Obj-C SDK Cleanup

This CL separates the files under sdk/objc into logical directories, replacing
the previous file layout under Framework/.

A long term goal is to have some system set up to generate the files under
sdk/objc/api (the PeerConnection API wrappers) from the C++ code. In the shorter
term the goal is to abstract out shared concepts from these classes in order to
make them as uniform as possible.

The separation into base/, components/, and helpers/ are to differentiate between
the base layer's common protocols, various utilities and the actual platform
specific components.

The old directory layout that resembled a framework's internal layout is not
necessary, since it is generated by the framework target when building it.

Bug: webrtc:9627
Change-Id: Ib084fd83f050ae980649ca99e841f4fb0580bd8f
Reviewed-on: https://webrtc-review.googlesource.com/94142
Reviewed-by: Kári Helgason <kthelgason@webrtc.org>
Reviewed-by: Mirko Bonadei <mbonadei@webrtc.org>
Reviewed-by: Rasmus Brandt <brandtr@webrtc.org>
Reviewed-by: Henrik Andreassson <henrika@webrtc.org>
Commit-Queue: Anders Carlsson <andersc@webrtc.org>
Cr-Commit-Position: refs/heads/master@{#24493}
This commit is contained in:
Anders Carlsson
2018-08-30 09:30:29 +02:00
committed by Commit Bot
parent 9ea5765f78
commit 7bca8ca4e2
470 changed files with 7255 additions and 5258 deletions

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@ -1,32 +0,0 @@
/*
* 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 "WebRTC/RTCAudioSource.h"
#import "RTCMediaSource+Private.h"
@interface RTCAudioSource ()
/**
* The AudioSourceInterface object passed to this RTCAudioSource during
* construction.
*/
@property(nonatomic, readonly) rtc::scoped_refptr<webrtc::AudioSourceInterface> nativeAudioSource;
/** Initialize an RTCAudioSource from a native AudioSourceInterface. */
- (instancetype)initWithFactory:(RTCPeerConnectionFactory*)factory
nativeAudioSource:(rtc::scoped_refptr<webrtc::AudioSourceInterface>)nativeAudioSource
NS_DESIGNATED_INITIALIZER;
- (instancetype)initWithFactory:(RTCPeerConnectionFactory*)factory
nativeMediaSource:(rtc::scoped_refptr<webrtc::MediaSourceInterface>)nativeMediaSource
type:(RTCMediaSourceType)type NS_UNAVAILABLE;
@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 "RTCAudioSource+Private.h"
#include "rtc_base/checks.h"
@implementation RTCAudioSource {
}
@synthesize volume = _volume;
@synthesize nativeAudioSource = _nativeAudioSource;
- (instancetype)initWithFactory:(RTCPeerConnectionFactory *)factory
nativeAudioSource:
(rtc::scoped_refptr<webrtc::AudioSourceInterface>)nativeAudioSource {
RTC_DCHECK(factory);
RTC_DCHECK(nativeAudioSource);
if (self = [super initWithFactory:factory
nativeMediaSource:nativeAudioSource
type:RTCMediaSourceTypeAudio]) {
_nativeAudioSource = nativeAudioSource;
}
return self;
}
- (instancetype)initWithFactory:(RTCPeerConnectionFactory *)factory
nativeMediaSource:(rtc::scoped_refptr<webrtc::MediaSourceInterface>)nativeMediaSource
type:(RTCMediaSourceType)type {
RTC_NOTREACHED();
return nil;
}
- (NSString *)description {
NSString *stateString = [[self class] stringForState:self.state];
return [NSString stringWithFormat:@"RTCAudioSource( %p ): %@", self, stateString];
}
- (void)setVolume:(double)volume {
_volume = volume;
_nativeAudioSource->SetVolume(volume);
}
@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/RTCAudioTrack.h"
#include "api/mediastreaminterface.h"
NS_ASSUME_NONNULL_BEGIN
@class RTCPeerConnectionFactory;
@interface RTCAudioTrack ()
/** AudioTrackInterface created or passed in at construction. */
@property(nonatomic, readonly) rtc::scoped_refptr<webrtc::AudioTrackInterface> nativeAudioTrack;
/** Initialize an RTCAudioTrack with an id. */
- (instancetype)initWithFactory:(RTCPeerConnectionFactory *)factory
source:(RTCAudioSource *)source
trackId:(NSString *)trackId;
@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 "RTCAudioTrack+Private.h"
#import "NSString+StdString.h"
#import "RTCAudioSource+Private.h"
#import "RTCMediaStreamTrack+Private.h"
#import "RTCPeerConnectionFactory+Private.h"
#include "rtc_base/checks.h"
@implementation RTCAudioTrack
@synthesize source = _source;
- (instancetype)initWithFactory:(RTCPeerConnectionFactory *)factory
source:(RTCAudioSource *)source
trackId:(NSString *)trackId {
RTC_DCHECK(factory);
RTC_DCHECK(source);
RTC_DCHECK(trackId.length);
std::string nativeId = [NSString stdStringForString:trackId];
rtc::scoped_refptr<webrtc::AudioTrackInterface> track =
factory.nativeFactory->CreateAudioTrack(nativeId, source.nativeAudioSource);
if (self = [self initWithFactory:factory nativeTrack:track type:RTCMediaStreamTrackTypeAudio]) {
_source = source;
}
return self;
}
- (instancetype)initWithFactory:(RTCPeerConnectionFactory *)factory
nativeTrack:(rtc::scoped_refptr<webrtc::MediaStreamTrackInterface>)nativeTrack
type:(RTCMediaStreamTrackType)type {
NSParameterAssert(factory);
NSParameterAssert(nativeTrack);
NSParameterAssert(type == RTCMediaStreamTrackTypeAudio);
return [super initWithFactory:factory nativeTrack:nativeTrack type:type];
}
- (RTCAudioSource *)source {
if (!_source) {
rtc::scoped_refptr<webrtc::AudioSourceInterface> source =
self.nativeAudioTrack->GetSource();
if (source) {
_source =
[[RTCAudioSource alloc] initWithFactory:self.factory nativeAudioSource:source.get()];
}
}
return _source;
}
#pragma mark - Private
- (rtc::scoped_refptr<webrtc::AudioTrackInterface>)nativeAudioTrack {
return static_cast<webrtc::AudioTrackInterface *>(self.nativeTrack.get());
}
@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/RTCCameraVideoCapturer.h"
#import "WebRTC/RTCLogging.h"
#import "WebRTC/RTCVideoFrameBuffer.h"
#if TARGET_OS_IPHONE
#import "WebRTC/UIDevice+RTCDevice.h"
#endif
#import "AVCaptureSession+DevicePosition.h"
#import "RTCDispatcher+Private.h"
const int64_t kNanosecondsPerSecond = 1000000000;
@interface RTCCameraVideoCapturer ()<AVCaptureVideoDataOutputSampleBufferDelegate>
@property(nonatomic, readonly) dispatch_queue_t frameQueue;
@end
@implementation RTCCameraVideoCapturer {
AVCaptureVideoDataOutput *_videoDataOutput;
AVCaptureSession *_captureSession;
AVCaptureDevice *_currentDevice;
FourCharCode _preferredOutputPixelFormat;
FourCharCode _outputPixelFormat;
BOOL _hasRetriedOnFatalError;
BOOL _isRunning;
// Will the session be running once all asynchronous operations have been completed?
BOOL _willBeRunning;
RTCVideoRotation _rotation;
#if TARGET_OS_IPHONE
UIDeviceOrientation _orientation;
#endif
}
@synthesize frameQueue = _frameQueue;
@synthesize captureSession = _captureSession;
- (instancetype)init {
return [self initWithDelegate:nil captureSession:[[AVCaptureSession alloc] init]];
}
- (instancetype)initWithDelegate:(__weak id<RTCVideoCapturerDelegate>)delegate {
return [self initWithDelegate:delegate captureSession:[[AVCaptureSession alloc] init]];
}
// This initializer is used for testing.
- (instancetype)initWithDelegate:(__weak id<RTCVideoCapturerDelegate>)delegate
captureSession:(AVCaptureSession *)captureSession {
if (self = [super initWithDelegate:delegate]) {
// Create the capture session and all relevant inputs and outputs. We need
// to do this in init because the application may want the capture session
// before we start the capturer for e.g. AVCapturePreviewLayer. All objects
// created here are retained until dealloc and never recreated.
if (![self setupCaptureSession:captureSession]) {
return nil;
}
NSNotificationCenter *center = [NSNotificationCenter defaultCenter];
#if TARGET_OS_IPHONE
_orientation = UIDeviceOrientationPortrait;
_rotation = RTCVideoRotation_90;
[center addObserver:self
selector:@selector(deviceOrientationDidChange:)
name:UIDeviceOrientationDidChangeNotification
object:nil];
[center addObserver:self
selector:@selector(handleCaptureSessionInterruption:)
name:AVCaptureSessionWasInterruptedNotification
object:_captureSession];
[center addObserver:self
selector:@selector(handleCaptureSessionInterruptionEnded:)
name:AVCaptureSessionInterruptionEndedNotification
object:_captureSession];
[center addObserver:self
selector:@selector(handleApplicationDidBecomeActive:)
name:UIApplicationDidBecomeActiveNotification
object:[UIApplication sharedApplication]];
#endif
[center addObserver:self
selector:@selector(handleCaptureSessionRuntimeError:)
name:AVCaptureSessionRuntimeErrorNotification
object:_captureSession];
[center addObserver:self
selector:@selector(handleCaptureSessionDidStartRunning:)
name:AVCaptureSessionDidStartRunningNotification
object:_captureSession];
[center addObserver:self
selector:@selector(handleCaptureSessionDidStopRunning:)
name:AVCaptureSessionDidStopRunningNotification
object:_captureSession];
}
return self;
}
- (void)dealloc {
NSAssert(
!_willBeRunning,
@"Session was still running in RTCCameraVideoCapturer dealloc. Forgot to call stopCapture?");
[[NSNotificationCenter defaultCenter] removeObserver:self];
}
+ (NSArray<AVCaptureDevice *> *)captureDevices {
#if defined(WEBRTC_IOS) && defined(__IPHONE_10_0) && \
__IPHONE_OS_VERSION_MIN_REQUIRED >= __IPHONE_10_0
AVCaptureDeviceDiscoverySession *session = [AVCaptureDeviceDiscoverySession
discoverySessionWithDeviceTypes:@[ AVCaptureDeviceTypeBuiltInWideAngleCamera ]
mediaType:AVMediaTypeVideo
position:AVCaptureDevicePositionUnspecified];
return session.devices;
#else
return [AVCaptureDevice devicesWithMediaType:AVMediaTypeVideo];
#endif
}
+ (NSArray<AVCaptureDeviceFormat *> *)supportedFormatsForDevice:(AVCaptureDevice *)device {
// Support opening the device in any format. We make sure it's converted to a format we
// can handle, if needed, in the method `-setupVideoDataOutput`.
return device.formats;
}
- (FourCharCode)preferredOutputPixelFormat {
return _preferredOutputPixelFormat;
}
- (void)startCaptureWithDevice:(AVCaptureDevice *)device
format:(AVCaptureDeviceFormat *)format
fps:(NSInteger)fps {
[self startCaptureWithDevice:device format:format fps:fps completionHandler:nil];
}
- (void)stopCapture {
[self stopCaptureWithCompletionHandler:nil];
}
- (void)startCaptureWithDevice:(AVCaptureDevice *)device
format:(AVCaptureDeviceFormat *)format
fps:(NSInteger)fps
completionHandler:(nullable void (^)(NSError *))completionHandler {
_willBeRunning = YES;
[RTCDispatcher
dispatchAsyncOnType:RTCDispatcherTypeCaptureSession
block:^{
RTCLogInfo("startCaptureWithDevice %@ @ %ld fps", format, (long)fps);
#if TARGET_OS_IPHONE
[[UIDevice currentDevice] beginGeneratingDeviceOrientationNotifications];
#endif
_currentDevice = device;
NSError *error = nil;
if (![_currentDevice lockForConfiguration:&error]) {
RTCLogError(
@"Failed to lock device %@. Error: %@", _currentDevice, error.userInfo);
if (completionHandler) {
completionHandler(error);
}
_willBeRunning = NO;
return;
}
[self reconfigureCaptureSessionInput];
[self updateOrientation];
[self updateDeviceCaptureFormat:format fps:fps];
[self updateVideoDataOutputPixelFormat:format];
[_captureSession startRunning];
[_currentDevice unlockForConfiguration];
_isRunning = YES;
if (completionHandler) {
completionHandler(nil);
}
}];
}
- (void)stopCaptureWithCompletionHandler:(nullable void (^)(void))completionHandler {
_willBeRunning = NO;
[RTCDispatcher
dispatchAsyncOnType:RTCDispatcherTypeCaptureSession
block:^{
RTCLogInfo("Stop");
_currentDevice = nil;
for (AVCaptureDeviceInput *oldInput in [_captureSession.inputs copy]) {
[_captureSession removeInput:oldInput];
}
[_captureSession stopRunning];
#if TARGET_OS_IPHONE
[[UIDevice currentDevice] endGeneratingDeviceOrientationNotifications];
#endif
_isRunning = NO;
if (completionHandler) {
completionHandler();
}
}];
}
#pragma mark iOS notifications
#if TARGET_OS_IPHONE
- (void)deviceOrientationDidChange:(NSNotification *)notification {
[RTCDispatcher dispatchAsyncOnType:RTCDispatcherTypeCaptureSession
block:^{
[self updateOrientation];
}];
}
#endif
#pragma mark AVCaptureVideoDataOutputSampleBufferDelegate
- (void)captureOutput:(AVCaptureOutput *)captureOutput
didOutputSampleBuffer:(CMSampleBufferRef)sampleBuffer
fromConnection:(AVCaptureConnection *)connection {
NSParameterAssert(captureOutput == _videoDataOutput);
if (CMSampleBufferGetNumSamples(sampleBuffer) != 1 || !CMSampleBufferIsValid(sampleBuffer) ||
!CMSampleBufferDataIsReady(sampleBuffer)) {
return;
}
CVPixelBufferRef pixelBuffer = CMSampleBufferGetImageBuffer(sampleBuffer);
if (pixelBuffer == nil) {
return;
}
#if TARGET_OS_IPHONE
// Default to portrait orientation on iPhone.
BOOL usingFrontCamera = NO;
// Check the image's EXIF for the camera the image came from as the image could have been
// delayed as we set alwaysDiscardsLateVideoFrames to NO.
AVCaptureDevicePosition cameraPosition =
[AVCaptureSession devicePositionForSampleBuffer:sampleBuffer];
if (cameraPosition != AVCaptureDevicePositionUnspecified) {
usingFrontCamera = AVCaptureDevicePositionFront == cameraPosition;
} else {
AVCaptureDeviceInput *deviceInput =
(AVCaptureDeviceInput *)((AVCaptureInputPort *)connection.inputPorts.firstObject).input;
usingFrontCamera = AVCaptureDevicePositionFront == deviceInput.device.position;
}
switch (_orientation) {
case UIDeviceOrientationPortrait:
_rotation = RTCVideoRotation_90;
break;
case UIDeviceOrientationPortraitUpsideDown:
_rotation = RTCVideoRotation_270;
break;
case UIDeviceOrientationLandscapeLeft:
_rotation = usingFrontCamera ? RTCVideoRotation_180 : RTCVideoRotation_0;
break;
case UIDeviceOrientationLandscapeRight:
_rotation = usingFrontCamera ? RTCVideoRotation_0 : RTCVideoRotation_180;
break;
case UIDeviceOrientationFaceUp:
case UIDeviceOrientationFaceDown:
case UIDeviceOrientationUnknown:
// Ignore.
break;
}
#else
// No rotation on Mac.
_rotation = RTCVideoRotation_0;
#endif
RTCCVPixelBuffer *rtcPixelBuffer = [[RTCCVPixelBuffer alloc] initWithPixelBuffer:pixelBuffer];
int64_t timeStampNs = CMTimeGetSeconds(CMSampleBufferGetPresentationTimeStamp(sampleBuffer)) *
kNanosecondsPerSecond;
RTCVideoFrame *videoFrame = [[RTCVideoFrame alloc] initWithBuffer:rtcPixelBuffer
rotation:_rotation
timeStampNs:timeStampNs];
[self.delegate capturer:self didCaptureVideoFrame:videoFrame];
}
- (void)captureOutput:(AVCaptureOutput *)captureOutput
didDropSampleBuffer:(CMSampleBufferRef)sampleBuffer
fromConnection:(AVCaptureConnection *)connection {
RTCLogError(@"Dropped sample buffer.");
}
#pragma mark - AVCaptureSession notifications
- (void)handleCaptureSessionInterruption:(NSNotification *)notification {
NSString *reasonString = nil;
#if TARGET_OS_IPHONE
NSNumber *reason = notification.userInfo[AVCaptureSessionInterruptionReasonKey];
if (reason) {
switch (reason.intValue) {
case AVCaptureSessionInterruptionReasonVideoDeviceNotAvailableInBackground:
reasonString = @"VideoDeviceNotAvailableInBackground";
break;
case AVCaptureSessionInterruptionReasonAudioDeviceInUseByAnotherClient:
reasonString = @"AudioDeviceInUseByAnotherClient";
break;
case AVCaptureSessionInterruptionReasonVideoDeviceInUseByAnotherClient:
reasonString = @"VideoDeviceInUseByAnotherClient";
break;
case AVCaptureSessionInterruptionReasonVideoDeviceNotAvailableWithMultipleForegroundApps:
reasonString = @"VideoDeviceNotAvailableWithMultipleForegroundApps";
break;
}
}
#endif
RTCLog(@"Capture session interrupted: %@", reasonString);
}
- (void)handleCaptureSessionInterruptionEnded:(NSNotification *)notification {
RTCLog(@"Capture session interruption ended.");
}
- (void)handleCaptureSessionRuntimeError:(NSNotification *)notification {
NSError *error = [notification.userInfo objectForKey:AVCaptureSessionErrorKey];
RTCLogError(@"Capture session runtime error: %@", error);
[RTCDispatcher dispatchAsyncOnType:RTCDispatcherTypeCaptureSession
block:^{
#if TARGET_OS_IPHONE
if (error.code == AVErrorMediaServicesWereReset) {
[self handleNonFatalError];
} else {
[self handleFatalError];
}
#else
[self handleFatalError];
#endif
}];
}
- (void)handleCaptureSessionDidStartRunning:(NSNotification *)notification {
RTCLog(@"Capture session started.");
[RTCDispatcher dispatchAsyncOnType:RTCDispatcherTypeCaptureSession
block:^{
// If we successfully restarted after an unknown error,
// allow future retries on fatal errors.
_hasRetriedOnFatalError = NO;
}];
}
- (void)handleCaptureSessionDidStopRunning:(NSNotification *)notification {
RTCLog(@"Capture session stopped.");
}
- (void)handleFatalError {
[RTCDispatcher
dispatchAsyncOnType:RTCDispatcherTypeCaptureSession
block:^{
if (!_hasRetriedOnFatalError) {
RTCLogWarning(@"Attempting to recover from fatal capture error.");
[self handleNonFatalError];
_hasRetriedOnFatalError = YES;
} else {
RTCLogError(@"Previous fatal error recovery failed.");
}
}];
}
- (void)handleNonFatalError {
[RTCDispatcher dispatchAsyncOnType:RTCDispatcherTypeCaptureSession
block:^{
RTCLog(@"Restarting capture session after error.");
if (_isRunning) {
[_captureSession startRunning];
}
}];
}
#if TARGET_OS_IPHONE
#pragma mark - UIApplication notifications
- (void)handleApplicationDidBecomeActive:(NSNotification *)notification {
[RTCDispatcher dispatchAsyncOnType:RTCDispatcherTypeCaptureSession
block:^{
if (_isRunning && !_captureSession.isRunning) {
RTCLog(@"Restarting capture session on active.");
[_captureSession startRunning];
}
}];
}
#endif // TARGET_OS_IPHONE
#pragma mark - Private
- (dispatch_queue_t)frameQueue {
if (!_frameQueue) {
_frameQueue =
dispatch_queue_create("org.webrtc.cameravideocapturer.video", DISPATCH_QUEUE_SERIAL);
dispatch_set_target_queue(_frameQueue,
dispatch_get_global_queue(DISPATCH_QUEUE_PRIORITY_HIGH, 0));
}
return _frameQueue;
}
- (BOOL)setupCaptureSession:(AVCaptureSession *)captureSession {
NSAssert(_captureSession == nil, @"Setup capture session called twice.");
_captureSession = captureSession;
#if defined(WEBRTC_IOS)
_captureSession.sessionPreset = AVCaptureSessionPresetInputPriority;
_captureSession.usesApplicationAudioSession = NO;
#endif
[self setupVideoDataOutput];
// Add the output.
if (![_captureSession canAddOutput:_videoDataOutput]) {
RTCLogError(@"Video data output unsupported.");
return NO;
}
[_captureSession addOutput:_videoDataOutput];
return YES;
}
- (void)setupVideoDataOutput {
NSAssert(_videoDataOutput == nil, @"Setup video data output called twice.");
AVCaptureVideoDataOutput *videoDataOutput = [[AVCaptureVideoDataOutput alloc] init];
// `videoDataOutput.availableVideoCVPixelFormatTypes` returns the pixel formats supported by the
// device with the most efficient output format first. Find the first format that we support.
NSSet<NSNumber *> *supportedPixelFormats = [RTCCVPixelBuffer supportedPixelFormats];
NSMutableOrderedSet *availablePixelFormats =
[NSMutableOrderedSet orderedSetWithArray:videoDataOutput.availableVideoCVPixelFormatTypes];
[availablePixelFormats intersectSet:supportedPixelFormats];
NSNumber *pixelFormat = availablePixelFormats.firstObject;
NSAssert(pixelFormat, @"Output device has no supported formats.");
_preferredOutputPixelFormat = [pixelFormat unsignedIntValue];
_outputPixelFormat = _preferredOutputPixelFormat;
videoDataOutput.videoSettings = @{(NSString *)kCVPixelBufferPixelFormatTypeKey : pixelFormat};
videoDataOutput.alwaysDiscardsLateVideoFrames = NO;
[videoDataOutput setSampleBufferDelegate:self queue:self.frameQueue];
_videoDataOutput = videoDataOutput;
}
- (void)updateVideoDataOutputPixelFormat:(AVCaptureDeviceFormat *)format {
FourCharCode mediaSubType = CMFormatDescriptionGetMediaSubType(format.formatDescription);
if (![[RTCCVPixelBuffer supportedPixelFormats] containsObject:@(mediaSubType)]) {
mediaSubType = _preferredOutputPixelFormat;
}
if (mediaSubType != _outputPixelFormat) {
_outputPixelFormat = mediaSubType;
_videoDataOutput.videoSettings =
@{ (NSString *)kCVPixelBufferPixelFormatTypeKey : @(mediaSubType) };
}
}
#pragma mark - Private, called inside capture queue
- (void)updateDeviceCaptureFormat:(AVCaptureDeviceFormat *)format fps:(NSInteger)fps {
NSAssert([RTCDispatcher isOnQueueForType:RTCDispatcherTypeCaptureSession],
@"updateDeviceCaptureFormat must be called on the capture queue.");
@try {
_currentDevice.activeFormat = format;
_currentDevice.activeVideoMinFrameDuration = CMTimeMake(1, fps);
} @catch (NSException *exception) {
RTCLogError(@"Failed to set active format!\n User info:%@", exception.userInfo);
return;
}
}
- (void)reconfigureCaptureSessionInput {
NSAssert([RTCDispatcher isOnQueueForType:RTCDispatcherTypeCaptureSession],
@"reconfigureCaptureSessionInput must be called on the capture queue.");
NSError *error = nil;
AVCaptureDeviceInput *input =
[AVCaptureDeviceInput deviceInputWithDevice:_currentDevice error:&error];
if (!input) {
RTCLogError(@"Failed to create front camera input: %@", error.localizedDescription);
return;
}
[_captureSession beginConfiguration];
for (AVCaptureDeviceInput *oldInput in [_captureSession.inputs copy]) {
[_captureSession removeInput:oldInput];
}
if ([_captureSession canAddInput:input]) {
[_captureSession addInput:input];
} else {
RTCLogError(@"Cannot add camera as an input to the session.");
}
[_captureSession commitConfiguration];
}
- (void)updateOrientation {
NSAssert([RTCDispatcher isOnQueueForType:RTCDispatcherTypeCaptureSession],
@"updateOrientation must be called on the capture queue.");
#if TARGET_OS_IPHONE
_orientation = [UIDevice currentDevice].orientation;
#endif
}
@end

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@ -1,70 +0,0 @@
/*
* 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 "WebRTC/RTCCertificate.h"
#import "WebRTC/RTCLogging.h"
#include "rtc_base/logging.h"
#include "rtc_base/rtccertificategenerator.h"
#include "rtc_base/sslidentity.h"
@implementation RTCCertificate
@synthesize private_key = _private_key;
@synthesize certificate = _certificate;
- (id)copyWithZone:(NSZone *)zone {
id copy = [[[self class] alloc] initWithPrivateKey:[self.private_key copyWithZone:zone]
certificate:[self.certificate copyWithZone:zone]];
return copy;
}
- (instancetype)initWithPrivateKey:(NSString *)private_key certificate:(NSString *)certificate {
if (self = [super init]) {
_private_key = [private_key copy];
_certificate = [certificate copy];
}
return self;
}
+ (nullable RTCCertificate *)generateCertificateWithParams:(NSDictionary *)params {
rtc::KeyType keyType = rtc::KT_ECDSA;
NSString *keyTypeString = [params valueForKey:@"name"];
if (keyTypeString && [keyTypeString isEqualToString:@"RSASSA-PKCS1-v1_5"]) {
keyType = rtc::KT_RSA;
}
NSNumber *expires = [params valueForKey:@"expires"];
rtc::scoped_refptr<rtc::RTCCertificate> cc_certificate = nullptr;
if (expires != nil) {
uint64_t expirationTimestamp = [expires unsignedLongLongValue];
cc_certificate = rtc::RTCCertificateGenerator::GenerateCertificate(rtc::KeyParams(keyType),
expirationTimestamp);
} else {
cc_certificate =
rtc::RTCCertificateGenerator::GenerateCertificate(rtc::KeyParams(keyType), absl::nullopt);
}
if (!cc_certificate) {
RTCLogError(@"Failed to generate certificate.");
return nullptr;
}
// grab PEMs and create an NS RTCCerticicate
rtc::RTCCertificatePEM pem = cc_certificate->ToPEM();
std::string pem_private_key = pem.private_key();
std::string pem_certificate = pem.certificate();
RTC_LOG(LS_INFO) << "CERT PEM ";
RTC_LOG(LS_INFO) << pem_certificate;
RTCCertificate *cert = [[RTCCertificate alloc] initWithPrivateKey:@(pem_private_key.c_str())
certificate:@(pem_certificate.c_str())];
return cert;
}
@end

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@ -8,21 +8,4 @@
* be found in the AUTHORS file in the root of the source tree.
*/
#import "WebRTC/RTCConfiguration.h"
#include "api/peerconnectioninterface.h"
NS_ASSUME_NONNULL_BEGIN
@interface RTCConfiguration ()
/** Optional TurnCustomizer.
* With this class one can modify outgoing TURN messages.
* The object passed in must remain valid until PeerConnection::Close() is
* called.
*/
@property(nonatomic, nullable) webrtc::TurnCustomizer* turnCustomizer;
@end
NS_ASSUME_NONNULL_END
#import "api/peerconnection/RTCConfiguration+Native.h"

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@ -1,78 +0,0 @@
/*
* 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/RTCConfiguration.h"
#include "api/peerconnectioninterface.h"
NS_ASSUME_NONNULL_BEGIN
@interface RTCConfiguration ()
+ (webrtc::PeerConnectionInterface::IceTransportsType)nativeTransportsTypeForTransportPolicy:
(RTCIceTransportPolicy)policy;
+ (RTCIceTransportPolicy)transportPolicyForTransportsType:
(webrtc::PeerConnectionInterface::IceTransportsType)nativeType;
+ (NSString *)stringForTransportPolicy:(RTCIceTransportPolicy)policy;
+ (webrtc::PeerConnectionInterface::BundlePolicy)nativeBundlePolicyForPolicy:
(RTCBundlePolicy)policy;
+ (RTCBundlePolicy)bundlePolicyForNativePolicy:
(webrtc::PeerConnectionInterface::BundlePolicy)nativePolicy;
+ (NSString *)stringForBundlePolicy:(RTCBundlePolicy)policy;
+ (webrtc::PeerConnectionInterface::RtcpMuxPolicy)nativeRtcpMuxPolicyForPolicy:
(RTCRtcpMuxPolicy)policy;
+ (RTCRtcpMuxPolicy)rtcpMuxPolicyForNativePolicy:
(webrtc::PeerConnectionInterface::RtcpMuxPolicy)nativePolicy;
+ (NSString *)stringForRtcpMuxPolicy:(RTCRtcpMuxPolicy)policy;
+ (webrtc::PeerConnectionInterface::TcpCandidatePolicy)nativeTcpCandidatePolicyForPolicy:
(RTCTcpCandidatePolicy)policy;
+ (RTCTcpCandidatePolicy)tcpCandidatePolicyForNativePolicy:
(webrtc::PeerConnectionInterface::TcpCandidatePolicy)nativePolicy;
+ (NSString *)stringForTcpCandidatePolicy:(RTCTcpCandidatePolicy)policy;
+ (webrtc::PeerConnectionInterface::CandidateNetworkPolicy)nativeCandidateNetworkPolicyForPolicy:
(RTCCandidateNetworkPolicy)policy;
+ (RTCCandidateNetworkPolicy)candidateNetworkPolicyForNativePolicy:
(webrtc::PeerConnectionInterface::CandidateNetworkPolicy)nativePolicy;
+ (NSString *)stringForCandidateNetworkPolicy:(RTCCandidateNetworkPolicy)policy;
+ (rtc::KeyType)nativeEncryptionKeyTypeForKeyType:(RTCEncryptionKeyType)keyType;
+ (webrtc::SdpSemantics)nativeSdpSemanticsForSdpSemantics:(RTCSdpSemantics)sdpSemantics;
+ (RTCSdpSemantics)sdpSemanticsForNativeSdpSemantics:(webrtc::SdpSemantics)sdpSemantics;
+ (NSString *)stringForSdpSemantics:(RTCSdpSemantics)sdpSemantics;
/**
* RTCConfiguration struct representation of this RTCConfiguration. This is
* needed to pass to the underlying C++ APIs.
*/
- (nullable webrtc::PeerConnectionInterface::RTCConfiguration *)createNativeConfiguration;
- (instancetype)initWithNativeConfiguration:
(const webrtc::PeerConnectionInterface::RTCConfiguration &)config NS_DESIGNATED_INITIALIZER;
@end
NS_ASSUME_NONNULL_END

