rhubarb-lip-sync/rhubarb/lib/webrtc-8d2248ff/talk/app/webrtc/objc/RTCPeerConnectionFactory.mm

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2016-06-21 20:13:05 +00:00
/*
* libjingle
* Copyright 2013 Google Inc.
*
* Redistribution and use in source and binary forms, with or without
* modification, are permitted provided that the following conditions are met:
*
* 1. Redistributions of source code must retain the above copyright notice,
* this list of conditions and the following disclaimer.
* 2. Redistributions in binary form must reproduce the above copyright notice,
* this list of conditions and the following disclaimer in the documentation
* and/or other materials provided with the distribution.
* 3. The name of the author may not be used to endorse or promote products
* derived from this software without specific prior written permission.
*
* THIS SOFTWARE IS PROVIDED BY THE AUTHOR ``AS IS'' AND ANY EXPRESS OR IMPLIED
* WARRANTIES, INCLUDING, BUT NOT LIMITED TO, THE IMPLIED WARRANTIES OF
* MERCHANTABILITY AND FITNESS FOR A PARTICULAR PURPOSE ARE DISCLAIMED. IN NO
* EVENT SHALL THE AUTHOR BE LIABLE FOR ANY DIRECT, INDIRECT, INCIDENTAL,
* SPECIAL, EXEMPLARY, OR CONSEQUENTIAL DAMAGES (INCLUDING, BUT NOT LIMITED TO,
* PROCUREMENT OF SUBSTITUTE GOODS OR SERVICES; LOSS OF USE, DATA, OR PROFITS;
* OR BUSINESS INTERRUPTION) HOWEVER CAUSED AND ON ANY THEORY OF LIABILITY,
* WHETHER IN CONTRACT, STRICT LIABILITY, OR TORT (INCLUDING NEGLIGENCE OR
* OTHERWISE) ARISING IN ANY WAY OUT OF THE USE OF THIS SOFTWARE, EVEN IF
* ADVISED OF THE POSSIBILITY OF SUCH DAMAGE.
*/
#if !defined(__has_feature) || !__has_feature(objc_arc)
#error "This file requires ARC support."
#endif
#import "RTCPeerConnectionFactory+Internal.h"
#include <memory>
#include <vector>
#import "RTCAudioTrack+Internal.h"
#import "RTCICEServer+Internal.h"
#import "RTCMediaConstraints+Internal.h"
#import "RTCMediaSource+Internal.h"
#import "RTCMediaStream+Internal.h"
#import "RTCMediaStreamTrack+Internal.h"
#import "RTCPeerConnection+Internal.h"
#import "RTCPeerConnectionDelegate.h"
#import "RTCPeerConnectionInterface+Internal.h"
#import "RTCVideoCapturer+Internal.h"
#import "RTCVideoSource+Internal.h"
#import "RTCVideoTrack+Internal.h"
#include "webrtc/api/audiotrack.h"
#include "webrtc/api/mediastreaminterface.h"
#include "webrtc/api/peerconnectioninterface.h"
#include "webrtc/api/videotrack.h"
#include "webrtc/base/logging.h"
#include "webrtc/base/ssladapter.h"
@implementation RTCPeerConnectionFactory {
std::unique_ptr<rtc::Thread> _networkThread;
std::unique_ptr<rtc::Thread> _workerThread;
std::unique_ptr<rtc::Thread> _signalingThread;
}
@synthesize nativeFactory = _nativeFactory;
+ (void)initializeSSL {
BOOL initialized = rtc::InitializeSSL();
NSAssert(initialized, @"Failed to initialize SSL library");
}
+ (void)deinitializeSSL {
BOOL deinitialized = rtc::CleanupSSL();
NSAssert(deinitialized, @"Failed to deinitialize SSL library");
}
- (id)init {
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.");
_nativeFactory = webrtc::CreatePeerConnectionFactory(
_networkThread.get(), _workerThread.get(), _signalingThread.get(),
nullptr, nullptr, nullptr);
NSAssert(_nativeFactory, @"Failed to initialize PeerConnectionFactory!");
// Uncomment to get sensitive logs emitted (to stderr or logcat).
// rtc::LogMessage::LogToDebug(rtc::LS_SENSITIVE);
}
return self;
}
- (RTCPeerConnection *)peerConnectionWithConfiguration:(RTCConfiguration *)configuration
constraints:(RTCMediaConstraints *)constraints
delegate:(id<RTCPeerConnectionDelegate>)delegate {
std::unique_ptr<webrtc::PeerConnectionInterface::RTCConfiguration> config(
[configuration createNativeConfiguration]);
if (!config) {
return nil;
}
return [[RTCPeerConnection alloc] initWithFactory:self.nativeFactory.get()
config:*config
constraints:constraints.constraints
delegate:delegate];
}
- (RTCPeerConnection*)
peerConnectionWithICEServers:(NSArray*)servers
constraints:(RTCMediaConstraints*)constraints
delegate:(id<RTCPeerConnectionDelegate>)delegate {
webrtc::PeerConnectionInterface::IceServers iceServers;
for (RTCICEServer* server in servers) {
iceServers.push_back(server.iceServer);
}
RTCPeerConnection* pc =
[[RTCPeerConnection alloc] initWithFactory:self.nativeFactory.get()
iceServers:iceServers
constraints:constraints.constraints];
pc.delegate = delegate;
return pc;
}
- (RTCMediaStream*)mediaStreamWithLabel:(NSString*)label {
rtc::scoped_refptr<webrtc::MediaStreamInterface> nativeMediaStream =
self.nativeFactory->CreateLocalMediaStream([label UTF8String]);
return [[RTCMediaStream alloc] initWithMediaStream:nativeMediaStream];
}
- (RTCVideoSource*)videoSourceWithCapturer:(RTCVideoCapturer*)capturer
constraints:(RTCMediaConstraints*)constraints {
if (!capturer) {
return nil;
}
rtc::scoped_refptr<webrtc::VideoTrackSourceInterface> source =
self.nativeFactory->CreateVideoSource([capturer takeNativeCapturer], constraints.constraints);
return [[RTCVideoSource alloc] initWithMediaSource:source];
}
- (RTCVideoTrack*)videoTrackWithID:(NSString*)videoId
source:(RTCVideoSource*)source {
rtc::scoped_refptr<webrtc::VideoTrackInterface> track =
self.nativeFactory->CreateVideoTrack([videoId UTF8String],
source.videoSource);
return [[RTCVideoTrack alloc] initWithMediaTrack:track];
}
- (RTCAudioTrack*)audioTrackWithID:(NSString*)audioId {
rtc::scoped_refptr<webrtc::AudioTrackInterface> track =
self.nativeFactory->CreateAudioTrack([audioId UTF8String], NULL);
return [[RTCAudioTrack alloc] initWithMediaTrack:track];
}
@end