC_OBJC_TYPE (RTCCallbackLogger) : NSObject
// The severity level to capture. The default is kRTCLoggingSeverityInfo.
@property(nonatomic, assign) RTCLoggingSeverity severity;
// The callback handler will be called on the same thread that does the
// logging, so if the logging callback can be slow it may be a good idea
// to implement dispatching to some other queue.
- (void)start:(nullable RTCCallbackLoggerMessageHandler)handler;
- (void)startWithMessageAndSeverityHandler:
(nullable RTCCallbackLoggerMessageAndSeverityHandler)handler;
- (void)stop;
@end
NS_ASSUME_NONNULL_END

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/*
* Copyright 2015 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 <Foundation/Foundation.h>
#import <UIKit/UIKit.h>
#import <WebRTC/RTCMacros.h>
@class AVCaptureSession;
/** RTCCameraPreviewView is a view that renders local video from an
* AVCaptureSession.
*/
RTC_OBJC_EXPORT
@interface RTC_OBJC_TYPE (RTCCameraPreviewView) : UIView
/** The capture session being rendered in the view. Capture session
* is assigned to AVCaptureVideoPreviewLayer async in the same
* queue that the AVCaptureSession is started/stopped.
*/
@property(nonatomic, strong) AVCaptureSession* captureSession;
@end

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/*
* Copyright 2017 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 <WebRTC/RTCMacros.h>
#import <WebRTC/RTCVideoCapturer.h>
NS_ASSUME_NONNULL_BEGIN
RTC_OBJC_EXPORT
// Camera capture that implements RTCVideoCapturer. Delivers frames to a
// RTCVideoCapturerDelegate (usually RTCVideoSource).
NS_EXTENSION_UNAVAILABLE_IOS("Camera not available in app extensions.")
@interface RTC_OBJC_TYPE (RTCCameraVideoCapturer) : RTC_OBJC_TYPE(RTCVideoCapturer)
// Capture session that is used for capturing. Valid from initialization to dealloc.
@property(readonly, nonatomic) AVCaptureSession *captureSession;
// Returns list of available capture devices that support video capture.
+ (NSArray<AVCaptureDevice *> *)captureDevices;
// Returns list of formats that are supported by this class for this device.
+ (NSArray<AVCaptureDeviceFormat *> *)supportedFormatsForDevice:(AVCaptureDevice *)device;
// Returns the most efficient supported output pixel format for this capturer.
- (FourCharCode)preferredOutputPixelFormat;
// Starts the capture session asynchronously and notifies callback on completion.
// The device will capture video in the format given in the `format` parameter. If the pixel format
// in `format` is supported by the WebRTC pipeline, the same pixel format will be used for the
// output. Otherwise, the format returned by `preferredOutputPixelFormat` will be used.
- (void)startCaptureWithDevice:(AVCaptureDevice *)device
format:(AVCaptureDeviceFormat *)format
fps:(NSInteger)fps
completionHandler:(nullable void (^)(NSError *_Nullable))completionHandler;
// Stops the capture session asynchronously and notifies callback on completion.
- (void)stopCaptureWithCompletionHandler:(nullable void (^)(void))completionHandler;
// Starts the capture session asynchronously.
- (void)startCaptureWithDevice:(AVCaptureDevice *)device
format:(AVCaptureDeviceFormat *)format
fps:(NSInteger)fps;
// Stops the capture session asynchronously.
- (void)stopCapture;
@end
NS_ASSUME_NONNULL_END

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/*
* 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 <Foundation/Foundation.h>
#import <WebRTC/RTCMacros.h>
NS_ASSUME_NONNULL_BEGIN
RTC_OBJC_EXPORT
@interface RTC_OBJC_TYPE (RTCCertificate) : NSObject <NSCopying>
/** Private key in PEM. */
@property(nonatomic, readonly, copy) NSString *private_key;
/** Public key in an x509 cert encoded in PEM. */
@property(nonatomic, readonly, copy) NSString *certificate;
/**
* Initialize an RTCCertificate with PEM strings for private_key and certificate.
*/
- (instancetype)initWithPrivateKey:(NSString *)private_key
certificate:(NSString *)certificate NS_DESIGNATED_INITIALIZER;
- (instancetype)init NS_UNAVAILABLE;
/** Generate a new certificate for 're' use.
*
* Optional dictionary of parameters. Defaults to KeyType ECDSA if none are
* provided.
* - name: "ECDSA" or "RSASSA-PKCS1-v1_5"
*/
+ (nullable RTC_OBJC_TYPE(RTCCertificate) *)generateCertificateWithParams:(NSDictionary *)params;
@end
NS_ASSUME_NONNULL_END

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Lib/WebRTC.xcframework/ios-arm64/WebRTC.framework/Headers/RTCCodecSpecificInfo.h Normal file
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/*
* Copyright 2017 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 <Foundation/Foundation.h>
#import <WebRTC/RTCMacros.h>
NS_ASSUME_NONNULL_BEGIN
/** Implement this protocol to pass codec specific info from the encoder.
* Corresponds to webrtc::CodecSpecificInfo.
*/
RTC_OBJC_EXPORT
@protocol RTC_OBJC_TYPE
(RTCCodecSpecificInfo)<NSObject> @end
NS_ASSUME_NONNULL_END

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/*
* Copyright 2017 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 <Foundation/Foundation.h>
#import <WebRTC/RTCCodecSpecificInfo.h>
#import <WebRTC/RTCMacros.h>
/** Class for H264 specific config. */
typedef NS_ENUM(NSUInteger, RTCH264PacketizationMode) {
RTCH264PacketizationModeNonInterleaved = 0, // Mode 1 - STAP-A, FU-A is allowed
RTCH264PacketizationModeSingleNalUnit // Mode 0 - only single NALU allowed
};
RTC_OBJC_EXPORT
@interface RTC_OBJC_TYPE (RTCCodecSpecificInfoH264) : NSObject <RTC_OBJC_TYPE(RTCCodecSpecificInfo)>
@property(nonatomic, assign) RTCH264PacketizationMode packetizationMode;
@end

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/*
* Copyright 2015 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 <Foundation/Foundation.h>
#import <WebRTC/RTCCertificate.h>
#import <WebRTC/RTCCryptoOptions.h>
#import <WebRTC/RTCMacros.h>
@class RTC_OBJC_TYPE(RTCIceServer);
/**
* Represents the ice transport policy. This exposes the same states in C++,
* which include one more state than what exists in the W3C spec.
*/
typedef NS_ENUM(NSInteger, RTCIceTransportPolicy) {
RTCIceTransportPolicyNone,
RTCIceTransportPolicyRelay,
RTCIceTransportPolicyNoHost,
RTCIceTransportPolicyAll
};
/** Represents the bundle policy. */
typedef NS_ENUM(NSInteger, RTCBundlePolicy) {
RTCBundlePolicyBalanced,
RTCBundlePolicyMaxCompat,
RTCBundlePolicyMaxBundle
};
/** Represents the rtcp mux policy. */
typedef NS_ENUM(NSInteger, RTCRtcpMuxPolicy) { RTCRtcpMuxPolicyNegotiate, RTCRtcpMuxPolicyRequire };
/** Represents the tcp candidate policy. */
typedef NS_ENUM(NSInteger, RTCTcpCandidatePolicy) {
RTCTcpCandidatePolicyEnabled,
RTCTcpCandidatePolicyDisabled
};
/** Represents the candidate network policy. */
typedef NS_ENUM(NSInteger, RTCCandidateNetworkPolicy) {
RTCCandidateNetworkPolicyAll,
RTCCandidateNetworkPolicyLowCost
};
/** Represents the continual gathering policy. */
typedef NS_ENUM(NSInteger, RTCContinualGatheringPolicy) {
RTCContinualGatheringPolicyGatherOnce,
RTCContinualGatheringPolicyGatherContinually
};
/** Represents the encryption key type. */
typedef NS_ENUM(NSInteger, RTCEncryptionKeyType) {
RTCEncryptionKeyTypeRSA,
RTCEncryptionKeyTypeECDSA,
};
/** Represents the chosen SDP semantics for the RTCPeerConnection. */
typedef NS_ENUM(NSInteger, RTCSdpSemantics) {
// TODO(https://crbug.com/webrtc/13528): Remove support for Plan B.
RTCSdpSemanticsPlanB,
RTCSdpSemanticsUnifiedPlan,
};
NS_ASSUME_NONNULL_BEGIN
RTC_OBJC_EXPORT
@interface RTC_OBJC_TYPE (RTCConfiguration) : NSObject
/** If true, allows DSCP codes to be set on outgoing packets, configured using
* networkPriority field of RTCRtpEncodingParameters. Defaults to false.
*/
@property(nonatomic, assign) BOOL enableDscp;
/** An array of Ice Servers available to be used by ICE. */
@property(nonatomic, copy) NSArray<RTC_OBJC_TYPE(RTCIceServer) *> *iceServers;
/** An RTCCertificate for 're' use. */
@property(nonatomic, nullable) RTC_OBJC_TYPE(RTCCertificate) * certificate;
/** Which candidates the ICE agent is allowed to use. The W3C calls it
* `iceTransportPolicy`, while in C++ it is called `type`. */
@property(nonatomic, assign) RTCIceTransportPolicy iceTransportPolicy;
/** The media-bundling policy to use when gathering ICE candidates. */
@property(nonatomic, assign) RTCBundlePolicy bundlePolicy;
/** The rtcp-mux policy to use when gathering ICE candidates. */
@property(nonatomic, assign) RTCRtcpMuxPolicy rtcpMuxPolicy;
@property(nonatomic, assign) RTCTcpCandidatePolicy tcpCandidatePolicy;
@property(nonatomic, assign) RTCCandidateNetworkPolicy candidateNetworkPolicy;
@property(nonatomic, assign) RTCContinualGatheringPolicy continualGatheringPolicy;
/** If set to YES, don't gather IPv6 ICE candidates on Wi-Fi.
* Only intended to be used on specific devices. Certain phones disable IPv6
* when the screen is turned off and it would be better to just disable the
* IPv6 ICE candidates on Wi-Fi in those cases.
* Default is NO.
*/
@property(nonatomic, assign) BOOL disableIPV6OnWiFi;
/** By default, the PeerConnection will use a limited number of IPv6 network
* interfaces, in order to avoid too many ICE candidate pairs being created
* and delaying ICE completion.
*
* Can be set to INT_MAX to effectively disable the limit.
*/
@property(nonatomic, assign) int maxIPv6Networks;
/** Exclude link-local network interfaces
* from considertaion for gathering ICE candidates.
* Defaults to NO.
*/
@property(nonatomic, assign) BOOL disableLinkLocalNetworks;
@property(nonatomic, assign) int audioJitterBufferMaxPackets;
@property(nonatomic, assign) BOOL audioJitterBufferFastAccelerate;
@property(nonatomic, assign) int iceConnectionReceivingTimeout;
@property(nonatomic, assign) int iceBackupCandidatePairPingInterval;
/** Key type used to generate SSL identity. Default is ECDSA. */
@property(nonatomic, assign) RTCEncryptionKeyType keyType;
/** ICE candidate pool size as defined in JSEP. Default is 0. */
@property(nonatomic, assign) int iceCandidatePoolSize;
/** Prune turn ports on the same network to the same turn server.
* Default is NO.
*/
@property(nonatomic, assign) BOOL shouldPruneTurnPorts;
/** If set to YES, this means the ICE transport should presume TURN-to-TURN
* candidate pairs will succeed, even before a binding response is received.
*/
@property(nonatomic, assign) BOOL shouldPresumeWritableWhenFullyRelayed;
/* This flag is only effective when `continualGatheringPolicy` is
* RTCContinualGatheringPolicyGatherContinually.
*
* If YES, after the ICE transport type is changed such that new types of
* ICE candidates are allowed by the new transport type, e.g. from
* RTCIceTransportPolicyRelay to RTCIceTransportPolicyAll, candidates that
* have been gathered by the ICE transport but not matching the previous
* transport type and as a result not observed by PeerConnectionDelegateAdapter,
* will be surfaced to the delegate.
*/
@property(nonatomic, assign) BOOL shouldSurfaceIceCandidatesOnIceTransportTypeChanged;
/** If set to non-nil, controls the minimal interval between consecutive ICE
* check packets.
*/
@property(nonatomic, copy, nullable) NSNumber *iceCheckMinInterval;
/**
* Configure the SDP semantics used by this PeerConnection. By default, this
* is RTCSdpSemanticsUnifiedPlan which is compliant to the WebRTC 1.0
* specification. It is possible to overrwite this to the deprecated
* RTCSdpSemanticsPlanB SDP format, but note that RTCSdpSemanticsPlanB will be
* deleted at some future date, see https://crbug.com/webrtc/13528.
*
* RTCSdpSemanticsUnifiedPlan will cause RTCPeerConnection to create offers and
* answers with multiple m= sections where each m= section maps to one
* RTCRtpSender and one RTCRtpReceiver (an RTCRtpTransceiver), either both audio
* or both video. This will also cause RTCPeerConnection to ignore all but the
* first a=ssrc lines that form a Plan B stream.
*
* RTCSdpSemanticsPlanB will cause RTCPeerConnection to create offers and
* answers with at most one audio and one video m= section with multiple
* RTCRtpSenders and RTCRtpReceivers specified as multiple a=ssrc lines within
* the section. This will also cause RTCPeerConnection to ignore all but the
* first m= section of the same media type.
*/
@property(nonatomic, assign) RTCSdpSemantics sdpSemantics;
/** Actively reset the SRTP parameters when the DTLS transports underneath are
* changed after offer/answer negotiation. This is only intended to be a
* workaround for crbug.com/835958
*/
@property(nonatomic, assign) BOOL activeResetSrtpParams;
/** If the remote side support mid-stream codec switches then allow encoder
* switching to be performed.
*/
@property(nonatomic, assign) BOOL allowCodecSwitching;
/**
* Defines advanced optional cryptographic settings related to SRTP and
* frame encryption for native WebRTC. Setting this will overwrite any
* options set through the PeerConnectionFactory (which is deprecated).
*/
@property(nonatomic, nullable) RTC_OBJC_TYPE(RTCCryptoOptions) * cryptoOptions;
/**
* An optional string that will be attached to the TURN_ALLOCATE_REQUEST which
* which can be used to correlate client logs with backend logs.
*/
@property(nonatomic, nullable, copy) NSString *turnLoggingId;
/**
* Time interval between audio RTCP reports.
*/
@property(nonatomic, assign) int rtcpAudioReportIntervalMs;
/**
* Time interval between video RTCP reports.
*/
@property(nonatomic, assign) int rtcpVideoReportIntervalMs;
/**
* Allow implicit rollback of local description when remote description
* conflicts with local description.
* See: https://w3c.github.io/webrtc-pc/#dom-peerconnection-setremotedescription
*/
@property(nonatomic, assign) BOOL enableImplicitRollback;
/**
* Control if "a=extmap-allow-mixed" is included in the offer.
* See: https://www.chromestatus.com/feature/6269234631933952
*/
@property(nonatomic, assign) BOOL offerExtmapAllowMixed;
/**
* Defines the interval applied to ALL candidate pairs
* when ICE is strongly connected, and it overrides the
* default value of this interval in the ICE implementation;
*/
@property(nonatomic, copy, nullable) NSNumber *iceCheckIntervalStrongConnectivity;
/**
* Defines the counterpart for ALL pairs when ICE is
* weakly connected, and it overrides the default value of
* this interval in the ICE implementation
*/
@property(nonatomic, copy, nullable) NSNumber *iceCheckIntervalWeakConnectivity;
/**
* The min time period for which a candidate pair must wait for response to
* connectivity checks before it becomes unwritable. This parameter
* overrides the default value in the ICE implementation if set.
*/
@property(nonatomic, copy, nullable) NSNumber *iceUnwritableTimeout;
/**
* The min number of connectivity checks that a candidate pair must sent
* without receiving response before it becomes unwritable. This parameter
* overrides the default value in the ICE implementation if set.
*/
@property(nonatomic, copy, nullable) NSNumber *iceUnwritableMinChecks;
/**
* The min time period for which a candidate pair must wait for response to
* connectivity checks it becomes inactive. This parameter overrides the
* default value in the ICE implementation if set.
*/
@property(nonatomic, copy, nullable) NSNumber *iceInactiveTimeout;
- (instancetype)init;
@end
NS_ASSUME_NONNULL_END

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/*
* 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 <Foundation/Foundation.h>
#import <WebRTC/RTCMacros.h>
NS_ASSUME_NONNULL_BEGIN
/**
* Objective-C bindings for webrtc::CryptoOptions. This API had to be flattened
* as Objective-C doesn't support nested structures.
*/
RTC_OBJC_EXPORT
@interface RTC_OBJC_TYPE (RTCCryptoOptions) : NSObject
/**
* Enable GCM crypto suites from RFC 7714 for SRTP. GCM will only be used
* if both sides enable it
*/
@property(nonatomic, assign) BOOL srtpEnableGcmCryptoSuites;
/**
* If set to true, the (potentially insecure) crypto cipher
* kSrtpAes128CmSha1_32 will be included in the list of supported ciphers
* during negotiation. It will only be used if both peers support it and no
* other ciphers get preferred.
*/
@property(nonatomic, assign) BOOL srtpEnableAes128Sha1_32CryptoCipher;
/**
* If set to true, encrypted RTP header extensions as defined in RFC 6904
* will be negotiated. They will only be used if both peers support them.
*/
@property(nonatomic, assign) BOOL srtpEnableEncryptedRtpHeaderExtensions;
/**
* If set all RtpSenders must have an FrameEncryptor attached to them before
* they are allowed to send packets. All RtpReceivers must have a
* FrameDecryptor attached to them before they are able to receive packets.
*/
@property(nonatomic, assign) BOOL sframeRequireFrameEncryption;
/**
* Initializes CryptoOptions with all possible options set explicitly. This
* is done when converting from a native RTCConfiguration.crypto_options.
*/
- (instancetype)initWithSrtpEnableGcmCryptoSuites:(BOOL)srtpEnableGcmCryptoSuites
srtpEnableAes128Sha1_32CryptoCipher:(BOOL)srtpEnableAes128Sha1_32CryptoCipher
srtpEnableEncryptedRtpHeaderExtensions:(BOOL)srtpEnableEncryptedRtpHeaderExtensions
sframeRequireFrameEncryption:(BOOL)sframeRequireFrameEncryption
NS_DESIGNATED_INITIALIZER;
- (instancetype)init NS_UNAVAILABLE;
@end
NS_ASSUME_NONNULL_END