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@ -1,460 +0,0 @@
/*
* 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 "RTCConfiguration+Private.h"
#include <memory>
#import "RTCConfiguration+Native.h"
#import "RTCIceServer+Private.h"
#import "RTCIntervalRange+Private.h"
#import "WebRTC/RTCLogging.h"
#include "rtc_base/rtccertificategenerator.h"
#include "rtc_base/sslidentity.h"
@implementation RTCConfiguration
@synthesize iceServers = _iceServers;
@synthesize certificate = _certificate;
@synthesize iceTransportPolicy = _iceTransportPolicy;
@synthesize bundlePolicy = _bundlePolicy;
@synthesize rtcpMuxPolicy = _rtcpMuxPolicy;
@synthesize tcpCandidatePolicy = _tcpCandidatePolicy;
@synthesize candidateNetworkPolicy = _candidateNetworkPolicy;
@synthesize continualGatheringPolicy = _continualGatheringPolicy;
@synthesize maxIPv6Networks = _maxIPv6Networks;
@synthesize disableLinkLocalNetworks = _disableLinkLocalNetworks;
@synthesize audioJitterBufferMaxPackets = _audioJitterBufferMaxPackets;
@synthesize audioJitterBufferFastAccelerate = _audioJitterBufferFastAccelerate;
@synthesize iceConnectionReceivingTimeout = _iceConnectionReceivingTimeout;
@synthesize iceBackupCandidatePairPingInterval =
_iceBackupCandidatePairPingInterval;
@synthesize keyType = _keyType;
@synthesize iceCandidatePoolSize = _iceCandidatePoolSize;
@synthesize shouldPruneTurnPorts = _shouldPruneTurnPorts;
@synthesize shouldPresumeWritableWhenFullyRelayed =
_shouldPresumeWritableWhenFullyRelayed;
@synthesize iceCheckMinInterval = _iceCheckMinInterval;
@synthesize iceRegatherIntervalRange = _iceRegatherIntervalRange;
@synthesize sdpSemantics = _sdpSemantics;
@synthesize turnCustomizer = _turnCustomizer;
@synthesize activeResetSrtpParams = _activeResetSrtpParams;
- (instancetype)init {
// Copy defaults.
webrtc::PeerConnectionInterface::RTCConfiguration config(
webrtc::PeerConnectionInterface::RTCConfigurationType::kAggressive);
return [self initWithNativeConfiguration:config];
}
- (instancetype)initWithNativeConfiguration:
(const webrtc::PeerConnectionInterface::RTCConfiguration &)config {
if (self = [super init]) {
NSMutableArray *iceServers = [NSMutableArray array];
for (const webrtc::PeerConnectionInterface::IceServer& server : config.servers) {
RTCIceServer *iceServer = [[RTCIceServer alloc] initWithNativeServer:server];
[iceServers addObject:iceServer];
}
_iceServers = iceServers;
if (!config.certificates.empty()) {
rtc::scoped_refptr<rtc::RTCCertificate> native_cert;
native_cert = config.certificates[0];
rtc::RTCCertificatePEM native_pem = native_cert->ToPEM();
_certificate =
[[RTCCertificate alloc] initWithPrivateKey:@(native_pem.private_key().c_str())
certificate:@(native_pem.certificate().c_str())];
}
_iceTransportPolicy =
[[self class] transportPolicyForTransportsType:config.type];
_bundlePolicy =
[[self class] bundlePolicyForNativePolicy:config.bundle_policy];
_rtcpMuxPolicy =
[[self class] rtcpMuxPolicyForNativePolicy:config.rtcp_mux_policy];
_tcpCandidatePolicy = [[self class] tcpCandidatePolicyForNativePolicy:
config.tcp_candidate_policy];
_candidateNetworkPolicy = [[self class]
candidateNetworkPolicyForNativePolicy:config.candidate_network_policy];
webrtc::PeerConnectionInterface::ContinualGatheringPolicy nativePolicy =
config.continual_gathering_policy;
_continualGatheringPolicy =
[[self class] continualGatheringPolicyForNativePolicy:nativePolicy];
_maxIPv6Networks = config.max_ipv6_networks;
_disableLinkLocalNetworks = config.disable_link_local_networks;
_audioJitterBufferMaxPackets = config.audio_jitter_buffer_max_packets;
_audioJitterBufferFastAccelerate = config.audio_jitter_buffer_fast_accelerate;
_iceConnectionReceivingTimeout = config.ice_connection_receiving_timeout;
_iceBackupCandidatePairPingInterval =
config.ice_backup_candidate_pair_ping_interval;
_keyType = RTCEncryptionKeyTypeECDSA;
_iceCandidatePoolSize = config.ice_candidate_pool_size;
_shouldPruneTurnPorts = config.prune_turn_ports;
_shouldPresumeWritableWhenFullyRelayed =
config.presume_writable_when_fully_relayed;
if (config.ice_check_min_interval) {
_iceCheckMinInterval =
[NSNumber numberWithInt:*config.ice_check_min_interval];
}
if (config.ice_regather_interval_range) {
const rtc::IntervalRange &nativeIntervalRange = config.ice_regather_interval_range.value();
_iceRegatherIntervalRange =
[[RTCIntervalRange alloc] initWithNativeIntervalRange:nativeIntervalRange];
}
_sdpSemantics = [[self class] sdpSemanticsForNativeSdpSemantics:config.sdp_semantics];
_turnCustomizer = config.turn_customizer;
_activeResetSrtpParams = config.active_reset_srtp_params;
}
return self;
}
- (NSString *)description {
static NSString *formatString =
@"RTCConfiguration: "
@"{\n%@\n%@\n%@\n%@\n%@\n%@\n%@\n%@\n%d\n%d\n%d\n%d\n%d\n%d\n%d\n%@\n%@\n%d\n%d\n%d\n}\n";
return [NSString
stringWithFormat:formatString,
_iceServers,
[[self class] stringForTransportPolicy:_iceTransportPolicy],
[[self class] stringForBundlePolicy:_bundlePolicy],
[[self class] stringForRtcpMuxPolicy:_rtcpMuxPolicy],
[[self class] stringForTcpCandidatePolicy:_tcpCandidatePolicy],
[[self class] stringForCandidateNetworkPolicy:_candidateNetworkPolicy],
[[self class] stringForContinualGatheringPolicy:_continualGatheringPolicy],
[[self class] stringForSdpSemantics:_sdpSemantics],
_audioJitterBufferMaxPackets,
_audioJitterBufferFastAccelerate,
_iceConnectionReceivingTimeout,
_iceBackupCandidatePairPingInterval,
_iceCandidatePoolSize,
_shouldPruneTurnPorts,
_shouldPresumeWritableWhenFullyRelayed,
_iceCheckMinInterval,
_iceRegatherIntervalRange,
_disableLinkLocalNetworks,
_maxIPv6Networks,
_activeResetSrtpParams];
}
#pragma mark - Private
- (webrtc::PeerConnectionInterface::RTCConfiguration *)
createNativeConfiguration {
std::unique_ptr<webrtc::PeerConnectionInterface::RTCConfiguration>
nativeConfig(new webrtc::PeerConnectionInterface::RTCConfiguration(
webrtc::PeerConnectionInterface::RTCConfigurationType::kAggressive));
for (RTCIceServer *iceServer in _iceServers) {
nativeConfig->servers.push_back(iceServer.nativeServer);
}
nativeConfig->type =
[[self class] nativeTransportsTypeForTransportPolicy:_iceTransportPolicy];
nativeConfig->bundle_policy =
[[self class] nativeBundlePolicyForPolicy:_bundlePolicy];
nativeConfig->rtcp_mux_policy =
[[self class] nativeRtcpMuxPolicyForPolicy:_rtcpMuxPolicy];
nativeConfig->tcp_candidate_policy =
[[self class] nativeTcpCandidatePolicyForPolicy:_tcpCandidatePolicy];
nativeConfig->candidate_network_policy = [[self class]
nativeCandidateNetworkPolicyForPolicy:_candidateNetworkPolicy];
nativeConfig->continual_gathering_policy = [[self class]
nativeContinualGatheringPolicyForPolicy:_continualGatheringPolicy];
nativeConfig->max_ipv6_networks = _maxIPv6Networks;
nativeConfig->disable_link_local_networks = _disableLinkLocalNetworks;
nativeConfig->audio_jitter_buffer_max_packets = _audioJitterBufferMaxPackets;
nativeConfig->audio_jitter_buffer_fast_accelerate =
_audioJitterBufferFastAccelerate ? true : false;
nativeConfig->ice_connection_receiving_timeout =
_iceConnectionReceivingTimeout;
nativeConfig->ice_backup_candidate_pair_ping_interval =
_iceBackupCandidatePairPingInterval;
rtc::KeyType keyType =
[[self class] nativeEncryptionKeyTypeForKeyType:_keyType];
if (_certificate != nullptr) {
// if offered a pemcert use it...
RTC_LOG(LS_INFO) << "Have configured cert - using it.";
std::string pem_private_key = [[_certificate private_key] UTF8String];
std::string pem_certificate = [[_certificate certificate] UTF8String];
rtc::RTCCertificatePEM pem = rtc::RTCCertificatePEM(pem_private_key, pem_certificate);
rtc::scoped_refptr<rtc::RTCCertificate> certificate = rtc::RTCCertificate::FromPEM(pem);
RTC_LOG(LS_INFO) << "Created cert from PEM strings.";
if (!certificate) {
RTC_LOG(LS_ERROR) << "Failed to generate certificate from PEM.";
return nullptr;
}
nativeConfig->certificates.push_back(certificate);
} else {
RTC_LOG(LS_INFO) << "Don't have configured cert.";
// Generate non-default certificate.
if (keyType != rtc::KT_DEFAULT) {
rtc::scoped_refptr<rtc::RTCCertificate> certificate =
rtc::RTCCertificateGenerator::GenerateCertificate(rtc::KeyParams(keyType),
absl::optional<uint64_t>());
if (!certificate) {
RTCLogError(@"Failed to generate certificate.");
return nullptr;
}
nativeConfig->certificates.push_back(certificate);
}
}
nativeConfig->ice_candidate_pool_size = _iceCandidatePoolSize;
nativeConfig->prune_turn_ports = _shouldPruneTurnPorts ? true : false;
nativeConfig->presume_writable_when_fully_relayed =
_shouldPresumeWritableWhenFullyRelayed ? true : false;
if (_iceCheckMinInterval != nil) {
nativeConfig->ice_check_min_interval = absl::optional<int>(_iceCheckMinInterval.intValue);
}
if (_iceRegatherIntervalRange != nil) {
std::unique_ptr<rtc::IntervalRange> nativeIntervalRange(
_iceRegatherIntervalRange.nativeIntervalRange);
nativeConfig->ice_regather_interval_range =
absl::optional<rtc::IntervalRange>(*nativeIntervalRange);
}
nativeConfig->sdp_semantics = [[self class] nativeSdpSemanticsForSdpSemantics:_sdpSemantics];
if (_turnCustomizer) {
nativeConfig->turn_customizer = _turnCustomizer;
}
nativeConfig->active_reset_srtp_params = _activeResetSrtpParams ? true : false;
return nativeConfig.release();
}
+ (webrtc::PeerConnectionInterface::IceTransportsType)
nativeTransportsTypeForTransportPolicy:(RTCIceTransportPolicy)policy {
switch (policy) {
case RTCIceTransportPolicyNone:
return webrtc::PeerConnectionInterface::kNone;
case RTCIceTransportPolicyRelay:
return webrtc::PeerConnectionInterface::kRelay;
case RTCIceTransportPolicyNoHost:
return webrtc::PeerConnectionInterface::kNoHost;
case RTCIceTransportPolicyAll:
return webrtc::PeerConnectionInterface::kAll;
}
}
+ (RTCIceTransportPolicy)transportPolicyForTransportsType:
(webrtc::PeerConnectionInterface::IceTransportsType)nativeType {
switch (nativeType) {
case webrtc::PeerConnectionInterface::kNone:
return RTCIceTransportPolicyNone;
case webrtc::PeerConnectionInterface::kRelay:
return RTCIceTransportPolicyRelay;
case webrtc::PeerConnectionInterface::kNoHost:
return RTCIceTransportPolicyNoHost;
case webrtc::PeerConnectionInterface::kAll:
return RTCIceTransportPolicyAll;
}
}
+ (NSString *)stringForTransportPolicy:(RTCIceTransportPolicy)policy {
switch (policy) {
case RTCIceTransportPolicyNone:
return @"NONE";
case RTCIceTransportPolicyRelay:
return @"RELAY";
case RTCIceTransportPolicyNoHost:
return @"NO_HOST";
case RTCIceTransportPolicyAll:
return @"ALL";
}
}
+ (webrtc::PeerConnectionInterface::BundlePolicy)nativeBundlePolicyForPolicy:
(RTCBundlePolicy)policy {
switch (policy) {
case RTCBundlePolicyBalanced:
return webrtc::PeerConnectionInterface::kBundlePolicyBalanced;
case RTCBundlePolicyMaxCompat:
return webrtc::PeerConnectionInterface::kBundlePolicyMaxCompat;
case RTCBundlePolicyMaxBundle:
return webrtc::PeerConnectionInterface::kBundlePolicyMaxBundle;
}
}
+ (RTCBundlePolicy)bundlePolicyForNativePolicy:
(webrtc::PeerConnectionInterface::BundlePolicy)nativePolicy {
switch (nativePolicy) {
case webrtc::PeerConnectionInterface::kBundlePolicyBalanced:
return RTCBundlePolicyBalanced;
case webrtc::PeerConnectionInterface::kBundlePolicyMaxCompat:
return RTCBundlePolicyMaxCompat;
case webrtc::PeerConnectionInterface::kBundlePolicyMaxBundle:
return RTCBundlePolicyMaxBundle;
}
}
+ (NSString *)stringForBundlePolicy:(RTCBundlePolicy)policy {
switch (policy) {
case RTCBundlePolicyBalanced:
return @"BALANCED";
case RTCBundlePolicyMaxCompat:
return @"MAX_COMPAT";
case RTCBundlePolicyMaxBundle:
return @"MAX_BUNDLE";
}
}
+ (webrtc::PeerConnectionInterface::RtcpMuxPolicy)nativeRtcpMuxPolicyForPolicy:
(RTCRtcpMuxPolicy)policy {
switch (policy) {
case RTCRtcpMuxPolicyNegotiate:
return webrtc::PeerConnectionInterface::kRtcpMuxPolicyNegotiate;
case RTCRtcpMuxPolicyRequire:
return webrtc::PeerConnectionInterface::kRtcpMuxPolicyRequire;
}
}
+ (RTCRtcpMuxPolicy)rtcpMuxPolicyForNativePolicy:
(webrtc::PeerConnectionInterface::RtcpMuxPolicy)nativePolicy {
switch (nativePolicy) {
case webrtc::PeerConnectionInterface::kRtcpMuxPolicyNegotiate:
return RTCRtcpMuxPolicyNegotiate;
case webrtc::PeerConnectionInterface::kRtcpMuxPolicyRequire:
return RTCRtcpMuxPolicyRequire;
}
}
+ (NSString *)stringForRtcpMuxPolicy:(RTCRtcpMuxPolicy)policy {
switch (policy) {
case RTCRtcpMuxPolicyNegotiate:
return @"NEGOTIATE";
case RTCRtcpMuxPolicyRequire:
return @"REQUIRE";
}
}
+ (webrtc::PeerConnectionInterface::TcpCandidatePolicy)
nativeTcpCandidatePolicyForPolicy:(RTCTcpCandidatePolicy)policy {
switch (policy) {
case RTCTcpCandidatePolicyEnabled:
return webrtc::PeerConnectionInterface::kTcpCandidatePolicyEnabled;
case RTCTcpCandidatePolicyDisabled:
return webrtc::PeerConnectionInterface::kTcpCandidatePolicyDisabled;
}
}
+ (webrtc::PeerConnectionInterface::CandidateNetworkPolicy)
nativeCandidateNetworkPolicyForPolicy:(RTCCandidateNetworkPolicy)policy {
switch (policy) {
case RTCCandidateNetworkPolicyAll:
return webrtc::PeerConnectionInterface::kCandidateNetworkPolicyAll;
case RTCCandidateNetworkPolicyLowCost:
return webrtc::PeerConnectionInterface::kCandidateNetworkPolicyLowCost;
}
}
+ (RTCTcpCandidatePolicy)tcpCandidatePolicyForNativePolicy:
(webrtc::PeerConnectionInterface::TcpCandidatePolicy)nativePolicy {
switch (nativePolicy) {
case webrtc::PeerConnectionInterface::kTcpCandidatePolicyEnabled:
return RTCTcpCandidatePolicyEnabled;
case webrtc::PeerConnectionInterface::kTcpCandidatePolicyDisabled:
return RTCTcpCandidatePolicyDisabled;
}
}
+ (NSString *)stringForTcpCandidatePolicy:(RTCTcpCandidatePolicy)policy {
switch (policy) {
case RTCTcpCandidatePolicyEnabled:
return @"TCP_ENABLED";
case RTCTcpCandidatePolicyDisabled:
return @"TCP_DISABLED";
}
}
+ (RTCCandidateNetworkPolicy)candidateNetworkPolicyForNativePolicy:
(webrtc::PeerConnectionInterface::CandidateNetworkPolicy)nativePolicy {
switch (nativePolicy) {
case webrtc::PeerConnectionInterface::kCandidateNetworkPolicyAll:
return RTCCandidateNetworkPolicyAll;
case webrtc::PeerConnectionInterface::kCandidateNetworkPolicyLowCost:
return RTCCandidateNetworkPolicyLowCost;
}
}
+ (NSString *)stringForCandidateNetworkPolicy:
(RTCCandidateNetworkPolicy)policy {
switch (policy) {
case RTCCandidateNetworkPolicyAll:
return @"CANDIDATE_ALL_NETWORKS";
case RTCCandidateNetworkPolicyLowCost:
return @"CANDIDATE_LOW_COST_NETWORKS";
}
}
+ (webrtc::PeerConnectionInterface::ContinualGatheringPolicy)
nativeContinualGatheringPolicyForPolicy:
(RTCContinualGatheringPolicy)policy {
switch (policy) {
case RTCContinualGatheringPolicyGatherOnce:
return webrtc::PeerConnectionInterface::GATHER_ONCE;
case RTCContinualGatheringPolicyGatherContinually:
return webrtc::PeerConnectionInterface::GATHER_CONTINUALLY;
}
}
+ (RTCContinualGatheringPolicy)continualGatheringPolicyForNativePolicy:
(webrtc::PeerConnectionInterface::ContinualGatheringPolicy)nativePolicy {
switch (nativePolicy) {
case webrtc::PeerConnectionInterface::GATHER_ONCE:
return RTCContinualGatheringPolicyGatherOnce;
case webrtc::PeerConnectionInterface::GATHER_CONTINUALLY:
return RTCContinualGatheringPolicyGatherContinually;
}
}
+ (NSString *)stringForContinualGatheringPolicy:
(RTCContinualGatheringPolicy)policy {
switch (policy) {
case RTCContinualGatheringPolicyGatherOnce:
return @"GATHER_ONCE";
case RTCContinualGatheringPolicyGatherContinually:
return @"GATHER_CONTINUALLY";
}
}
+ (rtc::KeyType)nativeEncryptionKeyTypeForKeyType:
(RTCEncryptionKeyType)keyType {
switch (keyType) {
case RTCEncryptionKeyTypeRSA:
return rtc::KT_RSA;
case RTCEncryptionKeyTypeECDSA:
return rtc::KT_ECDSA;
}
}
+ (webrtc::SdpSemantics)nativeSdpSemanticsForSdpSemantics:(RTCSdpSemantics)sdpSemantics {
switch (sdpSemantics) {
case RTCSdpSemanticsPlanB:
return webrtc::SdpSemantics::kPlanB;
case RTCSdpSemanticsUnifiedPlan:
return webrtc::SdpSemantics::kUnifiedPlan;
}
}
+ (RTCSdpSemantics)sdpSemanticsForNativeSdpSemantics:(webrtc::SdpSemantics)sdpSemantics {
switch (sdpSemantics) {
case webrtc::SdpSemantics::kPlanB:
return RTCSdpSemanticsPlanB;
case webrtc::SdpSemantics::kUnifiedPlan:
return RTCSdpSemanticsUnifiedPlan;
}
}
+ (NSString *)stringForSdpSemantics:(RTCSdpSemantics)sdpSemantics {
switch (sdpSemantics) {
case RTCSdpSemanticsPlanB:
return @"PLAN_B";
case RTCSdpSemanticsUnifiedPlan:
return @"UNIFIED_PLAN";
}
}
@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/RTCDataChannel.h"
#include "api/datachannelinterface.h"
#include "rtc_base/scoped_ref_ptr.h"
NS_ASSUME_NONNULL_BEGIN
@class RTCPeerConnectionFactory;
@interface RTCDataBuffer ()
/**
* The native DataBuffer representation of this RTCDatabuffer object. This is
* needed to pass to the underlying C++ APIs.
*/
@property(nonatomic, readonly) const webrtc::DataBuffer *nativeDataBuffer;
/** Initialize an RTCDataBuffer from a native DataBuffer. */
- (instancetype)initWithNativeBuffer:(const webrtc::DataBuffer &)nativeBuffer;
@end
@interface RTCDataChannel ()
/** Initialize an RTCDataChannel from a native DataChannelInterface. */
- (instancetype)initWithFactory:(RTCPeerConnectionFactory *)factory
nativeDataChannel:(rtc::scoped_refptr<webrtc::DataChannelInterface>)nativeDataChannel
NS_DESIGNATED_INITIALIZER;
+ (webrtc::DataChannelInterface::DataState)nativeDataChannelStateForState:
(RTCDataChannelState)state;
+ (RTCDataChannelState)dataChannelStateForNativeState:
(webrtc::DataChannelInterface::DataState)nativeState;
+ (NSString *)stringForState:(RTCDataChannelState)state;
@end
NS_ASSUME_NONNULL_END

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@ -1,223 +0,0 @@
/*
* 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 "RTCDataChannel+Private.h"
#import "NSString+StdString.h"
#include <memory>
namespace webrtc {
class DataChannelDelegateAdapter : public DataChannelObserver {
public:
DataChannelDelegateAdapter(RTCDataChannel *channel) { channel_ = channel; }
void OnStateChange() override {
[channel_.delegate dataChannelDidChangeState:channel_];
}
void OnMessage(const DataBuffer& buffer) override {
RTCDataBuffer *data_buffer =
[[RTCDataBuffer alloc] initWithNativeBuffer:buffer];
[channel_.delegate dataChannel:channel_
didReceiveMessageWithBuffer:data_buffer];
}
void OnBufferedAmountChange(uint64_t previousAmount) override {
id<RTCDataChannelDelegate> delegate = channel_.delegate;
SEL sel = @selector(dataChannel:didChangeBufferedAmount:);
if ([delegate respondsToSelector:sel]) {
[delegate dataChannel:channel_ didChangeBufferedAmount:previousAmount];
}
}
private:
__weak RTCDataChannel *channel_;
};
}
@implementation RTCDataBuffer {
std::unique_ptr<webrtc::DataBuffer> _dataBuffer;
}
- (instancetype)initWithData:(NSData *)data isBinary:(BOOL)isBinary {
NSParameterAssert(data);
if (self = [super init]) {
rtc::CopyOnWriteBuffer buffer(
reinterpret_cast<const uint8_t*>(data.bytes), data.length);
_dataBuffer.reset(new webrtc::DataBuffer(buffer, isBinary));
}
return self;
}
- (NSData *)data {
return [NSData dataWithBytes:_dataBuffer->data.data()
length:_dataBuffer->data.size()];
}
- (BOOL)isBinary {
return _dataBuffer->binary;
}
#pragma mark - Private
- (instancetype)initWithNativeBuffer:(const webrtc::DataBuffer&)nativeBuffer {
if (self = [super init]) {
_dataBuffer.reset(new webrtc::DataBuffer(nativeBuffer));
}
return self;
}
- (const webrtc::DataBuffer *)nativeDataBuffer {
return _dataBuffer.get();
}
@end
@implementation RTCDataChannel {
RTCPeerConnectionFactory *_factory;
rtc::scoped_refptr<webrtc::DataChannelInterface> _nativeDataChannel;
std::unique_ptr<webrtc::DataChannelDelegateAdapter> _observer;
BOOL _isObserverRegistered;
}
@synthesize delegate = _delegate;
- (void)dealloc {
// Handles unregistering the observer properly. We need to do this because
// there may still be other references to the underlying data channel.
_nativeDataChannel->UnregisterObserver();
}
- (NSString *)label {
return [NSString stringForStdString:_nativeDataChannel->label()];
}
- (BOOL)isReliable {
return _nativeDataChannel->reliable();
}
- (BOOL)isOrdered {
return _nativeDataChannel->ordered();
}
- (NSUInteger)maxRetransmitTime {
return self.maxPacketLifeTime;
}
- (uint16_t)maxPacketLifeTime {
return _nativeDataChannel->maxRetransmitTime();
}
- (uint16_t)maxRetransmits {
return _nativeDataChannel->maxRetransmits();
}
- (NSString *)protocol {
return [NSString stringForStdString:_nativeDataChannel->protocol()];
}
- (BOOL)isNegotiated {
return _nativeDataChannel->negotiated();
}
- (NSInteger)streamId {
return self.channelId;
}
- (int)channelId {
return _nativeDataChannel->id();
}
- (RTCDataChannelState)readyState {
return [[self class] dataChannelStateForNativeState:
_nativeDataChannel->state()];
}
- (uint64_t)bufferedAmount {
return _nativeDataChannel->buffered_amount();
}
- (void)close {
_nativeDataChannel->Close();
}
- (BOOL)sendData:(RTCDataBuffer *)data {
return _nativeDataChannel->Send(*data.nativeDataBuffer);
}
- (NSString *)description {
return [NSString stringWithFormat:@"RTCDataChannel:\n%ld\n%@\n%@",
(long)self.channelId,
self.label,
[[self class]
stringForState:self.readyState]];
}
#pragma mark - Private
- (instancetype)initWithFactory:(RTCPeerConnectionFactory *)factory
nativeDataChannel:
(rtc::scoped_refptr<webrtc::DataChannelInterface>)nativeDataChannel {
NSParameterAssert(nativeDataChannel);
if (self = [super init]) {
_factory = factory;
_nativeDataChannel = nativeDataChannel;
_observer.reset(new webrtc::DataChannelDelegateAdapter(self));
_nativeDataChannel->RegisterObserver(_observer.get());
}
return self;
}
+ (webrtc::DataChannelInterface::DataState)
nativeDataChannelStateForState:(RTCDataChannelState)state {
switch (state) {
case RTCDataChannelStateConnecting:
return webrtc::DataChannelInterface::DataState::kConnecting;
case RTCDataChannelStateOpen:
return webrtc::DataChannelInterface::DataState::kOpen;
case RTCDataChannelStateClosing:
return webrtc::DataChannelInterface::DataState::kClosing;
case RTCDataChannelStateClosed:
return webrtc::DataChannelInterface::DataState::kClosed;
}
}
+ (RTCDataChannelState)dataChannelStateForNativeState:
(webrtc::DataChannelInterface::DataState)nativeState {
switch (nativeState) {
case webrtc::DataChannelInterface::DataState::kConnecting:
return RTCDataChannelStateConnecting;
case webrtc::DataChannelInterface::DataState::kOpen:
return RTCDataChannelStateOpen;
case webrtc::DataChannelInterface::DataState::kClosing:
return RTCDataChannelStateClosing;
case webrtc::DataChannelInterface::DataState::kClosed:
return RTCDataChannelStateClosed;
}
}
+ (NSString *)stringForState:(RTCDataChannelState)state {
switch (state) {
case RTCDataChannelStateConnecting:
return @"Connecting";
case RTCDataChannelStateOpen:
return @"Open";
case RTCDataChannelStateClosing:
return @"Closing";
case RTCDataChannelStateClosed:
return @"Closed";
}
}
@end

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@ -1,23 +0,0 @@
/*
* 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/RTCDataChannelConfiguration.h"
#include "api/datachannelinterface.h"
NS_ASSUME_NONNULL_BEGIN
@interface RTCDataChannelConfiguration ()
@property(nonatomic, readonly) webrtc::DataChannelInit nativeDataChannelInit;
@end
NS_ASSUME_NONNULL_END

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@ -1,83 +0,0 @@
/*
* 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 "RTCDataChannelConfiguration+Private.h"
#import "NSString+StdString.h"
@implementation RTCDataChannelConfiguration
@synthesize nativeDataChannelInit = _nativeDataChannelInit;
- (BOOL)isOrdered {
return _nativeDataChannelInit.ordered;
}
- (void)setIsOrdered:(BOOL)isOrdered {
_nativeDataChannelInit.ordered = isOrdered;
}
- (NSInteger)maxRetransmitTimeMs {
return self.maxPacketLifeTime;
}
- (void)setMaxRetransmitTimeMs:(NSInteger)maxRetransmitTimeMs {
self.maxPacketLifeTime = maxRetransmitTimeMs;
}
- (int)maxPacketLifeTime {
return _nativeDataChannelInit.maxRetransmitTime;
}
- (void)setMaxPacketLifeTime:(int)maxPacketLifeTime {
_nativeDataChannelInit.maxRetransmitTime = maxPacketLifeTime;
}
- (int)maxRetransmits {
return _nativeDataChannelInit.maxRetransmits;
}
- (void)setMaxRetransmits:(int)maxRetransmits {
_nativeDataChannelInit.maxRetransmits = maxRetransmits;
}
- (NSString *)protocol {
return [NSString stringForStdString:_nativeDataChannelInit.protocol];
}
- (void)setProtocol:(NSString *)protocol {
_nativeDataChannelInit.protocol = [NSString stdStringForString:protocol];
}
- (BOOL)isNegotiated {
return _nativeDataChannelInit.negotiated;
}
- (void)setIsNegotiated:(BOOL)isNegotiated {
_nativeDataChannelInit.negotiated = isNegotiated;
}
- (int)streamId {
return self.channelId;
}
- (void)setStreamId:(int)streamId {
self.channelId = streamId;
}
- (int)channelId {
return _nativeDataChannelInit.id;
}
- (void)setChannelId:(int)channelId {
_nativeDataChannelInit.id = channelId;
}
@end

