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webrtc/sdk/objc/api/peerconnection/RTCPeerConnectionFactory.mm
Mirko Bonadei a81e9c82fc Wrap WebRTC OBJC API types with RTC_OBJC_TYPE. 
This CL introduced 2 new macros that affect the WebRTC OBJC API symbols:

- RTC_OBJC_TYPE_PREFIX:
  Macro used to prepend a prefix to the API types that are exported with
  RTC_OBJC_EXPORT.

  Clients can patch the definition of this macro locally and build
  WebRTC.framework with their own prefix in case symbol clashing is a
  problem.

  This macro must only be defined by changing the value in
  sdk/objc/base/RTCMacros.h  and not on via compiler flag to ensure
  it has a unique value.

- RCT_OBJC_TYPE:
  Macro used internally to reference API types. Declaring an API type
  without using this macro will not include the declared type in the
  set of types that will be affected by the configurable
  RTC_OBJC_TYPE_PREFIX.

Manual changes:
https://webrtc-review.googlesource.com/c/src/+/173781/5..10

The auto-generated changes in PS#5 have been done with:
https://webrtc-review.googlesource.com/c/src/+/174061.

Bug: None
Change-Id: I0d54ca94db764fb3b6cb4365873f79e14cd879b8
Reviewed-on: https://webrtc-review.googlesource.com/c/src/+/173781
Commit-Queue: Mirko Bonadei <mbonadei@webrtc.org>
Reviewed-by: Karl Wiberg <kwiberg@webrtc.org>
Reviewed-by: Kári Helgason <kthelgason@webrtc.org>
Cr-Commit-Position: refs/heads/master@{#31153}
2020-05-04 15:01:26 +00:00