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/*
* Copyright 2015 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 <AvailabilityMacros.h>
#import <Foundation/Foundation.h>
#import <WebRTC/RTCMacros.h>
NS_ASSUME_NONNULL_BEGIN
RTC_OBJC_EXPORT
@interface RTC_OBJC_TYPE (RTCDataBuffer) : NSObject
/** NSData representation of the underlying buffer. */
@property(nonatomic, readonly) NSData *data;
/** Indicates whether `data` contains UTF-8 or binary data. */
@property(nonatomic, readonly) BOOL isBinary;
- (instancetype)init NS_UNAVAILABLE;
/**
* Initialize an RTCDataBuffer from NSData. `isBinary` indicates whether `data`
* contains UTF-8 or binary data.
*/
- (instancetype)initWithData:(NSData *)data isBinary:(BOOL)isBinary;
@end
@class RTC_OBJC_TYPE(RTCDataChannel);
RTC_OBJC_EXPORT
@protocol RTC_OBJC_TYPE
(RTCDataChannelDelegate)<NSObject>
/** The data channel state changed. */
- (void)dataChannelDidChangeState : (RTC_OBJC_TYPE(RTCDataChannel) *)dataChannel;
/** The data channel successfully received a data buffer. */
- (void)dataChannel:(RTC_OBJC_TYPE(RTCDataChannel) *)dataChannel
didReceiveMessageWithBuffer:(RTC_OBJC_TYPE(RTCDataBuffer) *)buffer;
@optional
/** The data channel's `bufferedAmount` changed. */
- (void)dataChannel:(RTC_OBJC_TYPE(RTCDataChannel) *)dataChannel
didChangeBufferedAmount:(uint64_t)amount;
@end
/** Represents the state of the data channel. */
typedef NS_ENUM(NSInteger, RTCDataChannelState) {
RTCDataChannelStateConnecting,
RTCDataChannelStateOpen,
RTCDataChannelStateClosing,
RTCDataChannelStateClosed,
};
RTC_OBJC_EXPORT
@interface RTC_OBJC_TYPE (RTCDataChannel) : NSObject
/**
* A label that can be used to distinguish this data channel from other data
* channel objects.
*/
@property(nonatomic, readonly) NSString *label;
/** Whether the data channel can send messages in unreliable mode. */
@property(nonatomic, readonly) BOOL isReliable DEPRECATED_ATTRIBUTE;
/** Returns whether this data channel is ordered or not. */
@property(nonatomic, readonly) BOOL isOrdered;
/** Deprecated. Use maxPacketLifeTime. */
@property(nonatomic, readonly) NSUInteger maxRetransmitTime DEPRECATED_ATTRIBUTE;
/**
* The length of the time window (in milliseconds) during which transmissions
* and retransmissions may occur in unreliable mode.
*/
@property(nonatomic, readonly) uint16_t maxPacketLifeTime;
/**
* The maximum number of retransmissions that are attempted in unreliable mode.
*/
@property(nonatomic, readonly) uint16_t maxRetransmits;
/**
* The name of the sub-protocol used with this data channel, if any. Otherwise
* this returns an empty string.
*/
@property(nonatomic, readonly) NSString *protocol;
/**
* Returns whether this data channel was negotiated by the application or not.
*/
@property(nonatomic, readonly) BOOL isNegotiated;
/** Deprecated. Use channelId. */
@property(nonatomic, readonly) NSInteger streamId DEPRECATED_ATTRIBUTE;
/** The identifier for this data channel. */
@property(nonatomic, readonly) int channelId;
/** The state of the data channel. */
@property(nonatomic, readonly) RTCDataChannelState readyState;
/**
* The number of bytes of application data that have been queued using
* `sendData:` but that have not yet been transmitted to the network.
*/
@property(nonatomic, readonly) uint64_t bufferedAmount;
/** The delegate for this data channel. */
@property(nonatomic, weak) id<RTC_OBJC_TYPE(RTCDataChannelDelegate)> delegate;
- (instancetype)init NS_UNAVAILABLE;
/** Closes the data channel. */
- (void)close;
/** Attempt to send `data` on this data channel's underlying data transport. */
- (BOOL)sendData:(RTC_OBJC_TYPE(RTCDataBuffer) *)data;
@end
NS_ASSUME_NONNULL_END

52
Lib/WebRTC.xcframework/ios-arm64/WebRTC.framework/Headers/RTCDataChannelConfiguration.h Normal file
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/*
* Copyright 2015 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 <AvailabilityMacros.h>
#import <Foundation/Foundation.h>
#import <WebRTC/RTCMacros.h>
NS_ASSUME_NONNULL_BEGIN
RTC_OBJC_EXPORT
@interface RTC_OBJC_TYPE (RTCDataChannelConfiguration) : NSObject
/** Set to YES if ordered delivery is required. */
@property(nonatomic, assign) BOOL isOrdered;
/** Deprecated. Use maxPacketLifeTime. */
@property(nonatomic, assign) NSInteger maxRetransmitTimeMs DEPRECATED_ATTRIBUTE;
/**
* Max period in milliseconds in which retransmissions will be sent. After this
* time, no more retransmissions will be sent. -1 if unset.
*/
@property(nonatomic, assign) int maxPacketLifeTime;
/** The max number of retransmissions. -1 if unset. */
@property(nonatomic, assign) int maxRetransmits;
/** Set to YES if the channel has been externally negotiated and we do not send
* an in-band signalling in the form of an "open" message.
*/
@property(nonatomic, assign) BOOL isNegotiated;
/** Deprecated. Use channelId. */
@property(nonatomic, assign) int streamId DEPRECATED_ATTRIBUTE;
/** The id of the data channel. */
@property(nonatomic, assign) int channelId;
/** Set by the application and opaque to the WebRTC implementation. */
@property(nonatomic) NSString* protocol;
@end
NS_ASSUME_NONNULL_END

26
Lib/WebRTC.xcframework/ios-arm64/WebRTC.framework/Headers/RTCDefaultVideoDecoderFactory.h Normal file
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/*
* Copyright 2017 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 <Foundation/Foundation.h>
#import <WebRTC/RTCMacros.h>
#import <WebRTC/RTCVideoDecoderFactory.h>
NS_ASSUME_NONNULL_BEGIN
/** This decoder factory include support for all codecs bundled with WebRTC. If using custom
* codecs, create custom implementations of RTCVideoEncoderFactory and
* RTCVideoDecoderFactory.
*/
RTC_OBJC_EXPORT
@interface RTC_OBJC_TYPE (RTCDefaultVideoDecoderFactory) : NSObject <RTC_OBJC_TYPE(RTCVideoDecoderFactory)>
@end
NS_ASSUME_NONNULL_END

31
Lib/WebRTC.xcframework/ios-arm64/WebRTC.framework/Headers/RTCDefaultVideoEncoderFactory.h Normal file
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/*
* Copyright 2017 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 <Foundation/Foundation.h>
#import <WebRTC/RTCMacros.h>
#import <WebRTC/RTCVideoEncoderFactory.h>
NS_ASSUME_NONNULL_BEGIN
/** This encoder factory include support for all codecs bundled with WebRTC. If using custom
* codecs, create custom implementations of RTCVideoEncoderFactory and
* RTCVideoDecoderFactory.
*/
RTC_OBJC_EXPORT
@interface RTC_OBJC_TYPE (RTCDefaultVideoEncoderFactory) : NSObject <RTC_OBJC_TYPE(RTCVideoEncoderFactory)>
@property(nonatomic, retain) RTC_OBJC_TYPE(RTCVideoCodecInfo) *preferredCodec;
+ (NSArray<RTC_OBJC_TYPE(RTCVideoCodecInfo) *> *)supportedCodecs;
@end
NS_ASSUME_NONNULL_END

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Lib/WebRTC.xcframework/ios-arm64/WebRTC.framework/Headers/RTCDispatcher.h Normal file
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/*
* Copyright 2015 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 <Foundation/Foundation.h>
#import <WebRTC/RTCMacros.h>
typedef NS_ENUM(NSInteger, RTCDispatcherQueueType) {
// Main dispatcher queue.
RTCDispatcherTypeMain,
// Used for starting/stopping AVCaptureSession, and assigning
// capture session to AVCaptureVideoPreviewLayer.
RTCDispatcherTypeCaptureSession,
// Used for operations on AVAudioSession.
RTCDispatcherTypeAudioSession,
// Used for operations on NWPathMonitor.
RTCDispatcherTypeNetworkMonitor,
};
/** Dispatcher that asynchronously dispatches blocks to a specific
* shared dispatch queue.
*/
RTC_OBJC_EXPORT
@interface RTC_OBJC_TYPE (RTCDispatcher) : NSObject
- (instancetype)init NS_UNAVAILABLE;
/** Dispatch the block asynchronously on the queue for dispatchType.
* @param dispatchType The queue type to dispatch on.
* @param block The block to dispatch asynchronously.
*/
+ (void)dispatchAsyncOnType:(RTCDispatcherQueueType)dispatchType block:(dispatch_block_t)block;
/** Returns YES if run on queue for the dispatchType otherwise NO.
* Useful for asserting that a method is run on a correct queue.
*/
+ (BOOL)isOnQueueForType:(RTCDispatcherQueueType)dispatchType;
@end

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Lib/WebRTC.xcframework/ios-arm64/WebRTC.framework/Headers/RTCDtmfSender.h Normal file
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/*
* Copyright 2017 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 <Foundation/Foundation.h>
#import <WebRTC/RTCMacros.h>
NS_ASSUME_NONNULL_BEGIN
RTC_OBJC_EXPORT
@protocol RTC_OBJC_TYPE
(RTCDtmfSender)<NSObject>
/**
* Returns true if this RTCDtmfSender is capable of sending DTMF. Otherwise
* returns false. To be able to send DTMF, the associated RTCRtpSender must be
* able to send packets, and a "telephone-event" codec must be negotiated.
*/
@property(nonatomic, readonly) BOOL canInsertDtmf;
/**
* Queues a task that sends the DTMF tones. The tones parameter is treated
* as a series of characters. The characters 0 through 9, A through D, #, and *
* generate the associated DTMF tones. The characters a to d are equivalent
* to A to D. The character ',' indicates a delay of 2 seconds before
* processing the next character in the tones parameter.
*
* Unrecognized characters are ignored.
*
* @param duration The parameter indicates the duration to use for each
* character passed in the tones parameter. The duration cannot be more
* than 6000 or less than 70 ms.
*
* @param interToneGap The parameter indicates the gap between tones.
* This parameter must be at least 50 ms but should be as short as
* possible.
*
* If InsertDtmf is called on the same object while an existing task for this
* object to generate DTMF is still running, the previous task is canceled.
* Returns true on success and false on failure.
*/
- (BOOL)insertDtmf:(nonnull NSString *)tones
duration:(NSTimeInterval)duration
interToneGap:(NSTimeInterval)interToneGap;
/** The tones remaining to be played out */
- (nonnull NSString *)remainingTones;
/**
* The current tone duration value. This value will be the value last set via the
* insertDtmf method, or the default value of 100 ms if insertDtmf was never called.
*/
- (NSTimeInterval)duration;
/**
* The current value of the between-tone gap. This value will be the value last set
* via the insertDtmf() method, or the default value of 50 ms if insertDtmf() was never
* called.
*/
- (NSTimeInterval)interToneGap;
@end
NS_ASSUME_NONNULL_END

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Lib/WebRTC.xcframework/ios-arm64/WebRTC.framework/Headers/RTCEAGLVideoView.h Normal file
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/*
* Copyright 2015 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 <Foundation/Foundation.h>
#import <UIKit/UIKit.h>
#import <WebRTC/RTCMacros.h>
#import <WebRTC/RTCVideoRenderer.h>
#import <WebRTC/RTCVideoViewShading.h>
NS_ASSUME_NONNULL_BEGIN
@class RTC_OBJC_TYPE(RTCEAGLVideoView);
/**
* RTCEAGLVideoView is an RTCVideoRenderer which renders video frames
* in its bounds using OpenGLES 2.0 or OpenGLES 3.0.
*/
RTC_OBJC_EXPORT
NS_EXTENSION_UNAVAILABLE_IOS("Rendering not available in app extensions.")
@interface RTC_OBJC_TYPE (RTCEAGLVideoView) : UIView <RTC_OBJC_TYPE(RTCVideoRenderer)>
@property(nonatomic, weak) id<RTC_OBJC_TYPE(RTCVideoViewDelegate)> delegate;
- (instancetype)initWithFrame:(CGRect)frame
shader:(id<RTC_OBJC_TYPE(RTCVideoViewShading)>)shader
NS_DESIGNATED_INITIALIZER;
- (instancetype)initWithCoder:(NSCoder *)aDecoder
shader:(id<RTC_OBJC_TYPE(RTCVideoViewShading)>)shader
NS_DESIGNATED_INITIALIZER;
/** @abstract Wrapped RTCVideoRotation, or nil.
*/
@property(nonatomic, nullable) NSValue *rotationOverride;
@end
NS_ASSUME_NONNULL_END

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Lib/WebRTC.xcframework/ios-arm64/WebRTC.framework/Headers/RTCEncodedImage.h Normal file
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/*
* Copyright 2017 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 <Foundation/Foundation.h>
#import <WebRTC/RTCMacros.h>
#import <WebRTC/RTCVideoFrame.h>
NS_ASSUME_NONNULL_BEGIN
/** Represents an encoded frame's type. */
typedef NS_ENUM(NSUInteger, RTCFrameType) {
RTCFrameTypeEmptyFrame = 0,
RTCFrameTypeAudioFrameSpeech = 1,
RTCFrameTypeAudioFrameCN = 2,
RTCFrameTypeVideoFrameKey = 3,
RTCFrameTypeVideoFrameDelta = 4,
};
typedef NS_ENUM(NSUInteger, RTCVideoContentType) {
RTCVideoContentTypeUnspecified,
RTCVideoContentTypeScreenshare,
};
/** Represents an encoded frame. Corresponds to webrtc::EncodedImage. */
RTC_OBJC_EXPORT
@interface RTC_OBJC_TYPE (RTCEncodedImage) : NSObject
@property(nonatomic, strong) NSData *buffer;
@property(nonatomic, assign) int32_t encodedWidth;
@property(nonatomic, assign) int32_t encodedHeight;
@property(nonatomic, assign) uint32_t timeStamp;
@property(nonatomic, assign) int64_t captureTimeMs;
@property(nonatomic, assign) int64_t ntpTimeMs;
@property(nonatomic, assign) uint8_t flags;
@property(nonatomic, assign) int64_t encodeStartMs;
@property(nonatomic, assign) int64_t encodeFinishMs;
@property(nonatomic, assign) RTCFrameType frameType;
@property(nonatomic, assign) RTCVideoRotation rotation;
@property(nonatomic, strong) NSNumber *qp;
@property(nonatomic, assign) RTCVideoContentType contentType;
@end
NS_ASSUME_NONNULL_END

30
Lib/WebRTC.xcframework/ios-arm64/WebRTC.framework/Headers/RTCFieldTrials.h Normal file
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/*
* 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 <Foundation/Foundation.h>
#import <WebRTC/RTCMacros.h>
/** The only valid value for the following if set is kRTCFieldTrialEnabledValue. */
RTC_EXTERN NSString *const kRTCFieldTrialAudioForceABWENoTWCCKey;
RTC_EXTERN NSString *const kRTCFieldTrialFlexFec03AdvertisedKey;
RTC_EXTERN NSString *const kRTCFieldTrialFlexFec03Key;
RTC_EXTERN NSString *const kRTCFieldTrialH264HighProfileKey;
RTC_EXTERN NSString *const kRTCFieldTrialMinimizeResamplingOnMobileKey;
RTC_EXTERN NSString *const kRTCFieldTrialUseNWPathMonitor;
/** The valid value for field trials above. */
RTC_EXTERN NSString *const kRTCFieldTrialEnabledValue;
/** Initialize field trials using a dictionary mapping field trial keys to their
* values. See above for valid keys and values. Must be called before any other
* call into WebRTC. See: webrtc/system_wrappers/include/field_trial.h
*/
RTC_EXTERN void RTCInitFieldTrialDictionary(NSDictionary<NSString *, NSString *> *fieldTrials);

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Lib/WebRTC.xcframework/ios-arm64/WebRTC.framework/Headers/RTCFileLogger.h Normal file
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/*
* Copyright 2015 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 <Foundation/Foundation.h>
#import <WebRTC/RTCMacros.h>
typedef NS_ENUM(NSUInteger, RTCFileLoggerSeverity) {
RTCFileLoggerSeverityVerbose,
RTCFileLoggerSeverityInfo,
RTCFileLoggerSeverityWarning,
RTCFileLoggerSeverityError
};
typedef NS_ENUM(NSUInteger, RTCFileLoggerRotationType) {
RTCFileLoggerTypeCall,
RTCFileLoggerTypeApp,
};
NS_ASSUME_NONNULL_BEGIN
// This class intercepts WebRTC logs and saves them to a file. The file size
// will not exceed the given maximum bytesize. When the maximum bytesize is
// reached, logs are rotated according to the rotationType specified.
// For kRTCFileLoggerTypeCall, logs from the beginning and the end
// are preserved while the middle section is overwritten instead.
// For kRTCFileLoggerTypeApp, the oldest log is overwritten.
// This class is not threadsafe.
RTC_OBJC_EXPORT
@interface RTC_OBJC_TYPE (RTCFileLogger) : NSObject
// The severity level to capture. The default is kRTCFileLoggerSeverityInfo.
@property(nonatomic, assign) RTCFileLoggerSeverity severity;
// The rotation type for this file logger. The default is
// kRTCFileLoggerTypeCall.
@property(nonatomic, readonly) RTCFileLoggerRotationType rotationType;
// Disables buffering disk writes. Should be set before `start`. Buffering
// is enabled by default for performance.
@property(nonatomic, assign) BOOL shouldDisableBuffering;
// Default constructor provides default settings for dir path, file size and
// rotation type.
- (instancetype)init;
// Create file logger with default rotation type.
- (instancetype)initWithDirPath:(NSString *)dirPath maxFileSize:(NSUInteger)maxFileSize;
- (instancetype)initWithDirPath:(NSString *)dirPath
maxFileSize:(NSUInteger)maxFileSize
rotationType:(RTCFileLoggerRotationType)rotationType NS_DESIGNATED_INITIALIZER;
// Starts writing WebRTC logs to disk if not already started. Overwrites any
// existing file(s).
- (void)start;
// Stops writing WebRTC logs to disk. This method is also called on dealloc.
- (void)stop;
// Returns the current contents of the logs, or nil if start has been called
// without a stop.
- (nullable NSData *)logData;
@end
NS_ASSUME_NONNULL_END

51
Lib/WebRTC.xcframework/ios-arm64/WebRTC.framework/Headers/RTCFileVideoCapturer.h Normal file
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/*
* Copyright 2017 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 <Foundation/Foundation.h>
#import <WebRTC/RTCVideoCapturer.h>
NS_ASSUME_NONNULL_BEGIN
/**
* Error passing block.
*/
typedef void (^RTCFileVideoCapturerErrorBlock)(NSError *error);
/**
* Captures buffers from bundled video file.
*
* See @c RTCVideoCapturer for more info on capturers.
*/
RTC_OBJC_EXPORT
NS_CLASS_AVAILABLE_IOS(10)
@interface RTC_OBJC_TYPE (RTCFileVideoCapturer) : RTC_OBJC_TYPE(RTCVideoCapturer)
/**
* Starts asynchronous capture of frames from video file.
*
* Capturing is not started if error occurs. Underlying error will be
* relayed in the errorBlock if one is provided.
* Successfully captured video frames will be passed to the delegate.
*
* @param nameOfFile The name of the bundled video file to be read.
* @errorBlock block to be executed upon error.
*/
- (void)startCapturingFromFileNamed:(NSString *)nameOfFile
onError:(__nullable RTCFileVideoCapturerErrorBlock)errorBlock;
/**
* Immediately stops capture.
*/
- (void)stopCapture;
@end
NS_ASSUME_NONNULL_END

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Lib/WebRTC.xcframework/ios-arm64/WebRTC.framework/Headers/RTCH264ProfileLevelId.h Normal file
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/*
* Copyright 2017 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 <Foundation/Foundation.h>
#import <WebRTC/RTCMacros.h>
RTC_OBJC_EXPORT extern NSString *const kRTCVideoCodecH264Name;
RTC_OBJC_EXPORT extern NSString *const kRTCLevel31ConstrainedHigh;
RTC_OBJC_EXPORT extern NSString *const kRTCLevel31ConstrainedBaseline;
RTC_OBJC_EXPORT extern NSString *const kRTCMaxSupportedH264ProfileLevelConstrainedHigh;
RTC_OBJC_EXPORT extern NSString *const kRTCMaxSupportedH264ProfileLevelConstrainedBaseline;
/** H264 Profiles and levels. */
typedef NS_ENUM(NSUInteger, RTCH264Profile) {
RTCH264ProfileConstrainedBaseline,
RTCH264ProfileBaseline,
RTCH264ProfileMain,
RTCH264ProfileConstrainedHigh,
RTCH264ProfileHigh,
};
typedef NS_ENUM(NSUInteger, RTCH264Level) {
RTCH264Level1_b = 0,
RTCH264Level1 = 10,
RTCH264Level1_1 = 11,
RTCH264Level1_2 = 12,
RTCH264Level1_3 = 13,
RTCH264Level2 = 20,
RTCH264Level2_1 = 21,
RTCH264Level2_2 = 22,
RTCH264Level3 = 30,
RTCH264Level3_1 = 31,
RTCH264Level3_2 = 32,
RTCH264Level4 = 40,
RTCH264Level4_1 = 41,
RTCH264Level4_2 = 42,
RTCH264Level5 = 50,
RTCH264Level5_1 = 51,
RTCH264Level5_2 = 52
};
RTC_OBJC_EXPORT
@interface RTC_OBJC_TYPE (RTCH264ProfileLevelId) : NSObject
@property(nonatomic, readonly) RTCH264Profile profile;
@property(nonatomic, readonly) RTCH264Level level;
@property(nonatomic, readonly) NSString *hexString;
- (instancetype)initWithHexString:(NSString *)hexString;
- (instancetype)initWithProfile:(RTCH264Profile)profile level:(RTCH264Level)level;
@end