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@ -1,45 +0,0 @@
/*
* 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/RTCVideoCodecFactory.h"
#import "WebRTC/RTCVideoCodecH264.h"
#import "WebRTC/RTCVideoDecoderVP8.h"
#if !defined(RTC_DISABLE_VP9)
#import "WebRTC/RTCVideoDecoderVP9.h"
#endif
@implementation RTCDefaultVideoDecoderFactory
- (id<RTCVideoDecoder>)createDecoder:(RTCVideoCodecInfo *)info {
if ([info.name isEqualToString:kRTCVideoCodecH264Name]) {
return [[RTCVideoDecoderH264 alloc] init];
} else if ([info.name isEqualToString:kRTCVideoCodecVp8Name]) {
return [RTCVideoDecoderVP8 vp8Decoder];
#if !defined(RTC_DISABLE_VP9)
} else if ([info.name isEqualToString:kRTCVideoCodecVp9Name]) {
return [RTCVideoDecoderVP9 vp9Decoder];
#endif
}
return nil;
}
- (NSArray<RTCVideoCodecInfo *> *)supportedCodecs {
return @[
[[RTCVideoCodecInfo alloc] initWithName:kRTCVideoCodecH264Name],
[[RTCVideoCodecInfo alloc] initWithName:kRTCVideoCodecVp8Name],
#if !defined(RTC_DISABLE_VP9)
[[RTCVideoCodecInfo alloc] initWithName:kRTCVideoCodecVp9Name],
#endif
];
}
@end

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@ -1,87 +0,0 @@
/*
* 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/RTCVideoCodecFactory.h"
#import "WebRTC/RTCVideoCodec.h"
#import "WebRTC/RTCVideoCodecH264.h"
#import "WebRTC/RTCVideoEncoderVP8.h"
#if !defined(RTC_DISABLE_VP9)
#import "WebRTC/RTCVideoEncoderVP9.h"
#endif
@implementation RTCDefaultVideoEncoderFactory
@synthesize preferredCodec;
+ (NSArray<RTCVideoCodecInfo *> *)supportedCodecs {
NSDictionary<NSString *, NSString *> *constrainedHighParams = @{
@"profile-level-id" : kRTCMaxSupportedH264ProfileLevelConstrainedHigh,
@"level-asymmetry-allowed" : @"1",
@"packetization-mode" : @"1",
};
RTCVideoCodecInfo *constrainedHighInfo =
[[RTCVideoCodecInfo alloc] initWithName:kRTCVideoCodecH264Name
parameters:constrainedHighParams];
NSDictionary<NSString *, NSString *> *constrainedBaselineParams = @{
@"profile-level-id" : kRTCMaxSupportedH264ProfileLevelConstrainedBaseline,
@"level-asymmetry-allowed" : @"1",
@"packetization-mode" : @"1",
};
RTCVideoCodecInfo *constrainedBaselineInfo =
[[RTCVideoCodecInfo alloc] initWithName:kRTCVideoCodecH264Name
parameters:constrainedBaselineParams];
RTCVideoCodecInfo *vp8Info = [[RTCVideoCodecInfo alloc] initWithName:kRTCVideoCodecVp8Name];
#if !defined(RTC_DISABLE_VP9)
RTCVideoCodecInfo *vp9Info = [[RTCVideoCodecInfo alloc] initWithName:kRTCVideoCodecVp9Name];
#endif
return @[
constrainedHighInfo,
constrainedBaselineInfo,
vp8Info,
#if !defined(RTC_DISABLE_VP9)
vp9Info,
#endif
];
}
- (id<RTCVideoEncoder>)createEncoder:(RTCVideoCodecInfo *)info {
if ([info.name isEqualToString:kRTCVideoCodecH264Name]) {
return [[RTCVideoEncoderH264 alloc] initWithCodecInfo:info];
} else if ([info.name isEqualToString:kRTCVideoCodecVp8Name]) {
return [RTCVideoEncoderVP8 vp8Encoder];
#if !defined(RTC_DISABLE_VP9)
} else if ([info.name isEqualToString:kRTCVideoCodecVp9Name]) {
return [RTCVideoEncoderVP9 vp9Encoder];
#endif
}
return nil;
}
- (NSArray<RTCVideoCodecInfo *> *)supportedCodecs {
NSMutableArray<RTCVideoCodecInfo *> *codecs = [[[self class] supportedCodecs] mutableCopy];
NSMutableArray<RTCVideoCodecInfo *> *orderedCodecs = [NSMutableArray array];
NSUInteger index = [codecs indexOfObject:self.preferredCodec];
if (index != NSNotFound) {
[orderedCodecs addObject:[codecs objectAtIndex:index]];
[codecs removeObjectAtIndex:index];
}
[orderedCodecs addObjectsFromArray:codecs];
return [orderedCodecs copy];
}
@end

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@ -1,29 +0,0 @@
/*
* 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/RTCDtmfSender.h"
#include "api/dtmfsenderinterface.h"
NS_ASSUME_NONNULL_BEGIN
@interface RTCDtmfSender : NSObject <RTCDtmfSender>
@property(nonatomic, readonly) rtc::scoped_refptr<webrtc::DtmfSenderInterface> nativeDtmfSender;
- (instancetype)init NS_UNAVAILABLE;
/** Initialize an RTCDtmfSender with a native DtmfSenderInterface. */
- (instancetype)initWithNativeDtmfSender:
(rtc::scoped_refptr<webrtc::DtmfSenderInterface>)nativeDtmfSender NS_DESIGNATED_INITIALIZER;
@end
NS_ASSUME_NONNULL_END

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@ -1,74 +0,0 @@
/*
* 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 "RTCDtmfSender+Private.h"
#import "NSString+StdString.h"
#import "WebRTC/RTCLogging.h"
#include "rtc_base/timeutils.h"
@implementation RTCDtmfSender {
rtc::scoped_refptr<webrtc::DtmfSenderInterface> _nativeDtmfSender;
}
- (BOOL)canInsertDtmf {
return _nativeDtmfSender->CanInsertDtmf();
}
- (BOOL)insertDtmf:(nonnull NSString *)tones
duration:(NSTimeInterval)duration
interToneGap:(NSTimeInterval)interToneGap {
RTC_DCHECK(tones != nil);
int durationMs = static_cast<int>(duration * rtc::kNumMillisecsPerSec);
int interToneGapMs = static_cast<int>(interToneGap * rtc::kNumMillisecsPerSec);
return _nativeDtmfSender->InsertDtmf(
[NSString stdStringForString:tones], durationMs, interToneGapMs);
}
- (nonnull NSString *)remainingTones {
return [NSString stringForStdString:_nativeDtmfSender->tones()];
}
- (NSTimeInterval)duration {
return static_cast<NSTimeInterval>(_nativeDtmfSender->duration()) / rtc::kNumMillisecsPerSec;
}
- (NSTimeInterval)interToneGap {
return static_cast<NSTimeInterval>(_nativeDtmfSender->inter_tone_gap()) /
rtc::kNumMillisecsPerSec;
}
- (NSString *)description {
return [NSString
stringWithFormat:
@"RTCDtmfSender {\n remainingTones: %@\n duration: %f sec\n interToneGap: %f sec\n}",
[self remainingTones],
[self duration],
[self interToneGap]];
}
#pragma mark - Private
- (rtc::scoped_refptr<webrtc::DtmfSenderInterface>)nativeDtmfSender {
return _nativeDtmfSender;
}
- (instancetype)initWithNativeDtmfSender:
(rtc::scoped_refptr<webrtc::DtmfSenderInterface>)nativeDtmfSender {
NSParameterAssert(nativeDtmfSender);
if (self = [super init]) {
_nativeDtmfSender = nativeDtmfSender;
RTCLogInfo(@"RTCDtmfSender(%p): created DTMF sender: %@", self, self.description);
}
return self;
}
@end

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@ -1,83 +0,0 @@
/*
* 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/RTCVideoCodec.h"
#import "RTCVideoCodec+Private.h"
#include "rtc_base/numerics/safe_conversions.h"
@implementation RTCEncodedImage
@synthesize buffer = _buffer;
@synthesize encodedWidth = _encodedWidth;
@synthesize encodedHeight = _encodedHeight;
@synthesize timeStamp = _timeStamp;
@synthesize captureTimeMs = _captureTimeMs;
@synthesize ntpTimeMs = _ntpTimeMs;
@synthesize flags = _flags;
@synthesize encodeStartMs = _encodeStartMs;
@synthesize encodeFinishMs = _encodeFinishMs;
@synthesize frameType = _frameType;
@synthesize rotation = _rotation;
@synthesize completeFrame = _completeFrame;
@synthesize qp = _qp;
@synthesize contentType = _contentType;
- (instancetype)initWithNativeEncodedImage:(webrtc::EncodedImage)encodedImage {
if (self = [super init]) {
// Wrap the buffer in NSData without copying, do not take ownership.
_buffer = [NSData dataWithBytesNoCopy:encodedImage._buffer
length:encodedImage._length
freeWhenDone:NO];
_encodedWidth = rtc::dchecked_cast<int32_t>(encodedImage._encodedWidth);
_encodedHeight = rtc::dchecked_cast<int32_t>(encodedImage._encodedHeight);
_timeStamp = encodedImage.Timestamp();
_captureTimeMs = encodedImage.capture_time_ms_;
_ntpTimeMs = encodedImage.ntp_time_ms_;
_flags = encodedImage.timing_.flags;
_encodeStartMs = encodedImage.timing_.encode_start_ms;
_encodeFinishMs = encodedImage.timing_.encode_finish_ms;
_frameType = static_cast<RTCFrameType>(encodedImage._frameType);
_rotation = static_cast<RTCVideoRotation>(encodedImage.rotation_);
_completeFrame = encodedImage._completeFrame;
_qp = @(encodedImage.qp_);
_contentType = (encodedImage.content_type_ == webrtc::VideoContentType::SCREENSHARE) ?
RTCVideoContentTypeScreenshare :
RTCVideoContentTypeUnspecified;
}
return self;
}
- (webrtc::EncodedImage)nativeEncodedImage {
// Return the pointer without copying.
webrtc::EncodedImage encodedImage(
(uint8_t *)_buffer.bytes, (size_t)_buffer.length, (size_t)_buffer.length);
encodedImage._encodedWidth = rtc::dchecked_cast<uint32_t>(_encodedWidth);
encodedImage._encodedHeight = rtc::dchecked_cast<uint32_t>(_encodedHeight);
encodedImage.SetTimestamp(_timeStamp);
encodedImage.capture_time_ms_ = _captureTimeMs;
encodedImage.ntp_time_ms_ = _ntpTimeMs;
encodedImage.timing_.flags = _flags;
encodedImage.timing_.encode_start_ms = _encodeStartMs;
encodedImage.timing_.encode_finish_ms = _encodeFinishMs;
encodedImage._frameType = webrtc::FrameType(_frameType);
encodedImage.rotation_ = webrtc::VideoRotation(_rotation);
encodedImage._completeFrame = _completeFrame;
encodedImage.qp_ = _qp ? _qp.intValue : -1;
encodedImage.content_type_ = (_contentType == RTCVideoContentTypeScreenshare) ?
webrtc::VideoContentType::SCREENSHARE :
webrtc::VideoContentType::UNSPECIFIED;
return encodedImage;
}
@end

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@ -1,202 +0,0 @@
/**
* 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/RTCFileVideoCapturer.h"
#import "WebRTC/RTCLogging.h"
#import "WebRTC/RTCVideoFrameBuffer.h"
NSString *const kRTCFileVideoCapturerErrorDomain = @"org.webrtc.RTCFileVideoCapturer";
typedef NS_ENUM(NSInteger, RTCFileVideoCapturerErrorCode) {
RTCFileVideoCapturerErrorCode_CapturerRunning = 2000,
RTCFileVideoCapturerErrorCode_FileNotFound
};
typedef NS_ENUM(NSInteger, RTCFileVideoCapturerStatus) {
RTCFileVideoCapturerStatusNotInitialized,
RTCFileVideoCapturerStatusStarted,
RTCFileVideoCapturerStatusStopped
};
@implementation RTCFileVideoCapturer {
AVAssetReader *_reader;
AVAssetReaderTrackOutput *_outTrack;
RTCFileVideoCapturerStatus _status;
CMTime _lastPresentationTime;
dispatch_queue_t _frameQueue;
NSURL *_fileURL;
}
- (void)startCapturingFromFileNamed:(NSString *)nameOfFile
onError:(RTCFileVideoCapturerErrorBlock)errorBlock {
if (_status == RTCFileVideoCapturerStatusStarted) {
NSError *error =
[NSError errorWithDomain:kRTCFileVideoCapturerErrorDomain
code:RTCFileVideoCapturerErrorCode_CapturerRunning
userInfo:@{NSUnderlyingErrorKey : @"Capturer has been started."}];
errorBlock(error);
return;
} else {
_status = RTCFileVideoCapturerStatusStarted;
}
dispatch_async(dispatch_get_global_queue(DISPATCH_QUEUE_PRIORITY_DEFAULT, 0), ^{
NSString *pathForFile = [self pathForFileName:nameOfFile];
if (!pathForFile) {
NSString *errorString =
[NSString stringWithFormat:@"File %@ not found in bundle", nameOfFile];
NSError *error = [NSError errorWithDomain:kRTCFileVideoCapturerErrorDomain
code:RTCFileVideoCapturerErrorCode_FileNotFound
userInfo:@{NSUnderlyingErrorKey : errorString}];
errorBlock(error);
return;
}
_lastPresentationTime = CMTimeMake(0, 0);
_fileURL = [NSURL fileURLWithPath:pathForFile];
[self setupReaderOnError:errorBlock];
});
}
- (void)setupReaderOnError:(RTCFileVideoCapturerErrorBlock)errorBlock {
AVURLAsset *asset = [AVURLAsset URLAssetWithURL:_fileURL options:nil];
NSArray *allTracks = [asset tracksWithMediaType:AVMediaTypeVideo];
NSError *error = nil;
_reader = [[AVAssetReader alloc] initWithAsset:asset error:&error];
if (error) {
errorBlock(error);
return;
}
NSDictionary *options = @{
(NSString *)kCVPixelBufferPixelFormatTypeKey : @(kCVPixelFormatType_420YpCbCr8BiPlanarFullRange)
};
_outTrack =
[[AVAssetReaderTrackOutput alloc] initWithTrack:allTracks.firstObject outputSettings:options];
[_reader addOutput:_outTrack];
[_reader startReading];
RTCLog(@"File capturer started reading");
[self readNextBuffer];
}
- (void)stopCapture {
_status = RTCFileVideoCapturerStatusStopped;
RTCLog(@"File capturer stopped.");
}
#pragma mark - Private
- (nullable NSString *)pathForFileName:(NSString *)fileName {
NSArray *nameComponents = [fileName componentsSeparatedByString:@"."];
if (nameComponents.count != 2) {
return nil;
}
NSString *path =
[[NSBundle mainBundle] pathForResource:nameComponents[0] ofType:nameComponents[1]];
return path;
}
- (dispatch_queue_t)frameQueue {
if (!_frameQueue) {
_frameQueue = dispatch_queue_create("org.webrtc.filecapturer.video", DISPATCH_QUEUE_SERIAL);
dispatch_set_target_queue(_frameQueue,
dispatch_get_global_queue(DISPATCH_QUEUE_PRIORITY_BACKGROUND, 0));
}
return _frameQueue;
}
- (void)readNextBuffer {
if (_status == RTCFileVideoCapturerStatusStopped) {
[_reader cancelReading];
_reader = nil;
return;
}
if (_reader.status == AVAssetReaderStatusCompleted) {
[_reader cancelReading];
_reader = nil;
[self setupReaderOnError:nil];
return;
}
CMSampleBufferRef sampleBuffer = [_outTrack copyNextSampleBuffer];
if (!sampleBuffer) {
[self readNextBuffer];
return;
}
if (CMSampleBufferGetNumSamples(sampleBuffer) != 1 || !CMSampleBufferIsValid(sampleBuffer) ||
!CMSampleBufferDataIsReady(sampleBuffer)) {
CFRelease(sampleBuffer);
[self readNextBuffer];
return;
}
[self publishSampleBuffer:sampleBuffer];
}
- (void)publishSampleBuffer:(CMSampleBufferRef)sampleBuffer {
CMTime presentationTime = CMSampleBufferGetPresentationTimeStamp(sampleBuffer);
Float64 presentationDifference =
CMTimeGetSeconds(CMTimeSubtract(presentationTime, _lastPresentationTime));
_lastPresentationTime = presentationTime;
int64_t presentationDifferenceRound = lroundf(presentationDifference * NSEC_PER_SEC);
__block dispatch_source_t timer = [self createStrictTimer];
// Strict timer that will fire |presentationDifferenceRound| ns from now and never again.
dispatch_source_set_timer(timer,
dispatch_time(DISPATCH_TIME_NOW, presentationDifferenceRound),
DISPATCH_TIME_FOREVER,
0);
dispatch_source_set_event_handler(timer, ^{
dispatch_source_cancel(timer);
timer = nil;
CVPixelBufferRef pixelBuffer = CMSampleBufferGetImageBuffer(sampleBuffer);
if (!pixelBuffer) {
CFRelease(sampleBuffer);
dispatch_async(dispatch_get_global_queue(DISPATCH_QUEUE_PRIORITY_DEFAULT, 0), ^{
[self readNextBuffer];
});
return;
}
RTCCVPixelBuffer *rtcPixelBuffer = [[RTCCVPixelBuffer alloc] initWithPixelBuffer:pixelBuffer];
NSTimeInterval timeStampSeconds = CACurrentMediaTime();
int64_t timeStampNs = lroundf(timeStampSeconds * NSEC_PER_SEC);
RTCVideoFrame *videoFrame =
[[RTCVideoFrame alloc] initWithBuffer:rtcPixelBuffer rotation:0 timeStampNs:timeStampNs];
CFRelease(sampleBuffer);
dispatch_async(dispatch_get_global_queue(DISPATCH_QUEUE_PRIORITY_DEFAULT, 0), ^{
[self readNextBuffer];
});
[self.delegate capturer:self didCaptureVideoFrame:videoFrame];
});
dispatch_activate(timer);
}
- (dispatch_source_t)createStrictTimer {
dispatch_source_t timer = dispatch_source_create(
DISPATCH_SOURCE_TYPE_TIMER, 0, DISPATCH_TIMER_STRICT, [self frameQueue]);
return timer;
}
- (void)dealloc {
[self stopCapture];
}
@end

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@ -1,58 +0,0 @@
/*
* 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 "WebRTC/RTCVideoCodecH264.h"
#include "media/base/h264_profile_level_id.h"
@interface RTCH264ProfileLevelId ()
@property(nonatomic, assign) RTCH264Profile profile;
@property(nonatomic, assign) RTCH264Level level;
@property(nonatomic, strong) NSString *hexString;
@end
@implementation RTCH264ProfileLevelId
@synthesize profile = _profile;
@synthesize level = _level;
@synthesize hexString = _hexString;
- (instancetype)initWithHexString:(NSString *)hexString {
if (self = [super init]) {
self.hexString = hexString;
absl::optional<webrtc::H264::ProfileLevelId> profile_level_id =
webrtc::H264::ParseProfileLevelId([hexString cStringUsingEncoding:NSUTF8StringEncoding]);
if (profile_level_id.has_value()) {
self.profile = static_cast<RTCH264Profile>(profile_level_id->profile);
self.level = static_cast<RTCH264Level>(profile_level_id->level);
}
}
return self;
}
- (instancetype)initWithProfile:(RTCH264Profile)profile level:(RTCH264Level)level {
if (self = [super init]) {
self.profile = profile;
self.level = level;
absl::optional<std::string> hex_string =
webrtc::H264::ProfileLevelIdToString(webrtc::H264::ProfileLevelId(
static_cast<webrtc::H264::Profile>(profile), static_cast<webrtc::H264::Level>(level)));
self.hexString =
[NSString stringWithCString:hex_string.value_or("").c_str() encoding:NSUTF8StringEncoding];
}
return self;
}
@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/RTCIceCandidate.h"
#include <memory>
#include "api/jsep.h"
NS_ASSUME_NONNULL_BEGIN
@interface RTCIceCandidate ()
/**
* The native IceCandidateInterface representation of this RTCIceCandidate
* object. This is needed to pass to the underlying C++ APIs.
*/
@property(nonatomic, readonly) std::unique_ptr<webrtc::IceCandidateInterface> nativeCandidate;
/**
* Initialize an RTCIceCandidate from a native IceCandidateInterface. No
* ownership is taken of the native candidate.
*/
- (instancetype)initWithNativeCandidate:(const webrtc::IceCandidateInterface *)candidate;
@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 "RTCIceCandidate+Private.h"
#include <memory>
#import "NSString+StdString.h"
#import "WebRTC/RTCLogging.h"
@implementation RTCIceCandidate
@synthesize sdpMid = _sdpMid;
@synthesize sdpMLineIndex = _sdpMLineIndex;
@synthesize sdp = _sdp;
@synthesize serverUrl = _serverUrl;
- (instancetype)initWithSdp:(NSString *)sdp
sdpMLineIndex:(int)sdpMLineIndex
sdpMid:(NSString *)sdpMid {
NSParameterAssert(sdp.length);
if (self = [super init]) {
_sdpMid = [sdpMid copy];
_sdpMLineIndex = sdpMLineIndex;
_sdp = [sdp copy];
}
return self;
}
- (NSString *)description {
return [NSString stringWithFormat:@"RTCIceCandidate:\n%@\n%d\n%@\n%@",
_sdpMid,
_sdpMLineIndex,
_sdp,
_serverUrl];
}
#pragma mark - Private
- (instancetype)initWithNativeCandidate:
(const webrtc::IceCandidateInterface *)candidate {
NSParameterAssert(candidate);
std::string sdp;
candidate->ToString(&sdp);
RTCIceCandidate *rtcCandidate =
[self initWithSdp:[NSString stringForStdString:sdp]
sdpMLineIndex:candidate->sdp_mline_index()
sdpMid:[NSString stringForStdString:candidate->sdp_mid()]];
rtcCandidate->_serverUrl = [NSString stringForStdString:candidate->server_url()];
return rtcCandidate;
}
- (std::unique_ptr<webrtc::IceCandidateInterface>)nativeCandidate {
webrtc::SdpParseError error;
webrtc::IceCandidateInterface *candidate = webrtc::CreateIceCandidate(
_sdpMid.stdString, _sdpMLineIndex, _sdp.stdString, &error);
if (!candidate) {
RTCLog(@"Failed to create ICE candidate: %s\nline: %s",
error.description.c_str(),
error.line.c_str());
}
return std::unique_ptr<webrtc::IceCandidateInterface>(candidate);
}
@end

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@ -1,30 +0,0 @@
/*
* 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/RTCIceServer.h"
#include "api/peerconnectioninterface.h"
NS_ASSUME_NONNULL_BEGIN
@interface RTCIceServer ()
/**
* IceServer struct representation of this RTCIceServer object's data.
* This is needed to pass to the underlying C++ APIs.
*/
@property(nonatomic, readonly) webrtc::PeerConnectionInterface::IceServer nativeServer;
/** Initialize an RTCIceServer from a native IceServer. */
- (instancetype)initWithNativeServer:(webrtc::PeerConnectionInterface::IceServer)nativeServer;
@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 "RTCIceServer+Private.h"
#import "NSString+StdString.h"
@implementation RTCIceServer
@synthesize urlStrings = _urlStrings;
@synthesize username = _username;
@synthesize credential = _credential;
@synthesize tlsCertPolicy = _tlsCertPolicy;
@synthesize hostname = _hostname;
@synthesize tlsAlpnProtocols = _tlsAlpnProtocols;
@synthesize tlsEllipticCurves = _tlsEllipticCurves;
- (instancetype)initWithURLStrings:(NSArray<NSString *> *)urlStrings {
return [self initWithURLStrings:urlStrings
username:nil
credential:nil];
}
- (instancetype)initWithURLStrings:(NSArray<NSString *> *)urlStrings
username:(NSString *)username
credential:(NSString *)credential {
return [self initWithURLStrings:urlStrings
username:username
credential:credential
tlsCertPolicy:RTCTlsCertPolicySecure];
}
- (instancetype)initWithURLStrings:(NSArray<NSString *> *)urlStrings
username:(NSString *)username
credential:(NSString *)credential
tlsCertPolicy:(RTCTlsCertPolicy)tlsCertPolicy {
return [self initWithURLStrings:urlStrings
username:username
credential:credential
tlsCertPolicy:tlsCertPolicy
hostname:nil];
}
- (instancetype)initWithURLStrings:(NSArray<NSString *> *)urlStrings
username:(NSString *)username
credential:(NSString *)credential
tlsCertPolicy:(RTCTlsCertPolicy)tlsCertPolicy
hostname:(NSString *)hostname {
return [self initWithURLStrings:urlStrings
username:username
credential:credential
tlsCertPolicy:tlsCertPolicy
hostname:hostname
tlsAlpnProtocols:[NSArray array]];
}
- (instancetype)initWithURLStrings:(NSArray<NSString *> *)urlStrings
username:(NSString *)username
credential:(NSString *)credential
tlsCertPolicy:(RTCTlsCertPolicy)tlsCertPolicy
hostname:(NSString *)hostname
tlsAlpnProtocols:(NSArray<NSString *> *)tlsAlpnProtocols {
return [self initWithURLStrings:urlStrings
username:username
credential:credential
tlsCertPolicy:tlsCertPolicy
hostname:hostname
tlsAlpnProtocols:tlsAlpnProtocols
tlsEllipticCurves:[NSArray array]];
}
- (instancetype)initWithURLStrings:(NSArray<NSString *> *)urlStrings
username:(NSString *)username
credential:(NSString *)credential
tlsCertPolicy:(RTCTlsCertPolicy)tlsCertPolicy
hostname:(NSString *)hostname
tlsAlpnProtocols:(NSArray<NSString *> *)tlsAlpnProtocols
tlsEllipticCurves:(NSArray<NSString *> *)tlsEllipticCurves {
NSParameterAssert(urlStrings.count);
if (self = [super init]) {
_urlStrings = [[NSArray alloc] initWithArray:urlStrings copyItems:YES];
_username = [username copy];
_credential = [credential copy];
_tlsCertPolicy = tlsCertPolicy;
_hostname = [hostname copy];
_tlsAlpnProtocols = [[NSArray alloc] initWithArray:tlsAlpnProtocols copyItems:YES];
_tlsEllipticCurves = [[NSArray alloc] initWithArray:tlsEllipticCurves copyItems:YES];
}
return self;
}
- (NSString *)description {
return [NSString stringWithFormat:@"RTCIceServer:\n%@\n%@\n%@\n%@\n%@\n%@\n%@",
_urlStrings,
_username,
_credential,
[self stringForTlsCertPolicy:_tlsCertPolicy],
_hostname,
_tlsAlpnProtocols,
_tlsEllipticCurves];
}
#pragma mark - Private
- (NSString *)stringForTlsCertPolicy:(RTCTlsCertPolicy)tlsCertPolicy {
switch (tlsCertPolicy) {
case RTCTlsCertPolicySecure:
return @"RTCTlsCertPolicySecure";
case RTCTlsCertPolicyInsecureNoCheck:
return @"RTCTlsCertPolicyInsecureNoCheck";
}
}
- (webrtc::PeerConnectionInterface::IceServer)nativeServer {
__block webrtc::PeerConnectionInterface::IceServer iceServer;
iceServer.username = [NSString stdStringForString:_username];
iceServer.password = [NSString stdStringForString:_credential];
iceServer.hostname = [NSString stdStringForString:_hostname];
[_tlsAlpnProtocols enumerateObjectsUsingBlock:^(NSString *proto, NSUInteger idx, BOOL *stop) {
iceServer.tls_alpn_protocols.push_back(proto.stdString);
}];
[_tlsEllipticCurves enumerateObjectsUsingBlock:^(NSString *curve, NSUInteger idx, BOOL *stop) {
iceServer.tls_elliptic_curves.push_back(curve.stdString);
}];
[_urlStrings enumerateObjectsUsingBlock:^(NSString *url,
NSUInteger idx,
BOOL *stop) {
iceServer.urls.push_back(url.stdString);
}];
switch (_tlsCertPolicy) {
case RTCTlsCertPolicySecure:
iceServer.tls_cert_policy =
webrtc::PeerConnectionInterface::kTlsCertPolicySecure;
break;
case RTCTlsCertPolicyInsecureNoCheck:
iceServer.tls_cert_policy =
webrtc::PeerConnectionInterface::kTlsCertPolicyInsecureNoCheck;
break;
}
return iceServer;
}
- (instancetype)initWithNativeServer:
(webrtc::PeerConnectionInterface::IceServer)nativeServer {
NSMutableArray *urls =
[NSMutableArray arrayWithCapacity:nativeServer.urls.size()];
for (auto const &url : nativeServer.urls) {
[urls addObject:[NSString stringForStdString:url]];
}
NSString *username = [NSString stringForStdString:nativeServer.username];
NSString *credential = [NSString stringForStdString:nativeServer.password];
NSString *hostname = [NSString stringForStdString:nativeServer.hostname];
NSMutableArray *tlsAlpnProtocols =
[NSMutableArray arrayWithCapacity:nativeServer.tls_alpn_protocols.size()];
for (auto const &proto : nativeServer.tls_alpn_protocols) {
[tlsAlpnProtocols addObject:[NSString stringForStdString:proto]];
}
NSMutableArray *tlsEllipticCurves =
[NSMutableArray arrayWithCapacity:nativeServer.tls_elliptic_curves.size()];
for (auto const &curve : nativeServer.tls_elliptic_curves) {
[tlsEllipticCurves addObject:[NSString stringForStdString:curve]];
}
RTCTlsCertPolicy tlsCertPolicy;
switch (nativeServer.tls_cert_policy) {
case webrtc::PeerConnectionInterface::kTlsCertPolicySecure:
tlsCertPolicy = RTCTlsCertPolicySecure;
break;
case webrtc::PeerConnectionInterface::kTlsCertPolicyInsecureNoCheck:
tlsCertPolicy = RTCTlsCertPolicyInsecureNoCheck;
break;
}
self = [self initWithURLStrings:urls
username:username
credential:credential
tlsCertPolicy:tlsCertPolicy
hostname:hostname
tlsAlpnProtocols:tlsAlpnProtocols
tlsEllipticCurves:tlsEllipticCurves];
return self;
}
@end

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@ -1,25 +0,0 @@
/*
* 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/RTCIntervalRange.h"
#include "rtc_base/timeutils.h"
NS_ASSUME_NONNULL_BEGIN
@interface RTCIntervalRange ()
@property(nonatomic, readonly) std::unique_ptr<rtc::IntervalRange> nativeIntervalRange;
- (instancetype)initWithNativeIntervalRange:(const rtc::IntervalRange &)config;
@end
NS_ASSUME_NONNULL_END