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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.
*/
#include <memory>
#import "RTCPeerConnectionFactory+Native.h"
#import "RTCPeerConnectionFactory+Private.h"
#import "RTCPeerConnectionFactoryOptions+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 "base/RTCLogging.h"
#import "base/RTCVideoDecoderFactory.h"
#import "base/RTCVideoEncoderFactory.h"
#import "helpers/NSString+StdString.h"
#ifndef HAVE_NO_MEDIA
#import "components/video_codec/RTCVideoDecoderFactoryH264.h"
#import "components/video_codec/RTCVideoEncoderFactoryH264.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 "api/rtc_event_log/rtc_event_log_factory.h"
#include "api/task_queue/default_task_queue_factory.h"
#include "modules/audio_device/include/audio_device.h" // nogncheck
#include "modules/audio_processing/include/audio_processing.h" // nogncheck
#include "sdk/objc/native/api/video_decoder_factory.h"
#include "sdk/objc/native/api/video_encoder_factory.h"
#include "sdk/objc/native/src/objc_video_decoder_factory.h"
#include "sdk/objc/native/src/objc_video_encoder_factory.h"
#endif
#if defined(WEBRTC_IOS)
#import "sdk/objc/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 "api/transport/media/media_transport_interface.h"
#include "media/engine/webrtc_media_engine.h" // nogncheck
@implementation RTC_OBJC_TYPE (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([[RTC_OBJC_TYPE(
RTCVideoEncoderFactoryH264) alloc] init])
nativeVideoDecoderFactory:webrtc::ObjCToNativeVideoDecoderFactory([[RTC_OBJC_TYPE(
RTCVideoDecoderFactoryH264) alloc] init])
audioDeviceModule:[self audioDeviceModule]
audioProcessingModule:nullptr
mediaTransportFactory:nullptr];
#endif
}
- (instancetype)
initWithEncoderFactory:(nullable id<RTC_OBJC_TYPE(RTCVideoEncoderFactory)>)encoderFactory
decoderFactory:(nullable id<RTC_OBJC_TYPE(RTCVideoDecoderFactory)>)decoderFactory
mediaTransportFactory:(std::unique_ptr<webrtc::MediaTransportFactory>)mediaTransportFactory {
#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
mediaTransportFactory:std::move(mediaTransportFactory)];
#endif
}
- (instancetype)
initWithEncoderFactory:(nullable id<RTC_OBJC_TYPE(RTCVideoEncoderFactory)>)encoderFactory
decoderFactory:(nullable id<RTC_OBJC_TYPE(RTCVideoDecoderFactory)>)decoderFactory {
return [self initWithEncoderFactory:encoderFactory
decoderFactory:decoderFactory
mediaTransportFactory:nullptr];
}
- (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]) {
webrtc::PeerConnectionFactoryDependencies dependencies;
dependencies.network_thread = _networkThread.get();
dependencies.worker_thread = _workerThread.get();
dependencies.signaling_thread = _signalingThread.get();
_nativeFactory = webrtc::CreateModularPeerConnectionFactory(std::move(dependencies));
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:(webrtc::AudioDeviceModule *)audioDeviceModule
audioProcessingModule:
(rtc::scoped_refptr<webrtc::AudioProcessing>)audioProcessingModule {
return [self initWithNativeAudioEncoderFactory:audioEncoderFactory
nativeAudioDecoderFactory:audioDecoderFactory
nativeVideoEncoderFactory:std::move(videoEncoderFactory)
nativeVideoDecoderFactory:std::move(videoDecoderFactory)
audioDeviceModule:audioDeviceModule
audioProcessingModule:audioProcessingModule
mediaTransportFactory:nullptr];
}
- (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:(webrtc::AudioDeviceModule *)audioDeviceModule
audioProcessingModule:
(rtc::scoped_refptr<webrtc::AudioProcessing>)audioProcessingModule
mediaTransportFactory:(std::unique_ptr<webrtc::MediaTransportFactory>)
mediaTransportFactory {
return [self initWithNativeAudioEncoderFactory:audioEncoderFactory
nativeAudioDecoderFactory:audioDecoderFactory
nativeVideoEncoderFactory:std::move(videoEncoderFactory)
nativeVideoDecoderFactory:std::move(videoDecoderFactory)
audioDeviceModule:audioDeviceModule
audioProcessingModule:audioProcessingModule
networkControllerFactory:nullptr
mediaTransportFactory:std::move(mediaTransportFactory)];
}
- (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:(webrtc::AudioDeviceModule *)audioDeviceModule
audioProcessingModule:
(rtc::scoped_refptr<webrtc::AudioProcessing>)audioProcessingModule
networkControllerFactory:
(std::unique_ptr<webrtc::NetworkControllerFactoryInterface>)
networkControllerFactory