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Lib/WebRTC.xcframework/ios-arm64/WebRTC.framework/Headers/RTCI420Buffer.h Normal file
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/*
* 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 <AVFoundation/AVFoundation.h>
#import <WebRTC/RTCYUVPlanarBuffer.h>
NS_ASSUME_NONNULL_BEGIN
/** Protocol for RTCYUVPlanarBuffers containing I420 data */
RTC_OBJC_EXPORT
@protocol RTC_OBJC_TYPE
(RTCI420Buffer)<RTC_OBJC_TYPE(RTCYUVPlanarBuffer)> @end
NS_ASSUME_NONNULL_END

49
Lib/WebRTC.xcframework/ios-arm64/WebRTC.framework/Headers/RTCIceCandidate.h Normal file
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/*
* Copyright 2015 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 <Foundation/Foundation.h>
#import <WebRTC/RTCMacros.h>
NS_ASSUME_NONNULL_BEGIN
RTC_OBJC_EXPORT
@interface RTC_OBJC_TYPE (RTCIceCandidate) : NSObject
/**
* If present, the identifier of the "media stream identification" for the media
* component this candidate is associated with.
*/
@property(nonatomic, readonly, nullable) NSString *sdpMid;
/**
* The index (starting at zero) of the media description this candidate is
* associated with in the SDP.
*/
@property(nonatomic, readonly) int sdpMLineIndex;
/** The SDP string for this candidate. */
@property(nonatomic, readonly) NSString *sdp;
/** The URL of the ICE server which this candidate is gathered from. */
@property(nonatomic, readonly, nullable) NSString *serverUrl;
- (instancetype)init NS_UNAVAILABLE;
/**
* Initialize an RTCIceCandidate from SDP.
*/
- (instancetype)initWithSdp:(NSString *)sdp
sdpMLineIndex:(int)sdpMLineIndex
sdpMid:(nullable NSString *)sdpMid NS_DESIGNATED_INITIALIZER;
@end
NS_ASSUME_NONNULL_END

42
Lib/WebRTC.xcframework/ios-arm64/WebRTC.framework/Headers/RTCIceCandidateErrorEvent.h Normal file
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/*
* Copyright (c) 2021 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 <Foundation/Foundation.h>
#import <WebRTC/RTCMacros.h>
NS_ASSUME_NONNULL_BEGIN
RTC_OBJC_EXPORT
@interface RTC_OBJC_TYPE (RTCIceCandidateErrorEvent) : NSObject
/** The local IP address used to communicate with the STUN or TURN server. */
@property(nonatomic, readonly) NSString *address;
/** The port used to communicate with the STUN or TURN server. */
@property(nonatomic, readonly) int port;
/** The STUN or TURN URL that identifies the STUN or TURN server for which the failure occurred. */
@property(nonatomic, readonly) NSString *url;
/** The numeric STUN error code returned by the STUN or TURN server. If no host candidate can reach
* the server, errorCode will be set to the value 701 which is outside the STUN error code range.
* This error is only fired once per server URL while in the RTCIceGatheringState of "gathering". */
@property(nonatomic, readonly) int errorCode;
/** The STUN reason text returned by the STUN or TURN server. If the server could not be reached,
* errorText will be set to an implementation-specific value providing details about the error. */
@property(nonatomic, readonly) NSString *errorText;
- (instancetype)init NS_DESIGNATED_INITIALIZER;
@end
NS_ASSUME_NONNULL_END

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/*
* Copyright 2015 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 <Foundation/Foundation.h>
#import <WebRTC/RTCMacros.h>
typedef NS_ENUM(NSUInteger, RTCTlsCertPolicy) {
RTCTlsCertPolicySecure,
RTCTlsCertPolicyInsecureNoCheck
};
NS_ASSUME_NONNULL_BEGIN
RTC_OBJC_EXPORT
@interface RTC_OBJC_TYPE (RTCIceServer) : NSObject
/** URI(s) for this server represented as NSStrings. */
@property(nonatomic, readonly) NSArray<NSString *> *urlStrings;
/** Username to use if this RTCIceServer object is a TURN server. */
@property(nonatomic, readonly, nullable) NSString *username;
/** Credential to use if this RTCIceServer object is a TURN server. */
@property(nonatomic, readonly, nullable) NSString *credential;
/**
* TLS certificate policy to use if this RTCIceServer object is a TURN server.
*/
@property(nonatomic, readonly) RTCTlsCertPolicy tlsCertPolicy;
/**
If the URIs in `urls` only contain IP addresses, this field can be used
to indicate the hostname, which may be necessary for TLS (using the SNI
extension). If `urls` itself contains the hostname, this isn't necessary.
*/
@property(nonatomic, readonly, nullable) NSString *hostname;
/** List of protocols to be used in the TLS ALPN extension. */
@property(nonatomic, readonly) NSArray<NSString *> *tlsAlpnProtocols;
/**
List elliptic curves to be used in the TLS elliptic curves extension.
Only curve names supported by OpenSSL should be used (eg. "P-256","X25519").
*/
@property(nonatomic, readonly) NSArray<NSString *> *tlsEllipticCurves;
- (nonnull instancetype)init NS_UNAVAILABLE;
/** Convenience initializer for a server with no authentication (e.g. STUN). */
- (instancetype)initWithURLStrings:(NSArray<NSString *> *)urlStrings;
/**
* Initialize an RTCIceServer with its associated URLs, optional username,
* optional credential, and credentialType.
*/
- (instancetype)initWithURLStrings:(NSArray<NSString *> *)urlStrings
username:(nullable NSString *)username
credential:(nullable NSString *)credential;
/**
* Initialize an RTCIceServer with its associated URLs, optional username,
* optional credential, and TLS cert policy.
*/
- (instancetype)initWithURLStrings:(NSArray<NSString *> *)urlStrings
username:(nullable NSString *)username
credential:(nullable NSString *)credential
tlsCertPolicy:(RTCTlsCertPolicy)tlsCertPolicy;
/**
* Initialize an RTCIceServer with its associated URLs, optional username,
* optional credential, TLS cert policy and hostname.
*/
- (instancetype)initWithURLStrings:(NSArray<NSString *> *)urlStrings
username:(nullable NSString *)username
credential:(nullable NSString *)credential
tlsCertPolicy:(RTCTlsCertPolicy)tlsCertPolicy
hostname:(nullable NSString *)hostname;
/**
* Initialize an RTCIceServer with its associated URLs, optional username,
* optional credential, TLS cert policy, hostname and ALPN protocols.
*/
- (instancetype)initWithURLStrings:(NSArray<NSString *> *)urlStrings
username:(nullable NSString *)username
credential:(nullable NSString *)credential
tlsCertPolicy:(RTCTlsCertPolicy)tlsCertPolicy
hostname:(nullable NSString *)hostname
tlsAlpnProtocols:(NSArray<NSString *> *)tlsAlpnProtocols;
/**
* Initialize an RTCIceServer with its associated URLs, optional username,
* optional credential, TLS cert policy, hostname, ALPN protocols and
* elliptic curves.
*/
- (instancetype)initWithURLStrings:(NSArray<NSString *> *)urlStrings
username:(nullable NSString *)username
credential:(nullable NSString *)credential
tlsCertPolicy:(RTCTlsCertPolicy)tlsCertPolicy
hostname:(nullable NSString *)hostname
tlsAlpnProtocols:(nullable NSArray<NSString *> *)tlsAlpnProtocols
tlsEllipticCurves:(nullable NSArray<NSString *> *)tlsEllipticCurves
NS_DESIGNATED_INITIALIZER;
@end
NS_ASSUME_NONNULL_END

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/*
* Copyright 2015 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 <Foundation/Foundation.h>
#import <WebRTC/RTCMacros.h>
NS_ASSUME_NONNULL_BEGIN
/** This does not currently conform to the spec. */
RTC_OBJC_EXPORT
@interface RTC_OBJC_TYPE (RTCLegacyStatsReport) : NSObject
/** Time since 1970-01-01T00:00:00Z in milliseconds. */
@property(nonatomic, readonly) CFTimeInterval timestamp;
/** The type of stats held by this object. */
@property(nonatomic, readonly) NSString *type;
/** The identifier for this object. */
@property(nonatomic, readonly) NSString *reportId;
/** A dictionary holding the actual stats. */
@property(nonatomic, readonly) NSDictionary<NSString *, NSString *> *values;
- (instancetype)init NS_UNAVAILABLE;
@end
NS_ASSUME_NONNULL_END

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/*
* Copyright 2015 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 <Foundation/Foundation.h>
#import <WebRTC/RTCMacros.h>
// Subset of rtc::LoggingSeverity.
typedef NS_ENUM(NSInteger, RTCLoggingSeverity) {
RTCLoggingSeverityVerbose,
RTCLoggingSeverityInfo,
RTCLoggingSeverityWarning,
RTCLoggingSeverityError,
RTCLoggingSeverityNone,
};
// Wrapper for C++ RTC_LOG(sev) macros.
// Logs the log string to the webrtc logstream for the given severity.
RTC_EXTERN void RTCLogEx(RTCLoggingSeverity severity, NSString* log_string);
// Wrapper for rtc::LogMessage::LogToDebug.
// Sets the minimum severity to be logged to console.
RTC_EXTERN void RTCSetMinDebugLogLevel(RTCLoggingSeverity severity);
// Returns the filename with the path prefix removed.
RTC_EXTERN NSString* RTCFileName(const char* filePath);
// Some convenience macros.
#define RTCLogString(format, ...) \
[NSString stringWithFormat:@"(%@:%d %s): " format, \
RTCFileName(__FILE__), \
__LINE__, \
__FUNCTION__, \
##__VA_ARGS__]
#define RTCLogFormat(severity, format, ...) \
do { \
NSString* log_string = RTCLogString(format, ##__VA_ARGS__); \
RTCLogEx(severity, log_string); \
} while (false)
#define RTCLogVerbose(format, ...) RTCLogFormat(RTCLoggingSeverityVerbose, format, ##__VA_ARGS__)
#define RTCLogInfo(format, ...) RTCLogFormat(RTCLoggingSeverityInfo, format, ##__VA_ARGS__)
#define RTCLogWarning(format, ...) RTCLogFormat(RTCLoggingSeverityWarning, format, ##__VA_ARGS__)
#define RTCLogError(format, ...) RTCLogFormat(RTCLoggingSeverityError, format, ##__VA_ARGS__)
#if !defined(NDEBUG)
#define RTCLogDebug(format, ...) RTCLogInfo(format, ##__VA_ARGS__)
#else
#define RTCLogDebug(format, ...) \
do { \
} while (false)
#endif
#define RTCLog(format, ...) RTCLogInfo(format, ##__VA_ARGS__)

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Lib/WebRTC.xcframework/ios-arm64/WebRTC.framework/Headers/RTCMTLVideoView.h Normal file
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/*
* Copyright 2017 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 <Foundation/Foundation.h>
#import <WebRTC/RTCMacros.h>
#import <WebRTC/RTCVideoFrame.h>
#import <WebRTC/RTCVideoRenderer.h>
NS_ASSUME_NONNULL_BEGIN
/**
* RTCMTLVideoView is thin wrapper around MTKView.
*
* It has id<RTCVideoRenderer> property that renders video frames in the view's
* bounds using Metal.
*/
NS_CLASS_AVAILABLE_IOS(9)
RTC_OBJC_EXPORT
@interface RTC_OBJC_TYPE (RTCMTLVideoView) : UIView<RTC_OBJC_TYPE(RTCVideoRenderer)>
@property(nonatomic, weak) id<RTC_OBJC_TYPE(RTCVideoViewDelegate)> delegate;
@property(nonatomic) UIViewContentMode videoContentMode;
/** @abstract Enables/disables rendering.
*/
@property(nonatomic, getter=isEnabled) BOOL enabled;
/** @abstract Wrapped RTCVideoRotation, or nil.
*/
@property(nonatomic, nullable) NSValue* rotationOverride;
@end
NS_ASSUME_NONNULL_END

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/*
* 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 SDK_OBJC_BASE_RTCMACROS_H_
#define SDK_OBJC_BASE_RTCMACROS_H_
#ifdef WEBRTC_ENABLE_OBJC_SYMBOL_EXPORT
#if defined(WEBRTC_LIBRARY_IMPL)
#define RTC_OBJC_EXPORT __attribute__((visibility("default")))
#endif
#endif // WEBRTC_ENABLE_OBJC_SYMBOL_EXPORT
#ifndef RTC_OBJC_EXPORT
#define RTC_OBJC_EXPORT
#endif
// Internal macros used to correctly concatenate symbols.
#define RTC_SYMBOL_CONCAT_HELPER(a, b) a##b
#define RTC_SYMBOL_CONCAT(a, b) RTC_SYMBOL_CONCAT_HELPER(a, b)
// RTC_OBJC_TYPE_PREFIX
//
// Macro used to prepend a prefix to the API types that are exported with
// RTC_OBJC_EXPORT.
//
// Clients can patch the definition of this macro locally and build
// WebRTC.framework with their own prefix in case symbol clashing is a
// problem.
//
// This macro must be defined uniformily across all the translation units.
#ifndef RTC_OBJC_TYPE_PREFIX
#define RTC_OBJC_TYPE_PREFIX
#endif
// RCT_OBJC_TYPE
//
// Macro used internally to declare API types. Declaring an API type without
// using this macro will not include the declared type in the set of types
// that will be affected by the configurable RTC_OBJC_TYPE_PREFIX.
#define RTC_OBJC_TYPE(type_name) RTC_SYMBOL_CONCAT(RTC_OBJC_TYPE_PREFIX, type_name)
#if defined(__cplusplus)
#define RTC_EXTERN extern "C" RTC_OBJC_EXPORT
#else
#define RTC_EXTERN extern RTC_OBJC_EXPORT
#endif
#ifdef __OBJC__
#define RTC_FWD_DECL_OBJC_CLASS(classname) @class classname
#else
#define RTC_FWD_DECL_OBJC_CLASS(classname) typedef struct objc_object classname
#endif
#endif // SDK_OBJC_BASE_RTCMACROS_H_

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/*
* Copyright 2015 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 <Foundation/Foundation.h>
#import <WebRTC/RTCMacros.h>
NS_ASSUME_NONNULL_BEGIN
/** Constraint keys for media sources. */
/** The value for this key should be a base64 encoded string containing
* the data from the serialized configuration proto.
*/
RTC_EXTERN NSString *const kRTCMediaConstraintsAudioNetworkAdaptorConfig;
/** Constraint keys for generating offers and answers. */
RTC_EXTERN NSString *const kRTCMediaConstraintsIceRestart;
RTC_EXTERN NSString *const kRTCMediaConstraintsOfferToReceiveAudio;
RTC_EXTERN NSString *const kRTCMediaConstraintsOfferToReceiveVideo;
RTC_EXTERN NSString *const kRTCMediaConstraintsVoiceActivityDetection;
/** Constraint values for Boolean parameters. */
RTC_EXTERN NSString *const kRTCMediaConstraintsValueTrue;
RTC_EXTERN NSString *const kRTCMediaConstraintsValueFalse;
RTC_OBJC_EXPORT
@interface RTC_OBJC_TYPE (RTCMediaConstraints) : NSObject
- (instancetype)init NS_UNAVAILABLE;
/** Initialize with mandatory and/or optional constraints. */
- (instancetype)
initWithMandatoryConstraints:(nullable NSDictionary<NSString *, NSString *> *)mandatory
optionalConstraints:(nullable NSDictionary<NSString *, NSString *> *)optional
NS_DESIGNATED_INITIALIZER;
@end
NS_ASSUME_NONNULL_END

34
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/*
* 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 <Foundation/Foundation.h>
#import <WebRTC/RTCMacros.h>
typedef NS_ENUM(NSInteger, RTCSourceState) {
RTCSourceStateInitializing,
RTCSourceStateLive,
RTCSourceStateEnded,
RTCSourceStateMuted,
};
NS_ASSUME_NONNULL_BEGIN
RTC_OBJC_EXPORT
@interface RTC_OBJC_TYPE (RTCMediaSource) : NSObject
/** The current state of the RTCMediaSource. */
@property(nonatomic, readonly) RTCSourceState state;
- (instancetype)init NS_UNAVAILABLE;
@end
NS_ASSUME_NONNULL_END

49
Lib/WebRTC.xcframework/ios-arm64/WebRTC.framework/Headers/RTCMediaStream.h Normal file
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/*
* Copyright 2015 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 <Foundation/Foundation.h>
#import <WebRTC/RTCMacros.h>
NS_ASSUME_NONNULL_BEGIN
@class RTC_OBJC_TYPE(RTCAudioTrack);
@class RTC_OBJC_TYPE(RTCPeerConnectionFactory);
@class RTC_OBJC_TYPE(RTCVideoTrack);
RTC_OBJC_EXPORT
@interface RTC_OBJC_TYPE (RTCMediaStream) : NSObject
/** The audio tracks in this stream. */
@property(nonatomic, strong, readonly) NSArray<RTC_OBJC_TYPE(RTCAudioTrack) *> *audioTracks;
/** The video tracks in this stream. */
@property(nonatomic, strong, readonly) NSArray<RTC_OBJC_TYPE(RTCVideoTrack) *> *videoTracks;
/** An identifier for this media stream. */
@property(nonatomic, readonly) NSString *streamId;
- (instancetype)init NS_UNAVAILABLE;
/** Adds the given audio track to this media stream. */
- (void)addAudioTrack:(RTC_OBJC_TYPE(RTCAudioTrack) *)audioTrack;
/** Adds the given video track to this media stream. */
- (void)addVideoTrack:(RTC_OBJC_TYPE(RTCVideoTrack) *)videoTrack;
/** Removes the given audio track to this media stream. */
- (void)removeAudioTrack:(RTC_OBJC_TYPE(RTCAudioTrack) *)audioTrack;
/** Removes the given video track to this media stream. */
- (void)removeVideoTrack:(RTC_OBJC_TYPE(RTCVideoTrack) *)videoTrack;
@end
NS_ASSUME_NONNULL_END

50
Lib/WebRTC.xcframework/ios-arm64/WebRTC.framework/Headers/RTCMediaStreamTrack.h Normal file
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/*
* Copyright 2015 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 <Foundation/Foundation.h>
#import <WebRTC/RTCMacros.h>
/**
* Represents the state of the track. This exposes the same states in C++.
*/
typedef NS_ENUM(NSInteger, RTCMediaStreamTrackState) {
RTCMediaStreamTrackStateLive,
RTCMediaStreamTrackStateEnded
};
NS_ASSUME_NONNULL_BEGIN
RTC_EXTERN NSString *const kRTCMediaStreamTrackKindAudio;
RTC_EXTERN NSString *const kRTCMediaStreamTrackKindVideo;
RTC_OBJC_EXPORT
@interface RTC_OBJC_TYPE (RTCMediaStreamTrack) : NSObject
/**
* The kind of track. For example, "audio" if this track represents an audio
* track and "video" if this track represents a video track.
*/
@property(nonatomic, readonly) NSString *kind;
/** An identifier string. */
@property(nonatomic, readonly) NSString *trackId;
/** The enabled state of the track. */
@property(nonatomic, assign) BOOL isEnabled;
/** The state of the track. */
@property(nonatomic, readonly) RTCMediaStreamTrackState readyState;
- (instancetype)init NS_UNAVAILABLE;
@end
NS_ASSUME_NONNULL_END