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@ -1,50 +0,0 @@
/*
* 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 "RTCIntervalRange+Private.h"
#include "rtc_base/checks.h"
@implementation RTCIntervalRange
@synthesize min = _min;
@synthesize max = _max;
- (instancetype)init {
return [self initWithMin:0 max:0];
}
- (instancetype)initWithMin:(NSInteger)min
max:(NSInteger)max {
RTC_DCHECK_LE(min, max);
if (self = [super init]) {
_min = min;
_max = max;
}
return self;
}
- (instancetype)initWithNativeIntervalRange:(const rtc::IntervalRange &)config {
return [self initWithMin:config.min() max:config.max()];
}
- (NSString *)description {
return [NSString stringWithFormat:@"[%ld, %ld]", (long)_min, (long)_max];
}
#pragma mark - Private
- (std::unique_ptr<rtc::IntervalRange>)nativeIntervalRange {
std::unique_ptr<rtc::IntervalRange> nativeIntervalRange(
new rtc::IntervalRange((int)_min, (int)_max));
return nativeIntervalRange;
}
@end

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@ -1,24 +0,0 @@
/*
* 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/RTCLegacyStatsReport.h"
#include "api/statstypes.h"
NS_ASSUME_NONNULL_BEGIN
@interface RTCLegacyStatsReport ()
/** Initialize an RTCLegacyStatsReport object from a native StatsReport. */
- (instancetype)initWithNativeReport:(const webrtc::StatsReport &)nativeReport;
@end
NS_ASSUME_NONNULL_END

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@ -1,60 +0,0 @@
/*
* 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 "RTCLegacyStatsReport+Private.h"
#import "NSString+StdString.h"
#import "WebRTC/RTCLogging.h"
#include "rtc_base/checks.h"
@implementation RTCLegacyStatsReport
@synthesize timestamp = _timestamp;
@synthesize type = _type;
@synthesize reportId = _reportId;
@synthesize values = _values;
- (NSString *)description {
return [NSString stringWithFormat:@"RTCLegacyStatsReport:\n%@\n%@\n%f\n%@",
_reportId,
_type,
_timestamp,
_values];
}
#pragma mark - Private
- (instancetype)initWithNativeReport:(const webrtc::StatsReport &)nativeReport {
if (self = [super init]) {
_timestamp = nativeReport.timestamp();
_type = [NSString stringForStdString:nativeReport.TypeToString()];
_reportId = [NSString stringForStdString:
nativeReport.id()->ToString()];
NSUInteger capacity = nativeReport.values().size();
NSMutableDictionary *values =
[NSMutableDictionary dictionaryWithCapacity:capacity];
for (auto const &valuePair : nativeReport.values()) {
NSString *key = [NSString stringForStdString:
valuePair.second->display_name()];
NSString *value = [NSString stringForStdString:
valuePair.second->ToString()];
// Not expecting duplicate keys.
RTC_DCHECK(![values objectForKey:key]);
[values setObject:value forKey:key];
}
_values = values;
}
return self;
}
@end

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@ -1,51 +0,0 @@
/*
* 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/RTCMediaConstraints.h"
#include <memory>
#include "api/mediaconstraintsinterface.h"
namespace webrtc {
class MediaConstraints : public MediaConstraintsInterface {
public:
~MediaConstraints() override;
MediaConstraints();
MediaConstraints(const MediaConstraintsInterface::Constraints& mandatory,
const MediaConstraintsInterface::Constraints& optional);
const Constraints& GetMandatory() const override;
const Constraints& GetOptional() const override;
private:
MediaConstraintsInterface::Constraints mandatory_;
MediaConstraintsInterface::Constraints optional_;
};
} // namespace webrtc
NS_ASSUME_NONNULL_BEGIN
@interface RTCMediaConstraints ()
/**
* A MediaConstraints representation of this RTCMediaConstraints object. This is
* needed to pass to the underlying C++ APIs.
*/
- (std::unique_ptr<webrtc::MediaConstraints>)nativeConstraints;
/** Return a native Constraints object representing these constraints */
+ (webrtc::MediaConstraintsInterface::Constraints)nativeConstraintsForConstraints:
(NSDictionary<NSString*, NSString*>*)constraints;
@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 "RTCMediaConstraints+Private.h"
#import "NSString+StdString.h"
#include <memory>
NSString * const kRTCMediaConstraintsMinAspectRatio =
@(webrtc::MediaConstraintsInterface::kMinAspectRatio);
NSString * const kRTCMediaConstraintsMaxAspectRatio =
@(webrtc::MediaConstraintsInterface::kMaxAspectRatio);
NSString * const kRTCMediaConstraintsMinWidth =
@(webrtc::MediaConstraintsInterface::kMinWidth);
NSString * const kRTCMediaConstraintsMaxWidth =
@(webrtc::MediaConstraintsInterface::kMaxWidth);
NSString * const kRTCMediaConstraintsMinHeight =
@(webrtc::MediaConstraintsInterface::kMinHeight);
NSString * const kRTCMediaConstraintsMaxHeight =
@(webrtc::MediaConstraintsInterface::kMaxHeight);
NSString * const kRTCMediaConstraintsMinFrameRate =
@(webrtc::MediaConstraintsInterface::kMinFrameRate);
NSString * const kRTCMediaConstraintsMaxFrameRate =
@(webrtc::MediaConstraintsInterface::kMaxFrameRate);
NSString * const kRTCMediaConstraintsAudioNetworkAdaptorConfig =
@(webrtc::MediaConstraintsInterface::kAudioNetworkAdaptorConfig);
NSString * const kRTCMediaConstraintsIceRestart =
@(webrtc::MediaConstraintsInterface::kIceRestart);
NSString * const kRTCMediaConstraintsOfferToReceiveAudio =
@(webrtc::MediaConstraintsInterface::kOfferToReceiveAudio);
NSString * const kRTCMediaConstraintsOfferToReceiveVideo =
@(webrtc::MediaConstraintsInterface::kOfferToReceiveVideo);
NSString * const kRTCMediaConstraintsVoiceActivityDetection =
@(webrtc::MediaConstraintsInterface::kVoiceActivityDetection);
NSString * const kRTCMediaConstraintsValueTrue =
@(webrtc::MediaConstraintsInterface::kValueTrue);
NSString * const kRTCMediaConstraintsValueFalse =
@(webrtc::MediaConstraintsInterface::kValueFalse);
namespace webrtc {
MediaConstraints::~MediaConstraints() {}
MediaConstraints::MediaConstraints() {}
MediaConstraints::MediaConstraints(
const MediaConstraintsInterface::Constraints& mandatory,
const MediaConstraintsInterface::Constraints& optional)
: mandatory_(mandatory), optional_(optional) {}
const MediaConstraintsInterface::Constraints&
MediaConstraints::GetMandatory() const {
return mandatory_;
}
const MediaConstraintsInterface::Constraints&
MediaConstraints::GetOptional() const {
return optional_;
}
} // namespace webrtc
@implementation RTCMediaConstraints {
NSDictionary<NSString *, NSString *> *_mandatory;
NSDictionary<NSString *, NSString *> *_optional;
}
- (instancetype)initWithMandatoryConstraints:
(NSDictionary<NSString *, NSString *> *)mandatory
optionalConstraints:
(NSDictionary<NSString *, NSString *> *)optional {
if (self = [super init]) {
_mandatory = [[NSDictionary alloc] initWithDictionary:mandatory
copyItems:YES];
_optional = [[NSDictionary alloc] initWithDictionary:optional
copyItems:YES];
}
return self;
}
- (NSString *)description {
return [NSString stringWithFormat:@"RTCMediaConstraints:\n%@\n%@",
_mandatory,
_optional];
}
#pragma mark - Private
- (std::unique_ptr<webrtc::MediaConstraints>)nativeConstraints {
webrtc::MediaConstraintsInterface::Constraints mandatory =
[[self class] nativeConstraintsForConstraints:_mandatory];
webrtc::MediaConstraintsInterface::Constraints optional =
[[self class] nativeConstraintsForConstraints:_optional];
webrtc::MediaConstraints *nativeConstraints =
new webrtc::MediaConstraints(mandatory, optional);
return std::unique_ptr<webrtc::MediaConstraints>(nativeConstraints);
}
+ (webrtc::MediaConstraintsInterface::Constraints)
nativeConstraintsForConstraints:
(NSDictionary<NSString *, NSString *> *)constraints {
webrtc::MediaConstraintsInterface::Constraints nativeConstraints;
for (NSString *key in constraints) {
NSAssert([key isKindOfClass:[NSString class]],
@"%@ is not an NSString.", key);
NSString *value = [constraints objectForKey:key];
NSAssert([value isKindOfClass:[NSString class]],
@"%@ is not an NSString.", value);
if ([kRTCMediaConstraintsAudioNetworkAdaptorConfig isEqualToString:key]) {
// This value is base64 encoded.
NSData *charData = [[NSData alloc] initWithBase64EncodedString:value options:0];
std::string configValue =
std::string(reinterpret_cast<const char *>(charData.bytes), charData.length);
nativeConstraints.push_back(webrtc::MediaConstraintsInterface::Constraint(
key.stdString, configValue));
} else {
nativeConstraints.push_back(webrtc::MediaConstraintsInterface::Constraint(
key.stdString, value.stdString));
}
}
return nativeConstraints;
}
@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 "WebRTC/RTCMediaSource.h"
#include "api/mediastreaminterface.h"
NS_ASSUME_NONNULL_BEGIN
@class RTCPeerConnectionFactory;
typedef NS_ENUM(NSInteger, RTCMediaSourceType) {
RTCMediaSourceTypeAudio,
RTCMediaSourceTypeVideo,
};
@interface RTCMediaSource ()
@property(nonatomic, readonly) rtc::scoped_refptr<webrtc::MediaSourceInterface> nativeMediaSource;
- (instancetype)initWithFactory:(RTCPeerConnectionFactory *)factory
nativeMediaSource:(rtc::scoped_refptr<webrtc::MediaSourceInterface>)nativeMediaSource
type:(RTCMediaSourceType)type NS_DESIGNATED_INITIALIZER;
+ (webrtc::MediaSourceInterface::SourceState)nativeSourceStateForState:(RTCSourceState)state;
+ (RTCSourceState)sourceStateForNativeState:(webrtc::MediaSourceInterface::SourceState)nativeState;
+ (NSString *)stringForState:(RTCSourceState)state;
@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 "RTCMediaSource+Private.h"
#include "rtc_base/checks.h"
@implementation RTCMediaSource {
RTCPeerConnectionFactory *_factory;
RTCMediaSourceType _type;
}
@synthesize nativeMediaSource = _nativeMediaSource;
- (instancetype)initWithFactory:(RTCPeerConnectionFactory *)factory
nativeMediaSource:(rtc::scoped_refptr<webrtc::MediaSourceInterface>)nativeMediaSource
type:(RTCMediaSourceType)type {
RTC_DCHECK(factory);
RTC_DCHECK(nativeMediaSource);
if (self = [super init]) {
_factory = factory;
_nativeMediaSource = nativeMediaSource;
_type = type;
}
return self;
}
- (RTCSourceState)state {
return [[self class] sourceStateForNativeState:_nativeMediaSource->state()];
}
#pragma mark - Private
+ (webrtc::MediaSourceInterface::SourceState)nativeSourceStateForState:
(RTCSourceState)state {
switch (state) {
case RTCSourceStateInitializing:
return webrtc::MediaSourceInterface::kInitializing;
case RTCSourceStateLive:
return webrtc::MediaSourceInterface::kLive;
case RTCSourceStateEnded:
return webrtc::MediaSourceInterface::kEnded;
case RTCSourceStateMuted:
return webrtc::MediaSourceInterface::kMuted;
}
}
+ (RTCSourceState)sourceStateForNativeState:
(webrtc::MediaSourceInterface::SourceState)nativeState {
switch (nativeState) {
case webrtc::MediaSourceInterface::kInitializing:
return RTCSourceStateInitializing;
case webrtc::MediaSourceInterface::kLive:
return RTCSourceStateLive;
case webrtc::MediaSourceInterface::kEnded:
return RTCSourceStateEnded;
case webrtc::MediaSourceInterface::kMuted:
return RTCSourceStateMuted;
}
}
+ (NSString *)stringForState:(RTCSourceState)state {
switch (state) {
case RTCSourceStateInitializing:
return @"Initializing";
case RTCSourceStateLive:
return @"Live";
case RTCSourceStateEnded:
return @"Ended";
case RTCSourceStateMuted:
return @"Muted";
}
}
@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/RTCMediaStream.h"
#include "api/mediastreaminterface.h"
NS_ASSUME_NONNULL_BEGIN
@interface RTCMediaStream ()
/**
* MediaStreamInterface representation of this RTCMediaStream object. This is
* needed to pass to the underlying C++ APIs.
*/
@property(nonatomic, readonly) rtc::scoped_refptr<webrtc::MediaStreamInterface> nativeMediaStream;
/** Initialize an RTCMediaStream with an id. */
- (instancetype)initWithFactory:(RTCPeerConnectionFactory *)factory streamId:(NSString *)streamId;
/** Initialize an RTCMediaStream from a native MediaStreamInterface. */
- (instancetype)initWithFactory:(RTCPeerConnectionFactory *)factory
nativeMediaStream:(rtc::scoped_refptr<webrtc::MediaStreamInterface>)nativeMediaStream;
@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 "RTCMediaStream+Private.h"
#include <vector>
#import "NSString+StdString.h"
#import "RTCAudioTrack+Private.h"
#import "RTCMediaStreamTrack+Private.h"
#import "RTCPeerConnectionFactory+Private.h"
#import "RTCVideoTrack+Private.h"
@implementation RTCMediaStream {
RTCPeerConnectionFactory *_factory;
NSMutableArray *_audioTracks;
NSMutableArray *_videoTracks;
rtc::scoped_refptr<webrtc::MediaStreamInterface> _nativeMediaStream;
}
- (instancetype)initWithFactory:(RTCPeerConnectionFactory *)factory
streamId:(NSString *)streamId {
NSParameterAssert(factory);
NSParameterAssert(streamId.length);
std::string nativeId = [NSString stdStringForString:streamId];
rtc::scoped_refptr<webrtc::MediaStreamInterface> stream =
factory.nativeFactory->CreateLocalMediaStream(nativeId);
return [self initWithFactory:factory nativeMediaStream:stream];
}
- (NSArray<RTCAudioTrack *> *)audioTracks {
return [_audioTracks copy];
}
- (NSArray<RTCVideoTrack *> *)videoTracks {
return [_videoTracks copy];
}
- (NSString *)streamId {
return [NSString stringForStdString:_nativeMediaStream->id()];
}
- (void)addAudioTrack:(RTCAudioTrack *)audioTrack {
if (_nativeMediaStream->AddTrack(audioTrack.nativeAudioTrack)) {
[_audioTracks addObject:audioTrack];
}
}
- (void)addVideoTrack:(RTCVideoTrack *)videoTrack {
if (_nativeMediaStream->AddTrack(videoTrack.nativeVideoTrack)) {
[_videoTracks addObject:videoTrack];
}
}
- (void)removeAudioTrack:(RTCAudioTrack *)audioTrack {
NSUInteger index = [_audioTracks indexOfObjectIdenticalTo:audioTrack];
NSAssert(index != NSNotFound,
@"|removeAudioTrack| called on unexpected RTCAudioTrack");
if (index != NSNotFound &&
_nativeMediaStream->RemoveTrack(audioTrack.nativeAudioTrack)) {
[_audioTracks removeObjectAtIndex:index];
}
}
- (void)removeVideoTrack:(RTCVideoTrack *)videoTrack {
NSUInteger index = [_videoTracks indexOfObjectIdenticalTo:videoTrack];
NSAssert(index != NSNotFound,
@"|removeVideoTrack| called on unexpected RTCVideoTrack");
if (index != NSNotFound &&
_nativeMediaStream->RemoveTrack(videoTrack.nativeVideoTrack)) {
[_videoTracks removeObjectAtIndex:index];
}
}
- (NSString *)description {
return [NSString stringWithFormat:@"RTCMediaStream:\n%@\nA=%lu\nV=%lu",
self.streamId,
(unsigned long)self.audioTracks.count,
(unsigned long)self.videoTracks.count];
}
#pragma mark - Private
- (rtc::scoped_refptr<webrtc::MediaStreamInterface>)nativeMediaStream {
return _nativeMediaStream;
}
- (instancetype)initWithFactory:(RTCPeerConnectionFactory *)factory
nativeMediaStream:
(rtc::scoped_refptr<webrtc::MediaStreamInterface>)nativeMediaStream {
NSParameterAssert(nativeMediaStream);
if (self = [super init]) {
_factory = factory;
webrtc::AudioTrackVector audioTracks = nativeMediaStream->GetAudioTracks();
webrtc::VideoTrackVector videoTracks = nativeMediaStream->GetVideoTracks();
_audioTracks = [NSMutableArray arrayWithCapacity:audioTracks.size()];
_videoTracks = [NSMutableArray arrayWithCapacity:videoTracks.size()];
_nativeMediaStream = nativeMediaStream;
for (auto &track : audioTracks) {
RTCMediaStreamTrackType type = RTCMediaStreamTrackTypeAudio;
RTCAudioTrack *audioTrack =
[[RTCAudioTrack alloc] initWithFactory:_factory nativeTrack:track type:type];
[_audioTracks addObject:audioTrack];
}
for (auto &track : videoTracks) {
RTCMediaStreamTrackType type = RTCMediaStreamTrackTypeVideo;
RTCVideoTrack *videoTrack =
[[RTCVideoTrack alloc] initWithFactory:_factory nativeTrack:track type:type];
[_videoTracks addObject:videoTrack];
}
}
return self;
}
@end

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@ -1,60 +0,0 @@
/*
* 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"
#include "api/mediastreaminterface.h"
typedef NS_ENUM(NSInteger, RTCMediaStreamTrackType) {
RTCMediaStreamTrackTypeAudio,
RTCMediaStreamTrackTypeVideo,
};
NS_ASSUME_NONNULL_BEGIN
@class RTCPeerConnectionFactory;
@interface RTCMediaStreamTrack ()
@property(nonatomic, readonly) RTCPeerConnectionFactory *factory;
/**
* The native MediaStreamTrackInterface passed in or created during
* construction.
*/
@property(nonatomic, readonly) rtc::scoped_refptr<webrtc::MediaStreamTrackInterface> nativeTrack;
/**
* Initialize an RTCMediaStreamTrack from a native MediaStreamTrackInterface.
*/
- (instancetype)initWithFactory:(RTCPeerConnectionFactory *)factory
nativeTrack:(rtc::scoped_refptr<webrtc::MediaStreamTrackInterface>)nativeTrack
type:(RTCMediaStreamTrackType)type NS_DESIGNATED_INITIALIZER;
- (instancetype)initWithFactory:(RTCPeerConnectionFactory *)factory
nativeTrack:(rtc::scoped_refptr<webrtc::MediaStreamTrackInterface>)nativeTrack;
- (BOOL)isEqualToTrack:(RTCMediaStreamTrack *)track;
+ (webrtc::MediaStreamTrackInterface::TrackState)nativeTrackStateForState:
(RTCMediaStreamTrackState)state;
+ (RTCMediaStreamTrackState)trackStateForNativeState:
(webrtc::MediaStreamTrackInterface::TrackState)nativeState;
+ (NSString *)stringForState:(RTCMediaStreamTrackState)state;
+ (RTCMediaStreamTrack *)mediaTrackForNativeTrack:
(rtc::scoped_refptr<webrtc::MediaStreamTrackInterface>)nativeTrack
factory:(RTCPeerConnectionFactory *)factory;
@end
NS_ASSUME_NONNULL_END

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@ -1,160 +0,0 @@
/*
* 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 "RTCAudioTrack+Private.h"
#import "RTCMediaStreamTrack+Private.h"
#import "RTCVideoTrack+Private.h"
#import "NSString+StdString.h"
NSString * const kRTCMediaStreamTrackKindAudio =
@(webrtc::MediaStreamTrackInterface::kAudioKind);
NSString * const kRTCMediaStreamTrackKindVideo =
@(webrtc::MediaStreamTrackInterface::kVideoKind);
@implementation RTCMediaStreamTrack {
RTCPeerConnectionFactory *_factory;
rtc::scoped_refptr<webrtc::MediaStreamTrackInterface> _nativeTrack;
RTCMediaStreamTrackType _type;
}
- (NSString *)kind {
return [NSString stringForStdString:_nativeTrack->kind()];
}
- (NSString *)trackId {
return [NSString stringForStdString:_nativeTrack->id()];
}
- (BOOL)isEnabled {
return _nativeTrack->enabled();
}
- (void)setIsEnabled:(BOOL)isEnabled {
_nativeTrack->set_enabled(isEnabled);
}
- (RTCMediaStreamTrackState)readyState {
return [[self class] trackStateForNativeState:_nativeTrack->state()];
}
- (NSString *)description {
NSString *readyState = [[self class] stringForState:self.readyState];
return [NSString stringWithFormat:@"RTCMediaStreamTrack:\n%@\n%@\n%@\n%@",
self.kind,
self.trackId,
self.isEnabled ? @"enabled" : @"disabled",
readyState];
}
- (BOOL)isEqual:(id)object {
if (self == object) {
return YES;
}
if (![object isMemberOfClass:[self class]]) {
return NO;
}
return [self isEqualToTrack:(RTCMediaStreamTrack *)object];
}
- (NSUInteger)hash {
return (NSUInteger)_nativeTrack.get();
}
#pragma mark - Private
- (rtc::scoped_refptr<webrtc::MediaStreamTrackInterface>)nativeTrack {
return _nativeTrack;
}
@synthesize factory = _factory;
- (instancetype)initWithFactory:(RTCPeerConnectionFactory *)factory
nativeTrack:(rtc::scoped_refptr<webrtc::MediaStreamTrackInterface>)nativeTrack
type:(RTCMediaStreamTrackType)type {
NSParameterAssert(nativeTrack);
NSParameterAssert(factory);
if (self = [super init]) {
_factory = factory;
_nativeTrack = nativeTrack;
_type = type;
}
return self;
}
- (instancetype)initWithFactory:(RTCPeerConnectionFactory *)factory
nativeTrack:(rtc::scoped_refptr<webrtc::MediaStreamTrackInterface>)nativeTrack {
NSParameterAssert(nativeTrack);
if (nativeTrack->kind() ==
std::string(webrtc::MediaStreamTrackInterface::kAudioKind)) {
return [self initWithFactory:factory nativeTrack:nativeTrack type:RTCMediaStreamTrackTypeAudio];
}
if (nativeTrack->kind() ==
std::string(webrtc::MediaStreamTrackInterface::kVideoKind)) {
return [self initWithFactory:factory nativeTrack:nativeTrack type:RTCMediaStreamTrackTypeVideo];
}
return nil;
}
- (BOOL)isEqualToTrack:(RTCMediaStreamTrack *)track {
if (!track) {
return NO;
}
return _nativeTrack == track.nativeTrack;
}
+ (webrtc::MediaStreamTrackInterface::TrackState)nativeTrackStateForState:
(RTCMediaStreamTrackState)state {
switch (state) {
case RTCMediaStreamTrackStateLive:
return webrtc::MediaStreamTrackInterface::kLive;
case RTCMediaStreamTrackStateEnded:
return webrtc::MediaStreamTrackInterface::kEnded;
}
}
+ (RTCMediaStreamTrackState)trackStateForNativeState:
(webrtc::MediaStreamTrackInterface::TrackState)nativeState {
switch (nativeState) {
case webrtc::MediaStreamTrackInterface::kLive:
return RTCMediaStreamTrackStateLive;
case webrtc::MediaStreamTrackInterface::kEnded:
return RTCMediaStreamTrackStateEnded;
}
}
+ (NSString *)stringForState:(RTCMediaStreamTrackState)state {
switch (state) {
case RTCMediaStreamTrackStateLive:
return @"Live";
case RTCMediaStreamTrackStateEnded:
return @"Ended";
}
}
+ (RTCMediaStreamTrack *)mediaTrackForNativeTrack:
(rtc::scoped_refptr<webrtc::MediaStreamTrackInterface>)nativeTrack
factory:(RTCPeerConnectionFactory *)factory {
NSParameterAssert(nativeTrack);
NSParameterAssert(factory);
if (nativeTrack->kind() == webrtc::MediaStreamTrackInterface::kAudioKind) {
return [[RTCAudioTrack alloc] initWithFactory:factory
nativeTrack:nativeTrack
type:RTCMediaStreamTrackTypeAudio];
} else if (nativeTrack->kind() == webrtc::MediaStreamTrackInterface::kVideoKind) {
return [[RTCVideoTrack alloc] initWithFactory:factory
nativeTrack:nativeTrack
type:RTCMediaStreamTrackTypeVideo];
} else {
return [[RTCMediaStreamTrack alloc] initWithFactory:factory nativeTrack:nativeTrack];
}
}
@end

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@ -1,32 +0,0 @@
/*
* 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 "WebRTC/RTCMetrics.h"
#import "RTCMetricsSampleInfo+Private.h"
void RTCEnableMetrics(void) {
webrtc::metrics::Enable();
}
NSArray<RTCMetricsSampleInfo *> *RTCGetAndResetMetrics(void) {
std::map<std::string, std::unique_ptr<webrtc::metrics::SampleInfo>>
histograms;
webrtc::metrics::GetAndReset(&histograms);
NSMutableArray *metrics =
[NSMutableArray arrayWithCapacity:histograms.size()];
for (auto const &histogram : histograms) {
RTCMetricsSampleInfo *metric = [[RTCMetricsSampleInfo alloc]
initWithNativeSampleInfo:*histogram.second];
[metrics addObject:metric];
}
return metrics;
}

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@ -1,27 +0,0 @@
/*
* 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 "WebRTC/RTCMetricsSampleInfo.h"
// Adding 'nogncheck' to disable the gn include headers check.
// We don't want to depend on 'system_wrappers:metrics_default' because
// clients should be able to provide their own implementation.
#include "system_wrappers/include/metrics_default.h" // nogncheck
NS_ASSUME_NONNULL_BEGIN
@interface RTCMetricsSampleInfo ()
/** Initialize an RTCMetricsSampleInfo object from native SampleInfo. */
- (instancetype)initWithNativeSampleInfo:(const webrtc::metrics::SampleInfo &)info;
@end
NS_ASSUME_NONNULL_END

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@ -1,43 +0,0 @@
/*
* 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 "RTCMetricsSampleInfo+Private.h"
#import "NSString+StdString.h"
@implementation RTCMetricsSampleInfo
@synthesize name = _name;
@synthesize min = _min;
@synthesize max = _max;
@synthesize bucketCount = _bucketCount;
@synthesize samples = _samples;
#pragma mark - Private
- (instancetype)initWithNativeSampleInfo:
(const webrtc::metrics::SampleInfo &)info {
if (self = [super init]) {
_name = [NSString stringForStdString:info.name];
_min = info.min;
_max = info.max;
_bucketCount = info.bucket_count;
NSMutableDictionary *samples =
[NSMutableDictionary dictionaryWithCapacity:info.samples.size()];
for (auto const &sample : info.samples) {
[samples setObject:@(sample.second) forKey:@(sample.first)];
}
_samples = samples;
}
return self;
}
@end

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@ -1,33 +0,0 @@
/*
* 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 "RTCPeerConnection+Private.h"
#import "NSString+StdString.h"
#import "RTCDataChannel+Private.h"
#import "RTCDataChannelConfiguration+Private.h"
@implementation RTCPeerConnection (DataChannel)
- (nullable RTCDataChannel *)dataChannelForLabel:(NSString *)label
configuration:(RTCDataChannelConfiguration *)configuration {
std::string labelString = [NSString stdStringForString:label];
const webrtc::DataChannelInit nativeInit =
configuration.nativeDataChannelInit;
rtc::scoped_refptr<webrtc::DataChannelInterface> dataChannel =
self.nativePeerConnection->CreateDataChannel(labelString,
&nativeInit);
if (!dataChannel) {
return nil;
}
return [[RTCDataChannel alloc] initWithFactory:self.factory nativeDataChannel:dataChannel];
}
@end

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@ -8,27 +8,4 @@
* be found in the AUTHORS file in the root of the source tree.
*/
#import "WebRTC/RTCPeerConnection.h"
#include <memory>
namespace rtc {
class BitrateAllocationStrategy;
} // namespace rtc
NS_ASSUME_NONNULL_BEGIN
/**
* This class extension exposes methods that work directly with injectable C++ components.
*/
@interface RTCPeerConnection ()
/** Sets current strategy. If not set default WebRTC allocator will be used. May be changed during
* an active session.
*/
- (void)setBitrateAllocationStrategy:
(std::unique_ptr<rtc::BitrateAllocationStrategy>)bitrateAllocationStrategy;
@end
NS_ASSUME_NONNULL_END
#import "api/peerconnection/RTCPeerConnection+Native.h"

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@ -1,106 +0,0 @@
/*
* 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/RTCPeerConnection.h"
#include "api/peerconnectioninterface.h"
NS_ASSUME_NONNULL_BEGIN
namespace webrtc {
/**
* These objects are created by RTCPeerConnectionFactory to wrap an
* id<RTCPeerConnectionDelegate> and call methods on that interface.
*/
class PeerConnectionDelegateAdapter : public PeerConnectionObserver {
public:
PeerConnectionDelegateAdapter(RTCPeerConnection *peerConnection);
~PeerConnectionDelegateAdapter() override;
void OnSignalingChange(PeerConnectionInterface::SignalingState new_state) override;
void OnAddStream(rtc::scoped_refptr<MediaStreamInterface> stream) override;
void OnRemoveStream(rtc::scoped_refptr<MediaStreamInterface> stream) override;
void OnTrack(rtc::scoped_refptr<RtpTransceiverInterface> transceiver) override;
void OnDataChannel(rtc::scoped_refptr<DataChannelInterface> data_channel) override;
void OnRenegotiationNeeded() override;
void OnIceConnectionChange(PeerConnectionInterface::IceConnectionState new_state) override;
void OnIceGatheringChange(PeerConnectionInterface::IceGatheringState new_state) override;
void OnIceCandidate(const IceCandidateInterface *candidate) override;
void OnIceCandidatesRemoved(const std::vector<cricket::Candidate> &candidates) override;
void OnAddTrack(rtc::scoped_refptr<RtpReceiverInterface> receiver,
const std::vector<rtc::scoped_refptr<MediaStreamInterface>> &streams) override;
void OnRemoveTrack(rtc::scoped_refptr<RtpReceiverInterface> receiver) override;
private:
__weak RTCPeerConnection *peer_connection_;
};
} // namespace webrtc
@interface RTCPeerConnection ()
/** The factory used to create this RTCPeerConnection */
@property(nonatomic, readonly) RTCPeerConnectionFactory *factory;
/** The native PeerConnectionInterface created during construction. */
@property(nonatomic, readonly) rtc::scoped_refptr<webrtc::PeerConnectionInterface>
nativePeerConnection;
/** Initialize an RTCPeerConnection with a configuration, constraints, and
* delegate.
*/
- (instancetype)initWithFactory:(RTCPeerConnectionFactory *)factory
configuration:(RTCConfiguration *)configuration
constraints:(RTCMediaConstraints *)constraints
delegate:(nullable id<RTCPeerConnectionDelegate>)delegate
NS_DESIGNATED_INITIALIZER;
+ (webrtc::PeerConnectionInterface::SignalingState)nativeSignalingStateForState:
(RTCSignalingState)state;
+ (RTCSignalingState)signalingStateForNativeState:
(webrtc::PeerConnectionInterface::SignalingState)nativeState;
+ (NSString *)stringForSignalingState:(RTCSignalingState)state;
+ (webrtc::PeerConnectionInterface::IceConnectionState)nativeIceConnectionStateForState:
(RTCIceConnectionState)state;
+ (RTCIceConnectionState)iceConnectionStateForNativeState:
(webrtc::PeerConnectionInterface::IceConnectionState)nativeState;
+ (NSString *)stringForIceConnectionState:(RTCIceConnectionState)state;
+ (webrtc::PeerConnectionInterface::IceGatheringState)nativeIceGatheringStateForState:
(RTCIceGatheringState)state;
+ (RTCIceGatheringState)iceGatheringStateForNativeState:
(webrtc::PeerConnectionInterface::IceGatheringState)nativeState;
+ (NSString *)stringForIceGatheringState:(RTCIceGatheringState)state;
+ (webrtc::PeerConnectionInterface::StatsOutputLevel)nativeStatsOutputLevelForLevel:
(RTCStatsOutputLevel)level;
@end
NS_ASSUME_NONNULL_END

