webrtc_m130/webrtc/api/peerconnectionfactory.h
Henrik Kjellander 15583c19d7 Move talk/app/webrtc to webrtc/api
The previously disabled warnings that were inherited from
talk/build/common.gypi are now replaced by target-specific disabling
of only the failing warnings. Additional disabling was needed since the stricter
compilation warnings that applies to code in webrtc/.

License headers will be updated in a follow-up CL.

Other modifications:
* Updated the header guards.
* Sorted the includes using chromium/src/tools/sort-headers.py
  except for these files:
  talk/app/webrtc/peerconnectionendtoend_unittest.cc
  talk/app/webrtc/java/jni/androidmediadecoder_jni.cc
  talk/app/webrtc/java/jni/androidmediaencoder_jni.cc
  webrtc/media/devices/win32devicemanager.cc

The HAVE_SCTP define was added for the peerconnection_unittests target
in api_tests.gyp.

I also checked that none of
SRTP_RELATIVE_PATH
HAVE_SRTP
HAVE_WEBRTC_VIDEO
HAVE_WEBRTC_VOICE
were used by the talk/app/webrtc code.

For Chromium, the following changes will need to be applied to the roll CL that updates the
DEPS for WebRTC and libjingle:
https://codereview.chromium.org/1615433002

BUG=webrtc:5418
NOPRESUBMIT=True
R=deadbeef@webrtc.org, pthatcher@webrtc.org, tommi@webrtc.org

Review URL: https://codereview.webrtc.org/1610243002 .

Cr-Commit-Position: refs/heads/master@{#11545}
2016-02-10 09:53:26 +00:00

133 lines
5.1 KiB
C++

/*
* libjingle
* Copyright 2011 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.
*/
#ifndef WEBRTC_API_PEERCONNECTIONFACTORY_H_
#define WEBRTC_API_PEERCONNECTIONFACTORY_H_
#include <string>
#include "talk/session/media/channelmanager.h"
#include "webrtc/api/dtlsidentitystore.h"
#include "webrtc/api/mediacontroller.h"
#include "webrtc/api/mediastreaminterface.h"
#include "webrtc/api/peerconnectioninterface.h"
#include "webrtc/base/scoped_ptr.h"
#include "webrtc/base/scoped_ref_ptr.h"
#include "webrtc/base/thread.h"
namespace rtc {
class BasicNetworkManager;
class BasicPacketSocketFactory;
}
namespace webrtc {
typedef rtc::RefCountedObject<DtlsIdentityStoreImpl>
RefCountedDtlsIdentityStore;
class PeerConnectionFactory : public PeerConnectionFactoryInterface {
public:
virtual void SetOptions(const Options& options) {
options_ = options;
}
rtc::scoped_refptr<PeerConnectionInterface> CreatePeerConnection(
const PeerConnectionInterface::RTCConfiguration& configuration,
const MediaConstraintsInterface* constraints,
rtc::scoped_ptr<cricket::PortAllocator> allocator,
rtc::scoped_ptr<DtlsIdentityStoreInterface> dtls_identity_store,
PeerConnectionObserver* observer) override;
bool Initialize();
rtc::scoped_refptr<MediaStreamInterface>
CreateLocalMediaStream(const std::string& label) override;
rtc::scoped_refptr<AudioSourceInterface> CreateAudioSource(
const MediaConstraintsInterface* constraints) override;
rtc::scoped_refptr<VideoSourceInterface> CreateVideoSource(
cricket::VideoCapturer* capturer,
const MediaConstraintsInterface* constraints) override;
rtc::scoped_refptr<VideoTrackInterface>
CreateVideoTrack(const std::string& id,
VideoSourceInterface* video_source) override;
rtc::scoped_refptr<AudioTrackInterface>
CreateAudioTrack(const std::string& id,
AudioSourceInterface* audio_source) override;
bool StartAecDump(rtc::PlatformFile file, int64_t max_size_bytes) override;
void StopAecDump() override;
bool StartRtcEventLog(rtc::PlatformFile file) override;
void StopRtcEventLog() override;
virtual webrtc::MediaControllerInterface* CreateMediaController() const;
virtual rtc::Thread* signaling_thread();
virtual rtc::Thread* worker_thread();
const Options& options() const { return options_; }
protected:
PeerConnectionFactory();
PeerConnectionFactory(
rtc::Thread* worker_thread,
rtc::Thread* signaling_thread,
AudioDeviceModule* default_adm,
cricket::WebRtcVideoEncoderFactory* video_encoder_factory,
cricket::WebRtcVideoDecoderFactory* video_decoder_factory);
virtual ~PeerConnectionFactory();
private:
cricket::MediaEngineInterface* CreateMediaEngine_w();
bool owns_ptrs_;
bool wraps_current_thread_;
rtc::Thread* signaling_thread_;
rtc::Thread* worker_thread_;
Options options_;
// External Audio device used for audio playback.
rtc::scoped_refptr<AudioDeviceModule> default_adm_;
rtc::scoped_ptr<cricket::ChannelManager> channel_manager_;
// External Video encoder factory. This can be NULL if the client has not
// injected any. In that case, video engine will use the internal SW encoder.
rtc::scoped_ptr<cricket::WebRtcVideoEncoderFactory>
video_encoder_factory_;
// External Video decoder factory. This can be NULL if the client has not
// injected any. In that case, video engine will use the internal SW decoder.
rtc::scoped_ptr<cricket::WebRtcVideoDecoderFactory>
video_decoder_factory_;
rtc::scoped_ptr<rtc::BasicNetworkManager> default_network_manager_;
rtc::scoped_ptr<rtc::BasicPacketSocketFactory> default_socket_factory_;
rtc::scoped_refptr<RefCountedDtlsIdentityStore> dtls_identity_store_;
};
} // namespace webrtc
#endif // WEBRTC_API_PEERCONNECTIONFACTORY_H_