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/*
* 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 <Foundation/Foundation.h>
#import <WebRTC/RTCMacros.h>
#import <WebRTC/RTCMetricsSampleInfo.h>
/**
* Enables gathering of metrics (which can be fetched with
* RTCGetAndResetMetrics). Must be called before any other call into WebRTC.
*/
RTC_EXTERN void RTCEnableMetrics(void);
/** Gets and clears native histograms. */
RTC_EXTERN NSArray<RTC_OBJC_TYPE(RTCMetricsSampleInfo) *>* RTCGetAndResetMetrics(void);

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/*
* 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 <Foundation/Foundation.h>
#import <WebRTC/RTCMacros.h>
NS_ASSUME_NONNULL_BEGIN
RTC_OBJC_EXPORT
@interface RTC_OBJC_TYPE (RTCMetricsSampleInfo) : NSObject
/**
* Example of RTCMetricsSampleInfo:
* name: "WebRTC.Video.InputFramesPerSecond"
* min: 1
* max: 100
* bucketCount: 50
* samples: [29]:2 [30]:1
*/
/** The name of the histogram. */
@property(nonatomic, readonly) NSString *name;
/** The minimum bucket value. */
@property(nonatomic, readonly) int min;
/** The maximum bucket value. */
@property(nonatomic, readonly) int max;
/** The number of buckets. */
@property(nonatomic, readonly) int bucketCount;
/** A dictionary holding the samples <value, # of events>. */
@property(nonatomic, readonly) NSDictionary<NSNumber *, NSNumber *> *samples;
- (instancetype)init NS_UNAVAILABLE;
@end
NS_ASSUME_NONNULL_END

23
Lib/WebRTC.xcframework/ios-arm64/WebRTC.framework/Headers/RTCMutableI420Buffer.h Normal file
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/*
* 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 <AVFoundation/AVFoundation.h>
#import <WebRTC/RTCI420Buffer.h>
#import <WebRTC/RTCMutableYUVPlanarBuffer.h>
NS_ASSUME_NONNULL_BEGIN
/** Extension of the I420 buffer with mutable data access */
RTC_OBJC_EXPORT
@protocol RTC_OBJC_TYPE
(RTCMutableI420Buffer)<RTC_OBJC_TYPE(RTCI420Buffer), RTC_OBJC_TYPE(RTCMutableYUVPlanarBuffer)> @end
NS_ASSUME_NONNULL_END

28
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/*
* 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 <AVFoundation/AVFoundation.h>
#import <WebRTC/RTCYUVPlanarBuffer.h>
NS_ASSUME_NONNULL_BEGIN
/** Extension of the YUV planar data buffer with mutable data access */
RTC_OBJC_EXPORT
@protocol RTC_OBJC_TYPE
(RTCMutableYUVPlanarBuffer)<RTC_OBJC_TYPE(RTCYUVPlanarBuffer)>
@property(nonatomic, readonly) uint8_t *mutableDataY;
@property(nonatomic, readonly) uint8_t *mutableDataU;
@property(nonatomic, readonly) uint8_t *mutableDataV;
@end
NS_ASSUME_NONNULL_END

23
Lib/WebRTC.xcframework/ios-arm64/WebRTC.framework/Headers/RTCNativeI420Buffer.h Normal file
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/*
* 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 <AVFoundation/AVFoundation.h>
#import <WebRTC/RTCI420Buffer.h>
#import <WebRTC/RTCMacros.h>
NS_ASSUME_NONNULL_BEGIN
/** RTCI420Buffer implements the RTCI420Buffer protocol */
RTC_OBJC_EXPORT
@interface RTC_OBJC_TYPE (RTCI420Buffer) : NSObject<RTC_OBJC_TYPE(RTCI420Buffer)>
@end
NS_ASSUME_NONNULL_END

24
Lib/WebRTC.xcframework/ios-arm64/WebRTC.framework/Headers/RTCNativeMutableI420Buffer.h Normal file
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/*
* 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 <AVFoundation/AVFoundation.h>
#import <WebRTC/RTCMacros.h>
#import <WebRTC/RTCMutableI420Buffer.h>
#import <WebRTC/RTCNativeI420Buffer.h>
NS_ASSUME_NONNULL_BEGIN
/** Mutable version of RTCI420Buffer */
RTC_OBJC_EXPORT
@interface RTC_OBJC_TYPE (RTCMutableI420Buffer) : RTC_OBJC_TYPE(RTCI420Buffer)<RTC_OBJC_TYPE(RTCMutableI420Buffer)>
@end
NS_ASSUME_NONNULL_END

24
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/*
* Copyright 2020 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 <Foundation/Foundation.h>
NS_ASSUME_NONNULL_BEGIN
/** Listens for NWPathMonitor updates and forwards the results to a C++
* observer.
*/
@interface RTCNetworkMonitor : NSObject
- (instancetype)init NS_UNAVAILABLE;
@end
NS_ASSUME_NONNULL_END

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/*
* Copyright 2015 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 <Foundation/Foundation.h>
#import <WebRTC/RTCMacros.h>
@class RTC_OBJC_TYPE(RTCConfiguration);
@class RTC_OBJC_TYPE(RTCDataChannel);
@class RTC_OBJC_TYPE(RTCDataChannelConfiguration);
@class RTC_OBJC_TYPE(RTCIceCandidate);
@class RTC_OBJC_TYPE(RTCIceCandidateErrorEvent);
@class RTC_OBJC_TYPE(RTCMediaConstraints);
@class RTC_OBJC_TYPE(RTCMediaStream);
@class RTC_OBJC_TYPE(RTCMediaStreamTrack);
@class RTC_OBJC_TYPE(RTCPeerConnectionFactory);
@class RTC_OBJC_TYPE(RTCRtpReceiver);
@class RTC_OBJC_TYPE(RTCRtpSender);
@class RTC_OBJC_TYPE(RTCRtpTransceiver);
@class RTC_OBJC_TYPE(RTCRtpTransceiverInit);
@class RTC_OBJC_TYPE(RTCSessionDescription);
@class RTC_OBJC_TYPE(RTCStatisticsReport);
@class RTC_OBJC_TYPE(RTCLegacyStatsReport);
typedef NS_ENUM(NSInteger, RTCRtpMediaType);
NS_ASSUME_NONNULL_BEGIN
extern NSString *const kRTCPeerConnectionErrorDomain;
extern int const kRTCSessionDescriptionErrorCode;
/** Represents the signaling state of the peer connection. */
typedef NS_ENUM(NSInteger, RTCSignalingState) {
RTCSignalingStateStable,
RTCSignalingStateHaveLocalOffer,
RTCSignalingStateHaveLocalPrAnswer,
RTCSignalingStateHaveRemoteOffer,
RTCSignalingStateHaveRemotePrAnswer,
// Not an actual state, represents the total number of states.
RTCSignalingStateClosed,
};
/** Represents the ice connection state of the peer connection. */
typedef NS_ENUM(NSInteger, RTCIceConnectionState) {
RTCIceConnectionStateNew,
RTCIceConnectionStateChecking,
RTCIceConnectionStateConnected,
RTCIceConnectionStateCompleted,
RTCIceConnectionStateFailed,
RTCIceConnectionStateDisconnected,
RTCIceConnectionStateClosed,
RTCIceConnectionStateCount,
};
/** Represents the combined ice+dtls connection state of the peer connection. */
typedef NS_ENUM(NSInteger, RTCPeerConnectionState) {
RTCPeerConnectionStateNew,
RTCPeerConnectionStateConnecting,
RTCPeerConnectionStateConnected,
RTCPeerConnectionStateDisconnected,
RTCPeerConnectionStateFailed,
RTCPeerConnectionStateClosed,
};
/** Represents the ice gathering state of the peer connection. */
typedef NS_ENUM(NSInteger, RTCIceGatheringState) {
RTCIceGatheringStateNew,
RTCIceGatheringStateGathering,
RTCIceGatheringStateComplete,
};
/** Represents the stats output level. */
typedef NS_ENUM(NSInteger, RTCStatsOutputLevel) {
RTCStatsOutputLevelStandard,
RTCStatsOutputLevelDebug,
};
typedef void (^RTCCreateSessionDescriptionCompletionHandler)(
RTC_OBJC_TYPE(RTCSessionDescription) *_Nullable sdp, NSError *_Nullable error);
typedef void (^RTCSetSessionDescriptionCompletionHandler)(NSError *_Nullable error);
@class RTC_OBJC_TYPE(RTCPeerConnection);
RTC_OBJC_EXPORT
@protocol RTC_OBJC_TYPE
(RTCPeerConnectionDelegate)<NSObject>
/** Called when the SignalingState changed. */
- (void)peerConnection
: (RTC_OBJC_TYPE(RTCPeerConnection) *)peerConnection didChangeSignalingState
: (RTCSignalingState)stateChanged;
/** Called when media is received on a new stream from remote peer. */
- (void)peerConnection:(RTC_OBJC_TYPE(RTCPeerConnection) *)peerConnection
didAddStream:(RTC_OBJC_TYPE(RTCMediaStream) *)stream;
/** Called when a remote peer closes a stream.
* This is not called when RTCSdpSemanticsUnifiedPlan is specified.
*/
- (void)peerConnection:(RTC_OBJC_TYPE(RTCPeerConnection) *)peerConnection
didRemoveStream:(RTC_OBJC_TYPE(RTCMediaStream) *)stream;
/** Called when negotiation is needed, for example ICE has restarted. */
- (void)peerConnectionShouldNegotiate:(RTC_OBJC_TYPE(RTCPeerConnection) *)peerConnection;
/** Called any time the IceConnectionState changes. */
- (void)peerConnection:(RTC_OBJC_TYPE(RTCPeerConnection) *)peerConnection
didChangeIceConnectionState:(RTCIceConnectionState)newState;
/** Called any time the IceGatheringState changes. */
- (void)peerConnection:(RTC_OBJC_TYPE(RTCPeerConnection) *)peerConnection
didChangeIceGatheringState:(RTCIceGatheringState)newState;
/** New ice candidate has been found. */
- (void)peerConnection:(RTC_OBJC_TYPE(RTCPeerConnection) *)peerConnection
didGenerateIceCandidate:(RTC_OBJC_TYPE(RTCIceCandidate) *)candidate;
/** Called when a group of local Ice candidates have been removed. */
- (void)peerConnection:(RTC_OBJC_TYPE(RTCPeerConnection) *)peerConnection
didRemoveIceCandidates:(NSArray<RTC_OBJC_TYPE(RTCIceCandidate) *> *)candidates;
/** New data channel has been opened. */
- (void)peerConnection:(RTC_OBJC_TYPE(RTCPeerConnection) *)peerConnection
didOpenDataChannel:(RTC_OBJC_TYPE(RTCDataChannel) *)dataChannel;
/** Called when signaling indicates a transceiver will be receiving media from
* the remote endpoint.
* This is only called with RTCSdpSemanticsUnifiedPlan specified.
*/
@optional
/** Called any time the IceConnectionState changes following standardized
* transition. */
- (void)peerConnection:(RTC_OBJC_TYPE(RTCPeerConnection) *)peerConnection
didChangeStandardizedIceConnectionState:(RTCIceConnectionState)newState;
/** Called any time the PeerConnectionState changes. */
- (void)peerConnection:(RTC_OBJC_TYPE(RTCPeerConnection) *)peerConnection
didChangeConnectionState:(RTCPeerConnectionState)newState;
- (void)peerConnection:(RTC_OBJC_TYPE(RTCPeerConnection) *)peerConnection
didStartReceivingOnTransceiver:(RTC_OBJC_TYPE(RTCRtpTransceiver) *)transceiver;
/** Called when a receiver and its track are created. */
- (void)peerConnection:(RTC_OBJC_TYPE(RTCPeerConnection) *)peerConnection
didAddReceiver:(RTC_OBJC_TYPE(RTCRtpReceiver) *)rtpReceiver
streams:(NSArray<RTC_OBJC_TYPE(RTCMediaStream) *> *)mediaStreams;
/** Called when the receiver and its track are removed. */
- (void)peerConnection:(RTC_OBJC_TYPE(RTCPeerConnection) *)peerConnection
didRemoveReceiver:(RTC_OBJC_TYPE(RTCRtpReceiver) *)rtpReceiver;
/** Called when the selected ICE candidate pair is changed. */
- (void)peerConnection:(RTC_OBJC_TYPE(RTCPeerConnection) *)peerConnection
didChangeLocalCandidate:(RTC_OBJC_TYPE(RTCIceCandidate) *)local
remoteCandidate:(RTC_OBJC_TYPE(RTCIceCandidate) *)remote
lastReceivedMs:(int)lastDataReceivedMs
changeReason:(NSString *)reason;
/** Called when gathering of an ICE candidate failed. */
- (void)peerConnection:(RTC_OBJC_TYPE(RTCPeerConnection) *)peerConnection
didFailToGatherIceCandidate:(RTC_OBJC_TYPE(RTCIceCandidateErrorEvent) *)event;
@end
RTC_OBJC_EXPORT
@interface RTC_OBJC_TYPE (RTCPeerConnection) : NSObject
/** The object that will be notifed about events such as state changes and
* streams being added or removed.
*/
@property(nonatomic, weak, nullable) id<RTC_OBJC_TYPE(RTCPeerConnectionDelegate)> delegate;
/** This property is not available with RTCSdpSemanticsUnifiedPlan. Please use
* `senders` instead.
*/
@property(nonatomic, readonly) NSArray<RTC_OBJC_TYPE(RTCMediaStream) *> *localStreams;
@property(nonatomic, readonly, nullable) RTC_OBJC_TYPE(RTCSessionDescription) * localDescription;
@property(nonatomic, readonly, nullable) RTC_OBJC_TYPE(RTCSessionDescription) * remoteDescription;
@property(nonatomic, readonly) RTCSignalingState signalingState;
@property(nonatomic, readonly) RTCIceConnectionState iceConnectionState;
@property(nonatomic, readonly) RTCPeerConnectionState connectionState;
@property(nonatomic, readonly) RTCIceGatheringState iceGatheringState;
@property(nonatomic, readonly, copy) RTC_OBJC_TYPE(RTCConfiguration) * configuration;
/** Gets all RTCRtpSenders associated with this peer connection.
* Note: reading this property returns different instances of RTCRtpSender.
* Use isEqual: instead of == to compare RTCRtpSender instances.
*/
@property(nonatomic, readonly) NSArray<RTC_OBJC_TYPE(RTCRtpSender) *> *senders;
/** Gets all RTCRtpReceivers associated with this peer connection.
* Note: reading this property returns different instances of RTCRtpReceiver.
* Use isEqual: instead of == to compare RTCRtpReceiver instances.
*/
@property(nonatomic, readonly) NSArray<RTC_OBJC_TYPE(RTCRtpReceiver) *> *receivers;
/** Gets all RTCRtpTransceivers associated with this peer connection.
* Note: reading this property returns different instances of
* RTCRtpTransceiver. Use isEqual: instead of == to compare
* RTCRtpTransceiver instances. This is only available with
* RTCSdpSemanticsUnifiedPlan specified.
*/
@property(nonatomic, readonly) NSArray<RTC_OBJC_TYPE(RTCRtpTransceiver) *> *transceivers;
- (instancetype)init NS_UNAVAILABLE;
/** Sets the PeerConnection's global configuration to `configuration`.
* Any changes to STUN/TURN servers or ICE candidate policy will affect the
* next gathering phase, and cause the next call to createOffer to generate
* new ICE credentials. Note that the BUNDLE and RTCP-multiplexing policies
* cannot be changed with this method.
*/
- (BOOL)setConfiguration:(RTC_OBJC_TYPE(RTCConfiguration) *)configuration;
/** Terminate all media and close the transport. */
- (void)close;
/** Provide a remote candidate to the ICE Agent. */
- (void)addIceCandidate:(RTC_OBJC_TYPE(RTCIceCandidate) *)candidate
DEPRECATED_MSG_ATTRIBUTE("Please use addIceCandidate:completionHandler: instead");
/** Provide a remote candidate to the ICE Agent. */
- (void)addIceCandidate:(RTC_OBJC_TYPE(RTCIceCandidate) *)candidate
completionHandler:(void (^)(NSError *_Nullable error))completionHandler;
/** Remove a group of remote candidates from the ICE Agent. */
- (void)removeIceCandidates:(NSArray<RTC_OBJC_TYPE(RTCIceCandidate) *> *)candidates;
/** Add a new media stream to be sent on this peer connection.
* This method is not supported with RTCSdpSemanticsUnifiedPlan. Please use
* addTrack instead.
*/
- (void)addStream:(RTC_OBJC_TYPE(RTCMediaStream) *)stream;
/** Remove the given media stream from this peer connection.
* This method is not supported with RTCSdpSemanticsUnifiedPlan. Please use
* removeTrack instead.
*/
- (void)removeStream:(RTC_OBJC_TYPE(RTCMediaStream) *)stream;
/** Add a new media stream track to be sent on this peer connection, and return
* the newly created RTCRtpSender. The RTCRtpSender will be
* associated with the streams specified in the `streamIds` list.
*
* Errors: If an error occurs, returns nil. An error can occur if:
* - A sender already exists for the track.
* - The peer connection is closed.
*/
- (nullable RTC_OBJC_TYPE(RTCRtpSender) *)addTrack:(RTC_OBJC_TYPE(RTCMediaStreamTrack) *)track
streamIds:(NSArray<NSString *> *)streamIds;
/** With PlanB semantics, removes an RTCRtpSender from this peer connection.
*
* With UnifiedPlan semantics, sets sender's track to null and removes the
* send component from the associated RTCRtpTransceiver's direction.
*
* Returns YES on success.
*/
- (BOOL)removeTrack:(RTC_OBJC_TYPE(RTCRtpSender) *)sender;
/** addTransceiver creates a new RTCRtpTransceiver and adds it to the set of
* transceivers. Adding a transceiver will cause future calls to CreateOffer
* to add a media description for the corresponding transceiver.
*
* The initial value of `mid` in the returned transceiver is nil. Setting a
* new session description may change it to a non-nil value.
*
* https://w3c.github.io/webrtc-pc/#dom-rtcpeerconnection-addtransceiver
*
* Optionally, an RtpTransceiverInit structure can be specified to configure
* the transceiver from construction. If not specified, the transceiver will
* default to having a direction of kSendRecv and not be part of any streams.
*
* These methods are only available when Unified Plan is enabled (see
* RTCConfiguration).
*/
/** Adds a transceiver with a sender set to transmit the given track. The kind
* of the transceiver (and sender/receiver) will be derived from the kind of
* the track.
*/
- (nullable RTC_OBJC_TYPE(RTCRtpTransceiver) *)addTransceiverWithTrack:
(RTC_OBJC_TYPE(RTCMediaStreamTrack) *)track;
- (nullable RTC_OBJC_TYPE(RTCRtpTransceiver) *)
addTransceiverWithTrack:(RTC_OBJC_TYPE(RTCMediaStreamTrack) *)track
init:(RTC_OBJC_TYPE(RTCRtpTransceiverInit) *)init;
/** Adds a transceiver with the given kind. Can either be RTCRtpMediaTypeAudio
* or RTCRtpMediaTypeVideo.
*/
- (nullable RTC_OBJC_TYPE(RTCRtpTransceiver) *)addTransceiverOfType:(RTCRtpMediaType)mediaType;
- (nullable RTC_OBJC_TYPE(RTCRtpTransceiver) *)
addTransceiverOfType:(RTCRtpMediaType)mediaType
init:(RTC_OBJC_TYPE(RTCRtpTransceiverInit) *)init;
/** Tells the PeerConnection that ICE should be restarted. This triggers a need
* for negotiation and subsequent offerForConstraints:completionHandler call will act as if
* RTCOfferAnswerOptions::ice_restart is true.
*/
- (void)restartIce;
/** Generate an SDP offer. */
- (void)offerForConstraints:(RTC_OBJC_TYPE(RTCMediaConstraints) *)constraints
completionHandler:(RTCCreateSessionDescriptionCompletionHandler)completionHandler;
/** Generate an SDP answer. */
- (void)answerForConstraints:(RTC_OBJC_TYPE(RTCMediaConstraints) *)constraints
completionHandler:(RTCCreateSessionDescriptionCompletionHandler)completionHandler;
/** Apply the supplied RTCSessionDescription as the local description. */
- (void)setLocalDescription:(RTC_OBJC_TYPE(RTCSessionDescription) *)sdp
completionHandler:(RTCSetSessionDescriptionCompletionHandler)completionHandler;
/** Creates an offer or answer (depending on current signaling state) and sets
* it as the local session description. */
- (void)setLocalDescriptionWithCompletionHandler:
(RTCSetSessionDescriptionCompletionHandler)completionHandler;
/** Apply the supplied RTCSessionDescription as the remote description. */
- (void)setRemoteDescription:(RTC_OBJC_TYPE(RTCSessionDescription) *)sdp
completionHandler:(RTCSetSessionDescriptionCompletionHandler)completionHandler;
/** Limits the bandwidth allocated for all RTP streams sent by this
* PeerConnection. Nil parameters will be unchanged. Setting
* `currentBitrateBps` will force the available bitrate estimate to the given
* value. Returns YES if the parameters were successfully updated.
*/
- (BOOL)setBweMinBitrateBps:(nullable NSNumber *)minBitrateBps
currentBitrateBps:(nullable NSNumber *)currentBitrateBps
maxBitrateBps:(nullable NSNumber *)maxBitrateBps;
/** Start or stop recording an Rtc EventLog. */
- (BOOL)startRtcEventLogWithFilePath:(NSString *)filePath maxSizeInBytes:(int64_t)maxSizeInBytes;
- (void)stopRtcEventLog;
@end
@interface RTC_OBJC_TYPE (RTCPeerConnection)
(Media)
/** Create an RTCRtpSender with the specified kind and media stream ID.
* See RTCMediaStreamTrack.h for available kinds.
* This method is not supported with RTCSdpSemanticsUnifiedPlan. Please use
* addTransceiver instead.
*/
- (RTC_OBJC_TYPE(RTCRtpSender) *)senderWithKind : (NSString *)kind streamId
: (NSString *)streamId;
@end
@interface RTC_OBJC_TYPE (RTCPeerConnection)
(DataChannel)
/** Create a new data channel with the given label and configuration. */
- (nullable RTC_OBJC_TYPE(RTCDataChannel) *)dataChannelForLabel
: (NSString *)label configuration : (RTC_OBJC_TYPE(RTCDataChannelConfiguration) *)configuration;
@end
typedef void (^RTCStatisticsCompletionHandler)(RTC_OBJC_TYPE(RTCStatisticsReport) *);
@interface RTC_OBJC_TYPE (RTCPeerConnection)
(Stats)
/** Gather stats for the given RTCMediaStreamTrack. If `mediaStreamTrack` is nil
* statistics are gathered for all tracks.
*/
- (void)statsForTrack
: (nullable RTC_OBJC_TYPE(RTCMediaStreamTrack) *)mediaStreamTrack statsOutputLevel
: (RTCStatsOutputLevel)statsOutputLevel completionHandler
: (nullable void (^)(NSArray<RTC_OBJC_TYPE(RTCLegacyStatsReport) *> *stats))completionHandler;
/** Gather statistic through the v2 statistics API. */
- (void)statisticsWithCompletionHandler:(RTCStatisticsCompletionHandler)completionHandler;
/** Spec-compliant getStats() performing the stats selection algorithm with the
* sender.
*/
- (void)statisticsForSender:(RTC_OBJC_TYPE(RTCRtpSender) *)sender
completionHandler:(RTCStatisticsCompletionHandler)completionHandler;
/** Spec-compliant getStats() performing the stats selection algorithm with the
* receiver.
*/
- (void)statisticsForReceiver:(RTC_OBJC_TYPE(RTCRtpReceiver) *)receiver
completionHandler:(RTCStatisticsCompletionHandler)completionHandler;
@end
NS_ASSUME_NONNULL_END