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@ -1,62 +0,0 @@
/*
* 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 "RTCPeerConnection+Private.h"
#import "NSString+StdString.h"
#import "RTCMediaStreamTrack+Private.h"
#import "RTCLegacyStatsReport+Private.h"
#include "rtc_base/checks.h"
namespace webrtc {
class StatsObserverAdapter : public StatsObserver {
public:
StatsObserverAdapter(void (^completionHandler)
(NSArray<RTCLegacyStatsReport *> *stats)) {
completion_handler_ = completionHandler;
}
~StatsObserverAdapter() override { completion_handler_ = nil; }
void OnComplete(const StatsReports& reports) override {
RTC_DCHECK(completion_handler_);
NSMutableArray *stats = [NSMutableArray arrayWithCapacity:reports.size()];
for (const auto* report : reports) {
RTCLegacyStatsReport *statsReport =
[[RTCLegacyStatsReport alloc] initWithNativeReport:*report];
[stats addObject:statsReport];
}
completion_handler_(stats);
completion_handler_ = nil;
}
private:
void (^completion_handler_)(NSArray<RTCLegacyStatsReport *> *stats);
};
} // namespace webrtc
@implementation RTCPeerConnection (Stats)
- (void)statsForTrack:(RTCMediaStreamTrack *)mediaStreamTrack
statsOutputLevel:(RTCStatsOutputLevel)statsOutputLevel
completionHandler:
(void (^)(NSArray<RTCLegacyStatsReport *> *stats))completionHandler {
rtc::scoped_refptr<webrtc::StatsObserverAdapter> observer(
new rtc::RefCountedObject<webrtc::StatsObserverAdapter>
(completionHandler));
webrtc::PeerConnectionInterface::StatsOutputLevel nativeOutputLevel =
[[self class] nativeStatsOutputLevelForLevel:statsOutputLevel];
self.nativePeerConnection->GetStats(
observer, mediaStreamTrack.nativeTrack, nativeOutputLevel);
}
@end

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@ -1,748 +0,0 @@
/*
* 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 "RTCPeerConnection+Private.h"
#import "NSString+StdString.h"
#import "RTCConfiguration+Private.h"
#import "RTCDataChannel+Private.h"
#import "RTCIceCandidate+Private.h"
#import "RTCLegacyStatsReport+Private.h"
#import "RTCMediaConstraints+Private.h"
#import "RTCMediaStream+Private.h"
#import "RTCMediaStreamTrack+Private.h"
#import "RTCPeerConnection+Native.h"
#import "RTCPeerConnectionFactory+Private.h"
#import "RTCRtpReceiver+Private.h"
#import "RTCRtpSender+Private.h"
#import "RTCRtpTransceiver+Private.h"
#import "RTCSessionDescription+Private.h"
#import "WebRTC/RTCLogging.h"
#include <memory>
#include "api/jsepicecandidate.h"
#include "rtc_base/checks.h"
NSString * const kRTCPeerConnectionErrorDomain =
@"org.webrtc.RTCPeerConnection";
int const kRTCPeerConnnectionSessionDescriptionError = -1;
namespace webrtc {
class CreateSessionDescriptionObserverAdapter
: public CreateSessionDescriptionObserver {
public:
CreateSessionDescriptionObserverAdapter(
void (^completionHandler)(RTCSessionDescription *sessionDescription,
NSError *error)) {
completion_handler_ = completionHandler;
}
~CreateSessionDescriptionObserverAdapter() override { completion_handler_ = nil; }
void OnSuccess(SessionDescriptionInterface *desc) override {
RTC_DCHECK(completion_handler_);
std::unique_ptr<webrtc::SessionDescriptionInterface> description =
std::unique_ptr<webrtc::SessionDescriptionInterface>(desc);
RTCSessionDescription* session =
[[RTCSessionDescription alloc] initWithNativeDescription:
description.get()];
completion_handler_(session, nil);
completion_handler_ = nil;
}
void OnFailure(RTCError error) override {
RTC_DCHECK(completion_handler_);
// TODO(hta): Add handling of error.type()
NSString *str = [NSString stringForStdString:error.message()];
NSError* err =
[NSError errorWithDomain:kRTCPeerConnectionErrorDomain
code:kRTCPeerConnnectionSessionDescriptionError
userInfo:@{ NSLocalizedDescriptionKey : str }];
completion_handler_(nil, err);
completion_handler_ = nil;
}
private:
void (^completion_handler_)
(RTCSessionDescription *sessionDescription, NSError *error);
};
class SetSessionDescriptionObserverAdapter :
public SetSessionDescriptionObserver {
public:
SetSessionDescriptionObserverAdapter(void (^completionHandler)
(NSError *error)) {
completion_handler_ = completionHandler;
}
~SetSessionDescriptionObserverAdapter() override { completion_handler_ = nil; }
void OnSuccess() override {
RTC_DCHECK(completion_handler_);
completion_handler_(nil);
completion_handler_ = nil;
}
void OnFailure(RTCError error) override {
RTC_DCHECK(completion_handler_);
// TODO(hta): Add handling of error.type()
NSString *str = [NSString stringForStdString:error.message()];
NSError* err =
[NSError errorWithDomain:kRTCPeerConnectionErrorDomain
code:kRTCPeerConnnectionSessionDescriptionError
userInfo:@{ NSLocalizedDescriptionKey : str }];
completion_handler_(err);
completion_handler_ = nil;
}
private:
void (^completion_handler_)(NSError *error);
};
PeerConnectionDelegateAdapter::PeerConnectionDelegateAdapter(
RTCPeerConnection *peerConnection) {
peer_connection_ = peerConnection;
}
PeerConnectionDelegateAdapter::~PeerConnectionDelegateAdapter() {
peer_connection_ = nil;
}
void PeerConnectionDelegateAdapter::OnSignalingChange(
PeerConnectionInterface::SignalingState new_state) {
RTCSignalingState state =
[[RTCPeerConnection class] signalingStateForNativeState:new_state];
RTCPeerConnection *peer_connection = peer_connection_;
[peer_connection.delegate peerConnection:peer_connection
didChangeSignalingState:state];
}
void PeerConnectionDelegateAdapter::OnAddStream(
rtc::scoped_refptr<MediaStreamInterface> stream) {
RTCPeerConnection *peer_connection = peer_connection_;
RTCMediaStream *mediaStream =
[[RTCMediaStream alloc] initWithFactory:peer_connection.factory nativeMediaStream:stream];
[peer_connection.delegate peerConnection:peer_connection
didAddStream:mediaStream];
}
void PeerConnectionDelegateAdapter::OnRemoveStream(
rtc::scoped_refptr<MediaStreamInterface> stream) {
RTCPeerConnection *peer_connection = peer_connection_;
RTCMediaStream *mediaStream =
[[RTCMediaStream alloc] initWithFactory:peer_connection.factory nativeMediaStream:stream];
[peer_connection.delegate peerConnection:peer_connection
didRemoveStream:mediaStream];
}
void PeerConnectionDelegateAdapter::OnTrack(
rtc::scoped_refptr<RtpTransceiverInterface> nativeTransceiver) {
RTCPeerConnection *peer_connection = peer_connection_;
RTCRtpTransceiver *transceiver =
[[RTCRtpTransceiver alloc] initWithFactory:peer_connection.factory
nativeRtpTransceiver:nativeTransceiver];
if ([peer_connection.delegate
respondsToSelector:@selector(peerConnection:didStartReceivingOnTransceiver:)]) {
[peer_connection.delegate peerConnection:peer_connection
didStartReceivingOnTransceiver:transceiver];
}
}
void PeerConnectionDelegateAdapter::OnDataChannel(
rtc::scoped_refptr<DataChannelInterface> data_channel) {
RTCPeerConnection *peer_connection = peer_connection_;
RTCDataChannel *dataChannel = [[RTCDataChannel alloc] initWithFactory:peer_connection.factory
nativeDataChannel:data_channel];
[peer_connection.delegate peerConnection:peer_connection
didOpenDataChannel:dataChannel];
}
void PeerConnectionDelegateAdapter::OnRenegotiationNeeded() {
RTCPeerConnection *peer_connection = peer_connection_;
[peer_connection.delegate peerConnectionShouldNegotiate:peer_connection];
}
void PeerConnectionDelegateAdapter::OnIceConnectionChange(
PeerConnectionInterface::IceConnectionState new_state) {
RTCIceConnectionState state =
[[RTCPeerConnection class] iceConnectionStateForNativeState:new_state];
RTCPeerConnection *peer_connection = peer_connection_;
[peer_connection.delegate peerConnection:peer_connection
didChangeIceConnectionState:state];
}
void PeerConnectionDelegateAdapter::OnIceGatheringChange(
PeerConnectionInterface::IceGatheringState new_state) {
RTCIceGatheringState state =
[[RTCPeerConnection class] iceGatheringStateForNativeState:new_state];
RTCPeerConnection *peer_connection = peer_connection_;
[peer_connection.delegate peerConnection:peer_connection
didChangeIceGatheringState:state];
}
void PeerConnectionDelegateAdapter::OnIceCandidate(
const IceCandidateInterface *candidate) {
RTCIceCandidate *iceCandidate =
[[RTCIceCandidate alloc] initWithNativeCandidate:candidate];
RTCPeerConnection *peer_connection = peer_connection_;
[peer_connection.delegate peerConnection:peer_connection
didGenerateIceCandidate:iceCandidate];
}
void PeerConnectionDelegateAdapter::OnIceCandidatesRemoved(
const std::vector<cricket::Candidate>& candidates) {
NSMutableArray* ice_candidates =
[NSMutableArray arrayWithCapacity:candidates.size()];
for (const auto& candidate : candidates) {
std::unique_ptr<JsepIceCandidate> candidate_wrapper(
new JsepIceCandidate(candidate.transport_name(), -1, candidate));
RTCIceCandidate* ice_candidate = [[RTCIceCandidate alloc]
initWithNativeCandidate:candidate_wrapper.get()];
[ice_candidates addObject:ice_candidate];
}
RTCPeerConnection* peer_connection = peer_connection_;
[peer_connection.delegate peerConnection:peer_connection
didRemoveIceCandidates:ice_candidates];
}
void PeerConnectionDelegateAdapter::OnAddTrack(
rtc::scoped_refptr<RtpReceiverInterface> receiver,
const std::vector<rtc::scoped_refptr<MediaStreamInterface>>& streams) {
RTCPeerConnection *peer_connection = peer_connection_;
if ([peer_connection.delegate
respondsToSelector:@selector(peerConnection:didAddReceiver:streams:)]) {
NSMutableArray *mediaStreams = [NSMutableArray arrayWithCapacity:streams.size()];
for (const auto& nativeStream : streams) {
RTCMediaStream *mediaStream = [[RTCMediaStream alloc] initWithFactory:peer_connection.factory
nativeMediaStream:nativeStream];
[mediaStreams addObject:mediaStream];
}
RTCRtpReceiver *rtpReceiver =
[[RTCRtpReceiver alloc] initWithFactory:peer_connection.factory nativeRtpReceiver:receiver];
[peer_connection.delegate peerConnection:peer_connection
didAddReceiver:rtpReceiver
streams:mediaStreams];
}
}
void PeerConnectionDelegateAdapter::OnRemoveTrack(
rtc::scoped_refptr<RtpReceiverInterface> receiver) {
RTCPeerConnection *peer_connection = peer_connection_;
if ([peer_connection.delegate respondsToSelector:@selector(peerConnection:didRemoveReceiver:)]) {
RTCRtpReceiver *rtpReceiver =
[[RTCRtpReceiver alloc] initWithFactory:peer_connection.factory nativeRtpReceiver:receiver];
[peer_connection.delegate peerConnection:peer_connection didRemoveReceiver:rtpReceiver];
}
}
} // namespace webrtc
@implementation RTCPeerConnection {
RTCPeerConnectionFactory *_factory;
NSMutableArray<RTCMediaStream *> *_localStreams;
std::unique_ptr<webrtc::PeerConnectionDelegateAdapter> _observer;
rtc::scoped_refptr<webrtc::PeerConnectionInterface> _peerConnection;
std::unique_ptr<webrtc::MediaConstraints> _nativeConstraints;
BOOL _hasStartedRtcEventLog;
}
@synthesize delegate = _delegate;
@synthesize factory = _factory;
- (instancetype)initWithFactory:(RTCPeerConnectionFactory *)factory
configuration:(RTCConfiguration *)configuration
constraints:(RTCMediaConstraints *)constraints
delegate:(id<RTCPeerConnectionDelegate>)delegate {
NSParameterAssert(factory);
std::unique_ptr<webrtc::PeerConnectionInterface::RTCConfiguration> config(
[configuration createNativeConfiguration]);
if (!config) {
return nil;
}
if (self = [super init]) {
_observer.reset(new webrtc::PeerConnectionDelegateAdapter(self));
_nativeConstraints = constraints.nativeConstraints;
CopyConstraintsIntoRtcConfiguration(_nativeConstraints.get(),
config.get());
_peerConnection =
factory.nativeFactory->CreatePeerConnection(*config,
nullptr,
nullptr,
_observer.get());
if (!_peerConnection) {
return nil;
}
_factory = factory;
_localStreams = [[NSMutableArray alloc] init];
_delegate = delegate;
}
return self;
}
- (NSArray<RTCMediaStream *> *)localStreams {
return [_localStreams copy];
}
- (RTCSessionDescription *)localDescription {
const webrtc::SessionDescriptionInterface *description =
_peerConnection->local_description();
return description ?
[[RTCSessionDescription alloc] initWithNativeDescription:description]
: nil;
}
- (RTCSessionDescription *)remoteDescription {
const webrtc::SessionDescriptionInterface *description =
_peerConnection->remote_description();
return description ?
[[RTCSessionDescription alloc] initWithNativeDescription:description]
: nil;
}
- (RTCSignalingState)signalingState {
return [[self class]
signalingStateForNativeState:_peerConnection->signaling_state()];
}
- (RTCIceConnectionState)iceConnectionState {
return [[self class] iceConnectionStateForNativeState:
_peerConnection->ice_connection_state()];
}
- (RTCIceGatheringState)iceGatheringState {
return [[self class] iceGatheringStateForNativeState:
_peerConnection->ice_gathering_state()];
}
- (BOOL)setConfiguration:(RTCConfiguration *)configuration {
std::unique_ptr<webrtc::PeerConnectionInterface::RTCConfiguration> config(
[configuration createNativeConfiguration]);
if (!config) {
return NO;
}
CopyConstraintsIntoRtcConfiguration(_nativeConstraints.get(),
config.get());
return _peerConnection->SetConfiguration(*config);
}
- (RTCConfiguration *)configuration {
webrtc::PeerConnectionInterface::RTCConfiguration config =
_peerConnection->GetConfiguration();
return [[RTCConfiguration alloc] initWithNativeConfiguration:config];
}
- (void)close {
_peerConnection->Close();
}
- (void)addIceCandidate:(RTCIceCandidate *)candidate {
std::unique_ptr<const webrtc::IceCandidateInterface> iceCandidate(
candidate.nativeCandidate);
_peerConnection->AddIceCandidate(iceCandidate.get());
}
- (void)removeIceCandidates:(NSArray<RTCIceCandidate *> *)iceCandidates {
std::vector<cricket::Candidate> candidates;
for (RTCIceCandidate *iceCandidate in iceCandidates) {
std::unique_ptr<const webrtc::IceCandidateInterface> candidate(
iceCandidate.nativeCandidate);
if (candidate) {
candidates.push_back(candidate->candidate());
// Need to fill the transport name from the sdp_mid.
candidates.back().set_transport_name(candidate->sdp_mid());
}
}
if (!candidates.empty()) {
_peerConnection->RemoveIceCandidates(candidates);
}
}
- (void)addStream:(RTCMediaStream *)stream {
if (!_peerConnection->AddStream(stream.nativeMediaStream)) {
RTCLogError(@"Failed to add stream: %@", stream);
return;
}
[_localStreams addObject:stream];
}
- (void)removeStream:(RTCMediaStream *)stream {
_peerConnection->RemoveStream(stream.nativeMediaStream);
[_localStreams removeObject:stream];
}
- (RTCRtpSender *)addTrack:(RTCMediaStreamTrack *)track streamIds:(NSArray<NSString *> *)streamIds {
std::vector<std::string> nativeStreamIds;
for (NSString *streamId in streamIds) {
nativeStreamIds.push_back([streamId UTF8String]);
}
webrtc::RTCErrorOr<rtc::scoped_refptr<webrtc::RtpSenderInterface>> nativeSenderOrError =
_peerConnection->AddTrack(track.nativeTrack, nativeStreamIds);
if (!nativeSenderOrError.ok()) {
RTCLogError(@"Failed to add track %@: %s", track, nativeSenderOrError.error().message());
return nil;
}
return [[RTCRtpSender alloc] initWithFactory:self.factory
nativeRtpSender:nativeSenderOrError.MoveValue()];
}
- (BOOL)removeTrack:(RTCRtpSender *)sender {
bool result = _peerConnection->RemoveTrack(sender.nativeRtpSender);
if (!result) {
RTCLogError(@"Failed to remote track %@", sender);
}
return result;
}
- (RTCRtpTransceiver *)addTransceiverWithTrack:(RTCMediaStreamTrack *)track {
return [self addTransceiverWithTrack:track init:[[RTCRtpTransceiverInit alloc] init]];
}
- (RTCRtpTransceiver *)addTransceiverWithTrack:(RTCMediaStreamTrack *)track
init:(RTCRtpTransceiverInit *)init {
webrtc::RTCErrorOr<rtc::scoped_refptr<webrtc::RtpTransceiverInterface>> nativeTransceiverOrError =
_peerConnection->AddTransceiver(track.nativeTrack, init.nativeInit);
if (!nativeTransceiverOrError.ok()) {
RTCLogError(
@"Failed to add transceiver %@: %s", track, nativeTransceiverOrError.error().message());
return nil;
}
return [[RTCRtpTransceiver alloc] initWithFactory:self.factory
nativeRtpTransceiver:nativeTransceiverOrError.MoveValue()];
}
- (RTCRtpTransceiver *)addTransceiverOfType:(RTCRtpMediaType)mediaType {
return [self addTransceiverOfType:mediaType init:[[RTCRtpTransceiverInit alloc] init]];
}
- (RTCRtpTransceiver *)addTransceiverOfType:(RTCRtpMediaType)mediaType
init:(RTCRtpTransceiverInit *)init {
webrtc::RTCErrorOr<rtc::scoped_refptr<webrtc::RtpTransceiverInterface>> nativeTransceiverOrError =
_peerConnection->AddTransceiver([RTCRtpReceiver nativeMediaTypeForMediaType:mediaType],
init.nativeInit);
if (!nativeTransceiverOrError.ok()) {
RTCLogError(@"Failed to add transceiver %@: %s",
[RTCRtpReceiver stringForMediaType:mediaType],
nativeTransceiverOrError.error().message());
return nil;
}
return [[RTCRtpTransceiver alloc] initWithFactory:self.factory
nativeRtpTransceiver:nativeTransceiverOrError.MoveValue()];
}
- (void)offerForConstraints:(RTCMediaConstraints *)constraints
completionHandler:
(void (^)(RTCSessionDescription *sessionDescription,
NSError *error))completionHandler {
rtc::scoped_refptr<webrtc::CreateSessionDescriptionObserverAdapter>
observer(new rtc::RefCountedObject
<webrtc::CreateSessionDescriptionObserverAdapter>(completionHandler));
webrtc::PeerConnectionInterface::RTCOfferAnswerOptions options;
CopyConstraintsIntoOfferAnswerOptions(constraints.nativeConstraints.get(), &options);
_peerConnection->CreateOffer(observer, options);
}
- (void)answerForConstraints:(RTCMediaConstraints *)constraints
completionHandler:
(void (^)(RTCSessionDescription *sessionDescription,
NSError *error))completionHandler {
rtc::scoped_refptr<webrtc::CreateSessionDescriptionObserverAdapter>
observer(new rtc::RefCountedObject
<webrtc::CreateSessionDescriptionObserverAdapter>(completionHandler));
webrtc::PeerConnectionInterface::RTCOfferAnswerOptions options;
CopyConstraintsIntoOfferAnswerOptions(constraints.nativeConstraints.get(), &options);
_peerConnection->CreateAnswer(observer, options);
}
- (void)setLocalDescription:(RTCSessionDescription *)sdp
completionHandler:(void (^)(NSError *error))completionHandler {
rtc::scoped_refptr<webrtc::SetSessionDescriptionObserverAdapter> observer(
new rtc::RefCountedObject<webrtc::SetSessionDescriptionObserverAdapter>(
completionHandler));
_peerConnection->SetLocalDescription(observer, sdp.nativeDescription);
}
- (void)setRemoteDescription:(RTCSessionDescription *)sdp
completionHandler:(void (^)(NSError *error))completionHandler {
rtc::scoped_refptr<webrtc::SetSessionDescriptionObserverAdapter> observer(
new rtc::RefCountedObject<webrtc::SetSessionDescriptionObserverAdapter>(
completionHandler));
_peerConnection->SetRemoteDescription(observer, sdp.nativeDescription);
}
- (BOOL)setBweMinBitrateBps:(nullable NSNumber *)minBitrateBps
currentBitrateBps:(nullable NSNumber *)currentBitrateBps
maxBitrateBps:(nullable NSNumber *)maxBitrateBps {
webrtc::PeerConnectionInterface::BitrateParameters params;
if (minBitrateBps != nil) {
params.min_bitrate_bps = absl::optional<int>(minBitrateBps.intValue);
}
if (currentBitrateBps != nil) {
params.current_bitrate_bps = absl::optional<int>(currentBitrateBps.intValue);
}
if (maxBitrateBps != nil) {
params.max_bitrate_bps = absl::optional<int>(maxBitrateBps.intValue);
}
return _peerConnection->SetBitrate(params).ok();
}
- (void)setBitrateAllocationStrategy:
(std::unique_ptr<rtc::BitrateAllocationStrategy>)bitrateAllocationStrategy {
_peerConnection->SetBitrateAllocationStrategy(std::move(bitrateAllocationStrategy));
}
- (BOOL)startRtcEventLogWithFilePath:(NSString *)filePath
maxSizeInBytes:(int64_t)maxSizeInBytes {
RTC_DCHECK(filePath.length);
RTC_DCHECK_GT(maxSizeInBytes, 0);
RTC_DCHECK(!_hasStartedRtcEventLog);
if (_hasStartedRtcEventLog) {
RTCLogError(@"Event logging already started.");
return NO;
}
int fd = open(filePath.UTF8String, O_WRONLY | O_CREAT | O_TRUNC,
S_IRUSR | S_IWUSR);
if (fd < 0) {
RTCLogError(@"Error opening file: %@. Error: %d", filePath, errno);
return NO;
}
_hasStartedRtcEventLog =
_peerConnection->StartRtcEventLog(fd, maxSizeInBytes);
return _hasStartedRtcEventLog;
}
- (void)stopRtcEventLog {
_peerConnection->StopRtcEventLog();
_hasStartedRtcEventLog = NO;
}
- (RTCRtpSender *)senderWithKind:(NSString *)kind
streamId:(NSString *)streamId {
std::string nativeKind = [NSString stdStringForString:kind];
std::string nativeStreamId = [NSString stdStringForString:streamId];
rtc::scoped_refptr<webrtc::RtpSenderInterface> nativeSender(
_peerConnection->CreateSender(nativeKind, nativeStreamId));
return nativeSender ?
[[RTCRtpSender alloc] initWithFactory:self.factory nativeRtpSender:nativeSender] :
nil;
}
- (NSArray<RTCRtpSender *> *)senders {
std::vector<rtc::scoped_refptr<webrtc::RtpSenderInterface>> nativeSenders(
_peerConnection->GetSenders());
NSMutableArray *senders = [[NSMutableArray alloc] init];
for (const auto &nativeSender : nativeSenders) {
RTCRtpSender *sender =
[[RTCRtpSender alloc] initWithFactory:self.factory nativeRtpSender:nativeSender];
[senders addObject:sender];
}
return senders;
}
- (NSArray<RTCRtpReceiver *> *)receivers {
std::vector<rtc::scoped_refptr<webrtc::RtpReceiverInterface>> nativeReceivers(
_peerConnection->GetReceivers());
NSMutableArray *receivers = [[NSMutableArray alloc] init];
for (const auto &nativeReceiver : nativeReceivers) {
RTCRtpReceiver *receiver =
[[RTCRtpReceiver alloc] initWithFactory:self.factory nativeRtpReceiver:nativeReceiver];
[receivers addObject:receiver];
}
return receivers;
}
- (NSArray<RTCRtpTransceiver *> *)transceivers {
std::vector<rtc::scoped_refptr<webrtc::RtpTransceiverInterface>> nativeTransceivers(
_peerConnection->GetTransceivers());
NSMutableArray *transceivers = [[NSMutableArray alloc] init];
for (auto nativeTransceiver : nativeTransceivers) {
RTCRtpTransceiver *transceiver = [[RTCRtpTransceiver alloc] initWithFactory:self.factory
nativeRtpTransceiver:nativeTransceiver];
[transceivers addObject:transceiver];
}
return transceivers;
}
#pragma mark - Private
+ (webrtc::PeerConnectionInterface::SignalingState)nativeSignalingStateForState:
(RTCSignalingState)state {
switch (state) {
case RTCSignalingStateStable:
return webrtc::PeerConnectionInterface::kStable;
case RTCSignalingStateHaveLocalOffer:
return webrtc::PeerConnectionInterface::kHaveLocalOffer;
case RTCSignalingStateHaveLocalPrAnswer:
return webrtc::PeerConnectionInterface::kHaveLocalPrAnswer;
case RTCSignalingStateHaveRemoteOffer:
return webrtc::PeerConnectionInterface::kHaveRemoteOffer;
case RTCSignalingStateHaveRemotePrAnswer:
return webrtc::PeerConnectionInterface::kHaveRemotePrAnswer;
case RTCSignalingStateClosed:
return webrtc::PeerConnectionInterface::kClosed;
}
}
+ (RTCSignalingState)signalingStateForNativeState:
(webrtc::PeerConnectionInterface::SignalingState)nativeState {
switch (nativeState) {
case webrtc::PeerConnectionInterface::kStable:
return RTCSignalingStateStable;
case webrtc::PeerConnectionInterface::kHaveLocalOffer:
return RTCSignalingStateHaveLocalOffer;
case webrtc::PeerConnectionInterface::kHaveLocalPrAnswer:
return RTCSignalingStateHaveLocalPrAnswer;
case webrtc::PeerConnectionInterface::kHaveRemoteOffer:
return RTCSignalingStateHaveRemoteOffer;
case webrtc::PeerConnectionInterface::kHaveRemotePrAnswer:
return RTCSignalingStateHaveRemotePrAnswer;
case webrtc::PeerConnectionInterface::kClosed:
return RTCSignalingStateClosed;
}
}
+ (NSString *)stringForSignalingState:(RTCSignalingState)state {
switch (state) {
case RTCSignalingStateStable:
return @"STABLE";
case RTCSignalingStateHaveLocalOffer:
return @"HAVE_LOCAL_OFFER";
case RTCSignalingStateHaveLocalPrAnswer:
return @"HAVE_LOCAL_PRANSWER";
case RTCSignalingStateHaveRemoteOffer:
return @"HAVE_REMOTE_OFFER";
case RTCSignalingStateHaveRemotePrAnswer:
return @"HAVE_REMOTE_PRANSWER";
case RTCSignalingStateClosed:
return @"CLOSED";
}
}
+ (webrtc::PeerConnectionInterface::IceConnectionState)
nativeIceConnectionStateForState:(RTCIceConnectionState)state {
switch (state) {
case RTCIceConnectionStateNew:
return webrtc::PeerConnectionInterface::kIceConnectionNew;
case RTCIceConnectionStateChecking:
return webrtc::PeerConnectionInterface::kIceConnectionChecking;
case RTCIceConnectionStateConnected:
return webrtc::PeerConnectionInterface::kIceConnectionConnected;
case RTCIceConnectionStateCompleted:
return webrtc::PeerConnectionInterface::kIceConnectionCompleted;
case RTCIceConnectionStateFailed:
return webrtc::PeerConnectionInterface::kIceConnectionFailed;
case RTCIceConnectionStateDisconnected:
return webrtc::PeerConnectionInterface::kIceConnectionDisconnected;
case RTCIceConnectionStateClosed:
return webrtc::PeerConnectionInterface::kIceConnectionClosed;
case RTCIceConnectionStateCount:
return webrtc::PeerConnectionInterface::kIceConnectionMax;
}
}
+ (RTCIceConnectionState)iceConnectionStateForNativeState:
(webrtc::PeerConnectionInterface::IceConnectionState)nativeState {
switch (nativeState) {
case webrtc::PeerConnectionInterface::kIceConnectionNew:
return RTCIceConnectionStateNew;
case webrtc::PeerConnectionInterface::kIceConnectionChecking:
return RTCIceConnectionStateChecking;
case webrtc::PeerConnectionInterface::kIceConnectionConnected:
return RTCIceConnectionStateConnected;
case webrtc::PeerConnectionInterface::kIceConnectionCompleted:
return RTCIceConnectionStateCompleted;
case webrtc::PeerConnectionInterface::kIceConnectionFailed:
return RTCIceConnectionStateFailed;
case webrtc::PeerConnectionInterface::kIceConnectionDisconnected:
return RTCIceConnectionStateDisconnected;
case webrtc::PeerConnectionInterface::kIceConnectionClosed:
return RTCIceConnectionStateClosed;
case webrtc::PeerConnectionInterface::kIceConnectionMax:
return RTCIceConnectionStateCount;
}
}
+ (NSString *)stringForIceConnectionState:(RTCIceConnectionState)state {
switch (state) {
case RTCIceConnectionStateNew:
return @"NEW";
case RTCIceConnectionStateChecking:
return @"CHECKING";
case RTCIceConnectionStateConnected:
return @"CONNECTED";
case RTCIceConnectionStateCompleted:
return @"COMPLETED";
case RTCIceConnectionStateFailed:
return @"FAILED";
case RTCIceConnectionStateDisconnected:
return @"DISCONNECTED";
case RTCIceConnectionStateClosed:
return @"CLOSED";
case RTCIceConnectionStateCount:
return @"COUNT";
}
}
+ (webrtc::PeerConnectionInterface::IceGatheringState)
nativeIceGatheringStateForState:(RTCIceGatheringState)state {
switch (state) {
case RTCIceGatheringStateNew:
return webrtc::PeerConnectionInterface::kIceGatheringNew;
case RTCIceGatheringStateGathering:
return webrtc::PeerConnectionInterface::kIceGatheringGathering;
case RTCIceGatheringStateComplete:
return webrtc::PeerConnectionInterface::kIceGatheringComplete;
}
}
+ (RTCIceGatheringState)iceGatheringStateForNativeState:
(webrtc::PeerConnectionInterface::IceGatheringState)nativeState {
switch (nativeState) {
case webrtc::PeerConnectionInterface::kIceGatheringNew:
return RTCIceGatheringStateNew;
case webrtc::PeerConnectionInterface::kIceGatheringGathering:
return RTCIceGatheringStateGathering;
case webrtc::PeerConnectionInterface::kIceGatheringComplete:
return RTCIceGatheringStateComplete;
}
}
+ (NSString *)stringForIceGatheringState:(RTCIceGatheringState)state {
switch (state) {
case RTCIceGatheringStateNew:
return @"NEW";
case RTCIceGatheringStateGathering:
return @"GATHERING";
case RTCIceGatheringStateComplete:
return @"COMPLETE";
}
}
+ (webrtc::PeerConnectionInterface::StatsOutputLevel)
nativeStatsOutputLevelForLevel:(RTCStatsOutputLevel)level {
switch (level) {
case RTCStatsOutputLevelStandard:
return webrtc::PeerConnectionInterface::kStatsOutputLevelStandard;
case RTCStatsOutputLevelDebug:
return webrtc::PeerConnectionInterface::kStatsOutputLevelDebug;
}
}
- (rtc::scoped_refptr<webrtc::PeerConnectionInterface>)nativePeerConnection {
return _peerConnection;
}
@end

