mirror of
https://github.com/mollyim/webrtc.git
synced 2025-05-15 06:40:43 +01:00

Exposed setOptions API for iOS SDK via RTCPeerConnectionFactory method to provide ability to disable encryption and control which network adapters are ignored. Only subset of webrtc::PeerConnectionFactoryInterface::Options options are exposed via iOS SDK, additional options can be exposed as requested. Android SDK has already exposed setOption API via Java's PeerConnection constructor, there changes provide similar functionaly to iOS SDK. Bug: webrtc:8712 Change-Id: Ia2de38cf382afc1bad9bbec6c6eac21ad29aee89 Reviewed-on: https://webrtc-review.googlesource.com/34900 Commit-Queue: Taylor Brandstetter <deadbeef@webrtc.org> Reviewed-by: Taylor Brandstetter <deadbeef@webrtc.org> Reviewed-by: Magnus Jedvert <magjed@webrtc.org> Cr-Commit-Position: refs/heads/master@{#21504}
274 lines
12 KiB
Text
274 lines
12 KiB
Text
/*
|
|
* 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 "RTCAVFoundationVideoSource+Private.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
|
|
#include "VideoToolbox/objc_video_decoder_factory.h"
|
|
#include "VideoToolbox/objc_video_encoder_factory.h"
|
|
#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 "modules/audio_device/include/audio_device.h" // nogncheck
|
|
#include "modules/audio_processing/include/audio_processing.h" // nogncheck
|
|
#endif
|
|
|
|
#include "Video/objcvideotracksource.h"
|
|
#include "api/videosourceproxy.h"
|
|
// 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 "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;
|
|
|
|
- (instancetype)init {
|
|
#ifdef HAVE_NO_MEDIA
|
|
return [self initWithNoMedia];
|
|
#else
|
|
return [self initWithNativeAudioEncoderFactory:webrtc::CreateBuiltinAudioEncoderFactory()
|
|
nativeAudioDecoderFactory:webrtc::CreateBuiltinAudioDecoderFactory()
|
|
legacyNativeVideoEncoderFactory:new webrtc::ObjCVideoEncoderFactory(
|
|
[[RTCVideoEncoderFactoryH264 alloc] init])
|
|
legacyNativeVideoDecoderFactory:new webrtc::ObjCVideoDecoderFactory(
|
|
[[RTCVideoDecoderFactoryH264 alloc] init])
|
|
audioDeviceModule: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.reset(new webrtc::ObjCVideoEncoderFactory(encoderFactory));
|
|
}
|
|
if (decoderFactory) {
|
|
native_decoder_factory.reset(new webrtc::ObjCVideoDecoderFactory(decoderFactory));
|
|
}
|
|
return [self initWithNativeAudioEncoderFactory:webrtc::CreateBuiltinAudioEncoderFactory()
|
|
nativeAudioDecoderFactory:webrtc::CreateBuiltinAudioDecoderFactory()
|
|
nativeVideoEncoderFactory:std::move(native_encoder_factory)
|
|
nativeVideoDecoderFactory:std::move(native_decoder_factory)
|
|
audioDeviceModule:nullptr
|
|
audioProcessingModule:nullptr];
|
|
#endif
|
|
}
|
|
|
|
- (instancetype)initNative {
|
|
if (self = [super init]) {
|
|
_networkThread = rtc::Thread::CreateWithSocketServer();
|
|
BOOL result = _networkThread->Start();
|
|
NSAssert(result, @"Failed to start network thread.");
|
|
|
|
_workerThread = rtc::Thread::Create();
|
|
result = _workerThread->Start();
|
|
NSAssert(result, @"Failed to start worker thread.");
|
|
|
|
_signalingThread = rtc::Thread::Create();
|
|
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
|
|
}
|
|
|
|
- (instancetype)
|
|
initWithNativeAudioEncoderFactory:
|
|
(rtc::scoped_refptr<webrtc::AudioEncoderFactory>)audioEncoderFactory
|
|
nativeAudioDecoderFactory:
|
|
(rtc::scoped_refptr<webrtc::AudioDecoderFactory>)audioDecoderFactory
|
|
legacyNativeVideoEncoderFactory:(cricket::WebRtcVideoEncoderFactory *)videoEncoderFactory
|
|
legacyNativeVideoDecoderFactory:(cricket::WebRtcVideoDecoderFactory *)videoDecoderFactory
|
|
audioDeviceModule:(nullable webrtc::AudioDeviceModule *)audioDeviceModule {
|
|
#ifdef HAVE_NO_MEDIA
|
|
return [self initWithNoMedia];
|
|
#else
|
|
if (self = [self initNative]) {
|
|
_nativeFactory = webrtc::CreatePeerConnectionFactory(_networkThread.get(),
|
|
_workerThread.get(),
|
|
_signalingThread.get(),
|
|
audioDeviceModule,
|
|
audioEncoderFactory,
|
|
audioDecoderFactory,
|
|
videoEncoderFactory,
|
|
videoDecoderFactory);
|
|
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;
|
|
}
|
|
rtc::scoped_refptr<webrtc::AudioSourceInterface> source =
|
|
_nativeFactory->CreateAudioSource(nativeConstraints.get());
|
|
return [[RTCAudioSource alloc] initWithNativeAudioSource: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];
|
|
}
|
|
|
|
- (RTCAVFoundationVideoSource *)avFoundationVideoSourceWithConstraints:
|
|
(nullable RTCMediaConstraints *)constraints {
|
|
#ifdef HAVE_NO_MEDIA
|
|
return nil;
|
|
#else
|
|
return [[RTCAVFoundationVideoSource alloc] initWithFactory:self constraints:constraints];
|
|
#endif
|
|
}
|
|
|
|
- (RTCVideoSource *)videoSource {
|
|
rtc::scoped_refptr<webrtc::ObjcVideoTrackSource> objcVideoTrackSource(
|
|
new rtc::RefCountedObject<webrtc::ObjcVideoTrackSource>());
|
|
return [[RTCVideoSource alloc]
|
|
initWithNativeVideoSource:webrtc::VideoTrackSourceProxy::Create(_signalingThread.get(),
|
|
_workerThread.get(),
|
|
objcVideoTrackSource)];
|
|
}
|
|
|
|
- (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
|