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/*
* Copyright 2015 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 <Foundation/Foundation.h>
#import <WebRTC/RTCMacros.h>
NS_ASSUME_NONNULL_BEGIN
@class RTC_OBJC_TYPE(RTCAudioSource);
@class RTC_OBJC_TYPE(RTCAudioTrack);
@class RTC_OBJC_TYPE(RTCConfiguration);
@class RTC_OBJC_TYPE(RTCMediaConstraints);
@class RTC_OBJC_TYPE(RTCMediaStream);
@class RTC_OBJC_TYPE(RTCPeerConnection);
@class RTC_OBJC_TYPE(RTCVideoSource);
@class RTC_OBJC_TYPE(RTCVideoTrack);
@class RTC_OBJC_TYPE(RTCPeerConnectionFactoryOptions);
@protocol RTC_OBJC_TYPE
(RTCPeerConnectionDelegate);
@protocol RTC_OBJC_TYPE
(RTCVideoDecoderFactory);
@protocol RTC_OBJC_TYPE
(RTCVideoEncoderFactory);
@protocol RTC_OBJC_TYPE
(RTCSSLCertificateVerifier);
@protocol RTC_OBJC_TYPE
(RTCAudioDevice);
RTC_OBJC_EXPORT
@interface RTC_OBJC_TYPE (RTCPeerConnectionFactory) : NSObject
/* Initialize object with default H264 video encoder/decoder factories and default ADM */
- (instancetype)init;
/* Initialize object with injectable video encoder/decoder factories and default ADM */
- (instancetype)
initWithEncoderFactory:(nullable id<RTC_OBJC_TYPE(RTCVideoEncoderFactory)>)encoderFactory
decoderFactory:(nullable id<RTC_OBJC_TYPE(RTCVideoDecoderFactory)>)decoderFactory;
/* Initialize object with injectable video encoder/decoder factories and injectable ADM */
- (instancetype)
initWithEncoderFactory:(nullable id<RTC_OBJC_TYPE(RTCVideoEncoderFactory)>)encoderFactory
decoderFactory:(nullable id<RTC_OBJC_TYPE(RTCVideoDecoderFactory)>)decoderFactory
audioDevice:(nullable id<RTC_OBJC_TYPE(RTCAudioDevice)>)audioDevice;
/** Initialize an RTCAudioSource with constraints. */
- (RTC_OBJC_TYPE(RTCAudioSource) *)audioSourceWithConstraints:
(nullable RTC_OBJC_TYPE(RTCMediaConstraints) *)constraints;
/** Initialize an RTCAudioTrack with an id. Convenience ctor to use an audio source
* with no constraints.
*/
- (RTC_OBJC_TYPE(RTCAudioTrack) *)audioTrackWithTrackId:(NSString *)trackId;
/** Initialize an RTCAudioTrack with a source and an id. */
- (RTC_OBJC_TYPE(RTCAudioTrack) *)audioTrackWithSource:(RTC_OBJC_TYPE(RTCAudioSource) *)source
trackId:(NSString *)trackId;
/** Initialize a generic RTCVideoSource. The RTCVideoSource should be
* passed to a RTCVideoCapturer implementation, e.g.
* RTCCameraVideoCapturer, in order to produce frames.
*/
- (RTC_OBJC_TYPE(RTCVideoSource) *)videoSource;
/** Initialize a generic RTCVideoSource with he posibility of marking
* it as usable for screen sharing. The RTCVideoSource should be
* passed to a RTCVideoCapturer implementation, e.g.
* RTCCameraVideoCapturer, in order to produce frames.
*/
- (RTC_OBJC_TYPE(RTCVideoSource) *)videoSourceForScreenCast:(BOOL)forScreenCast;
/** Initialize an RTCVideoTrack with a source and an id. */
- (RTC_OBJC_TYPE(RTCVideoTrack) *)videoTrackWithSource:(RTC_OBJC_TYPE(RTCVideoSource) *)source
trackId:(NSString *)trackId;
/** Initialize an RTCMediaStream with an id. */
- (RTC_OBJC_TYPE(RTCMediaStream) *)mediaStreamWithStreamId:(NSString *)streamId;
/** Initialize an RTCPeerConnection with a configuration, constraints, and
* delegate.
*/
- (nullable RTC_OBJC_TYPE(RTCPeerConnection) *)
peerConnectionWithConfiguration:(RTC_OBJC_TYPE(RTCConfiguration) *)configuration
constraints:(RTC_OBJC_TYPE(RTCMediaConstraints) *)constraints
delegate:(nullable id<RTC_OBJC_TYPE(RTCPeerConnectionDelegate)>)delegate;
- (nullable RTC_OBJC_TYPE(RTCPeerConnection) *)
peerConnectionWithConfiguration:(RTC_OBJC_TYPE(RTCConfiguration) *)configuration
constraints:(RTC_OBJC_TYPE(RTCMediaConstraints) *)constraints
certificateVerifier:
(id<RTC_OBJC_TYPE(RTCSSLCertificateVerifier)>)certificateVerifier
delegate:(nullable id<RTC_OBJC_TYPE(RTCPeerConnectionDelegate)>)delegate;
/** Set the options to be used for subsequently created RTCPeerConnections */
- (void)setOptions:(nonnull RTC_OBJC_TYPE(RTCPeerConnectionFactoryOptions) *)options;
/** Start an AecDump recording. This API call will likely change in the future. */
- (BOOL)startAecDumpWithFilePath:(NSString *)filePath maxSizeInBytes:(int64_t)maxSizeInBytes;
/* Stop an active AecDump recording */
- (void)stopAecDump;
@end
NS_ASSUME_NONNULL_END

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/*
* Copyright 2017 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 <Foundation/Foundation.h>
#import <WebRTC/RTCMacros.h>
NS_ASSUME_NONNULL_BEGIN
RTC_OBJC_EXPORT
@interface RTC_OBJC_TYPE (RTCPeerConnectionFactoryOptions) : NSObject
@property(nonatomic, assign) BOOL disableEncryption;
@property(nonatomic, assign) BOOL disableNetworkMonitor;
@property(nonatomic, assign) BOOL ignoreLoopbackNetworkAdapter;
@property(nonatomic, assign) BOOL ignoreVPNNetworkAdapter;
@property(nonatomic, assign) BOOL ignoreCellularNetworkAdapter;
@property(nonatomic, assign) BOOL ignoreWiFiNetworkAdapter;
@property(nonatomic, assign) BOOL ignoreEthernetNetworkAdapter;
- (instancetype)init NS_DESIGNATED_INITIALIZER;
@end
NS_ASSUME_NONNULL_END

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Lib/WebRTC.xcframework/ios-arm64/WebRTC.framework/Headers/RTCRtcpParameters.h Normal file
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/*
* 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 <Foundation/Foundation.h>
#import <WebRTC/RTCMacros.h>
NS_ASSUME_NONNULL_BEGIN
RTC_OBJC_EXPORT
@interface RTC_OBJC_TYPE (RTCRtcpParameters) : NSObject
/** The Canonical Name used by RTCP. */
@property(nonatomic, readonly, copy) NSString *cname;
/** Whether reduced size RTCP is configured or compound RTCP. */
@property(nonatomic, assign) BOOL isReducedSize;
- (instancetype)init;
@end
NS_ASSUME_NONNULL_END

73
Lib/WebRTC.xcframework/ios-arm64/WebRTC.framework/Headers/RTCRtpCodecParameters.h Normal file
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/*
* 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 <Foundation/Foundation.h>
#import <WebRTC/RTCMacros.h>
NS_ASSUME_NONNULL_BEGIN
RTC_EXTERN const NSString *const kRTCRtxCodecName;
RTC_EXTERN const NSString *const kRTCRedCodecName;
RTC_EXTERN const NSString *const kRTCUlpfecCodecName;
RTC_EXTERN const NSString *const kRTCFlexfecCodecName;
RTC_EXTERN const NSString *const kRTCOpusCodecName;
RTC_EXTERN const NSString *const kRTCIsacCodecName;
RTC_EXTERN const NSString *const kRTCL16CodecName;
RTC_EXTERN const NSString *const kRTCG722CodecName;
RTC_EXTERN const NSString *const kRTCIlbcCodecName;
RTC_EXTERN const NSString *const kRTCPcmuCodecName;
RTC_EXTERN const NSString *const kRTCPcmaCodecName;
RTC_EXTERN const NSString *const kRTCDtmfCodecName;
RTC_EXTERN const NSString *const kRTCComfortNoiseCodecName;
RTC_EXTERN const NSString *const kRTCVp8CodecName;
RTC_EXTERN const NSString *const kRTCVp9CodecName;
RTC_EXTERN const NSString *const kRTCH264CodecName;
/** Defined in https://www.w3.org/TR/webrtc/#idl-def-rtcrtpcodecparameters */
RTC_OBJC_EXPORT
@interface RTC_OBJC_TYPE (RTCRtpCodecParameters) : NSObject
/** The RTP payload type. */
@property(nonatomic, assign) int payloadType;
/**
* The codec MIME subtype. Valid types are listed in:
* http://www.iana.org/assignments/rtp-parameters/rtp-parameters.xhtml#rtp-parameters-2
*
* Several supported types are represented by the constants above.
*/
@property(nonatomic, readonly, nonnull) NSString *name;
/**
* The media type of this codec. Equivalent to MIME top-level type.
*
* Valid values are kRTCMediaStreamTrackKindAudio and
* kRTCMediaStreamTrackKindVideo.
*/
@property(nonatomic, readonly, nonnull) NSString *kind;
/** The codec clock rate expressed in Hertz. */
@property(nonatomic, readonly, nullable) NSNumber *clockRate;
/**
* The number of channels (mono=1, stereo=2).
* Set to null for video codecs.
**/
@property(nonatomic, readonly, nullable) NSNumber *numChannels;
/** The "format specific parameters" field from the "a=fmtp" line in the SDP */
@property(nonatomic, readonly, nonnull) NSDictionary *parameters;
- (instancetype)init;
@end
NS_ASSUME_NONNULL_END

76
Lib/WebRTC.xcframework/ios-arm64/WebRTC.framework/Headers/RTCRtpEncodingParameters.h Normal file
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/*
* 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 <Foundation/Foundation.h>
#import <WebRTC/RTCMacros.h>
NS_ASSUME_NONNULL_BEGIN
/** Corresponds to webrtc::Priority. */
typedef NS_ENUM(NSInteger, RTCPriority) {
RTCPriorityVeryLow,
RTCPriorityLow,
RTCPriorityMedium,
RTCPriorityHigh
};
RTC_OBJC_EXPORT
@interface RTC_OBJC_TYPE (RTCRtpEncodingParameters) : NSObject
/** The idenfifier for the encoding layer. This is used in simulcast. */
@property(nonatomic, copy, nullable) NSString *rid;
/** Controls whether the encoding is currently transmitted. */
@property(nonatomic, assign) BOOL isActive;
/** The maximum bitrate to use for the encoding, or nil if there is no
* limit.
*/
@property(nonatomic, copy, nullable) NSNumber *maxBitrateBps;
/** The minimum bitrate to use for the encoding, or nil if there is no
* limit.
*/
@property(nonatomic, copy, nullable) NSNumber *minBitrateBps;
/** The maximum framerate to use for the encoding, or nil if there is no
* limit.
*/
@property(nonatomic, copy, nullable) NSNumber *maxFramerate;
/** The requested number of temporal layers to use for the encoding, or nil
* if the default should be used.
*/
@property(nonatomic, copy, nullable) NSNumber *numTemporalLayers;
/** Scale the width and height down by this factor for video. If nil,
* implementation default scaling factor will be used.
*/
@property(nonatomic, copy, nullable) NSNumber *scaleResolutionDownBy;
/** The SSRC being used by this encoding. */
@property(nonatomic, readonly, nullable) NSNumber *ssrc;
/** The relative bitrate priority. */
@property(nonatomic, assign) double bitratePriority;
/** The relative DiffServ Code Point priority. */
@property(nonatomic, assign) RTCPriority networkPriority;
/** Allow dynamic frame length changes for audio:
https://w3c.github.io/webrtc-extensions/#dom-rtcrtpencodingparameters-adaptiveptime */
@property(nonatomic, assign) BOOL adaptiveAudioPacketTime;
- (instancetype)init;
@end
NS_ASSUME_NONNULL_END

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Lib/WebRTC.xcframework/ios-arm64/WebRTC.framework/Headers/RTCRtpHeaderExtension.h Normal file
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/*
* 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 <Foundation/Foundation.h>
#import <WebRTC/RTCMacros.h>
NS_ASSUME_NONNULL_BEGIN
RTC_OBJC_EXPORT
@interface RTC_OBJC_TYPE (RTCRtpHeaderExtension) : NSObject
/** The URI of the RTP header extension, as defined in RFC5285. */
@property(nonatomic, readonly, copy) NSString *uri;
/** The value put in the RTP packet to identify the header extension. */
@property(nonatomic, readonly) int id;
/** Whether the header extension is encrypted or not. */
@property(nonatomic, readonly, getter=isEncrypted) BOOL encrypted;
- (instancetype)init;
@end
NS_ASSUME_NONNULL_END

58
Lib/WebRTC.xcframework/ios-arm64/WebRTC.framework/Headers/RTCRtpParameters.h Normal file
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/*
* 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 <Foundation/Foundation.h>
#import <WebRTC/RTCMacros.h>
#import <WebRTC/RTCRtcpParameters.h>
#import <WebRTC/RTCRtpCodecParameters.h>
#import <WebRTC/RTCRtpEncodingParameters.h>
#import <WebRTC/RTCRtpHeaderExtension.h>
NS_ASSUME_NONNULL_BEGIN
/** Corresponds to webrtc::DegradationPreference. */
typedef NS_ENUM(NSInteger, RTCDegradationPreference) {
RTCDegradationPreferenceDisabled,
RTCDegradationPreferenceMaintainFramerate,
RTCDegradationPreferenceMaintainResolution,
RTCDegradationPreferenceBalanced
};
RTC_OBJC_EXPORT
@interface RTC_OBJC_TYPE (RTCRtpParameters) : NSObject
/** A unique identifier for the last set of parameters applied. */
@property(nonatomic, copy) NSString *transactionId;
/** Parameters used for RTCP. */
@property(nonatomic, readonly, copy) RTC_OBJC_TYPE(RTCRtcpParameters) * rtcp;
/** An array containing parameters for RTP header extensions. */
@property(nonatomic, readonly, copy)
NSArray<RTC_OBJC_TYPE(RTCRtpHeaderExtension) *> *headerExtensions;
/** The currently active encodings in the order of preference. */
@property(nonatomic, copy) NSArray<RTC_OBJC_TYPE(RTCRtpEncodingParameters) *> *encodings;
/** The negotiated set of send codecs in order of preference. */
@property(nonatomic, copy) NSArray<RTC_OBJC_TYPE(RTCRtpCodecParameters) *> *codecs;
/**
* Degradation preference in case of CPU adaptation or constrained bandwidth.
* If nil, implementation default degradation preference will be used.
*/
@property(nonatomic, copy, nullable) NSNumber *degradationPreference;
- (instancetype)init;
@end
NS_ASSUME_NONNULL_END