View File

@ -8,47 +8,4 @@
* be found in the AUTHORS file in the root of the source tree.
*/
#import "WebRTC/RTCPeerConnectionFactory.h"
#include "rtc_base/scoped_ref_ptr.h"
namespace webrtc {
class AudioDeviceModule;
class AudioEncoderFactory;
class AudioDecoderFactory;
class VideoEncoderFactory;
class VideoDecoderFactory;
class AudioProcessing;
} // namespace webrtc
NS_ASSUME_NONNULL_BEGIN
/**
* This class extension exposes methods that work directly with injectable C++ components.
*/
@interface RTCPeerConnectionFactory ()
- (instancetype)initNative NS_DESIGNATED_INITIALIZER;
/* Initializer used when WebRTC is compiled with no media support */
- (instancetype)initWithNoMedia;
/* Initialize object with injectable native audio/video encoder/decoder factories */
- (instancetype)initWithNativeAudioEncoderFactory:
(rtc::scoped_refptr<webrtc::AudioEncoderFactory>)audioEncoderFactory
nativeAudioDecoderFactory:
(rtc::scoped_refptr<webrtc::AudioDecoderFactory>)audioDecoderFactory
nativeVideoEncoderFactory:
(std::unique_ptr<webrtc::VideoEncoderFactory>)videoEncoderFactory
nativeVideoDecoderFactory:
(std::unique_ptr<webrtc::VideoDecoderFactory>)videoDecoderFactory
audioDeviceModule:
(nullable webrtc::AudioDeviceModule *)audioDeviceModule
audioProcessingModule:
(rtc::scoped_refptr<webrtc::AudioProcessing>)audioProcessingModule;
@end
NS_ASSUME_NONNULL_END
#import "api/peerconnection/RTCPeerConnectionFactory+Native.h"

View File

@ -1,31 +0,0 @@
/*
* 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/RTCPeerConnectionFactory.h"
#include "api/peerconnectioninterface.h"
#include "rtc_base/scoped_ref_ptr.h"
NS_ASSUME_NONNULL_BEGIN
@interface RTCPeerConnectionFactory ()
/**
* PeerConnectionFactoryInterface created and held by this
* RTCPeerConnectionFactory object. This is needed to pass to the underlying
* C++ APIs.
*/
@property(nonatomic, readonly)
rtc::scoped_refptr<webrtc::PeerConnectionFactoryInterface>
nativeFactory;
@end
NS_ASSUME_NONNULL_END

View File

@ -1,256 +0,0 @@
/*
* 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 "RTCPeerConnectionFactory+Native.h"
#import "RTCPeerConnectionFactory+Private.h"
#import "RTCPeerConnectionFactoryOptions+Private.h"
#import "NSString+StdString.h"
#import "RTCAudioSource+Private.h"
#import "RTCAudioTrack+Private.h"
#import "RTCMediaConstraints+Private.h"
#import "RTCMediaStream+Private.h"
#import "RTCPeerConnection+Private.h"
#import "RTCVideoSource+Private.h"
#import "RTCVideoTrack+Private.h"
#import "WebRTC/RTCLogging.h"
#import "WebRTC/RTCVideoCodecFactory.h"
#ifndef HAVE_NO_MEDIA
#import "WebRTC/RTCVideoCodecH264.h"
// The no-media version PeerConnectionFactory doesn't depend on these files, but the gn check tool
// is not smart enough to take the #ifdef into account.
#include "api/audio_codecs/builtin_audio_decoder_factory.h" // nogncheck
#include "api/audio_codecs/builtin_audio_encoder_factory.h" // nogncheck
#include "media/engine/convert_legacy_video_factory.h" // nogncheck
#include "modules/audio_device/include/audio_device.h" // nogncheck
#include "modules/audio_processing/include/audio_processing.h" // nogncheck
#include "sdk/objc/Framework/Native/api/video_decoder_factory.h"
#include "sdk/objc/Framework/Native/api/video_encoder_factory.h"
#include "sdk/objc/Framework/Native/src/objc_video_decoder_factory.h"
#include "sdk/objc/Framework/Native/src/objc_video_encoder_factory.h"
#endif
#if defined(WEBRTC_IOS)
#import "sdk/objc/Framework/Native/api/audio_device_module.h"
#endif
// Adding the nogncheck to disable the including header check.
// The no-media version PeerConnectionFactory doesn't depend on media related
// C++ target.
// TODO(zhihuang): Remove nogncheck once MediaEngineInterface is moved to C++
// API layer.
#include "absl/memory/memory.h"
#include "media/engine/webrtcmediaengine.h" // nogncheck
@implementation RTCPeerConnectionFactory {
std::unique_ptr<rtc::Thread> _networkThread;
std::unique_ptr<rtc::Thread> _workerThread;
std::unique_ptr<rtc::Thread> _signalingThread;
BOOL _hasStartedAecDump;
}
@synthesize nativeFactory = _nativeFactory;
- (rtc::scoped_refptr<webrtc::AudioDeviceModule>)audioDeviceModule {
#if defined(WEBRTC_IOS)
return webrtc::CreateAudioDeviceModule();
#else
return nullptr;
#endif
}
- (instancetype)init {
#ifdef HAVE_NO_MEDIA
return [self initWithNoMedia];
#else
return [self initWithNativeAudioEncoderFactory:webrtc::CreateBuiltinAudioEncoderFactory()
nativeAudioDecoderFactory:webrtc::CreateBuiltinAudioDecoderFactory()
nativeVideoEncoderFactory:webrtc::ObjCToNativeVideoEncoderFactory(
[[RTCVideoEncoderFactoryH264 alloc] init])
nativeVideoDecoderFactory:webrtc::ObjCToNativeVideoDecoderFactory(
[[RTCVideoDecoderFactoryH264 alloc] init])
audioDeviceModule:[self audioDeviceModule]
audioProcessingModule:nullptr];
#endif
}
- (instancetype)initWithEncoderFactory:(nullable id<RTCVideoEncoderFactory>)encoderFactory
decoderFactory:(nullable id<RTCVideoDecoderFactory>)decoderFactory {
#ifdef HAVE_NO_MEDIA
return [self initWithNoMedia];
#else
std::unique_ptr<webrtc::VideoEncoderFactory> native_encoder_factory;
std::unique_ptr<webrtc::VideoDecoderFactory> native_decoder_factory;
if (encoderFactory) {
native_encoder_factory = webrtc::ObjCToNativeVideoEncoderFactory(encoderFactory);
}
if (decoderFactory) {
native_decoder_factory = webrtc::ObjCToNativeVideoDecoderFactory(decoderFactory);
}
return [self initWithNativeAudioEncoderFactory:webrtc::CreateBuiltinAudioEncoderFactory()
nativeAudioDecoderFactory:webrtc::CreateBuiltinAudioDecoderFactory()
nativeVideoEncoderFactory:std::move(native_encoder_factory)
nativeVideoDecoderFactory:std::move(native_decoder_factory)
audioDeviceModule:[self audioDeviceModule]
audioProcessingModule:nullptr];
#endif
}
- (instancetype)initNative {
if (self = [super init]) {
_networkThread = rtc::Thread::CreateWithSocketServer();
_networkThread->SetName("network_thread", _networkThread.get());
BOOL result = _networkThread->Start();
NSAssert(result, @"Failed to start network thread.");
_workerThread = rtc::Thread::Create();
_workerThread->SetName("worker_thread", _workerThread.get());
result = _workerThread->Start();
NSAssert(result, @"Failed to start worker thread.");
_signalingThread = rtc::Thread::Create();
_signalingThread->SetName("signaling_thread", _signalingThread.get());
result = _signalingThread->Start();
NSAssert(result, @"Failed to start signaling thread.");
}
return self;
}
- (instancetype)initWithNoMedia {
if (self = [self initNative]) {
_nativeFactory = webrtc::CreateModularPeerConnectionFactory(
_networkThread.get(),
_workerThread.get(),
_signalingThread.get(),
std::unique_ptr<cricket::MediaEngineInterface>(),
std::unique_ptr<webrtc::CallFactoryInterface>(),
std::unique_ptr<webrtc::RtcEventLogFactoryInterface>());
NSAssert(_nativeFactory, @"Failed to initialize PeerConnectionFactory!");
}
return self;
}
- (instancetype)initWithNativeAudioEncoderFactory:
(rtc::scoped_refptr<webrtc::AudioEncoderFactory>)audioEncoderFactory
nativeAudioDecoderFactory:
(rtc::scoped_refptr<webrtc::AudioDecoderFactory>)audioDecoderFactory
nativeVideoEncoderFactory:
(std::unique_ptr<webrtc::VideoEncoderFactory>)videoEncoderFactory
nativeVideoDecoderFactory:
(std::unique_ptr<webrtc::VideoDecoderFactory>)videoDecoderFactory
audioDeviceModule:
(nullable webrtc::AudioDeviceModule *)audioDeviceModule
audioProcessingModule:
(rtc::scoped_refptr<webrtc::AudioProcessing>)audioProcessingModule {
#ifdef HAVE_NO_MEDIA
return [self initWithNoMedia];
#else
if (self = [self initNative]) {
_nativeFactory = webrtc::CreatePeerConnectionFactory(_networkThread.get(),
_workerThread.get(),
_signalingThread.get(),
audioDeviceModule,
audioEncoderFactory,
audioDecoderFactory,
std::move(videoEncoderFactory),
std::move(videoDecoderFactory),
nullptr, // audio mixer
audioProcessingModule);
NSAssert(_nativeFactory, @"Failed to initialize PeerConnectionFactory!");
}
return self;
#endif
}
- (RTCAudioSource *)audioSourceWithConstraints:(nullable RTCMediaConstraints *)constraints {
std::unique_ptr<webrtc::MediaConstraints> nativeConstraints;
if (constraints) {
nativeConstraints = constraints.nativeConstraints;
}
cricket::AudioOptions options;
CopyConstraintsIntoAudioOptions(nativeConstraints.get(), &options);
rtc::scoped_refptr<webrtc::AudioSourceInterface> source =
_nativeFactory->CreateAudioSource(options);
return [[RTCAudioSource alloc] initWithFactory:self nativeAudioSource:source];
}
- (RTCAudioTrack *)audioTrackWithTrackId:(NSString *)trackId {
RTCAudioSource *audioSource = [self audioSourceWithConstraints:nil];
return [self audioTrackWithSource:audioSource trackId:trackId];
}
- (RTCAudioTrack *)audioTrackWithSource:(RTCAudioSource *)source
trackId:(NSString *)trackId {
return [[RTCAudioTrack alloc] initWithFactory:self
source:source
trackId:trackId];
}
- (RTCVideoSource *)videoSource {
return [[RTCVideoSource alloc] initWithFactory:self
signalingThread:_signalingThread.get()
workerThread:_workerThread.get()];
}
- (RTCVideoTrack *)videoTrackWithSource:(RTCVideoSource *)source
trackId:(NSString *)trackId {
return [[RTCVideoTrack alloc] initWithFactory:self
source:source
trackId:trackId];
}
- (RTCMediaStream *)mediaStreamWithStreamId:(NSString *)streamId {
return [[RTCMediaStream alloc] initWithFactory:self
streamId:streamId];
}
- (RTCPeerConnection *)peerConnectionWithConfiguration:
(RTCConfiguration *)configuration
constraints:
(RTCMediaConstraints *)constraints
delegate:
(nullable id<RTCPeerConnectionDelegate>)delegate {
return [[RTCPeerConnection alloc] initWithFactory:self
configuration:configuration
constraints:constraints
delegate:delegate];
}
- (void)setOptions:(nonnull RTCPeerConnectionFactoryOptions *)options {
RTC_DCHECK(options != nil);
_nativeFactory->SetOptions(options.nativeOptions);
}
- (BOOL)startAecDumpWithFilePath:(NSString *)filePath
maxSizeInBytes:(int64_t)maxSizeInBytes {
RTC_DCHECK(filePath.length);
RTC_DCHECK_GT(maxSizeInBytes, 0);
if (_hasStartedAecDump) {
RTCLogError(@"Aec dump already started.");
return NO;
}
int fd = open(filePath.UTF8String, O_WRONLY | O_CREAT | O_TRUNC, S_IRUSR | S_IWUSR);
if (fd < 0) {
RTCLogError(@"Error opening file: %@. Error: %d", filePath, errno);
return NO;
}
_hasStartedAecDump = _nativeFactory->StartAecDump(fd, maxSizeInBytes);
return _hasStartedAecDump;
}
- (void)stopAecDump {
_nativeFactory->StopAecDump();
_hasStartedAecDump = NO;
}
@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 "RTCPeerConnectionFactoryBuilder.h"
NS_ASSUME_NONNULL_BEGIN
@interface RTCPeerConnectionFactoryBuilder (DefaultComponents)
+ (RTCPeerConnectionFactoryBuilder *)defaultBuilder;
@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 "RTCPeerConnectionFactory+Native.h"
#import "RTCPeerConnectionFactoryBuilder+DefaultComponents.h"
#import "WebRTC/RTCVideoCodecH264.h"
#include "api/audio_codecs/builtin_audio_decoder_factory.h"
#include "api/audio_codecs/builtin_audio_encoder_factory.h"
#include "sdk/objc/Framework/Native/api/video_decoder_factory.h"
#include "sdk/objc/Framework/Native/api/video_encoder_factory.h"
#if defined(WEBRTC_IOS)
#import "sdk/objc/Framework/Native/api/audio_device_module.h"
#endif
@implementation RTCPeerConnectionFactoryBuilder (DefaultComponents)
+ (RTCPeerConnectionFactoryBuilder *)defaultBuilder {
RTCPeerConnectionFactoryBuilder *builder = [[RTCPeerConnectionFactoryBuilder alloc] init];
auto audioEncoderFactory = webrtc::CreateBuiltinAudioEncoderFactory();
[builder setAudioEncoderFactory:audioEncoderFactory];
auto audioDecoderFactory = webrtc::CreateBuiltinAudioDecoderFactory();
[builder setAudioDecoderFactory:audioDecoderFactory];
auto videoEncoderFactory =
webrtc::ObjCToNativeVideoEncoderFactory([[RTCVideoEncoderFactoryH264 alloc] init]);
[builder setVideoEncoderFactory:std::move(videoEncoderFactory)];
auto videoDecoderFactory =
webrtc::ObjCToNativeVideoDecoderFactory([[RTCVideoDecoderFactoryH264 alloc] init]);
[builder setVideoDecoderFactory:std::move(videoDecoderFactory)];
#if defined(WEBRTC_IOS)
[builder setAudioDeviceModule:webrtc::CreateAudioDeviceModule()];
#endif
return builder;
}
@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 "WebRTC/RTCPeerConnectionFactory.h"
#include "rtc_base/scoped_ref_ptr.h"
namespace webrtc {
class AudioDeviceModule;
class AudioEncoderFactory;
class AudioDecoderFactory;
class VideoEncoderFactory;
class VideoDecoderFactory;
class AudioProcessing;
} // namespace webrtc
NS_ASSUME_NONNULL_BEGIN
@interface RTCPeerConnectionFactoryBuilder : NSObject
+ (RTCPeerConnectionFactoryBuilder *)builder;
- (RTCPeerConnectionFactory *)createPeerConnectionFactory;
- (void)setVideoEncoderFactory:(std::unique_ptr<webrtc::VideoEncoderFactory>)videoEncoderFactory;
- (void)setVideoDecoderFactory:(std::unique_ptr<webrtc::VideoDecoderFactory>)videoDecoderFactory;
- (void)setAudioEncoderFactory:(rtc::scoped_refptr<webrtc::AudioEncoderFactory>)audioEncoderFactory;
- (void)setAudioDecoderFactory:(rtc::scoped_refptr<webrtc::AudioDecoderFactory>)audioDecoderFactory;
- (void)setAudioDeviceModule:(rtc::scoped_refptr<webrtc::AudioDeviceModule>)audioDeviceModule;
- (void)setAudioProcessingModule:(rtc::scoped_refptr<webrtc::AudioProcessing>)audioProcessingModule;
@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 "RTCPeerConnectionFactoryBuilder.h"
#import "RTCPeerConnectionFactory+Native.h"
#include "api/audio_codecs/audio_decoder_factory.h"
#include "api/audio_codecs/audio_encoder_factory.h"
#include "api/video_codecs/video_decoder_factory.h"
#include "api/video_codecs/video_encoder_factory.h"
#include "modules/audio_device/include/audio_device.h"
#include "modules/audio_processing/include/audio_processing.h"
@implementation RTCPeerConnectionFactoryBuilder {
std::unique_ptr<webrtc::VideoEncoderFactory> _videoEncoderFactory;
std::unique_ptr<webrtc::VideoDecoderFactory> _videoDecoderFactory;
rtc::scoped_refptr<webrtc::AudioEncoderFactory> _audioEncoderFactory;
rtc::scoped_refptr<webrtc::AudioDecoderFactory> _audioDecoderFactory;
rtc::scoped_refptr<webrtc::AudioDeviceModule> _audioDeviceModule;
rtc::scoped_refptr<webrtc::AudioProcessing> _audioProcessingModule;
}
+ (RTCPeerConnectionFactoryBuilder *)builder {
return [[RTCPeerConnectionFactoryBuilder alloc] init];
}
- (RTCPeerConnectionFactory *)createPeerConnectionFactory {
RTCPeerConnectionFactory *factory = [RTCPeerConnectionFactory alloc];
return [factory initWithNativeAudioEncoderFactory:_audioEncoderFactory
nativeAudioDecoderFactory:_audioDecoderFactory
nativeVideoEncoderFactory:std::move(_videoEncoderFactory)
nativeVideoDecoderFactory:std::move(_videoDecoderFactory)
audioDeviceModule:_audioDeviceModule
audioProcessingModule:_audioProcessingModule];
}
- (void)setVideoEncoderFactory:(std::unique_ptr<webrtc::VideoEncoderFactory>)videoEncoderFactory {
_videoEncoderFactory = std::move(videoEncoderFactory);
}
- (void)setVideoDecoderFactory:(std::unique_ptr<webrtc::VideoDecoderFactory>)videoDecoderFactory {
_videoDecoderFactory = std::move(videoDecoderFactory);
}
- (void)setAudioEncoderFactory:
(rtc::scoped_refptr<webrtc::AudioEncoderFactory>)audioEncoderFactory {
_audioEncoderFactory = audioEncoderFactory;
}
- (void)setAudioDecoderFactory:
(rtc::scoped_refptr<webrtc::AudioDecoderFactory>)audioDecoderFactory {
_audioDecoderFactory = audioDecoderFactory;
}
- (void)setAudioDeviceModule:(rtc::scoped_refptr<webrtc::AudioDeviceModule>)audioDeviceModule {
_audioDeviceModule = audioDeviceModule;
}
- (void)setAudioProcessingModule:
(rtc::scoped_refptr<webrtc::AudioProcessing>)audioProcessingModule {
_audioProcessingModule = audioProcessingModule;
}
@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 "WebRTC/RTCPeerConnectionFactoryOptions.h"
#include "api/peerconnectioninterface.h"
NS_ASSUME_NONNULL_BEGIN
@interface RTCPeerConnectionFactoryOptions ()
/** Returns the equivalent native PeerConnectionFactoryInterface::Options
* structure. */
@property(nonatomic, readonly)
webrtc::PeerConnectionFactoryInterface::Options nativeOptions;
@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 "RTCPeerConnectionFactoryOptions+Private.h"
#include "rtc_base/network_constants.h"
namespace {
void setNetworkBit(webrtc::PeerConnectionFactoryInterface::Options* options,
rtc::AdapterType type,
bool ignore) {
if (ignore) {
options->network_ignore_mask |= type;
} else {
options->network_ignore_mask &= ~type;
}
}
} // namespace
@implementation RTCPeerConnectionFactoryOptions
@synthesize disableEncryption = _disableEncryption;
@synthesize disableNetworkMonitor = _disableNetworkMonitor;
@synthesize ignoreLoopbackNetworkAdapter = _ignoreLoopbackNetworkAdapter;
@synthesize ignoreVPNNetworkAdapter = _ignoreVPNNetworkAdapter;
@synthesize ignoreCellularNetworkAdapter = _ignoreCellularNetworkAdapter;
@synthesize ignoreWiFiNetworkAdapter = _ignoreWiFiNetworkAdapter;
@synthesize ignoreEthernetNetworkAdapter = _ignoreEthernetNetworkAdapter;
@synthesize enableAes128Sha1_32CryptoCipher = _enableAes128Sha1_32CryptoCipher;
@synthesize enableGcmCryptoSuites = _enableGcmCryptoSuites;
- (instancetype)init {
return [super init];
}
- (webrtc::PeerConnectionFactoryInterface::Options)nativeOptions {
webrtc::PeerConnectionFactoryInterface::Options options;
options.disable_encryption = self.disableEncryption;
options.disable_network_monitor = self.disableNetworkMonitor;
setNetworkBit(&options, rtc::ADAPTER_TYPE_LOOPBACK, self.ignoreLoopbackNetworkAdapter);
setNetworkBit(&options, rtc::ADAPTER_TYPE_VPN, self.ignoreVPNNetworkAdapter);
setNetworkBit(&options, rtc::ADAPTER_TYPE_CELLULAR, self.ignoreCellularNetworkAdapter);
setNetworkBit(&options, rtc::ADAPTER_TYPE_WIFI, self.ignoreWiFiNetworkAdapter);
setNetworkBit(&options, rtc::ADAPTER_TYPE_ETHERNET, self.ignoreEthernetNetworkAdapter);
options.crypto_options.enable_aes128_sha1_32_crypto_cipher = self.enableAes128Sha1_32CryptoCipher;
options.crypto_options.enable_gcm_crypto_suites = self.enableGcmCryptoSuites;
return options;
}
@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 "WebRTC/RTCRtcpParameters.h"
#include "api/rtpparameters.h"
NS_ASSUME_NONNULL_BEGIN
@interface RTCRtcpParameters ()
/** Returns the equivalent native RtcpParameters structure. */
@property(nonatomic, readonly) webrtc::RtcpParameters nativeParameters;
/** Initialize the object with a native RtcpParameters structure. */
- (instancetype)initWithNativeParameters:(const webrtc::RtcpParameters &)nativeParameters;
@end
NS_ASSUME_NONNULL_END

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@ -1,39 +0,0 @@
/*
* 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 "RTCRtcpParameters+Private.h"
#import "NSString+StdString.h"
@implementation RTCRtcpParameters
@synthesize cname = _cname;
@synthesize isReducedSize = _isReducedSize;
- (instancetype)init {
return [super init];
}
- (instancetype)initWithNativeParameters:(const webrtc::RtcpParameters &)nativeParameters {
if (self = [self init]) {
_cname = [NSString stringForStdString:nativeParameters.cname];
_isReducedSize = nativeParameters.reduced_size;
}
return self;
}
- (webrtc::RtcpParameters)nativeParameters {
webrtc::RtcpParameters parameters;
parameters.cname = [NSString stdStringForString:_cname];
parameters.reduced_size = _isReducedSize;
return parameters;
}
@end

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@ -1,27 +0,0 @@
/*
* 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 "WebRTC/RTCRtpCodecParameters.h"
#include "api/rtpparameters.h"
NS_ASSUME_NONNULL_BEGIN
@interface RTCRtpCodecParameters ()
/** Returns the equivalent native RtpCodecParameters structure. */
@property(nonatomic, readonly) webrtc::RtpCodecParameters nativeParameters;
/** Initialize the object with a native RtpCodecParameters structure. */
- (instancetype)initWithNativeParameters:(const webrtc::RtpCodecParameters &)nativeParameters;
@end
NS_ASSUME_NONNULL_END

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@ -1,109 +0,0 @@
/*
* 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 "RTCRtpCodecParameters+Private.h"
#import "NSString+StdString.h"
#import "WebRTC/RTCMediaStreamTrack.h" // For "kind" strings.
#include "media/base/mediaconstants.h"
#include "rtc_base/checks.h"
const NSString * const kRTCRtxCodecName = @(cricket::kRtxCodecName);
const NSString * const kRTCRedCodecName = @(cricket::kRedCodecName);
const NSString * const kRTCUlpfecCodecName = @(cricket::kUlpfecCodecName);
const NSString * const kRTCFlexfecCodecName = @(cricket::kFlexfecCodecName);
const NSString * const kRTCOpusCodecName = @(cricket::kOpusCodecName);
const NSString * const kRTCIsacCodecName = @(cricket::kIsacCodecName);
const NSString * const kRTCL16CodecName = @(cricket::kL16CodecName);
const NSString * const kRTCG722CodecName = @(cricket::kG722CodecName);
const NSString * const kRTCIlbcCodecName = @(cricket::kIlbcCodecName);
const NSString * const kRTCPcmuCodecName = @(cricket::kPcmuCodecName);
const NSString * const kRTCPcmaCodecName = @(cricket::kPcmaCodecName);
const NSString * const kRTCDtmfCodecName = @(cricket::kDtmfCodecName);
const NSString * const kRTCComfortNoiseCodecName =
@(cricket::kComfortNoiseCodecName);
const NSString * const kRTCVp8CodecName = @(cricket::kVp8CodecName);
const NSString * const kRTCVp9CodecName = @(cricket::kVp9CodecName);
const NSString * const kRTCH264CodecName = @(cricket::kH264CodecName);
@implementation RTCRtpCodecParameters
@synthesize payloadType = _payloadType;
@synthesize name = _name;
@synthesize kind = _kind;
@synthesize clockRate = _clockRate;
@synthesize numChannels = _numChannels;
@synthesize parameters = _parameters;
- (instancetype)init {
return [super init];
}
- (instancetype)initWithNativeParameters:
(const webrtc::RtpCodecParameters &)nativeParameters {
if (self = [self init]) {
_payloadType = nativeParameters.payload_type;
_name = [NSString stringForStdString:nativeParameters.name];
switch (nativeParameters.kind) {
case cricket::MEDIA_TYPE_AUDIO:
_kind = kRTCMediaStreamTrackKindAudio;
break;
case cricket::MEDIA_TYPE_VIDEO:
_kind = kRTCMediaStreamTrackKindVideo;
break;
case cricket::MEDIA_TYPE_DATA:
RTC_NOTREACHED();
break;
}
if (nativeParameters.clock_rate) {
_clockRate = [NSNumber numberWithInt:*nativeParameters.clock_rate];
}
if (nativeParameters.num_channels) {
_numChannels = [NSNumber numberWithInt:*nativeParameters.num_channels];
}
NSMutableDictionary *parameters = [NSMutableDictionary dictionary];
for (const auto &parameter : nativeParameters.parameters) {
[parameters setObject:[NSString stringForStdString:parameter.second]
forKey:[NSString stringForStdString:parameter.first]];
}
_parameters = parameters;
}
return self;
}
- (webrtc::RtpCodecParameters)nativeParameters {
webrtc::RtpCodecParameters parameters;
parameters.payload_type = _payloadType;
parameters.name = [NSString stdStringForString:_name];
// NSString pointer comparison is safe here since "kind" is readonly and only
// populated above.
if (_kind == kRTCMediaStreamTrackKindAudio) {
parameters.kind = cricket::MEDIA_TYPE_AUDIO;
} else if (_kind == kRTCMediaStreamTrackKindVideo) {
parameters.kind = cricket::MEDIA_TYPE_VIDEO;
} else {
RTC_NOTREACHED();
}
if (_clockRate != nil) {
parameters.clock_rate = absl::optional<int>(_clockRate.intValue);
}
if (_numChannels != nil) {
parameters.num_channels = absl::optional<int>(_numChannels.intValue);
}
for (NSString *paramKey in _parameters.allKeys) {
std::string key = [NSString stdStringForString:paramKey];
std::string value = [NSString stdStringForString:_parameters[paramKey]];
parameters.parameters[key] = value;
}
return parameters;
}
@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 "WebRTC/RTCRtpEncodingParameters.h"
#include "api/rtpparameters.h"
NS_ASSUME_NONNULL_BEGIN
@interface RTCRtpEncodingParameters ()
/** Returns the equivalent native RtpEncodingParameters structure. */
@property(nonatomic, readonly) webrtc::RtpEncodingParameters nativeParameters;
/** Initialize the object with a native RtpEncodingParameters structure. */
- (instancetype)initWithNativeParameters:(const webrtc::RtpEncodingParameters &)nativeParameters;
@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 "RTCRtpEncodingParameters+Private.h"
@implementation RTCRtpEncodingParameters
@synthesize isActive = _isActive;
@synthesize maxBitrateBps = _maxBitrateBps;
@synthesize minBitrateBps = _minBitrateBps;
@synthesize ssrc = _ssrc;
- (instancetype)init {
return [super init];
}
- (instancetype)initWithNativeParameters:
(const webrtc::RtpEncodingParameters &)nativeParameters {
if (self = [self init]) {
_isActive = nativeParameters.active;
if (nativeParameters.max_bitrate_bps) {
_maxBitrateBps =
[NSNumber numberWithInt:*nativeParameters.max_bitrate_bps];
}
if (nativeParameters.min_bitrate_bps) {
_minBitrateBps =
[NSNumber numberWithInt:*nativeParameters.min_bitrate_bps];
}
if (nativeParameters.ssrc) {
_ssrc = [NSNumber numberWithUnsignedLong:*nativeParameters.ssrc];
}
}
return self;
}
- (webrtc::RtpEncodingParameters)nativeParameters {
webrtc::RtpEncodingParameters parameters;
parameters.active = _isActive;
if (_maxBitrateBps != nil) {
parameters.max_bitrate_bps = absl::optional<int>(_maxBitrateBps.intValue);
}
if (_minBitrateBps != nil) {
parameters.min_bitrate_bps = absl::optional<int>(_minBitrateBps.intValue);
}
if (_ssrc != nil) {
parameters.ssrc = absl::optional<uint32_t>(_ssrc.unsignedLongValue);
}
return parameters;
}
@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 "WebRTC/RTCVideoCodec.h"
#include "modules/include/module_common_types.h"
@implementation RTCRtpFragmentationHeader
@synthesize fragmentationOffset = _fragmentationOffset;
@synthesize fragmentationLength = _fragmentationLength;
@synthesize fragmentationTimeDiff = _fragmentationTimeDiff;
@synthesize fragmentationPlType = _fragmentationPlType;
- (instancetype)initWithNativeFragmentationHeader:
(const webrtc::RTPFragmentationHeader *)fragmentationHeader {
if (self = [super init]) {
if (fragmentationHeader) {
int count = fragmentationHeader->fragmentationVectorSize;
NSMutableArray *offsets = [NSMutableArray array];
NSMutableArray *lengths = [NSMutableArray array];
NSMutableArray *timeDiffs = [NSMutableArray array];
NSMutableArray *plTypes = [NSMutableArray array];
for (int i = 0; i < count; ++i) {
[offsets addObject:@(fragmentationHeader->fragmentationOffset[i])];
[lengths addObject:@(fragmentationHeader->fragmentationLength[i])];
[timeDiffs addObject:@(fragmentationHeader->fragmentationTimeDiff[i])];
[plTypes addObject:@(fragmentationHeader->fragmentationPlType[i])];
}
_fragmentationOffset = [offsets copy];
_fragmentationLength = [lengths copy];
_fragmentationTimeDiff = [timeDiffs copy];
_fragmentationPlType = [plTypes copy];
}
}
return self;
}
- (std::unique_ptr<webrtc::RTPFragmentationHeader>)createNativeFragmentationHeader {
auto fragmentationHeader =
std::unique_ptr<webrtc::RTPFragmentationHeader>(new webrtc::RTPFragmentationHeader);
fragmentationHeader->VerifyAndAllocateFragmentationHeader(_fragmentationOffset.count);
for (NSUInteger i = 0; i < _fragmentationOffset.count; ++i) {
fragmentationHeader->fragmentationOffset[i] = (size_t)_fragmentationOffset[i].unsignedIntValue;
fragmentationHeader->fragmentationLength[i] = (size_t)_fragmentationLength[i].unsignedIntValue;
fragmentationHeader->fragmentationTimeDiff[i] =
(uint16_t)_fragmentationOffset[i].unsignedIntValue;
fragmentationHeader->fragmentationPlType[i] = (uint8_t)_fragmentationOffset[i].unsignedIntValue;
}
return fragmentationHeader;
}
@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 "WebRTC/RTCRtpHeaderExtension.h"
#include "api/rtpparameters.h"
NS_ASSUME_NONNULL_BEGIN
@interface RTCRtpHeaderExtension ()
/** Returns the equivalent native RtpExtension structure. */
@property(nonatomic, readonly) webrtc::RtpExtension nativeParameters;
/** Initialize the object with a native RtpExtension structure. */
- (instancetype)initWithNativeParameters:(const webrtc::RtpExtension &)nativeParameters;
@end
NS_ASSUME_NONNULL_END

