webrtc/sdk/objc/Framework/Classes/PeerConnection/RTCPeerConnectionFactory.mm
Yura Yaroshevich bf567120bd Exposed setOptions via RTCPeerConnectionFactory
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}
2018-01-05 18:10:29 +00:00

274 lines
12 KiB
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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 "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