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Lib/WebRTC.xcframework/ios-arm64/WebRTC.framework/Headers/RTCRtpReceiver.h Normal file
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/*
* 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 <Foundation/Foundation.h>
#import <WebRTC/RTCMacros.h>
#import <WebRTC/RTCMediaStreamTrack.h>
#import <WebRTC/RTCRtpParameters.h>
NS_ASSUME_NONNULL_BEGIN
/** Represents the media type of the RtpReceiver. */
typedef NS_ENUM(NSInteger, RTCRtpMediaType) {
RTCRtpMediaTypeAudio,
RTCRtpMediaTypeVideo,
RTCRtpMediaTypeData,
RTCRtpMediaTypeUnsupported,
};
@class RTC_OBJC_TYPE(RTCRtpReceiver);
RTC_OBJC_EXPORT
@protocol RTC_OBJC_TYPE
(RTCRtpReceiverDelegate)<NSObject>
/** Called when the first RTP packet is received.
*
* Note: Currently if there are multiple RtpReceivers of the same media type,
* they will all call OnFirstPacketReceived at once.
*
* For example, if we create three audio receivers, A/B/C, they will listen to
* the same signal from the underneath network layer. Whenever the first audio packet
* is received, the underneath signal will be fired. All the receivers A/B/C will be
* notified and the callback of the receiver's delegate will be called.
*
* The process is the same for video receivers.
*/
- (void)rtpReceiver
: (RTC_OBJC_TYPE(RTCRtpReceiver) *)rtpReceiver didReceiveFirstPacketForMediaType
: (RTCRtpMediaType)mediaType;
@end
RTC_OBJC_EXPORT
@protocol RTC_OBJC_TYPE
(RTCRtpReceiver)<NSObject>
/** A unique identifier for this receiver. */
@property(nonatomic, readonly) NSString *receiverId;
/** The currently active RTCRtpParameters, as defined in
* https://www.w3.org/TR/webrtc/#idl-def-RTCRtpParameters.
*
* The WebRTC specification only defines RTCRtpParameters in terms of senders,
* but this API also applies them to receivers, similar to ORTC:
* http://ortc.org/wp-content/uploads/2016/03/ortc.html#rtcrtpparameters*.
*/
@property(nonatomic, readonly) RTC_OBJC_TYPE(RTCRtpParameters) * parameters;
/** The RTCMediaStreamTrack associated with the receiver.
* Note: reading this property returns a new instance of
* RTCMediaStreamTrack. Use isEqual: instead of == to compare
* RTCMediaStreamTrack instances.
*/
@property(nonatomic, readonly, nullable) RTC_OBJC_TYPE(RTCMediaStreamTrack) * track;
/** The delegate for this RtpReceiver. */
@property(nonatomic, weak) id<RTC_OBJC_TYPE(RTCRtpReceiverDelegate)> delegate;
@end
RTC_OBJC_EXPORT
@interface RTC_OBJC_TYPE (RTCRtpReceiver) : NSObject <RTC_OBJC_TYPE(RTCRtpReceiver)>
- (instancetype)init NS_UNAVAILABLE;
@end
NS_ASSUME_NONNULL_END

54
Lib/WebRTC.xcframework/ios-arm64/WebRTC.framework/Headers/RTCRtpSender.h Normal file
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/*
* 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 <Foundation/Foundation.h>
#import <WebRTC/RTCDtmfSender.h>
#import <WebRTC/RTCMacros.h>
#import <WebRTC/RTCMediaStreamTrack.h>
#import <WebRTC/RTCRtpParameters.h>
NS_ASSUME_NONNULL_BEGIN
RTC_OBJC_EXPORT
@protocol RTC_OBJC_TYPE
(RTCRtpSender)<NSObject>
/** A unique identifier for this sender. */
@property(nonatomic, readonly) NSString *senderId;
/** The currently active RTCRtpParameters, as defined in
* https://www.w3.org/TR/webrtc/#idl-def-RTCRtpParameters.
*/
@property(nonatomic, copy) RTC_OBJC_TYPE(RTCRtpParameters) * parameters;
/** The RTCMediaStreamTrack associated with the sender.
* Note: reading this property returns a new instance of
* RTCMediaStreamTrack. Use isEqual: instead of == to compare
* RTCMediaStreamTrack instances.
*/
@property(nonatomic, copy, nullable) RTC_OBJC_TYPE(RTCMediaStreamTrack) * track;
/** IDs of streams associated with the RTP sender */
@property(nonatomic, copy) NSArray<NSString *> *streamIds;
/** The RTCDtmfSender accociated with the RTP sender. */
@property(nonatomic, readonly, nullable) id<RTC_OBJC_TYPE(RTCDtmfSender)> dtmfSender;
@end
RTC_OBJC_EXPORT
@interface RTC_OBJC_TYPE (RTCRtpSender) : NSObject <RTC_OBJC_TYPE(RTCRtpSender)>
- (instancetype)init NS_UNAVAILABLE;
@end
NS_ASSUME_NONNULL_END

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Lib/WebRTC.xcframework/ios-arm64/WebRTC.framework/Headers/RTCRtpTransceiver.h Normal file
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/*
* 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 <Foundation/Foundation.h>
#import <WebRTC/RTCMacros.h>
#import <WebRTC/RTCRtpReceiver.h>
#import <WebRTC/RTCRtpSender.h>
NS_ASSUME_NONNULL_BEGIN
extern NSString *const kRTCRtpTransceiverErrorDomain;
/** https://w3c.github.io/webrtc-pc/#dom-rtcrtptransceiverdirection */
typedef NS_ENUM(NSInteger, RTCRtpTransceiverDirection) {
RTCRtpTransceiverDirectionSendRecv,
RTCRtpTransceiverDirectionSendOnly,
RTCRtpTransceiverDirectionRecvOnly,
RTCRtpTransceiverDirectionInactive,
RTCRtpTransceiverDirectionStopped
};
/** Structure for initializing an RTCRtpTransceiver in a call to
* RTCPeerConnection.addTransceiver.
* https://w3c.github.io/webrtc-pc/#dom-rtcrtptransceiverinit
*/
RTC_OBJC_EXPORT
@interface RTC_OBJC_TYPE (RTCRtpTransceiverInit) : NSObject
/** Direction of the RTCRtpTransceiver. See RTCRtpTransceiver.direction. */
@property(nonatomic) RTCRtpTransceiverDirection direction;
/** The added RTCRtpTransceiver will be added to these streams. */
@property(nonatomic) NSArray<NSString *> *streamIds;
/** TODO(bugs.webrtc.org/7600): Not implemented. */
@property(nonatomic) NSArray<RTC_OBJC_TYPE(RTCRtpEncodingParameters) *> *sendEncodings;
@end
@class RTC_OBJC_TYPE(RTCRtpTransceiver);
/** The RTCRtpTransceiver maps to the RTCRtpTransceiver defined by the
* WebRTC specification. A transceiver represents a combination of an RTCRtpSender
* and an RTCRtpReceiver that share a common mid. As defined in JSEP, an
* RTCRtpTransceiver is said to be associated with a media description if its
* mid property is non-nil; otherwise, it is said to be disassociated.
* JSEP: https://tools.ietf.org/html/draft-ietf-rtcweb-jsep-24
*
* Note that RTCRtpTransceivers are only supported when using
* RTCPeerConnection with Unified Plan SDP.
*
* WebRTC specification for RTCRtpTransceiver, the JavaScript analog:
* https://w3c.github.io/webrtc-pc/#dom-rtcrtptransceiver
*/
RTC_OBJC_EXPORT
@protocol RTC_OBJC_TYPE
(RTCRtpTransceiver)<NSObject>
/** Media type of the transceiver. The sender and receiver will also have this
* type.
*/
@property(nonatomic, readonly) RTCRtpMediaType mediaType;
/** The mid attribute is the mid negotiated and present in the local and
* remote descriptions. Before negotiation is complete, the mid value may be
* nil. After rollbacks, the value may change from a non-nil value to nil.
* https://w3c.github.io/webrtc-pc/#dom-rtcrtptransceiver-mid
*/
@property(nonatomic, readonly) NSString *mid;
/** The sender attribute exposes the RTCRtpSender corresponding to the RTP
* media that may be sent with the transceiver's mid. The sender is always
* present, regardless of the direction of media.
* https://w3c.github.io/webrtc-pc/#dom-rtcrtptransceiver-sender
*/
@property(nonatomic, readonly) RTC_OBJC_TYPE(RTCRtpSender) * sender;
/** The receiver attribute exposes the RTCRtpReceiver corresponding to the RTP
* media that may be received with the transceiver's mid. The receiver is
* always present, regardless of the direction of media.
* https://w3c.github.io/webrtc-pc/#dom-rtcrtptransceiver-receiver
*/
@property(nonatomic, readonly) RTC_OBJC_TYPE(RTCRtpReceiver) * receiver;
/** The isStopped attribute indicates that the sender of this transceiver will
* no longer send, and that the receiver will no longer receive. It is true if
* either stop has been called or if setting the local or remote description
* has caused the RTCRtpTransceiver to be stopped.
* https://w3c.github.io/webrtc-pc/#dom-rtcrtptransceiver-stopped
*/
@property(nonatomic, readonly) BOOL isStopped;
/** The direction attribute indicates the preferred direction of this
* transceiver, which will be used in calls to createOffer and createAnswer.
* https://w3c.github.io/webrtc-pc/#dom-rtcrtptransceiver-direction
*/
@property(nonatomic, readonly) RTCRtpTransceiverDirection direction;
/** The currentDirection attribute indicates the current direction negotiated
* for this transceiver. If this transceiver has never been represented in an
* offer/answer exchange, or if the transceiver is stopped, the value is not
* present and this method returns NO.
* https://w3c.github.io/webrtc-pc/#dom-rtcrtptransceiver-currentdirection
*/
- (BOOL)currentDirection:(RTCRtpTransceiverDirection *)currentDirectionOut;
/** The stop method irreversibly stops the RTCRtpTransceiver. The sender of
* this transceiver will no longer send, the receiver will no longer receive.
* https://w3c.github.io/webrtc-pc/#dom-rtcrtptransceiver-stop
*/
- (void)stopInternal;
/** An update of directionality does not take effect immediately. Instead,
* future calls to createOffer and createAnswer mark the corresponding media
* descriptions as sendrecv, sendonly, recvonly, or inactive.
* https://w3c.github.io/webrtc-pc/#dom-rtcrtptransceiver-direction
*/
- (void)setDirection:(RTCRtpTransceiverDirection)direction error:(NSError **)error;
@end
RTC_OBJC_EXPORT
@interface RTC_OBJC_TYPE (RTCRtpTransceiver) : NSObject <RTC_OBJC_TYPE(RTCRtpTransceiver)>
- (instancetype)init NS_UNAVAILABLE;
@end
NS_ASSUME_NONNULL_END

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Lib/WebRTC.xcframework/ios-arm64/WebRTC.framework/Headers/RTCSSLAdapter.h Normal file
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/*
* 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 <Foundation/Foundation.h>
#import <WebRTC/RTCMacros.h>
/**
* Initialize and clean up the SSL library. Failure is fatal. These call the
* corresponding functions in webrtc/rtc_base/ssladapter.h.
*/
RTC_EXTERN BOOL RTCInitializeSSL(void);
RTC_EXTERN BOOL RTCCleanupSSL(void);

25
Lib/WebRTC.xcframework/ios-arm64/WebRTC.framework/Headers/RTCSSLCertificateVerifier.h Normal file
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/*
* Copyright 2022 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 <Foundation/Foundation.h>
#import <WebRTC/RTCMacros.h>
NS_ASSUME_NONNULL_BEGIN
RTC_OBJC_EXPORT @protocol RTC_OBJC_TYPE
(RTCSSLCertificateVerifier)<NSObject>
/** The certificate to verify */
- (BOOL)verify : (NSData *)derCertificate;
@end
NS_ASSUME_NONNULL_END

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Lib/WebRTC.xcframework/ios-arm64/WebRTC.framework/Headers/RTCSessionDescription.h Normal file
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/*
* Copyright 2015 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 <Foundation/Foundation.h>
#import <WebRTC/RTCMacros.h>
/**
* Represents the session description type. This exposes the same types that are
* in C++, which doesn't include the rollback type that is in the W3C spec.
*/
typedef NS_ENUM(NSInteger, RTCSdpType) {
RTCSdpTypeOffer,
RTCSdpTypePrAnswer,
RTCSdpTypeAnswer,
RTCSdpTypeRollback,
};
NS_ASSUME_NONNULL_BEGIN
RTC_OBJC_EXPORT
@interface RTC_OBJC_TYPE (RTCSessionDescription) : NSObject
/** The type of session description. */
@property(nonatomic, readonly) RTCSdpType type;
/** The SDP string representation of this session description. */
@property(nonatomic, readonly) NSString *sdp;
- (instancetype)init NS_UNAVAILABLE;
/** Initialize a session description with a type and SDP string. */
- (instancetype)initWithType:(RTCSdpType)type sdp:(NSString *)sdp NS_DESIGNATED_INITIALIZER;
+ (NSString *)stringForType:(RTCSdpType)type;
+ (RTCSdpType)typeForString:(NSString *)string;
@end
NS_ASSUME_NONNULL_END

55
Lib/WebRTC.xcframework/ios-arm64/WebRTC.framework/Headers/RTCStatisticsReport.h Normal file
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/*
* Copyright 2019 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 <Foundation/Foundation.h>
#import <WebRTC/RTCMacros.h>
@class RTC_OBJC_TYPE(RTCStatistics);
NS_ASSUME_NONNULL_BEGIN
/** A statistics report. Encapsulates a number of RTCStatistics objects. */
RTC_OBJC_EXPORT
@interface RTC_OBJC_TYPE (RTCStatisticsReport) : NSObject
/** The timestamp of the report in microseconds since 1970-01-01T00:00:00Z. */
@property(nonatomic, readonly) CFTimeInterval timestamp_us;
/** RTCStatistics objects by id. */
@property(nonatomic, readonly) NSDictionary<NSString *, RTC_OBJC_TYPE(RTCStatistics) *> *statistics;
- (instancetype)init NS_UNAVAILABLE;
@end
/** A part of a report (a subreport) covering a certain area. */
RTC_OBJC_EXPORT
@interface RTC_OBJC_TYPE (RTCStatistics) : NSObject
/** The id of this subreport, e.g. "RTCMediaStreamTrack_receiver_2". */
@property(nonatomic, readonly) NSString *id;
/** The timestamp of the subreport in microseconds since 1970-01-01T00:00:00Z. */
@property(nonatomic, readonly) CFTimeInterval timestamp_us;
/** The type of the subreport, e.g. "track", "codec". */
@property(nonatomic, readonly) NSString *type;
/** The keys and values of the subreport, e.g. "totalFramesDuration = 5.551".
The values are either NSNumbers or NSStrings or NSArrays encapsulating NSNumbers
or NSStrings, or NSDictionary of NSString keys to NSNumber values. */
@property(nonatomic, readonly) NSDictionary<NSString *, NSObject *> *values;
- (instancetype)init NS_UNAVAILABLE;
@end
NS_ASSUME_NONNULL_END

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Lib/WebRTC.xcframework/ios-arm64/WebRTC.framework/Headers/RTCTracing.h Normal file
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/*
* 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 <Foundation/Foundation.h>
#import <WebRTC/RTCMacros.h>
RTC_EXTERN void RTCSetupInternalTracer(void);
/** Starts capture to specified file. Must be a valid writable path.
* Returns YES if capture starts.
*/
RTC_EXTERN BOOL RTCStartInternalCapture(NSString* filePath);
RTC_EXTERN void RTCStopInternalCapture(void);
RTC_EXTERN void RTCShutdownInternalTracer(void);

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Lib/WebRTC.xcframework/ios-arm64/WebRTC.framework/Headers/RTCVideoCapturer.h Normal file
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/*
* Copyright 2017 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 <WebRTC/RTCVideoFrame.h>
#import <WebRTC/RTCMacros.h>
NS_ASSUME_NONNULL_BEGIN
@class RTC_OBJC_TYPE(RTCVideoCapturer);
RTC_OBJC_EXPORT
@protocol RTC_OBJC_TYPE
(RTCVideoCapturerDelegate)<NSObject> -
(void)capturer : (RTC_OBJC_TYPE(RTCVideoCapturer) *)capturer didCaptureVideoFrame
: (RTC_OBJC_TYPE(RTCVideoFrame) *)frame;
@end
RTC_OBJC_EXPORT
@interface RTC_OBJC_TYPE (RTCVideoCapturer) : NSObject
@property(nonatomic, weak) id<RTC_OBJC_TYPE(RTCVideoCapturerDelegate)> delegate;
- (instancetype)initWithDelegate:(id<RTC_OBJC_TYPE(RTCVideoCapturerDelegate)>)delegate;
@end
NS_ASSUME_NONNULL_END

17
Lib/WebRTC.xcframework/ios-arm64/WebRTC.framework/Headers/RTCVideoCodecConstants.h Normal file
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/*
* 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 <Foundation/Foundation.h>
#import <WebRTC/RTCMacros.h>
RTC_EXTERN NSString* const kRTCVideoCodecVp8Name;
RTC_EXTERN NSString* const kRTCVideoCodecVp9Name;
RTC_EXTERN NSString* const kRTCVideoCodecAv1Name;

36
Lib/WebRTC.xcframework/ios-arm64/WebRTC.framework/Headers/RTCVideoCodecInfo.h Normal file
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/*
* Copyright 2017 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 <Foundation/Foundation.h>
#import <WebRTC/RTCMacros.h>
NS_ASSUME_NONNULL_BEGIN
/** Holds information to identify a codec. Corresponds to webrtc::SdpVideoFormat. */
RTC_OBJC_EXPORT
@interface RTC_OBJC_TYPE (RTCVideoCodecInfo) : NSObject <NSCoding>
- (instancetype)init NS_UNAVAILABLE;
- (instancetype)initWithName:(NSString *)name;
- (instancetype)initWithName:(NSString *)name
parameters:(nullable NSDictionary<NSString *, NSString *> *)parameters
NS_DESIGNATED_INITIALIZER;
- (BOOL)isEqualToCodecInfo:(RTC_OBJC_TYPE(RTCVideoCodecInfo) *)info;
@property(nonatomic, readonly) NSString *name;
@property(nonatomic, readonly) NSDictionary<NSString *, NSString *> *parameters;
@end
NS_ASSUME_NONNULL_END

41
Lib/WebRTC.xcframework/ios-arm64/WebRTC.framework/Headers/RTCVideoDecoder.h Normal file
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/*
* Copyright 2017 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 <Foundation/Foundation.h>
#import <WebRTC/RTCCodecSpecificInfo.h>
#import <WebRTC/RTCEncodedImage.h>
#import <WebRTC/RTCMacros.h>
#import <WebRTC/RTCVideoEncoderSettings.h>
#import <WebRTC/RTCVideoFrame.h>
NS_ASSUME_NONNULL_BEGIN
/** Callback block for decoder. */
typedef void (^RTCVideoDecoderCallback)(RTC_OBJC_TYPE(RTCVideoFrame) * frame);
/** Protocol for decoder implementations. */
RTC_OBJC_EXPORT
@protocol RTC_OBJC_TYPE
(RTCVideoDecoder)<NSObject>
- (void)setCallback : (RTCVideoDecoderCallback)callback;
- (NSInteger)startDecodeWithNumberOfCores:(int)numberOfCores;
- (NSInteger)releaseDecoder;
// TODO(bugs.webrtc.org/15444): Remove obsolete missingFrames param.
- (NSInteger)decode:(RTC_OBJC_TYPE(RTCEncodedImage) *)encodedImage
missingFrames:(BOOL)missingFrames
codecSpecificInfo:(nullable id<RTC_OBJC_TYPE(RTCCodecSpecificInfo)>)info
renderTimeMs:(int64_t)renderTimeMs;
- (NSString *)implementationName;
@end
NS_ASSUME_NONNULL_END

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/*
* Copyright 2021 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 <Foundation/Foundation.h>
#import <WebRTC/RTCMacros.h>
#import <WebRTC/RTCVideoDecoder.h>
RTC_OBJC_EXPORT
@interface RTC_OBJC_TYPE (RTCVideoDecoderAV1) : NSObject
/* This returns a AV1 decoder that can be returned from a RTCVideoDecoderFactory injected into
* RTCPeerConnectionFactory. Even though it implements the RTCVideoDecoder protocol, it can not be
* used independently from the RTCPeerConnectionFactory.
*/
+ (id<RTC_OBJC_TYPE(RTCVideoDecoder)>)av1Decoder;
@end