mediaTransportFactory:(std::unique_ptr<webrtc::MediaTransportFactory>)
mediaTransportFactory {
if (self = [self initNative]) {
webrtc::PeerConnectionFactoryDependencies dependencies;
dependencies.network_thread = _networkThread.get();
dependencies.worker_thread = _workerThread.get();
dependencies.signaling_thread = _signalingThread.get();
#ifndef HAVE_NO_MEDIA
dependencies.task_queue_factory = webrtc::CreateDefaultTaskQueueFactory();
cricket::MediaEngineDependencies media_deps;
media_deps.adm = std::move(audioDeviceModule);
media_deps.task_queue_factory = dependencies.task_queue_factory.get();
media_deps.audio_encoder_factory = std::move(audioEncoderFactory);
media_deps.audio_decoder_factory = std::move(audioDecoderFactory);
media_deps.video_encoder_factory = std::move(videoEncoderFactory);
media_deps.video_decoder_factory = std::move(videoDecoderFactory);
if (audioProcessingModule) {
media_deps.audio_processing = std::move(audioProcessingModule);
} else {
media_deps.audio_processing = webrtc::AudioProcessingBuilder().Create();
}
dependencies.media_engine = cricket::CreateMediaEngine(std::move(media_deps));
dependencies.call_factory = webrtc::CreateCallFactory();
dependencies.event_log_factory =
std::make_unique<webrtc::RtcEventLogFactory>(dependencies.task_queue_factory.get());
dependencies.network_controller_factory = std::move(networkControllerFactory);
dependencies.media_transport_factory = std::move(mediaTransportFactory);
#endif
_nativeFactory = webrtc::CreateModularPeerConnectionFactory(std::move(dependencies));
NSAssert(_nativeFactory, @"Failed to initialize PeerConnectionFactory!");
}
return self;
}
- (RTC_OBJC_TYPE(RTCAudioSource) *)audioSourceWithConstraints:
(nullable RTC_OBJC_TYPE(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 [[RTC_OBJC_TYPE(RTCAudioSource) alloc] initWithFactory:self nativeAudioSource:source];
}
- (RTC_OBJC_TYPE(RTCAudioTrack) *)audioTrackWithTrackId:(NSString *)trackId {
RTC_OBJC_TYPE(RTCAudioSource) *audioSource = [self audioSourceWithConstraints:nil];
return [self audioTrackWithSource:audioSource trackId:trackId];
}
- (RTC_OBJC_TYPE(RTCAudioTrack) *)audioTrackWithSource:(RTC_OBJC_TYPE(RTCAudioSource) *)source
trackId:(NSString *)trackId {
return [[RTC_OBJC_TYPE(RTCAudioTrack) alloc] initWithFactory:self source:source trackId:trackId];
}
- (RTC_OBJC_TYPE(RTCVideoSource) *)videoSource {
return [[RTC_OBJC_TYPE(RTCVideoSource) alloc] initWithFactory:self
signalingThread:_signalingThread.get()
workerThread:_workerThread.get()];
}
- (RTC_OBJC_TYPE(RTCVideoTrack) *)videoTrackWithSource:(RTC_OBJC_TYPE(RTCVideoSource) *)source
trackId:(NSString *)trackId {
return [[RTC_OBJC_TYPE(RTCVideoTrack) alloc] initWithFactory:self source:source trackId:trackId];
}
- (RTC_OBJC_TYPE(RTCMediaStream) *)mediaStreamWithStreamId:(NSString *)streamId {
return [[RTC_OBJC_TYPE(RTCMediaStream) alloc] initWithFactory:self streamId:streamId];
}
- (RTC_OBJC_TYPE(RTCPeerConnection) *)
peerConnectionWithConfiguration:(RTC_OBJC_TYPE(RTCConfiguration) *)configuration
constraints:(RTC_OBJC_TYPE(RTCMediaConstraints) *)constraints
delegate:
(nullable id<RTC_OBJC_TYPE(RTCPeerConnectionDelegate)>)delegate {
return [[RTC_OBJC_TYPE(RTCPeerConnection) alloc] initWithFactory:self
configuration:configuration
constraints:constraints
delegate:delegate];
}
- (RTC_OBJC_TYPE(RTCPeerConnection) *)
peerConnectionWithDependencies:(RTC_OBJC_TYPE(RTCConfiguration) *)configuration
constraints:(RTC_OBJC_TYPE(RTCMediaConstraints) *)constraints
dependencies:(std::unique_ptr<webrtc::PeerConnectionDependencies>)dependencies
delegate:(id<RTC_OBJC_TYPE(RTCPeerConnectionDelegate)>)delegate {
return [[RTC_OBJC_TYPE(RTCPeerConnection) alloc] initWithDependencies:self
configuration:configuration
constraints:constraints
dependencies:std::move(dependencies)
delegate:delegate];
}
- (void)setOptions:(nonnull RTC_OBJC_TYPE(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;
}
FILE *f = fopen(filePath.UTF8String, "wb");
if (!f) {
RTCLogError(@"Error opening file: %@. Error: %s", filePath, strerror(errno));
return NO;
}
_hasStartedAecDump = _nativeFactory->StartAecDump(f, maxSizeInBytes);
return _hasStartedAecDump;
}
- (void)stopAecDump {
_nativeFactory->StopAecDump();
_hasStartedAecDump = NO;
}
@end
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