Starting with patch 34 from https://codereview.webrtc.org/2854123003/

webrtc/api/peerconnectioninterface.h

 /*
  *  Copyright 2012 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.
  */
 
@@ skipping 72 matching lines @@
 #include "webrtc/api/rtpsenderinterface.h"
 #include "webrtc/api/stats/rtcstatscollectorcallback.h"
 #include "webrtc/api/statstypes.h"
 #include "webrtc/api/umametrics.h"
 #include "webrtc/base/fileutils.h"
 #include "webrtc/base/network.h"
 #include "webrtc/base/rtccertificate.h"
 #include "webrtc/base/rtccertificategenerator.h"
 #include "webrtc/base/socketaddress.h"
 #include "webrtc/base/sslstreamadapter.h"
+#include "webrtc/call/callfactoryinterface.h"
+#include "webrtc/logging/rtc_event_log/rtc_event_log_factory_interface.h"
 #include "webrtc/media/base/mediachannel.h"
 #include "webrtc/media/base/videocapturer.h"
 #include "webrtc/p2p/base/portallocator.h"
 
 namespace rtc {
 class SSLIdentity;
 class Thread;
 }
 
 namespace cricket {
+class MediaEngineInterface;
 class WebRtcVideoDecoderFactory;
 class WebRtcVideoEncoderFactory;
 }
 
 namespace webrtc {
 class AudioDeviceModule;
 class AudioMixer;
+class CallFactoryInterface;
 class MediaConstraintsInterface;
 
 // MediaStream container interface.
 class StreamCollectionInterface : public rtc::RefCountInterface {
  public:
   // TODO(ronghuawu): Update the function names to c++ style, e.g. find -> Find.
   virtual size_t count() = 0;
   virtual MediaStreamInterface* at(size_t index) = 0;
   virtual MediaStreamInterface* find(const std::string& label) = 0;
   virtual MediaStreamTrackInterface* FindAudioTrack(
@@ skipping 905 matching lines @@
 //
 // As such, if the current thread is not already running an rtc::Thread message
 // loop, an application using this method must eventually either call
 // rtc::Thread::Current()->Run(), or call
 // rtc::Thread::Current()->ProcessMessages() within the application's own
 // message loop.
 rtc::scoped_refptr<PeerConnectionFactoryInterface> CreatePeerConnectionFactory(
     rtc::scoped_refptr<AudioEncoderFactory> audio_encoder_factory,
     rtc::scoped_refptr<AudioDecoderFactory> audio_decoder_factory);
 
-// Deprecated variant of the above.
+// Deprecated. Use the CreateModularPeerConnectionFactory instead.
 // TODO(kwiberg): Remove.
 rtc::scoped_refptr<PeerConnectionFactoryInterface>
 CreatePeerConnectionFactory();
 
+// Deprecated. Use the CreateModularPeerConnectionFactory instead.
+//
 // Create a new instance of PeerConnectionFactoryInterface.
 //
 // |network_thread|, |worker_thread| and |signaling_thread| are
 // the only mandatory parameters.
 //
 // If non-null, a reference is added to |default_adm|, and ownership of
 // |video_encoder_factory| and |video_decoder_factory| is transferred to the
 // returned factory.
 // TODO(deadbeef): Use rtc::scoped_refptr<> and std::unique_ptr<> to make this
 // ownership transfer and ref counting more obvious.
@@ skipping 69 matching lines @@
     rtc::Thread* worker_and_network_thread,
     rtc::Thread* signaling_thread,
     AudioDeviceModule* default_adm,
     cricket::WebRtcVideoEncoderFactory* encoder_factory,
     cricket::WebRtcVideoDecoderFactory* decoder_factory) {
   return CreatePeerConnectionFactory(
       worker_and_network_thread, worker_and_network_thread, signaling_thread,
       default_adm, encoder_factory, decoder_factory);
 }
 
+// This is a lower-level version of the CreatePeerConnectionFactory functions
+// above. It's implemented in the "peerconnection" build target, whereas the
+// above methods are only implemented in the broader "libjingle_peerconnection"
+// build target, which pulls in the implementations of every module webrtc may
+// use.
+//
+// If an application knows it will only require certain modules, it can reduce
+// webrtc's impact on its binary size by depending only on the "peerconnection"
+// target and the modules the application requires, using
+// CreateModularPeerConnectionFactory instead of one of the
+// CreatePeerConnectionFactory methods above. For example, if an application
+// only uses WebRTC for audio, it can pass in null pointers for the
+// video-specific interfaces, and omit the corresponding modules from its
+// build.
+//
+// If |network_thread| or |worker_thread| are null, the PeerConnectionFactory
+// will create the necessary thread internally. If |signaling_thread| is null,
+// the PeerConnectionFactory will use the thread on which this method is called
+// as the signaling thread, wrapping it in an rtc::Thread object if needed.
+//
+// If non-null, a reference is added to |default_adm|, and ownership of
+// |video_encoder_factory| and |video_decoder_factory| is transferred to the
+// returned factory.
+//
+// TODO(deadbeef): Use rtc::scoped_refptr<> and std::unique_ptr<> to make this
+// ownership transfer and ref counting more obvious.
+//
+// TODO(deadbeef): Encapsulate these modules in a struct, so that when a new
+// module is inevitably exposed, we can just add a field to the struct instead
+// of adding a whole new CreateModularPeerConnectionFactory overload.
+rtc::scoped_refptr<PeerConnectionFactoryInterface>
+CreateModularPeerConnectionFactory(
+    rtc::Thread* network_thread,
+    rtc::Thread* worker_thread,
+    rtc::Thread* signaling_thread,
+    AudioDeviceModule* default_adm,
+    rtc::scoped_refptr<AudioEncoderFactory> audio_encoder_factory,
+    rtc::scoped_refptr<AudioDecoderFactory> audio_decoder_factory,
+    cricket::WebRtcVideoEncoderFactory* video_encoder_factory,
+    cricket::WebRtcVideoDecoderFactory* video_decoder_factory,
+    rtc::scoped_refptr<AudioMixer> audio_mixer,
+    std::unique_ptr<cricket::MediaEngineInterface> media_engine,
+    std::unique_ptr<CallFactoryInterface> call_factory,
+    std::unique_ptr<RtcEventLogFactoryInterface> event_log_factory);
+
 }  // namespace webrtc
 
 #endif  // WEBRTC_API_PEERCONNECTIONINTERFACE_H_