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/*
* Copyright 2017 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 <Foundation/Foundation.h>
#import <WebRTC/RTCMacros.h>
#import <WebRTC/RTCVideoCodecInfo.h>
#import <WebRTC/RTCVideoDecoder.h>
NS_ASSUME_NONNULL_BEGIN
/** RTCVideoDecoderFactory is an Objective-C version of webrtc::VideoDecoderFactory.
*/
RTC_OBJC_EXPORT
@protocol RTC_OBJC_TYPE
(RTCVideoDecoderFactory)<NSObject>
- (nullable id<RTC_OBJC_TYPE(RTCVideoDecoder)>)createDecoder
: (RTC_OBJC_TYPE(RTCVideoCodecInfo) *)info;
- (NSArray<RTC_OBJC_TYPE(RTCVideoCodecInfo) *> *)
supportedCodecs; // TODO(andersc): "supportedFormats" instead?
@end
NS_ASSUME_NONNULL_END

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/*
* Copyright 2017 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 <Foundation/Foundation.h>
#import <WebRTC/RTCMacros.h>
#import <WebRTC/RTCVideoDecoderFactory.h>
RTC_OBJC_EXPORT
@interface RTC_OBJC_TYPE (RTCVideoDecoderFactoryH264) : NSObject <RTC_OBJC_TYPE(RTCVideoDecoderFactory)>
@end

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/*
* Copyright 2017 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 <Foundation/Foundation.h>
#import <WebRTC/RTCMacros.h>
#import <WebRTC/RTCVideoDecoder.h>
RTC_OBJC_EXPORT
@interface RTC_OBJC_TYPE (RTCVideoDecoderH264) : NSObject <RTC_OBJC_TYPE(RTCVideoDecoder)>
@end

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/*
* Copyright 2017 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 <Foundation/Foundation.h>
#import <WebRTC/RTCMacros.h>
#import <WebRTC/RTCVideoDecoder.h>
RTC_OBJC_EXPORT
@interface RTC_OBJC_TYPE (RTCVideoDecoderVP8) : NSObject
/* This returns a VP8 decoder that can be returned from a RTCVideoDecoderFactory injected into
* RTCPeerConnectionFactory. Even though it implements the RTCVideoDecoder protocol, it can not be
* used independently from the RTCPeerConnectionFactory.
*/
+ (id<RTC_OBJC_TYPE(RTCVideoDecoder)>)vp8Decoder;
@end

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/*
* Copyright 2017 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 <Foundation/Foundation.h>
#import <WebRTC/RTCMacros.h>
#import <WebRTC/RTCVideoDecoder.h>
RTC_OBJC_EXPORT
@interface RTC_OBJC_TYPE (RTCVideoDecoderVP9) : NSObject
/* This returns a VP9 decoder that can be returned from a RTCVideoDecoderFactory injected into
* RTCPeerConnectionFactory. Even though it implements the RTCVideoDecoder protocol, it can not be
* used independently from the RTCPeerConnectionFactory.
*/
+ (id<RTC_OBJC_TYPE(RTCVideoDecoder)>)vp9Decoder;
+ (bool)isSupported;
@end

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/*
* Copyright 2017 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 <Foundation/Foundation.h>
#import <WebRTC/RTCCodecSpecificInfo.h>
#import <WebRTC/RTCEncodedImage.h>
#import <WebRTC/RTCMacros.h>
#import <WebRTC/RTCVideoEncoderQpThresholds.h>
#import <WebRTC/RTCVideoEncoderSettings.h>
#import <WebRTC/RTCVideoFrame.h>
NS_ASSUME_NONNULL_BEGIN
/** Callback block for encoder. */
typedef BOOL (^RTCVideoEncoderCallback)(RTC_OBJC_TYPE(RTCEncodedImage) * frame,
id<RTC_OBJC_TYPE(RTCCodecSpecificInfo)> info);
/** Protocol for encoder implementations. */
RTC_OBJC_EXPORT
@protocol RTC_OBJC_TYPE
(RTCVideoEncoder)<NSObject>
- (void)setCallback : (nullable RTCVideoEncoderCallback)callback;
- (NSInteger)startEncodeWithSettings:(RTC_OBJC_TYPE(RTCVideoEncoderSettings) *)settings
numberOfCores:(int)numberOfCores;
- (NSInteger)releaseEncoder;
- (NSInteger)encode:(RTC_OBJC_TYPE(RTCVideoFrame) *)frame
codecSpecificInfo:(nullable id<RTC_OBJC_TYPE(RTCCodecSpecificInfo)>)info
frameTypes:(NSArray<NSNumber *> *)frameTypes;
- (int)setBitrate:(uint32_t)bitrateKbit framerate:(uint32_t)framerate;
- (NSString *)implementationName;
/** Returns QP scaling settings for encoder. The quality scaler adjusts the resolution in order to
* keep the QP from the encoded images within the given range. Returning nil from this function
* disables quality scaling. */
- (nullable RTC_OBJC_TYPE(RTCVideoEncoderQpThresholds) *)scalingSettings;
/** Resolutions should be aligned to this value. */
@property(nonatomic, readonly) NSInteger resolutionAlignment;
/** If enabled, resolution alignment is applied to all simulcast layers simultaneously so that when
scaled, all resolutions comply with 'resolutionAlignment'. */
@property(nonatomic, readonly) BOOL applyAlignmentToAllSimulcastLayers;
/** If YES, the receiver is expected to resample/scale the source texture to the expected output
size. */
@property(nonatomic, readonly) BOOL supportsNativeHandle;
@end
NS_ASSUME_NONNULL_END

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Lib/WebRTC.xcframework/ios-arm64/WebRTC.framework/Headers/RTCVideoEncoderAV1.h Normal file
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/*
* Copyright 2021 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 <Foundation/Foundation.h>
#import <WebRTC/RTCMacros.h>
#import <WebRTC/RTCVideoEncoder.h>
RTC_OBJC_EXPORT
@interface RTC_OBJC_TYPE (RTCVideoEncoderAV1) : NSObject
/* This returns a AV1 encoder that can be returned from a RTCVideoEncoderFactory injected into
* RTCPeerConnectionFactory. Even though it implements the RTCVideoEncoder protocol, it can not be
* used independently from the RTCPeerConnectionFactory.
*/
+ (id<RTC_OBJC_TYPE(RTCVideoEncoder)>)av1Encoder;
+ (bool)isSupported;
@end

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/*
* Copyright 2017 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 <Foundation/Foundation.h>
#import <WebRTC/RTCMacros.h>
#import <WebRTC/RTCVideoCodecInfo.h>
#import <WebRTC/RTCVideoEncoder.h>
NS_ASSUME_NONNULL_BEGIN
/** RTCVideoEncoderFactory is an Objective-C version of
webrtc::VideoEncoderFactory::VideoEncoderSelector.
*/
RTC_OBJC_EXPORT
@protocol RTC_OBJC_TYPE
(RTCVideoEncoderSelector)<NSObject>
- (void)registerCurrentEncoderInfo : (RTC_OBJC_TYPE(RTCVideoCodecInfo) *)info;
- (nullable RTC_OBJC_TYPE(RTCVideoCodecInfo) *)encoderForBitrate:(NSInteger)bitrate;
- (nullable RTC_OBJC_TYPE(RTCVideoCodecInfo) *)encoderForBrokenEncoder;
@optional
- (nullable RTC_OBJC_TYPE(RTCVideoCodecInfo) *)encoderForResolutionChangeBySize:(CGSize)size;
@end
/** RTCVideoEncoderFactory is an Objective-C version of webrtc::VideoEncoderFactory.
*/
RTC_OBJC_EXPORT
@protocol RTC_OBJC_TYPE
(RTCVideoEncoderFactory)<NSObject>
- (nullable id<RTC_OBJC_TYPE(RTCVideoEncoder)>)createEncoder
: (RTC_OBJC_TYPE(RTCVideoCodecInfo) *)info;
- (NSArray<RTC_OBJC_TYPE(RTCVideoCodecInfo) *> *)
supportedCodecs; // TODO(andersc): "supportedFormats" instead?
@optional
- (NSArray<RTC_OBJC_TYPE(RTCVideoCodecInfo) *> *)implementations;
- (nullable id<RTC_OBJC_TYPE(RTCVideoEncoderSelector)>)encoderSelector;
@end
NS_ASSUME_NONNULL_END

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Lib/WebRTC.xcframework/ios-arm64/WebRTC.framework/Headers/RTCVideoEncoderFactoryH264.h Normal file
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/*
* Copyright 2017 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 <Foundation/Foundation.h>
#import <WebRTC/RTCMacros.h>
#import <WebRTC/RTCVideoEncoderFactory.h>
RTC_OBJC_EXPORT
@interface RTC_OBJC_TYPE (RTCVideoEncoderFactoryH264) : NSObject <RTC_OBJC_TYPE(RTCVideoEncoderFactory)>
@end

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/*
* Copyright 2017 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 <Foundation/Foundation.h>
#import <WebRTC/RTCMacros.h>
#import <WebRTC/RTCVideoCodecInfo.h>
#import <WebRTC/RTCVideoEncoder.h>
RTC_OBJC_EXPORT
@interface RTC_OBJC_TYPE (RTCVideoEncoderH264) : NSObject <RTC_OBJC_TYPE(RTCVideoEncoder)>
- (instancetype)initWithCodecInfo:(RTC_OBJC_TYPE(RTCVideoCodecInfo) *)codecInfo;
@end

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/*
* Copyright 2017 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 <Foundation/Foundation.h>
#import <WebRTC/RTCMacros.h>
NS_ASSUME_NONNULL_BEGIN
/** QP thresholds for encoder. Corresponds to webrtc::VideoEncoder::QpThresholds. */
RTC_OBJC_EXPORT
@interface RTC_OBJC_TYPE (RTCVideoEncoderQpThresholds) : NSObject
- (instancetype)initWithThresholdsLow:(NSInteger)low high:(NSInteger)high;
@property(nonatomic, readonly) NSInteger low;
@property(nonatomic, readonly) NSInteger high;
@end
NS_ASSUME_NONNULL_END

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Lib/WebRTC.xcframework/ios-arm64/WebRTC.framework/Headers/RTCVideoEncoderSettings.h Normal file
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/*
* Copyright 2017 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 <Foundation/Foundation.h>
#import <WebRTC/RTCMacros.h>
NS_ASSUME_NONNULL_BEGIN
typedef NS_ENUM(NSUInteger, RTCVideoCodecMode) {
RTCVideoCodecModeRealtimeVideo,
RTCVideoCodecModeScreensharing,
};
/** Settings for encoder. Corresponds to webrtc::VideoCodec. */
RTC_OBJC_EXPORT
@interface RTC_OBJC_TYPE (RTCVideoEncoderSettings) : NSObject
@property(nonatomic, strong) NSString *name;
@property(nonatomic, assign) unsigned short width;
@property(nonatomic, assign) unsigned short height;
@property(nonatomic, assign) unsigned int startBitrate; // kilobits/sec.
@property(nonatomic, assign) unsigned int maxBitrate;
@property(nonatomic, assign) unsigned int minBitrate;
@property(nonatomic, assign) uint32_t maxFramerate;
@property(nonatomic, assign) unsigned int qpMax;
@property(nonatomic, assign) RTCVideoCodecMode mode;
@end
NS_ASSUME_NONNULL_END

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Lib/WebRTC.xcframework/ios-arm64/WebRTC.framework/Headers/RTCVideoEncoderVP8.h Normal file
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/*
* Copyright 2017 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 <Foundation/Foundation.h>
#import <WebRTC/RTCMacros.h>
#import <WebRTC/RTCVideoEncoder.h>
RTC_OBJC_EXPORT
@interface RTC_OBJC_TYPE (RTCVideoEncoderVP8) : NSObject
/* This returns a VP8 encoder that can be returned from a RTCVideoEncoderFactory injected into
* RTCPeerConnectionFactory. Even though it implements the RTCVideoEncoder protocol, it can not be
* used independently from the RTCPeerConnectionFactory.
*/
+ (id<RTC_OBJC_TYPE(RTCVideoEncoder)>)vp8Encoder;
@end

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Lib/WebRTC.xcframework/ios-arm64/WebRTC.framework/Headers/RTCVideoEncoderVP9.h Normal file
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/*
* Copyright 2017 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 <Foundation/Foundation.h>
#import <WebRTC/RTCMacros.h>
#import <WebRTC/RTCVideoEncoder.h>
RTC_OBJC_EXPORT
@interface RTC_OBJC_TYPE (RTCVideoEncoderVP9) : NSObject
/* This returns a VP9 encoder that can be returned from a RTCVideoEncoderFactory injected into
* RTCPeerConnectionFactory. Even though it implements the RTCVideoEncoder protocol, it can not be
* used independently from the RTCPeerConnectionFactory.
*/
+ (id<RTC_OBJC_TYPE(RTCVideoEncoder)>)vp9Encoder;
+ (bool)isSupported;
@end

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/*
* Copyright 2015 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 <WebRTC/RTCMacros.h>
NS_ASSUME_NONNULL_BEGIN
typedef NS_ENUM(NSInteger, RTCVideoRotation) {
RTCVideoRotation_0 = 0,
RTCVideoRotation_90 = 90,
RTCVideoRotation_180 = 180,
RTCVideoRotation_270 = 270,
};
@protocol RTC_OBJC_TYPE
(RTCVideoFrameBuffer);
// RTCVideoFrame is an ObjectiveC version of webrtc::VideoFrame.
RTC_OBJC_EXPORT
@interface RTC_OBJC_TYPE (RTCVideoFrame) : NSObject
/** Width without rotation applied. */
@property(nonatomic, readonly) int width;
/** Height without rotation applied. */
@property(nonatomic, readonly) int height;
@property(nonatomic, readonly) RTCVideoRotation rotation;
/** Timestamp in nanoseconds. */
@property(nonatomic, readonly) int64_t timeStampNs;
/** Timestamp 90 kHz. */
@property(nonatomic, assign) int32_t timeStamp;
@property(nonatomic, readonly) id<RTC_OBJC_TYPE(RTCVideoFrameBuffer)> buffer;
- (instancetype)init NS_UNAVAILABLE;
- (instancetype)new NS_UNAVAILABLE;
/** Initialize an RTCVideoFrame from a pixel buffer, rotation, and timestamp.
* Deprecated - initialize with a RTCCVPixelBuffer instead
*/
- (instancetype)initWithPixelBuffer:(CVPixelBufferRef)pixelBuffer
rotation:(RTCVideoRotation)rotation
timeStampNs:(int64_t)timeStampNs
DEPRECATED_MSG_ATTRIBUTE("use initWithBuffer instead");
/** Initialize an RTCVideoFrame from a pixel buffer combined with cropping and
* scaling. Cropping will be applied first on the pixel buffer, followed by
* scaling to the final resolution of scaledWidth x scaledHeight.
*/
- (instancetype)initWithPixelBuffer:(CVPixelBufferRef)pixelBuffer
scaledWidth:(int)scaledWidth
scaledHeight:(int)scaledHeight
cropWidth:(int)cropWidth
cropHeight:(int)cropHeight
cropX:(int)cropX
cropY:(int)cropY
rotation:(RTCVideoRotation)rotation
timeStampNs:(int64_t)timeStampNs
DEPRECATED_MSG_ATTRIBUTE("use initWithBuffer instead");
/** Initialize an RTCVideoFrame from a frame buffer, rotation, and timestamp.
*/
- (instancetype)initWithBuffer:(id<RTC_OBJC_TYPE(RTCVideoFrameBuffer)>)frameBuffer
rotation:(RTCVideoRotation)rotation
timeStampNs:(int64_t)timeStampNs;
/** Return a frame that is guaranteed to be I420, i.e. it is possible to access
* the YUV data on it.
*/
- (RTC_OBJC_TYPE(RTCVideoFrame) *)newI420VideoFrame;
@end
NS_ASSUME_NONNULL_END

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/*
* Copyright 2017 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 <WebRTC/RTCMacros.h>
NS_ASSUME_NONNULL_BEGIN
@protocol RTC_OBJC_TYPE
(RTCI420Buffer);
// RTCVideoFrameBuffer is an ObjectiveC version of webrtc::VideoFrameBuffer.
RTC_OBJC_EXPORT
@protocol RTC_OBJC_TYPE
(RTCVideoFrameBuffer)<NSObject>
@property(nonatomic, readonly) int width;
@property(nonatomic, readonly) int height;
- (id<RTC_OBJC_TYPE(RTCI420Buffer)>)toI420;
@optional
- (id<RTC_OBJC_TYPE(RTCVideoFrameBuffer)>)cropAndScaleWith:(int)offsetX
offsetY:(int)offsetY
cropWidth:(int)cropWidth
cropHeight:(int)cropHeight
scaleWidth:(int)scaleWidth
scaleHeight:(int)scaleHeight;
@end
NS_ASSUME_NONNULL_END

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/*
* Copyright 2015 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 <Foundation/Foundation.h>
#if TARGET_OS_IPHONE
#import <UIKit/UIKit.h>
#endif
#import <WebRTC/RTCMacros.h>
NS_ASSUME_NONNULL_BEGIN
@class RTC_OBJC_TYPE(RTCVideoFrame);
RTC_OBJC_EXPORT
@protocol RTC_OBJC_TYPE
(RTCVideoRenderer)<NSObject>
/** The size of the frame. */
- (void)setSize : (CGSize)size;
/** The frame to be displayed. */
- (void)renderFrame:(nullable RTC_OBJC_TYPE(RTCVideoFrame) *)frame;
@end
RTC_OBJC_EXPORT
@protocol RTC_OBJC_TYPE
(RTCVideoViewDelegate)
- (void)videoView : (id<RTC_OBJC_TYPE(RTCVideoRenderer)>)videoView didChangeVideoSize
: (CGSize)size;
@end
NS_ASSUME_NONNULL_END

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/*
* Copyright 2015 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 <Foundation/Foundation.h>
#import <WebRTC/RTCMacros.h>
#import <WebRTC/RTCMediaSource.h>
#import <WebRTC/RTCVideoCapturer.h>
NS_ASSUME_NONNULL_BEGIN
RTC_OBJC_EXPORT
@interface RTC_OBJC_TYPE (RTCVideoSource) : RTC_OBJC_TYPE(RTCMediaSource) <RTC_OBJC_TYPE(RTCVideoCapturerDelegate)>
- (instancetype)init NS_UNAVAILABLE;
/**
* Calling this function will cause frames to be scaled down to the
* requested resolution. Also, frames will be cropped to match the
* requested aspect ratio, and frames will be dropped to match the
* requested fps. The requested aspect ratio is orientation agnostic and
* will be adjusted to maintain the input orientation, so it doesn't
* matter if e.g. 1280x720 or 720x1280 is requested.
*/
- (void)adaptOutputFormatToWidth:(int)width height:(int)height fps:(int)fps;
@end
NS_ASSUME_NONNULL_END

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/*
* Copyright 2015 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 <WebRTC/RTCMediaStreamTrack.h>
#import <WebRTC/RTCMacros.h>
NS_ASSUME_NONNULL_BEGIN
@protocol RTC_OBJC_TYPE
(RTCVideoRenderer);
@class RTC_OBJC_TYPE(RTCPeerConnectionFactory);
@class RTC_OBJC_TYPE(RTCVideoSource);
RTC_OBJC_EXPORT
@interface RTC_OBJC_TYPE (RTCVideoTrack) : RTC_OBJC_TYPE(RTCMediaStreamTrack)
/** The video source for this video track. */
@property(nonatomic, readonly) RTC_OBJC_TYPE(RTCVideoSource) *source;
- (instancetype)init NS_UNAVAILABLE;
/** Register a renderer that will render all frames received on this track. */
- (void)addRenderer:(id<RTC_OBJC_TYPE(RTCVideoRenderer)>)renderer;
/** Deregister a renderer. */
- (void)removeRenderer:(id<RTC_OBJC_TYPE(RTCVideoRenderer)>)renderer;
@end
NS_ASSUME_NONNULL_END