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@ -1,42 +0,0 @@
/*
* 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 "RTCRtpHeaderExtension+Private.h"
#import "NSString+StdString.h"
@implementation RTCRtpHeaderExtension
@synthesize uri = _uri;
@synthesize id = _id;
@synthesize encrypted = _encrypted;
- (instancetype)init {
return [super init];
}
- (instancetype)initWithNativeParameters:(const webrtc::RtpExtension &)nativeParameters {
if (self = [self init]) {
_uri = [NSString stringForStdString:nativeParameters.uri];
_id = nativeParameters.id;
_encrypted = nativeParameters.encrypt;
}
return self;
}
- (webrtc::RtpExtension)nativeParameters {
webrtc::RtpExtension extension;
extension.uri = [NSString stdStringForString:_uri];
extension.id = _id;
extension.encrypt = _encrypted;
return extension;
}
@end

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@ -1,27 +0,0 @@
/*
* 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 "WebRTC/RTCRtpParameters.h"
#include "api/rtpparameters.h"
NS_ASSUME_NONNULL_BEGIN
@interface RTCRtpParameters ()
/** Returns the equivalent native RtpParameters structure. */
@property(nonatomic, readonly) webrtc::RtpParameters nativeParameters;
/** Initialize the object with a native RtpParameters structure. */
- (instancetype)initWithNativeParameters:(const webrtc::RtpParameters &)nativeParameters;
@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 "RTCRtpParameters+Private.h"
#import "NSString+StdString.h"
#import "RTCRtcpParameters+Private.h"
#import "RTCRtpCodecParameters+Private.h"
#import "RTCRtpEncodingParameters+Private.h"
#import "RTCRtpHeaderExtension+Private.h"
@implementation RTCRtpParameters
@synthesize transactionId = _transactionId;
@synthesize rtcp = _rtcp;
@synthesize headerExtensions = _headerExtensions;
@synthesize encodings = _encodings;
@synthesize codecs = _codecs;
- (instancetype)init {
return [super init];
}
- (instancetype)initWithNativeParameters:
(const webrtc::RtpParameters &)nativeParameters {
if (self = [self init]) {
_transactionId = [NSString stringForStdString:nativeParameters.transaction_id];
_rtcp = [[RTCRtcpParameters alloc] initWithNativeParameters:nativeParameters.rtcp];
NSMutableArray *headerExtensions = [[NSMutableArray alloc] init];
for (const auto &headerExtension : nativeParameters.header_extensions) {
[headerExtensions
addObject:[[RTCRtpHeaderExtension alloc] initWithNativeParameters:headerExtension]];
}
_headerExtensions = headerExtensions;
NSMutableArray *encodings = [[NSMutableArray alloc] init];
for (const auto &encoding : nativeParameters.encodings) {
[encodings addObject:[[RTCRtpEncodingParameters alloc]
initWithNativeParameters:encoding]];
}
_encodings = encodings;
NSMutableArray *codecs = [[NSMutableArray alloc] init];
for (const auto &codec : nativeParameters.codecs) {
[codecs addObject:[[RTCRtpCodecParameters alloc]
initWithNativeParameters:codec]];
}
_codecs = codecs;
}
return self;
}
- (webrtc::RtpParameters)nativeParameters {
webrtc::RtpParameters parameters;
parameters.transaction_id = [NSString stdStringForString:_transactionId];
parameters.rtcp = [_rtcp nativeParameters];
for (RTCRtpHeaderExtension *headerExtension in _headerExtensions) {
parameters.header_extensions.push_back(headerExtension.nativeParameters);
}
for (RTCRtpEncodingParameters *encoding in _encodings) {
parameters.encodings.push_back(encoding.nativeParameters);
}
for (RTCRtpCodecParameters *codec in _codecs) {
parameters.codecs.push_back(codec.nativeParameters);
}
return parameters;
}
@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 "WebRTC/RTCRtpReceiver.h"
#include "api/rtpreceiverinterface.h"
NS_ASSUME_NONNULL_BEGIN
@class RTCPeerConnectionFactory;
namespace webrtc {
class RtpReceiverDelegateAdapter : public RtpReceiverObserverInterface {
public:
RtpReceiverDelegateAdapter(RTCRtpReceiver* receiver);
void OnFirstPacketReceived(cricket::MediaType media_type) override;
private:
__weak RTCRtpReceiver* receiver_;
};
} // namespace webrtc
@interface RTCRtpReceiver ()
@property(nonatomic, readonly) rtc::scoped_refptr<webrtc::RtpReceiverInterface> nativeRtpReceiver;
/** Initialize an RTCRtpReceiver with a native RtpReceiverInterface. */
- (instancetype)initWithFactory:(RTCPeerConnectionFactory*)factory
nativeRtpReceiver:(rtc::scoped_refptr<webrtc::RtpReceiverInterface>)nativeRtpReceiver
NS_DESIGNATED_INITIALIZER;
+ (RTCRtpMediaType)mediaTypeForNativeMediaType:(cricket::MediaType)nativeMediaType;
+ (cricket::MediaType)nativeMediaTypeForMediaType:(RTCRtpMediaType)mediaType;
+ (NSString*)stringForMediaType:(RTCRtpMediaType)mediaType;
@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 "RTCRtpReceiver+Private.h"
#import "NSString+StdString.h"
#import "RTCMediaStreamTrack+Private.h"
#import "RTCRtpParameters+Private.h"
#import "WebRTC/RTCLogging.h"
#include "api/mediastreaminterface.h"
namespace webrtc {
RtpReceiverDelegateAdapter::RtpReceiverDelegateAdapter(
RTCRtpReceiver *receiver) {
RTC_CHECK(receiver);
receiver_ = receiver;
}
void RtpReceiverDelegateAdapter::OnFirstPacketReceived(
cricket::MediaType media_type) {
RTCRtpMediaType packet_media_type =
[RTCRtpReceiver mediaTypeForNativeMediaType:media_type];
RTCRtpReceiver *receiver = receiver_;
[receiver.delegate rtpReceiver:receiver didReceiveFirstPacketForMediaType:packet_media_type];
}
} // namespace webrtc
@implementation RTCRtpReceiver {
RTCPeerConnectionFactory *_factory;
rtc::scoped_refptr<webrtc::RtpReceiverInterface> _nativeRtpReceiver;
std::unique_ptr<webrtc::RtpReceiverDelegateAdapter> _observer;
}
@synthesize delegate = _delegate;
- (NSString *)receiverId {
return [NSString stringForStdString:_nativeRtpReceiver->id()];
}
- (RTCRtpParameters *)parameters {
return [[RTCRtpParameters alloc]
initWithNativeParameters:_nativeRtpReceiver->GetParameters()];
}
- (void)setParameters:(RTCRtpParameters *)parameters {
if (!_nativeRtpReceiver->SetParameters(parameters.nativeParameters)) {
RTCLogError(@"RTCRtpReceiver(%p): Failed to set parameters: %@", self,
parameters);
}
}
- (nullable RTCMediaStreamTrack *)track {
rtc::scoped_refptr<webrtc::MediaStreamTrackInterface> nativeTrack(
_nativeRtpReceiver->track());
if (nativeTrack) {
return [RTCMediaStreamTrack mediaTrackForNativeTrack:nativeTrack factory:_factory];
}
return nil;
}
- (NSString *)description {
return [NSString stringWithFormat:@"RTCRtpReceiver {\n receiverId: %@\n}",
self.receiverId];
}
- (void)dealloc {
if (_nativeRtpReceiver) {
_nativeRtpReceiver->SetObserver(nullptr);
}
}
- (BOOL)isEqual:(id)object {
if (self == object) {
return YES;
}
if (object == nil) {
return NO;
}
if (![object isMemberOfClass:[self class]]) {
return NO;
}
RTCRtpReceiver *receiver = (RTCRtpReceiver *)object;
return _nativeRtpReceiver == receiver.nativeRtpReceiver;
}
- (NSUInteger)hash {
return (NSUInteger)_nativeRtpReceiver.get();
}
#pragma mark - Private
- (rtc::scoped_refptr<webrtc::RtpReceiverInterface>)nativeRtpReceiver {
return _nativeRtpReceiver;
}
- (instancetype)initWithFactory:(RTCPeerConnectionFactory *)factory
nativeRtpReceiver:
(rtc::scoped_refptr<webrtc::RtpReceiverInterface>)nativeRtpReceiver {
if (self = [super init]) {
_factory = factory;
_nativeRtpReceiver = nativeRtpReceiver;
RTCLogInfo(
@"RTCRtpReceiver(%p): created receiver: %@", self, self.description);
_observer.reset(new webrtc::RtpReceiverDelegateAdapter(self));
_nativeRtpReceiver->SetObserver(_observer.get());
}
return self;
}
+ (RTCRtpMediaType)mediaTypeForNativeMediaType:
(cricket::MediaType)nativeMediaType {
switch (nativeMediaType) {
case cricket::MEDIA_TYPE_AUDIO:
return RTCRtpMediaTypeAudio;
case cricket::MEDIA_TYPE_VIDEO:
return RTCRtpMediaTypeVideo;
case cricket::MEDIA_TYPE_DATA:
return RTCRtpMediaTypeData;
}
}
+ (cricket::MediaType)nativeMediaTypeForMediaType:(RTCRtpMediaType)mediaType {
switch (mediaType) {
case RTCRtpMediaTypeAudio:
return cricket::MEDIA_TYPE_AUDIO;
case RTCRtpMediaTypeVideo:
return cricket::MEDIA_TYPE_VIDEO;
case RTCRtpMediaTypeData:
return cricket::MEDIA_TYPE_DATA;
}
}
+ (NSString *)stringForMediaType:(RTCRtpMediaType)mediaType {
switch (mediaType) {
case RTCRtpMediaTypeAudio:
return @"AUDIO";
case RTCRtpMediaTypeVideo:
return @"VIDEO";
case RTCRtpMediaTypeData:
return @"DATA";
}
}
@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 "WebRTC/RTCRtpSender.h"
#include "api/rtpsenderinterface.h"
NS_ASSUME_NONNULL_BEGIN
@class RTCPeerConnectionFactory;
@interface RTCRtpSender ()
@property(nonatomic, readonly) rtc::scoped_refptr<webrtc::RtpSenderInterface> nativeRtpSender;
/** Initialize an RTCRtpSender with a native RtpSenderInterface. */
- (instancetype)initWithFactory:(RTCPeerConnectionFactory*)factory
nativeRtpSender:(rtc::scoped_refptr<webrtc::RtpSenderInterface>)nativeRtpSender
NS_DESIGNATED_INITIALIZER;
@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 "RTCRtpSender+Private.h"
#import "NSString+StdString.h"
#import "RTCDtmfSender+Private.h"
#import "RTCMediaStreamTrack+Private.h"
#import "RTCRtpParameters+Private.h"
#import "WebRTC/RTCLogging.h"
#include "api/mediastreaminterface.h"
@implementation RTCRtpSender {
RTCPeerConnectionFactory *_factory;
rtc::scoped_refptr<webrtc::RtpSenderInterface> _nativeRtpSender;
}
@synthesize dtmfSender = _dtmfSender;
- (NSString *)senderId {
return [NSString stringForStdString:_nativeRtpSender->id()];
}
- (RTCRtpParameters *)parameters {
return [[RTCRtpParameters alloc]
initWithNativeParameters:_nativeRtpSender->GetParameters()];
}
- (void)setParameters:(RTCRtpParameters *)parameters {
if (!_nativeRtpSender->SetParameters(parameters.nativeParameters).ok()) {
RTCLogError(@"RTCRtpSender(%p): Failed to set parameters: %@", self,
parameters);
}
}
- (RTCMediaStreamTrack *)track {
rtc::scoped_refptr<webrtc::MediaStreamTrackInterface> nativeTrack(
_nativeRtpSender->track());
if (nativeTrack) {
return [RTCMediaStreamTrack mediaTrackForNativeTrack:nativeTrack factory:_factory];
}
return nil;
}
- (void)setTrack:(RTCMediaStreamTrack *)track {
if (!_nativeRtpSender->SetTrack(track.nativeTrack)) {
RTCLogError(@"RTCRtpSender(%p): Failed to set track %@", self, track);
}
}
- (NSString *)description {
return [NSString stringWithFormat:@"RTCRtpSender {\n senderId: %@\n}",
self.senderId];
}
- (BOOL)isEqual:(id)object {
if (self == object) {
return YES;
}
if (object == nil) {
return NO;
}
if (![object isMemberOfClass:[self class]]) {
return NO;
}
RTCRtpSender *sender = (RTCRtpSender *)object;
return _nativeRtpSender == sender.nativeRtpSender;
}
- (NSUInteger)hash {
return (NSUInteger)_nativeRtpSender.get();
}
#pragma mark - Private
- (rtc::scoped_refptr<webrtc::RtpSenderInterface>)nativeRtpSender {
return _nativeRtpSender;
}
- (instancetype)initWithFactory:(RTCPeerConnectionFactory *)factory
nativeRtpSender:(rtc::scoped_refptr<webrtc::RtpSenderInterface>)nativeRtpSender {
NSParameterAssert(factory);
NSParameterAssert(nativeRtpSender);
if (self = [super init]) {
_factory = factory;
_nativeRtpSender = nativeRtpSender;
rtc::scoped_refptr<webrtc::DtmfSenderInterface> nativeDtmfSender(
_nativeRtpSender->GetDtmfSender());
if (nativeDtmfSender) {
_dtmfSender = [[RTCDtmfSender alloc] initWithNativeDtmfSender:nativeDtmfSender];
}
RTCLogInfo(@"RTCRtpSender(%p): created sender: %@", self, self.description);
}
return self;
}
@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 "WebRTC/RTCRtpTransceiver.h"
#include "api/rtptransceiverinterface.h"
NS_ASSUME_NONNULL_BEGIN
@class RTCPeerConnectionFactory;
@interface RTCRtpTransceiverInit ()
@property(nonatomic, readonly) webrtc::RtpTransceiverInit nativeInit;
@end
@interface RTCRtpTransceiver ()
@property(nonatomic, readonly) rtc::scoped_refptr<webrtc::RtpTransceiverInterface>
nativeRtpTransceiver;
/** Initialize an RTCRtpTransceiver with a native RtpTransceiverInterface. */
- (instancetype)initWithFactory:(RTCPeerConnectionFactory*)factory
nativeRtpTransceiver:
(rtc::scoped_refptr<webrtc::RtpTransceiverInterface>)nativeRtpTransceiver
NS_DESIGNATED_INITIALIZER;
+ (webrtc::RtpTransceiverDirection)nativeRtpTransceiverDirectionFromDirection:
(RTCRtpTransceiverDirection)direction;
+ (RTCRtpTransceiverDirection)rtpTransceiverDirectionFromNativeDirection:
(webrtc::RtpTransceiverDirection)nativeDirection;
@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 "RTCRtpTransceiver+Private.h"
#import "NSString+StdString.h"
#import "RTCRtpEncodingParameters+Private.h"
#import "RTCRtpParameters+Private.h"
#import "RTCRtpReceiver+Private.h"
#import "RTCRtpSender+Private.h"
#import "WebRTC/RTCLogging.h"
@implementation RTCRtpTransceiverInit
@synthesize direction = _direction;
@synthesize streamIds = _streamIds;
@synthesize sendEncodings = _sendEncodings;
- (instancetype)init {
if (self = [super init]) {
_direction = RTCRtpTransceiverDirectionSendRecv;
}
return self;
}
- (webrtc::RtpTransceiverInit)nativeInit {
webrtc::RtpTransceiverInit init;
init.direction = [RTCRtpTransceiver nativeRtpTransceiverDirectionFromDirection:_direction];
for (NSString *streamId in _streamIds) {
init.stream_ids.push_back([streamId UTF8String]);
}
for (RTCRtpEncodingParameters *sendEncoding in _sendEncodings) {
init.send_encodings.push_back(sendEncoding.nativeParameters);
}
return init;
}
@end
@implementation RTCRtpTransceiver {
RTCPeerConnectionFactory *_factory;
rtc::scoped_refptr<webrtc::RtpTransceiverInterface> _nativeRtpTransceiver;
}
- (RTCRtpMediaType)mediaType {
return [RTCRtpReceiver mediaTypeForNativeMediaType:_nativeRtpTransceiver->media_type()];
}
- (NSString *)mid {
if (_nativeRtpTransceiver->mid()) {
return [NSString stringForStdString:*_nativeRtpTransceiver->mid()];
} else {
return nil;
}
}
@synthesize sender = _sender;
@synthesize receiver = _receiver;
- (BOOL)isStopped {
return _nativeRtpTransceiver->stopped();
}
- (RTCRtpTransceiverDirection)direction {
return [RTCRtpTransceiver
rtpTransceiverDirectionFromNativeDirection:_nativeRtpTransceiver->direction()];
}
- (void)setDirection:(RTCRtpTransceiverDirection)direction {
_nativeRtpTransceiver->SetDirection(
[RTCRtpTransceiver nativeRtpTransceiverDirectionFromDirection:direction]);
}
- (BOOL)currentDirection:(RTCRtpTransceiverDirection *)currentDirectionOut {
if (_nativeRtpTransceiver->current_direction()) {
*currentDirectionOut = [RTCRtpTransceiver
rtpTransceiverDirectionFromNativeDirection:*_nativeRtpTransceiver->current_direction()];
return YES;
} else {
return NO;
}
}
- (void)stop {
_nativeRtpTransceiver->Stop();
}
- (NSString *)description {
return [NSString
stringWithFormat:@"RTCRtpTransceiver {\n sender: %@\n receiver: %@\n}", _sender, _receiver];
}
- (BOOL)isEqual:(id)object {
if (self == object) {
return YES;
}
if (object == nil) {
return NO;
}
if (![object isMemberOfClass:[self class]]) {
return NO;
}
RTCRtpTransceiver *transceiver = (RTCRtpTransceiver *)object;
return _nativeRtpTransceiver == transceiver.nativeRtpTransceiver;
}
- (NSUInteger)hash {
return (NSUInteger)_nativeRtpTransceiver.get();
}
#pragma mark - Private
- (rtc::scoped_refptr<webrtc::RtpTransceiverInterface>)nativeRtpTransceiver {
return _nativeRtpTransceiver;
}
- (instancetype)initWithFactory:(RTCPeerConnectionFactory *)factory
nativeRtpTransceiver:
(rtc::scoped_refptr<webrtc::RtpTransceiverInterface>)nativeRtpTransceiver {
NSParameterAssert(factory);
NSParameterAssert(nativeRtpTransceiver);
if (self = [super init]) {
_factory = factory;
_nativeRtpTransceiver = nativeRtpTransceiver;
_sender = [[RTCRtpSender alloc] initWithFactory:_factory
nativeRtpSender:nativeRtpTransceiver->sender()];
_receiver = [[RTCRtpReceiver alloc] initWithFactory:_factory
nativeRtpReceiver:nativeRtpTransceiver->receiver()];
RTCLogInfo(@"RTCRtpTransceiver(%p): created transceiver: %@", self, self.description);
}
return self;
}
+ (webrtc::RtpTransceiverDirection)nativeRtpTransceiverDirectionFromDirection:
(RTCRtpTransceiverDirection)direction {
switch (direction) {
case RTCRtpTransceiverDirectionSendRecv:
return webrtc::RtpTransceiverDirection::kSendRecv;
case RTCRtpTransceiverDirectionSendOnly:
return webrtc::RtpTransceiverDirection::kSendOnly;
case RTCRtpTransceiverDirectionRecvOnly:
return webrtc::RtpTransceiverDirection::kRecvOnly;
case RTCRtpTransceiverDirectionInactive:
return webrtc::RtpTransceiverDirection::kInactive;
}
}
+ (RTCRtpTransceiverDirection)rtpTransceiverDirectionFromNativeDirection:
(webrtc::RtpTransceiverDirection)nativeDirection {
switch (nativeDirection) {
case webrtc::RtpTransceiverDirection::kSendRecv:
return RTCRtpTransceiverDirectionSendRecv;
case webrtc::RtpTransceiverDirection::kSendOnly:
return RTCRtpTransceiverDirectionSendOnly;
case webrtc::RtpTransceiverDirection::kRecvOnly:
return RTCRtpTransceiverDirectionRecvOnly;
case webrtc::RtpTransceiverDirection::kInactive:
return RTCRtpTransceiverDirectionInactive;
}
}
@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 "WebRTC/RTCSSLAdapter.h"
#include "rtc_base/checks.h"
#include "rtc_base/ssladapter.h"
BOOL RTCInitializeSSL(void) {
BOOL initialized = rtc::InitializeSSL();
RTC_DCHECK(initialized);
return initialized;
}
BOOL RTCCleanupSSL(void) {
BOOL cleanedUp = rtc::CleanupSSL();
RTC_DCHECK(cleanedUp);
return cleanedUp;
}

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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/RTCSessionDescription.h"
#include "api/jsep.h"
NS_ASSUME_NONNULL_BEGIN
@interface RTCSessionDescription ()
/**
* The native SessionDescriptionInterface representation of this
* RTCSessionDescription object. This is needed to pass to the underlying C++
* APIs.
*/
@property(nonatomic, readonly, nullable) webrtc::SessionDescriptionInterface *nativeDescription;
/**
* Initialize an RTCSessionDescription from a native
* SessionDescriptionInterface. No ownership is taken of the native session
* description.
*/
- (instancetype)initWithNativeDescription:
(const webrtc::SessionDescriptionInterface *)nativeDescription;
+ (std::string)stdStringForType:(RTCSdpType)type;
+ (RTCSdpType)typeForStdString:(const std::string &)string;
@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 "RTCSessionDescription+Private.h"
#import "NSString+StdString.h"
#import "WebRTC/RTCLogging.h"
#include "rtc_base/checks.h"
@implementation RTCSessionDescription
@synthesize type = _type;
@synthesize sdp = _sdp;
+ (NSString *)stringForType:(RTCSdpType)type {
std::string string = [[self class] stdStringForType:type];
return [NSString stringForStdString:string];
}
+ (RTCSdpType)typeForString:(NSString *)string {
std::string typeString = string.stdString;
return [[self class] typeForStdString:typeString];
}
- (instancetype)initWithType:(RTCSdpType)type sdp:(NSString *)sdp {
NSParameterAssert(sdp.length);
if (self = [super init]) {
_type = type;
_sdp = [sdp copy];
}
return self;
}
- (NSString *)description {
return [NSString stringWithFormat:@"RTCSessionDescription:\n%@\n%@",
[[self class] stringForType:_type],
_sdp];
}
#pragma mark - Private
- (webrtc::SessionDescriptionInterface *)nativeDescription {
webrtc::SdpParseError error;
webrtc::SessionDescriptionInterface *description =
webrtc::CreateSessionDescription([[self class] stdStringForType:_type],
_sdp.stdString,
&error);
if (!description) {
RTCLogError(@"Failed to create session description: %s\nline: %s",
error.description.c_str(),
error.line.c_str());
}
return description;
}
- (instancetype)initWithNativeDescription:
(const webrtc::SessionDescriptionInterface *)nativeDescription {
NSParameterAssert(nativeDescription);
std::string sdp;
nativeDescription->ToString(&sdp);
RTCSdpType type = [[self class] typeForStdString:nativeDescription->type()];
return [self initWithType:type
sdp:[NSString stringForStdString:sdp]];
}
+ (std::string)stdStringForType:(RTCSdpType)type {
switch (type) {
case RTCSdpTypeOffer:
return webrtc::SessionDescriptionInterface::kOffer;
case RTCSdpTypePrAnswer:
return webrtc::SessionDescriptionInterface::kPrAnswer;
case RTCSdpTypeAnswer:
return webrtc::SessionDescriptionInterface::kAnswer;
}
}
+ (RTCSdpType)typeForStdString:(const std::string &)string {
if (string == webrtc::SessionDescriptionInterface::kOffer) {
return RTCSdpTypeOffer;
} else if (string == webrtc::SessionDescriptionInterface::kPrAnswer) {
return RTCSdpTypePrAnswer;
} else if (string == webrtc::SessionDescriptionInterface::kAnswer) {
return RTCSdpTypeAnswer;
} else {
RTC_NOTREACHED();
return RTCSdpTypeOffer;
}
}
@end

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@ -1,29 +0,0 @@
/*
* 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 "WebRTC/RTCTracing.h"
#include "rtc_base/event_tracer.h"
void RTCSetupInternalTracer(void) {
rtc::tracing::SetupInternalTracer();
}
BOOL RTCStartInternalCapture(NSString *filePath) {
return rtc::tracing::StartInternalCapture(filePath.UTF8String);
}
void RTCStopInternalCapture(void) {
rtc::tracing::StopInternalCapture();
}
void RTCShutdownInternalTracer(void) {
rtc::tracing::ShutdownInternalTracer();
}

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@ -1,24 +0,0 @@
/*
* 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/RTCVideoCapturer.h"
@implementation RTCVideoCapturer
@synthesize delegate = _delegate;
- (instancetype)initWithDelegate:(id<RTCVideoCapturerDelegate>)delegate {
if (self = [super init]) {
_delegate = delegate;
}
return self;
}
@end

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@ -8,51 +8,8 @@
* be found in the AUTHORS file in the root of the source tree.
*/
#import "WebRTC/RTCVideoCodec.h"
#import "WebRTC/RTCVideoCodecH264.h"
#include "api/video_codecs/sdp_video_format.h"
#include "common_video/include/video_frame.h"
#include "media/base/codec.h"
#include "modules/video_coding/include/video_codec_interface.h"
NS_ASSUME_NONNULL_BEGIN
/* Interfaces for converting to/from internal C++ formats. */
@interface RTCEncodedImage ()
- (instancetype)initWithNativeEncodedImage:(webrtc::EncodedImage)encodedImage;
- (webrtc::EncodedImage)nativeEncodedImage;
@end
@interface RTCVideoEncoderSettings ()
- (instancetype)initWithNativeVideoCodec:(const webrtc::VideoCodec *__nullable)videoCodec;
- (webrtc::VideoCodec)nativeVideoCodec;
@end
@interface RTCCodecSpecificInfoH264 ()
- (webrtc::CodecSpecificInfo)nativeCodecSpecificInfo;
@end
@interface RTCRtpFragmentationHeader ()
- (instancetype)initWithNativeFragmentationHeader:
(const webrtc::RTPFragmentationHeader *__nullable)fragmentationHeader;
- (std::unique_ptr<webrtc::RTPFragmentationHeader>)createNativeFragmentationHeader;
@end
@interface RTCVideoCodecInfo ()
- (instancetype)initWithNativeSdpVideoFormat:(webrtc::SdpVideoFormat)format;
- (webrtc::SdpVideoFormat)nativeSdpVideoFormat;
@end
NS_ASSUME_NONNULL_END
#import "api/peerconnection/RTCEncodedImage+Private.h"
#import "api/peerconnection/RTCRtpFragmentationHeader+Private.h"
#import "api/peerconnection/RTCVideoCodecInfo+Private.h"
#import "api/peerconnection/RTCVideoEncoderSettings+Private.h"
#import "components/video_codec/RTCCodecSpecificInfoH264+Private.h"

