Support building WebRTC without audio and video for IOS

webrtc/sdk/objc/Framework/Classes/PeerConnection/RTCPeerConnectionFactory+Media.mm

Index: webrtc/sdk/objc/Framework/Classes/PeerConnection/RTCPeerConnectionFactory+Media.mm
diff --git a/webrtc/sdk/objc/Framework/Classes/PeerConnection/RTCPeerConnectionFactory+Media.mm b/webrtc/sdk/objc/Framework/Classes/PeerConnection/RTCPeerConnectionFactory+Media.mm
new file mode 100644
index 0000000000000000000000000000000000000000..f0124c09a0ad0f8aeca5ac5805d0e7b6dd960932
--- /dev/null
+++ b/webrtc/sdk/objc/Framework/Classes/PeerConnection/RTCPeerConnectionFactory+Media.mm
@@ -0,0 +1,99 @@
+/*
+ *  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 "RTCPeerConnectionFactory+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"
+
+#include "Video/objcvideotracksource.h"
+#include "VideoToolbox/videocodecfactory.h"
+#include "webrtc/api/videosourceproxy.h"
+
+#if !defined(HAVE_RTC_AUDIO) && !defined(HAVE_RTC_VIDEO)
+#include "webrtc/media/engine/webrtcmediaengine.h"
+namespace webrtc {
+rtc::scoped_refptr<webrtc::PeerConnectionFactoryInterface> CreatePeerConnectionFactory(

[tkchin_webrtc, 2017/06/20 21:51:23]

why is this method declared only if !defined(HAVE_RTC_AUDIO) &&
!defined(HAVE_RTC_VIDEO)?

Shouldn't this also be an if / else block inside the init method?

+    rtc::Thread* network_thread,
+    rtc::Thread* worker_thread,
+    rtc::Thread* signaling_thread,
+    webrtc::AudioDeviceModule* default_adm,
+    cricket::WebRtcVideoEncoderFactory* video_encoder_factory,
+    cricket::WebRtcVideoDecoderFactory* video_decoder_factory) {
+  return CreateModularPeerConnectionFactory(network_thread,
+                                            worker_thread,
+                                            signaling_thread,
+                                            default_adm,
+                                            nullptr,
+                                            nullptr,
+                                            video_encoder_factory,
+                                            video_decoder_factory,
+                                            nullptr,
+                                            std::unique_ptr<cricket::MediaEngineInterface>(),
+                                            std::unique_ptr<webrtc::CallFactoryInterface>(),
+                                            std::unique_ptr<webrtc::RtcEventLogFactoryInterface>());
+}
+}  // namespace webrtc
+#endif
+
+@implementation RTCPeerConnectionFactory (Media)

[tkchin_webrtc, 2017/06/20 21:51:23]

same question: why does this need to be in its own file?

+
+- (instancetype)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.");
+
+    webrtc::VideoToolboxVideoEncoderFactory* encoder_factory = nullptr;
+    webrtc::VideoToolboxVideoDecoderFactory* decoder_factory = nullptr;
+
+#if defined(HAVE_RTC_VIDEO)
+    encoder_factory = new webrtc::VideoToolboxVideoEncoderFactory();
+    decoder_factory = new webrtc::VideoToolboxVideoDecoderFactory();
+#endif
+
+    // Ownership of encoder/decoder factories is passed on to the
+    // peerconnectionfactory, that handles deleting them.
+    _nativeFactory = webrtc::CreatePeerConnectionFactory(_networkThread.get(),
+                                                         _workerThread.get(),
+                                                         _signalingThread.get(),
+                                                         nullptr,
+                                                         encoder_factory,
+                                                         decoder_factory);
+    NSAssert(_nativeFactory, @"Failed to initialize PeerConnectionFactory!");
+  }
+  return self;
+}
+
+- (RTCMediaStream*)mediaStreamWithStreamId:(NSString*)streamId {
+#if defined(HAVE_RTC_AUDIO) || defined(HAVE_RTC_VIDEO)
+  return [[RTCMediaStream alloc] initWithFactory:self streamId:streamId];
+#else
+  return nil;
+#endif
+}
+
+@end