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/*
* Copyright 2017 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 <Foundation/Foundation.h>
#import <WebRTC/RTCVideoFrame.h>
NS_ASSUME_NONNULL_BEGIN
/**
* RTCVideoViewShading provides a way for apps to customize the OpenGL(ES shaders
* used in rendering for the RTCEAGLVideoView/RTCNSGLVideoView.
*/
RTC_OBJC_EXPORT
@protocol RTC_OBJC_TYPE
(RTCVideoViewShading)<NSObject>
/** Callback for I420 frames. Each plane is given as a texture. */
- (void)applyShadingForFrameWithWidth : (int)width height : (int)height rotation
: (RTCVideoRotation)rotation yPlane : (GLuint)yPlane uPlane : (GLuint)uPlane vPlane
: (GLuint)vPlane;
/** Callback for NV12 frames. Each plane is given as a texture. */
- (void)applyShadingForFrameWithWidth:(int)width
height:(int)height
rotation:(RTCVideoRotation)rotation
yPlane:(GLuint)yPlane
uvPlane:(GLuint)uvPlane;
@end
NS_ASSUME_NONNULL_END

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/*
* 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 <AVFoundation/AVFoundation.h>
#import <WebRTC/RTCMacros.h>
#import <WebRTC/RTCVideoFrameBuffer.h>
NS_ASSUME_NONNULL_BEGIN
/** Protocol for RTCVideoFrameBuffers containing YUV planar data. */
RTC_OBJC_EXPORT
@protocol RTC_OBJC_TYPE
(RTCYUVPlanarBuffer)<RTC_OBJC_TYPE(RTCVideoFrameBuffer)>
@property(nonatomic, readonly) int chromaWidth;
@property(nonatomic, readonly) int chromaHeight;
@property(nonatomic, readonly) const uint8_t *dataY;
@property(nonatomic, readonly) const uint8_t *dataU;
@property(nonatomic, readonly) const uint8_t *dataV;
@property(nonatomic, readonly) int strideY;
@property(nonatomic, readonly) int strideU;
@property(nonatomic, readonly) int strideV;
- (instancetype)initWithWidth:(int)width
height:(int)height
dataY:(const uint8_t *)dataY
dataU:(const uint8_t *)dataU
dataV:(const uint8_t *)dataV;
- (instancetype)initWithWidth:(int)width height:(int)height;
- (instancetype)initWithWidth:(int)width
height:(int)height
strideY:(int)strideY
strideU:(int)strideU
strideV:(int)strideV;
@end
NS_ASSUME_NONNULL_END

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/*
* 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 <UIKit/UIKit.h>
@interface UIDevice (RTCDevice)
+ (NSString *)machineName;
@end

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/*
* Copyright 2023 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 <WebRTC/RTCCodecSpecificInfo.h>
#import <WebRTC/RTCEncodedImage.h>
#import <WebRTC/RTCI420Buffer.h>
#import <WebRTC/RTCLogging.h>
#import <WebRTC/RTCMacros.h>
#import <WebRTC/RTCMutableI420Buffer.h>
#import <WebRTC/RTCMutableYUVPlanarBuffer.h>
#import <WebRTC/RTCSSLCertificateVerifier.h>
#import <WebRTC/RTCVideoCapturer.h>
#import <WebRTC/RTCVideoCodecInfo.h>
#import <WebRTC/RTCVideoDecoder.h>
#import <WebRTC/RTCVideoDecoderFactory.h>
#import <WebRTC/RTCVideoEncoder.h>
#import <WebRTC/RTCVideoEncoderFactory.h>
#import <WebRTC/RTCVideoEncoderQpThresholds.h>
#import <WebRTC/RTCVideoEncoderSettings.h>
#import <WebRTC/RTCVideoFrame.h>
#import <WebRTC/RTCVideoFrameBuffer.h>
#import <WebRTC/RTCVideoRenderer.h>
#import <WebRTC/RTCYUVPlanarBuffer.h>
#import <WebRTC/RTCAudioDevice.h>
#import <WebRTC/RTCAudioSession.h>
#import <WebRTC/RTCAudioSessionConfiguration.h>
#import <WebRTC/RTCCameraVideoCapturer.h>
#import <WebRTC/RTCFileVideoCapturer.h>
#import <WebRTC/RTCNetworkMonitor.h>
#import <WebRTC/RTCMTLVideoView.h>
#import <WebRTC/RTCEAGLVideoView.h>
#import <WebRTC/RTCVideoViewShading.h>
#import <WebRTC/RTCCodecSpecificInfoH264.h>
#import <WebRTC/RTCDefaultVideoDecoderFactory.h>
#import <WebRTC/RTCDefaultVideoEncoderFactory.h>
#import <WebRTC/RTCH264ProfileLevelId.h>
#import <WebRTC/RTCVideoDecoderFactoryH264.h>
#import <WebRTC/RTCVideoDecoderH264.h>
#import <WebRTC/RTCVideoEncoderFactoryH264.h>
#import <WebRTC/RTCVideoEncoderH264.h>
#import <WebRTC/RTCCVPixelBuffer.h>
#import <WebRTC/RTCCameraPreviewView.h>
#import <WebRTC/RTCDispatcher.h>
#import <WebRTC/UIDevice+RTCDevice.h>
#import <WebRTC/RTCAudioSource.h>
#import <WebRTC/RTCAudioTrack.h>
#import <WebRTC/RTCConfiguration.h>
#import <WebRTC/RTCDataChannel.h>
#import <WebRTC/RTCDataChannelConfiguration.h>
#import <WebRTC/RTCFieldTrials.h>
#import <WebRTC/RTCIceCandidate.h>
#import <WebRTC/RTCIceCandidateErrorEvent.h>
#import <WebRTC/RTCIceServer.h>
#import <WebRTC/RTCLegacyStatsReport.h>
#import <WebRTC/RTCMediaConstraints.h>
#import <WebRTC/RTCMediaSource.h>
#import <WebRTC/RTCMediaStream.h>
#import <WebRTC/RTCMediaStreamTrack.h>
#import <WebRTC/RTCMetrics.h>
#import <WebRTC/RTCMetricsSampleInfo.h>
#import <WebRTC/RTCPeerConnection.h>
#import <WebRTC/RTCPeerConnectionFactory.h>
#import <WebRTC/RTCPeerConnectionFactoryOptions.h>
#import <WebRTC/RTCRtcpParameters.h>
#import <WebRTC/RTCRtpCodecParameters.h>
#import <WebRTC/RTCRtpEncodingParameters.h>
#import <WebRTC/RTCRtpHeaderExtension.h>
#import <WebRTC/RTCRtpParameters.h>
#import <WebRTC/RTCRtpReceiver.h>
#import <WebRTC/RTCRtpSender.h>
#import <WebRTC/RTCRtpTransceiver.h>
#import <WebRTC/RTCDtmfSender.h>
#import <WebRTC/RTCSSLAdapter.h>
#import <WebRTC/RTCSessionDescription.h>
#import <WebRTC/RTCStatisticsReport.h>
#import <WebRTC/RTCTracing.h>
#import <WebRTC/RTCCertificate.h>
#import <WebRTC/RTCCryptoOptions.h>
#import <WebRTC/RTCVideoSource.h>
#import <WebRTC/RTCVideoTrack.h>
#import <WebRTC/RTCVideoCodecConstants.h>
#import <WebRTC/RTCVideoDecoderVP8.h>
#import <WebRTC/RTCVideoDecoderVP9.h>
#import <WebRTC/RTCVideoDecoderAV1.h>
#import <WebRTC/RTCVideoEncoderVP8.h>
#import <WebRTC/RTCVideoEncoderVP9.h>
#import <WebRTC/RTCVideoEncoderAV1.h>
#import <WebRTC/RTCNativeI420Buffer.h>
#import <WebRTC/RTCNativeMutableI420Buffer.h>
#import <WebRTC/RTCCallbackLogger.h>
#import <WebRTC/RTCFileLogger.h>

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framework module WebRTC {
umbrella header "WebRTC.h"
export *
module * { export * }
}

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/*
* Copyright 2022 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 <AudioUnit/AudioUnit.h>
#import <Foundation/Foundation.h>
#import <WebRTC/RTCMacros.h>
NS_ASSUME_NONNULL_BEGIN
typedef OSStatus (^RTC_OBJC_TYPE(RTCAudioDeviceGetPlayoutDataBlock))(
AudioUnitRenderActionFlags *_Nonnull actionFlags,
const AudioTimeStamp *_Nonnull timestamp,
NSInteger inputBusNumber,
UInt32 frameCount,
AudioBufferList *_Nonnull outputData);
typedef OSStatus (^RTC_OBJC_TYPE(RTCAudioDeviceRenderRecordedDataBlock))(
AudioUnitRenderActionFlags *_Nonnull actionFlags,
const AudioTimeStamp *_Nonnull timestamp,
NSInteger inputBusNumber,
UInt32 frameCount,
AudioBufferList *_Nonnull inputData,
void *_Nullable renderContext);
typedef OSStatus (^RTC_OBJC_TYPE(RTCAudioDeviceDeliverRecordedDataBlock))(
AudioUnitRenderActionFlags *_Nonnull actionFlags,
const AudioTimeStamp *_Nonnull timestamp,
NSInteger inputBusNumber,
UInt32 frameCount,
const AudioBufferList *_Nullable inputData,
void *_Nullable renderContext,
NS_NOESCAPE RTC_OBJC_TYPE(RTCAudioDeviceRenderRecordedDataBlock) _Nullable renderBlock);
/**
* Delegate object provided by native ADM during RTCAudioDevice initialization.
* Provides blocks to poll playback audio samples from native ADM and to feed
* recorded audio samples into native ADM.
*/
RTC_OBJC_EXPORT @protocol RTC_OBJC_TYPE
(RTCAudioDeviceDelegate)<NSObject>
/**
* Implementation of RTCAudioSource should call this block to feed recorded PCM (16-bit integer)
* into native ADM. Stereo data is expected to be interleaved starting with the left channel.
* Either `inputData` with pre-filled audio data must be provided during block
* call or `renderBlock` must be provided which must fill provided audio buffer with recorded
* samples.
*
* NOTE: Implementation of RTCAudioDevice is expected to call the block on the same thread until
* `notifyAudioInterrupted` is called. When `notifyAudioInterrupted` is called implementation
* can call the block on a different thread.
*/
@property(readonly, nonnull)
RTC_OBJC_TYPE(RTCAudioDeviceDeliverRecordedDataBlock) deliverRecordedData;
/**
* Provides input sample rate preference as it preferred by native ADM.
*/
@property(readonly) double preferredInputSampleRate;
/**
* Provides input IO buffer duration preference as it preferred by native ADM.
*/
@property(readonly) NSTimeInterval preferredInputIOBufferDuration;
/**
* Provides output sample rate preference as it preferred by native ADM.
*/
@property(readonly) double preferredOutputSampleRate;
/**
* Provides output IO buffer duration preference as it preferred by native ADM.
*/
@property(readonly) NSTimeInterval preferredOutputIOBufferDuration;
/**
* Implementation of RTCAudioDevice should call this block to request PCM (16-bit integer)
* from native ADM to play. Stereo data is interleaved starting with the left channel.
*
* NOTE: Implementation of RTCAudioDevice is expected to invoke of this block on the
* same thread until `notifyAudioInterrupted` is called. When `notifyAudioInterrupted` is called
* implementation can call the block from a different thread.
*/
@property(readonly, nonnull) RTC_OBJC_TYPE(RTCAudioDeviceGetPlayoutDataBlock) getPlayoutData;
/**
* Notifies native ADM that some of the audio input parameters of RTCAudioDevice like
* samle rate and/or IO buffer duration and/or IO latency had possibly changed.
* Native ADM will adjust its audio input buffer to match current parameters of audio device.
*
* NOTE: Must be called within block executed via `dispatchAsync` or `dispatchSync`.
*/
- (void)notifyAudioInputParametersChange;
/**
* Notifies native ADM that some of the audio output parameters of RTCAudioDevice like
* samle rate and/or IO buffer duration and/or IO latency had possibly changed.
* Native ADM will adjust its audio output buffer to match current parameters of audio device.
*
* NOTE: Must be called within block executed via `dispatchAsync` or `dispatchSync`.
*/
- (void)notifyAudioOutputParametersChange;
/**
* Notifies native ADM that audio input is interrupted and further audio playout
* and recording might happen on a different thread.
*
* NOTE: Must be called within block executed via `dispatchAsync` or `dispatchSync`.
*/
- (void)notifyAudioInputInterrupted;
/**
* Notifies native ADM that audio output is interrupted and further audio playout
* and recording might happen on a different thread.
*
* NOTE: Must be called within block executed via `dispatchAsync` or `dispatchSync`.
*/
- (void)notifyAudioOutputInterrupted;
/**
* Asynchronously execute block of code within the context of
* thread which owns native ADM.
*
* NOTE: Intended to be used to invoke `notifyAudioInputParametersChange`,
* `notifyAudioOutputParametersChange`, `notifyAudioInputInterrupted`,
* `notifyAudioOutputInterrupted` on native ADM thread.
* Also could be used by `RTCAudioDevice` implementation to tie
* mutations of underlying audio objects (AVAudioEngine, AudioUnit, etc)
* to the native ADM thread. Could be useful to handle events like audio route change, which
* could lead to audio parameters change.
*/
- (void)dispatchAsync:(dispatch_block_t)block;
/**
* Synchronously execute block of code within the context of
* thread which owns native ADM. Allows reentrancy.
*
* NOTE: Intended to be used to invoke `notifyAudioInputParametersChange`,
* `notifyAudioOutputParametersChange`, `notifyAudioInputInterrupted`,
* `notifyAudioOutputInterrupted` on native ADM thread and make sure
* aforementioned is completed before `dispatchSync` returns. Could be useful
* when implementation of `RTCAudioDevice` tie mutation to underlying audio objects (AVAudioEngine,
* AudioUnit, etc) to own thread to satisfy requirement that native ADM audio parameters
* must be kept in sync with current audio parameters before audio is actually played or recorded.
*/
- (void)dispatchSync:(dispatch_block_t)block;
@end
/**
* Protocol to abstract platform specific ways to implement playback and recording.
*
* NOTE: All the members of protocol are called by native ADM from the same thread
* between calls to `initializeWithDelegate` and `terminate`.
* NOTE: Implementation is fully responsible for configuring application's AVAudioSession.
* An example implementation of RTCAudioDevice: https://github.com/mstyura/RTCAudioDevice
* TODO(yura.yaroshevich): Implement custom RTCAudioDevice for AppRTCMobile demo app.
*/
RTC_OBJC_EXPORT @protocol RTC_OBJC_TYPE
(RTCAudioDevice)<NSObject>
/**
* Indicates current sample rate of audio recording. Changes to this property
* must be notified back to native ADM via `-[RTCAudioDeviceDelegate
* notifyAudioParametersChange]`.
*/
@property(readonly) double deviceInputSampleRate;
/**
* Indicates current size of record buffer. Changes to this property
* must be notified back to native ADM via `-[RTCAudioDeviceDelegate notifyAudioParametersChange]`.
*/
@property(readonly) NSTimeInterval inputIOBufferDuration;
/**
* Indicates current number of recorded audio channels. Changes to this property
* must be notified back to native ADM via `-[RTCAudioDeviceDelegate notifyAudioParametersChange]`.
*/
@property(readonly) NSInteger inputNumberOfChannels;
/**
* Indicates current input latency
*/
@property(readonly) NSTimeInterval inputLatency;
/**
* Indicates current sample rate of audio playback. Changes to this property
* must be notified back to native ADM via `-[RTCAudioDeviceDelegate notifyAudioParametersChange]`.
*/
@property(readonly) double deviceOutputSampleRate;
/**
* Indicates current size of playback buffer. Changes to this property
* must be notified back to native ADM via `-[RTCAudioDeviceDelegate notifyAudioParametersChange]`.
*/
@property(readonly) NSTimeInterval outputIOBufferDuration;
/**
* Indicates current number of playback audio channels. Changes to this property
* must be notified back to WebRTC via `[RTCAudioDeviceDelegate notifyAudioParametersChange]`.
*/
@property(readonly) NSInteger outputNumberOfChannels;
/**
* Indicates current output latency
*/
@property(readonly) NSTimeInterval outputLatency;
/**
* Indicates if invocation of `initializeWithDelegate` required before usage of RTCAudioDevice.
* YES indicates that `initializeWithDelegate` was called earlier without subsequent call to
* `terminate`. NO indicates that either `initializeWithDelegate` not called or `terminate` called.
*/
@property(readonly) BOOL isInitialized;
/**
* Initializes RTCAudioDevice with RTCAudioDeviceDelegate.
* Implementation must return YES if RTCAudioDevice initialized successfully and NO otherwise.
*/
- (BOOL)initializeWithDelegate:(id<RTC_OBJC_TYPE(RTCAudioDeviceDelegate)>)delegate;
/**
* De-initializes RTCAudioDevice. Implementation should forget about `delegate` provided in
* `initializeWithDelegate`.
*/
- (BOOL)terminateDevice;
/**
* Property to indicate if `initializePlayout` call required before invocation of `startPlayout`.
* YES indicates that `initializePlayout` was successfully invoked earlier or not necessary,
* NO indicates that `initializePlayout` invocation required.
*/
@property(readonly) BOOL isPlayoutInitialized;
/**
* Prepares RTCAudioDevice to play audio.
* Called by native ADM before invocation of `startPlayout`.
* Implementation is expected to return YES in case of successful playout initialization and NO
* otherwise.
*/
- (BOOL)initializePlayout;
/**
* Property to indicate if RTCAudioDevice should be playing according to
* earlier calls of `startPlayout` and `stopPlayout`.
*/
@property(readonly) BOOL isPlaying;
/**
* Method is called when native ADM wants to play audio.
* Implementation is expected to return YES if playback start request
* successfully handled and NO otherwise.
*/
- (BOOL)startPlayout;
/**
* Method is called when native ADM no longer needs to play audio.
* Implementation is expected to return YES if playback stop request
* successfully handled and NO otherwise.
*/
- (BOOL)stopPlayout;
/**
* Property to indicate if `initializeRecording` call required before usage of `startRecording`.
* YES indicates that `initializeRecording` was successfully invoked earlier or not necessary,
* NO indicates that `initializeRecording` invocation required.
*/
@property(readonly) BOOL isRecordingInitialized;
/**
* Prepares RTCAudioDevice to record audio.
* Called by native ADM before invocation of `startRecording`.
* Implementation may use this method to prepare resources required to record audio.
* Implementation is expected to return YES in case of successful record initialization and NO
* otherwise.
*/
- (BOOL)initializeRecording;
/**
* Property to indicate if RTCAudioDevice should record audio according to
* earlier calls to `startRecording` and `stopRecording`.
*/
@property(readonly) BOOL isRecording;
/**
* Method is called when native ADM wants to record audio.
* Implementation is expected to return YES if recording start request
* successfully handled and NO otherwise.
*/
- (BOOL)startRecording;
/**
* Method is called when native ADM no longer needs to record audio.
* Implementation is expected to return YES if recording stop request
* successfully handled and NO otherwise.
*/
- (BOOL)stopRecording;
@end
NS_ASSUME_NONNULL_END

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/*
* 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 <WebRTC/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:(AVAudioSessionCategory)category
mode:(AVAudioSessionMode)mode
options:(AVAudioSessionCategoryOptions)options
error:(NSError **)outError;
- (BOOL)setCategory:(AVAudioSessionCategory)category
withOptions:(AVAudioSessionCategoryOptions)options
error:(NSError **)outError;
- (BOOL)setMode:(AVAudioSessionMode)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;
@end
NS_ASSUME_NONNULL_END

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