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@ -1,167 +0,0 @@
/*
* 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/RTCVideoCodec.h"
#import "NSString+StdString.h"
#import "RTCVideoCodec+Private.h"
#if defined(WEBRTC_IOS)
#import "UIDevice+H264Profile.h"
#endif
#import "WebRTC/RTCVideoCodecFactory.h"
#include "media/base/mediaconstants.h"
namespace {
NSString *MaxSupportedProfileLevelConstrainedHigh();
NSString *MaxSupportedProfileLevelConstrainedBaseline();
} // namespace
NSString *const kRTCVideoCodecVp8Name = @(cricket::kVp8CodecName);
NSString *const kRTCVideoCodecVp9Name = @(cricket::kVp9CodecName);
NSString *const kRTCVideoCodecH264Name = @(cricket::kH264CodecName);
NSString *const kRTCLevel31ConstrainedHigh = @"640c1f";
NSString *const kRTCLevel31ConstrainedBaseline = @"42e01f";
NSString *const kRTCMaxSupportedH264ProfileLevelConstrainedHigh =
MaxSupportedProfileLevelConstrainedHigh();
NSString *const kRTCMaxSupportedH264ProfileLevelConstrainedBaseline =
MaxSupportedProfileLevelConstrainedBaseline();
namespace {
#if defined(WEBRTC_IOS)
using namespace webrtc::H264;
NSString *MaxSupportedLevelForProfile(Profile profile) {
const absl::optional<ProfileLevelId> profileLevelId = [UIDevice maxSupportedH264Profile];
if (profileLevelId && profileLevelId->profile >= profile) {
const absl::optional<std::string> profileString =
ProfileLevelIdToString(ProfileLevelId(profile, profileLevelId->level));
if (profileString) {
return [NSString stringForStdString:*profileString];
}
}
return nil;
}
#endif
NSString *MaxSupportedProfileLevelConstrainedBaseline() {
#if defined(WEBRTC_IOS)
NSString *profile = MaxSupportedLevelForProfile(webrtc::H264::kProfileConstrainedBaseline);
if (profile != nil) {
return profile;
}
#endif
return kRTCLevel31ConstrainedBaseline;
}
NSString *MaxSupportedProfileLevelConstrainedHigh() {
#if defined(WEBRTC_IOS)
NSString *profile = MaxSupportedLevelForProfile(webrtc::H264::kProfileConstrainedHigh);
if (profile != nil) {
return profile;
}
#endif
return kRTCLevel31ConstrainedHigh;
}
} // namespace
@implementation RTCVideoCodecInfo
@synthesize name = _name;
@synthesize parameters = _parameters;
- (instancetype)initWithName:(NSString *)name {
return [self initWithName:name parameters:nil];
}
- (instancetype)initWithName:(NSString *)name
parameters:(nullable NSDictionary<NSString *, NSString *> *)parameters {
if (self = [super init]) {
_name = name;
_parameters = (parameters ? parameters : @{});
}
return self;
}
- (instancetype)initWithNativeSdpVideoFormat:(webrtc::SdpVideoFormat)format {
NSMutableDictionary *params = [NSMutableDictionary dictionary];
for (auto it = format.parameters.begin(); it != format.parameters.end(); ++it) {
[params setObject:[NSString stringForStdString:it->second]
forKey:[NSString stringForStdString:it->first]];
}
return [self initWithName:[NSString stringForStdString:format.name] parameters:params];
}
- (BOOL)isEqualToCodecInfo:(RTCVideoCodecInfo *)info {
if (!info ||
![self.name isEqualToString:info.name] ||
![self.parameters isEqualToDictionary:info.parameters]) {
return NO;
}
return YES;
}
- (BOOL)isEqual:(id)object {
if (self == object)
return YES;
if (![object isKindOfClass:[self class]])
return NO;
return [self isEqualToCodecInfo:object];
}
- (NSUInteger)hash {
return [self.name hash] ^ [self.parameters hash];
}
- (webrtc::SdpVideoFormat)nativeSdpVideoFormat {
std::map<std::string, std::string> parameters;
for (NSString *paramKey in _parameters.allKeys) {
std::string key = [NSString stdStringForString:paramKey];
std::string value = [NSString stdStringForString:_parameters[paramKey]];
parameters[key] = value;
}
return webrtc::SdpVideoFormat([NSString stdStringForString:_name], parameters);
}
#pragma mark - NSCoding
- (instancetype)initWithCoder:(NSCoder *)decoder {
return [self initWithName:[decoder decodeObjectForKey:@"name"]
parameters:[decoder decodeObjectForKey:@"parameters"]];
}
- (void)encodeWithCoder:(NSCoder *)encoder {
[encoder encodeObject:_name forKey:@"name"];
[encoder encodeObject:_parameters forKey:@"parameters"];
}
@end
@implementation RTCVideoEncoderQpThresholds
@synthesize low = _low;
@synthesize high = _high;
- (instancetype)initWithThresholdsLow:(NSInteger)low high:(NSInteger)high {
if (self = [super init]) {
_low = low;
_high = high;
}
return self;
}
@end

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@ -1,83 +0,0 @@
/*
* 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/RTCVideoCodecH264.h"
#include <vector>
#import "RTCVideoCodec+Private.h"
#import "WebRTC/RTCVideoCodec.h"
#include "rtc_base/timeutils.h"
// H264 specific settings.
@implementation RTCCodecSpecificInfoH264
@synthesize packetizationMode = _packetizationMode;
- (webrtc::CodecSpecificInfo)nativeCodecSpecificInfo {
webrtc::CodecSpecificInfo codecSpecificInfo;
codecSpecificInfo.codecType = webrtc::kVideoCodecH264;
codecSpecificInfo.codec_name = [kRTCVideoCodecH264Name cStringUsingEncoding:NSUTF8StringEncoding];
codecSpecificInfo.codecSpecific.H264.packetization_mode =
(webrtc::H264PacketizationMode)_packetizationMode;
return codecSpecificInfo;
}
@end
// Encoder factory.
@implementation RTCVideoEncoderFactoryH264
- (NSArray<RTCVideoCodecInfo *> *)supportedCodecs {
NSMutableArray<RTCVideoCodecInfo *> *codecs = [NSMutableArray array];
NSString *codecName = kRTCVideoCodecH264Name;
NSDictionary<NSString *, NSString *> *constrainedHighParams = @{
@"profile-level-id" : kRTCMaxSupportedH264ProfileLevelConstrainedHigh,
@"level-asymmetry-allowed" : @"1",
@"packetization-mode" : @"1",
};
RTCVideoCodecInfo *constrainedHighInfo =
[[RTCVideoCodecInfo alloc] initWithName:codecName parameters:constrainedHighParams];
[codecs addObject:constrainedHighInfo];
NSDictionary<NSString *, NSString *> *constrainedBaselineParams = @{
@"profile-level-id" : kRTCMaxSupportedH264ProfileLevelConstrainedBaseline,
@"level-asymmetry-allowed" : @"1",
@"packetization-mode" : @"1",
};
RTCVideoCodecInfo *constrainedBaselineInfo =
[[RTCVideoCodecInfo alloc] initWithName:codecName parameters:constrainedBaselineParams];
[codecs addObject:constrainedBaselineInfo];
return [codecs copy];
}
- (id<RTCVideoEncoder>)createEncoder:(RTCVideoCodecInfo *)info {
return [[RTCVideoEncoderH264 alloc] initWithCodecInfo:info];
}
@end
// Decoder factory.
@implementation RTCVideoDecoderFactoryH264
- (id<RTCVideoDecoder>)createDecoder:(RTCVideoCodecInfo *)info {
return [[RTCVideoDecoderH264 alloc] init];
}
- (NSArray<RTCVideoCodecInfo *> *)supportedCodecs {
NSString *codecName = kRTCVideoCodecH264Name;
return @[ [[RTCVideoCodecInfo alloc] initWithName:codecName parameters:nil] ];
}
@end

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@ -1,41 +0,0 @@
/*
* Copyright (c) 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 "RTCWrappedNativeVideoDecoder.h"
#import "RTCWrappedNativeVideoEncoder.h"
#import "WebRTC/RTCVideoDecoderVP8.h"
#import "WebRTC/RTCVideoEncoderVP8.h"
#include "modules/video_coding/codecs/vp8/include/vp8.h"
#pragma mark - Encoder
@implementation RTCVideoEncoderVP8
+ (id<RTCVideoEncoder>)vp8Encoder {
return [[RTCWrappedNativeVideoEncoder alloc]
initWithNativeEncoder:std::unique_ptr<webrtc::VideoEncoder>(webrtc::VP8Encoder::Create())];
}
@end
#pragma mark - Decoder
@implementation RTCVideoDecoderVP8
+ (id<RTCVideoDecoder>)vp8Decoder {
return [[RTCWrappedNativeVideoDecoder alloc]
initWithNativeDecoder:std::unique_ptr<webrtc::VideoDecoder>(webrtc::VP8Decoder::Create())];
}
@end

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@ -1,41 +0,0 @@
/*
* Copyright (c) 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 "RTCWrappedNativeVideoDecoder.h"
#import "RTCWrappedNativeVideoEncoder.h"
#import "WebRTC/RTCVideoDecoderVP9.h"
#import "WebRTC/RTCVideoEncoderVP9.h"
#include "modules/video_coding/codecs/vp9/include/vp9.h"
#pragma mark - Encoder
@implementation RTCVideoEncoderVP9
+ (id<RTCVideoEncoder>)vp9Encoder {
return [[RTCWrappedNativeVideoEncoder alloc]
initWithNativeEncoder:std::unique_ptr<webrtc::VideoEncoder>(webrtc::VP9Encoder::Create())];
}
@end
#pragma mark - Decoder
@implementation RTCVideoDecoderVP9
+ (id<RTCVideoDecoder>)vp9Decoder {
return [[RTCWrappedNativeVideoDecoder alloc]
initWithNativeDecoder:std::unique_ptr<webrtc::VideoDecoder>(webrtc::VP9Decoder::Create())];
}
@end

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@ -1,66 +0,0 @@
/*
* 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/RTCVideoCodec.h"
#import "NSString+StdString.h"
#import "RTCVideoCodec+Private.h"
#import "WebRTC/RTCVideoCodecFactory.h"
@implementation RTCVideoEncoderSettings
@synthesize name = _name;
@synthesize width = _width;
@synthesize height = _height;
@synthesize startBitrate = _startBitrate;
@synthesize maxBitrate = _maxBitrate;
@synthesize minBitrate = _minBitrate;
@synthesize targetBitrate = _targetBitrate;
@synthesize maxFramerate = _maxFramerate;
@synthesize qpMax = _qpMax;
@synthesize mode = _mode;
- (instancetype)initWithNativeVideoCodec:(const webrtc::VideoCodec *)videoCodec {
if (self = [super init]) {
if (videoCodec) {
const char *codecName = CodecTypeToPayloadString(videoCodec->codecType);
_name = [NSString stringWithUTF8String:codecName];
_width = videoCodec->width;
_height = videoCodec->height;
_startBitrate = videoCodec->startBitrate;
_maxBitrate = videoCodec->maxBitrate;
_minBitrate = videoCodec->minBitrate;
_targetBitrate = videoCodec->targetBitrate;
_maxFramerate = videoCodec->maxFramerate;
_qpMax = videoCodec->qpMax;
_mode = (RTCVideoCodecMode)videoCodec->mode;
}
}
return self;
}
- (webrtc::VideoCodec)nativeVideoCodec {
webrtc::VideoCodec videoCodec;
videoCodec.width = _width;
videoCodec.height = _height;
videoCodec.startBitrate = _startBitrate;
videoCodec.maxBitrate = _maxBitrate;
videoCodec.minBitrate = _minBitrate;
videoCodec.targetBitrate = _targetBitrate;
videoCodec.maxBitrate = _maxBitrate;
videoCodec.qpMax = _qpMax;
videoCodec.mode = (webrtc::VideoCodecMode)_mode;
return videoCodec;
}
@end

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@ -1,84 +0,0 @@
/*
* 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/RTCVideoFrame.h"
#import "WebRTC/RTCVideoFrameBuffer.h"
@implementation RTCVideoFrame {
RTCVideoRotation _rotation;
int64_t _timeStampNs;
}
@synthesize buffer = _buffer;
@synthesize timeStamp;
- (int)width {
return _buffer.width;
}
- (int)height {
return _buffer.height;
}
- (RTCVideoRotation)rotation {
return _rotation;
}
- (int64_t)timeStampNs {
return _timeStampNs;
}
- (RTCVideoFrame *)newI420VideoFrame {
return [[RTCVideoFrame alloc] initWithBuffer:[_buffer toI420]
rotation:_rotation
timeStampNs:_timeStampNs];
}
- (instancetype)initWithPixelBuffer:(CVPixelBufferRef)pixelBuffer
rotation:(RTCVideoRotation)rotation
timeStampNs:(int64_t)timeStampNs {
return [self initWithBuffer:[[RTCCVPixelBuffer alloc] initWithPixelBuffer:pixelBuffer]
rotation:rotation
timeStampNs:timeStampNs];
}
- (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 {
RTCCVPixelBuffer *rtcPixelBuffer = [[RTCCVPixelBuffer alloc] initWithPixelBuffer:pixelBuffer
adaptedWidth:scaledWidth
adaptedHeight:scaledHeight
cropWidth:cropWidth
cropHeight:cropHeight
cropX:cropX
cropY:cropY];
return [self initWithBuffer:rtcPixelBuffer rotation:rotation timeStampNs:timeStampNs];
}
- (instancetype)initWithBuffer:(id<RTCVideoFrameBuffer>)buffer
rotation:(RTCVideoRotation)rotation
timeStampNs:(int64_t)timeStampNs {
if (self = [super init]) {
_buffer = buffer;
_rotation = rotation;
_timeStampNs = timeStampNs;
}
return self;
}
@end

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@ -1,41 +0,0 @@
/*
* 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 "RTCVideoRendererAdapter.h"
#import "WebRTC/RTCVideoRenderer.h"
#include "api/mediastreaminterface.h"
NS_ASSUME_NONNULL_BEGIN
@interface RTCVideoRendererAdapter ()
/**
* The Objective-C video renderer passed to this adapter during construction.
* Calls made to the webrtc::VideoRenderInterface will be adapted and passed to
* this video renderer.
*/
@property(nonatomic, readonly) id<RTCVideoRenderer> videoRenderer;
/**
* The native VideoSinkInterface surface exposed by this adapter. Calls made
* to this interface will be adapted and passed to the RTCVideoRenderer supplied
* during construction. This pointer is unsafe and owned by this class.
*/
@property(nonatomic, readonly) rtc::VideoSinkInterface<webrtc::VideoFrame> *nativeVideoRenderer;
/** Initialize an RTCVideoRendererAdapter with an RTCVideoRenderer. */
- (instancetype)initWithNativeRenderer:(id<RTCVideoRenderer>)videoRenderer
NS_DESIGNATED_INITIALIZER;
@end
NS_ASSUME_NONNULL_END

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@ -1,27 +0,0 @@
/*
* 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>
NS_ASSUME_NONNULL_BEGIN
/*
* Creates a rtc::VideoSinkInterface surface for an RTCVideoRenderer. The
* rtc::VideoSinkInterface is used by WebRTC rendering code - this
* adapter adapts calls made to that interface to the RTCVideoRenderer supplied
* during construction.
*/
@interface RTCVideoRendererAdapter : NSObject
- (instancetype)init NS_UNAVAILABLE;
@end
NS_ASSUME_NONNULL_END

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@ -1,69 +0,0 @@
/*
* 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 "RTCI420Buffer+Private.h"
#import "RTCVideoRendererAdapter+Private.h"
#import "WebRTC/RTCVideoFrame.h"
#import "WebRTC/RTCVideoFrameBuffer.h"
#include <memory>
#include "sdk/objc/Framework/Native/api/video_frame.h"
namespace webrtc {
class VideoRendererAdapter
: public rtc::VideoSinkInterface<webrtc::VideoFrame> {
public:
VideoRendererAdapter(RTCVideoRendererAdapter* adapter) {
adapter_ = adapter;
size_ = CGSizeZero;
}
void OnFrame(const webrtc::VideoFrame& nativeVideoFrame) override {
RTCVideoFrame* videoFrame = NativeToObjCVideoFrame(nativeVideoFrame);
CGSize current_size = (videoFrame.rotation % 180 == 0)
? CGSizeMake(videoFrame.width, videoFrame.height)
: CGSizeMake(videoFrame.height, videoFrame.width);
if (!CGSizeEqualToSize(size_, current_size)) {
size_ = current_size;
[adapter_.videoRenderer setSize:size_];
}
[adapter_.videoRenderer renderFrame:videoFrame];
}
private:
__weak RTCVideoRendererAdapter *adapter_;
CGSize size_;
};
}
@implementation RTCVideoRendererAdapter {
std::unique_ptr<webrtc::VideoRendererAdapter> _adapter;
}
@synthesize videoRenderer = _videoRenderer;
- (instancetype)initWithNativeRenderer:(id<RTCVideoRenderer>)videoRenderer {
NSParameterAssert(videoRenderer);
if (self = [super init]) {
_videoRenderer = videoRenderer;
_adapter.reset(new webrtc::VideoRendererAdapter(self));
}
return self;
}
- (rtc::VideoSinkInterface<webrtc::VideoFrame> *)nativeVideoRenderer {
return _adapter.get();
}
@end

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@ -1,44 +0,0 @@
/*
* 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/RTCVideoSource.h"
#import "RTCMediaSource+Private.h"
#include "api/mediastreaminterface.h"
NS_ASSUME_NONNULL_BEGIN
@interface RTCVideoSource ()
/**
* The VideoTrackSourceInterface object passed to this RTCVideoSource during
* construction.
*/
@property(nonatomic, readonly) rtc::scoped_refptr<webrtc::VideoTrackSourceInterface>
nativeVideoSource;
/** Initialize an RTCVideoSource from a native VideoTrackSourceInterface. */
- (instancetype)initWithFactory:(RTCPeerConnectionFactory *)factory
nativeVideoSource:
(rtc::scoped_refptr<webrtc::VideoTrackSourceInterface>)nativeVideoSource
NS_DESIGNATED_INITIALIZER;
- (instancetype)initWithFactory:(RTCPeerConnectionFactory *)factory
nativeMediaSource:(rtc::scoped_refptr<webrtc::MediaSourceInterface>)nativeMediaSource
type:(RTCMediaSourceType)type NS_UNAVAILABLE;
- (instancetype)initWithFactory:(RTCPeerConnectionFactory *)factory
signalingThread:(rtc::Thread *)signalingThread
workerThread:(rtc::Thread *)workerThread;
@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 "RTCVideoSource+Private.h"
#include "api/videosourceproxy.h"
#include "rtc_base/checks.h"
#include "sdk/objc/Framework/Native/src/objc_video_track_source.h"
static webrtc::ObjCVideoTrackSource *getObjCVideoSource(
const rtc::scoped_refptr<webrtc::VideoTrackSourceInterface> nativeSource) {
webrtc::VideoTrackSourceProxy *proxy_source =
static_cast<webrtc::VideoTrackSourceProxy *>(nativeSource.get());
return static_cast<webrtc::ObjCVideoTrackSource *>(proxy_source->internal());
}
// TODO(magjed): Refactor this class and target ObjCVideoTrackSource only once
// RTCAVFoundationVideoSource is gone. See http://crbug/webrtc/7177 for more
// info.
@implementation RTCVideoSource {
rtc::scoped_refptr<webrtc::VideoTrackSourceInterface> _nativeVideoSource;
}
- (instancetype)initWithFactory:(RTCPeerConnectionFactory *)factory
nativeVideoSource:
(rtc::scoped_refptr<webrtc::VideoTrackSourceInterface>)nativeVideoSource {
RTC_DCHECK(factory);
RTC_DCHECK(nativeVideoSource);
if (self = [super initWithFactory:factory
nativeMediaSource:nativeVideoSource
type:RTCMediaSourceTypeVideo]) {
_nativeVideoSource = nativeVideoSource;
}
return self;
}
- (instancetype)initWithFactory:(RTCPeerConnectionFactory *)factory
nativeMediaSource:(rtc::scoped_refptr<webrtc::MediaSourceInterface>)nativeMediaSource
type:(RTCMediaSourceType)type {
RTC_NOTREACHED();
return nil;
}
- (instancetype)initWithFactory:(RTCPeerConnectionFactory *)factory
signalingThread:(rtc::Thread *)signalingThread
workerThread:(rtc::Thread *)workerThread {
rtc::scoped_refptr<webrtc::ObjCVideoTrackSource> objCVideoTrackSource(
new rtc::RefCountedObject<webrtc::ObjCVideoTrackSource>());
return [self initWithFactory:factory
nativeVideoSource:webrtc::VideoTrackSourceProxy::Create(
signalingThread, workerThread, objCVideoTrackSource)];
}
- (NSString *)description {
NSString *stateString = [[self class] stringForState:self.state];
return [NSString stringWithFormat:@"RTCVideoSource( %p ): %@", self, stateString];
}
- (void)capturer:(RTCVideoCapturer *)capturer didCaptureVideoFrame:(RTCVideoFrame *)frame {
getObjCVideoSource(_nativeVideoSource)->OnCapturedFrame(frame);
}
- (void)adaptOutputFormatToWidth:(int)width height:(int)height fps:(int)fps {
getObjCVideoSource(_nativeVideoSource)->OnOutputFormatRequest(width, height, fps);
}
#pragma mark - Private
- (rtc::scoped_refptr<webrtc::VideoTrackSourceInterface>)nativeVideoSource {
return _nativeVideoSource;
}
@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/RTCVideoTrack.h"
#include "api/mediastreaminterface.h"
NS_ASSUME_NONNULL_BEGIN
@interface RTCVideoTrack ()
/** VideoTrackInterface created or passed in at construction. */
@property(nonatomic, readonly) rtc::scoped_refptr<webrtc::VideoTrackInterface> nativeVideoTrack;
/** Initialize an RTCVideoTrack with its source and an id. */
- (instancetype)initWithFactory:(RTCPeerConnectionFactory *)factory
source:(RTCVideoSource *)source
trackId:(NSString *)trackId;
@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 "RTCVideoTrack+Private.h"
#import "NSString+StdString.h"
#import "RTCMediaStreamTrack+Private.h"
#import "RTCPeerConnectionFactory+Private.h"
#import "RTCVideoRendererAdapter+Private.h"
#import "RTCVideoSource+Private.h"
@implementation RTCVideoTrack {
NSMutableArray *_adapters;
}
@synthesize source = _source;
- (instancetype)initWithFactory:(RTCPeerConnectionFactory *)factory
source:(RTCVideoSource *)source
trackId:(NSString *)trackId {
NSParameterAssert(factory);
NSParameterAssert(source);
NSParameterAssert(trackId.length);
std::string nativeId = [NSString stdStringForString:trackId];
rtc::scoped_refptr<webrtc::VideoTrackInterface> track =
factory.nativeFactory->CreateVideoTrack(nativeId,
source.nativeVideoSource);
if (self = [self initWithFactory:factory nativeTrack:track type:RTCMediaStreamTrackTypeVideo]) {
_source = source;
}
return self;
}
- (instancetype)initWithFactory:(RTCPeerConnectionFactory *)factory
nativeTrack:
(rtc::scoped_refptr<webrtc::MediaStreamTrackInterface>)nativeMediaTrack
type:(RTCMediaStreamTrackType)type {
NSParameterAssert(factory);
NSParameterAssert(nativeMediaTrack);
NSParameterAssert(type == RTCMediaStreamTrackTypeVideo);
if (self = [super initWithFactory:factory nativeTrack:nativeMediaTrack type:type]) {
_adapters = [NSMutableArray array];
}
return self;
}
- (void)dealloc {
for (RTCVideoRendererAdapter *adapter in _adapters) {
self.nativeVideoTrack->RemoveSink(adapter.nativeVideoRenderer);
}
}
- (RTCVideoSource *)source {
if (!_source) {
rtc::scoped_refptr<webrtc::VideoTrackSourceInterface> source =
self.nativeVideoTrack->GetSource();
if (source) {
_source =
[[RTCVideoSource alloc] initWithFactory:self.factory nativeVideoSource:source.get()];
}
}
return _source;
}
- (void)addRenderer:(id<RTCVideoRenderer>)renderer {
// Make sure we don't have this renderer yet.
for (RTCVideoRendererAdapter *adapter in _adapters) {
if (adapter.videoRenderer == renderer) {
NSAssert(NO, @"|renderer| is already attached to this track");
return;
}
}
// Create a wrapper that provides a native pointer for us.
RTCVideoRendererAdapter* adapter =
[[RTCVideoRendererAdapter alloc] initWithNativeRenderer:renderer];
[_adapters addObject:adapter];
self.nativeVideoTrack->AddOrUpdateSink(adapter.nativeVideoRenderer,
rtc::VideoSinkWants());
}
- (void)removeRenderer:(id<RTCVideoRenderer>)renderer {
__block NSUInteger indexToRemove = NSNotFound;
[_adapters enumerateObjectsUsingBlock:^(RTCVideoRendererAdapter *adapter,
NSUInteger idx,
BOOL *stop) {
if (adapter.videoRenderer == renderer) {
indexToRemove = idx;
*stop = YES;
}
}];
if (indexToRemove == NSNotFound) {
return;
}
RTCVideoRendererAdapter *adapterToRemove =
[_adapters objectAtIndex:indexToRemove];
self.nativeVideoTrack->RemoveSink(adapterToRemove.nativeVideoRenderer);
[_adapters removeObjectAtIndex:indexToRemove];
}
#pragma mark - Private
- (rtc::scoped_refptr<webrtc::VideoTrackInterface>)nativeVideoTrack {
return static_cast<webrtc::VideoTrackInterface *>(self.nativeTrack.get());
}
@end

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/*
* Copyright (c) 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/RTCVideoCodec.h"
#include "api/video_codecs/video_decoder.h"
#include "media/base/codec.h"
@interface RTCWrappedNativeVideoDecoder : NSObject <RTCVideoDecoder>
- (instancetype)initWithNativeDecoder:(std::unique_ptr<webrtc::VideoDecoder>)decoder;
/* This moves the ownership of the wrapped decoder to the caller. */
- (std::unique_ptr<webrtc::VideoDecoder>)releaseWrappedDecoder;
@end

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/*
* Copyright (c) 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 "NSString+StdString.h"
#import "RTCWrappedNativeVideoDecoder.h"
@implementation RTCWrappedNativeVideoDecoder {
std::unique_ptr<webrtc::VideoDecoder> _wrappedDecoder;
}
- (instancetype)initWithNativeDecoder:(std::unique_ptr<webrtc::VideoDecoder>)decoder {
if (self = [super init]) {
_wrappedDecoder = std::move(decoder);
}
return self;
}
- (std::unique_ptr<webrtc::VideoDecoder>)releaseWrappedDecoder {
return std::move(_wrappedDecoder);
}
#pragma mark - RTCVideoDecoder
- (void)setCallback:(RTCVideoDecoderCallback)callback {
RTC_NOTREACHED();
}
- (NSInteger)startDecodeWithNumberOfCores:(int)numberOfCores {
RTC_NOTREACHED();
return 0;
}
- (NSInteger)startDecodeWithSettings:(RTCVideoEncoderSettings *)settings
numberOfCores:(int)numberOfCores {
RTC_NOTREACHED();
return 0;
}
- (NSInteger)releaseDecoder {
RTC_NOTREACHED();
return 0;
}
- (NSInteger)decode:(RTCEncodedImage *)encodedImage
missingFrames:(BOOL)missingFrames
codecSpecificInfo:(nullable id<RTCCodecSpecificInfo>)info
renderTimeMs:(int64_t)renderTimeMs {
RTC_NOTREACHED();
return 0;
}
- (NSString *)implementationName {
RTC_NOTREACHED();
return nil;
}
@end

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/*
* Copyright (c) 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/RTCVideoCodec.h"
#include "api/video_codecs/sdp_video_format.h"
#include "api/video_codecs/video_encoder.h"
#include "media/base/codec.h"
@interface RTCWrappedNativeVideoEncoder : NSObject <RTCVideoEncoder>
- (instancetype)initWithNativeEncoder:(std::unique_ptr<webrtc::VideoEncoder>)encoder;
/* This moves the ownership of the wrapped encoder to the caller. */
- (std::unique_ptr<webrtc::VideoEncoder>)releaseWrappedEncoder;
@end

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/*
* Copyright (c) 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 "NSString+StdString.h"
#import "RTCWrappedNativeVideoEncoder.h"
@implementation RTCWrappedNativeVideoEncoder {
std::unique_ptr<webrtc::VideoEncoder> _wrappedEncoder;
}
- (instancetype)initWithNativeEncoder:(std::unique_ptr<webrtc::VideoEncoder>)encoder {
if (self = [super init]) {
_wrappedEncoder = std::move(encoder);
}
return self;
}
- (std::unique_ptr<webrtc::VideoEncoder>)releaseWrappedEncoder {
return std::move(_wrappedEncoder);
}
#pragma mark - RTCVideoEncoder
- (void)setCallback:(RTCVideoEncoderCallback)callback {
RTC_NOTREACHED();
}
- (NSInteger)startEncodeWithSettings:(RTCVideoEncoderSettings *)settings
numberOfCores:(int)numberOfCores {
RTC_NOTREACHED();
return 0;
}
- (NSInteger)releaseEncoder {
RTC_NOTREACHED();
return 0;
}
- (NSInteger)encode:(RTCVideoFrame *)frame
codecSpecificInfo:(nullable id<RTCCodecSpecificInfo>)info
frameTypes:(NSArray<NSNumber *> *)frameTypes {
RTC_NOTREACHED();
return 0;
}
- (int)setBitrate:(uint32_t)bitrateKbit framerate:(uint32_t)framerate {
RTC_NOTREACHED();
return 0;
}
- (NSString *)implementationName {
RTC_NOTREACHED();
return nil;
}
- (RTCVideoEncoderQpThresholds *)scalingSettings {
RTC_NOTREACHED();
return nil;
}
@end