webrtc_m130/talk/app/webrtc/objc/RTCPeerConnectionObserver.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

76 lines
3.0 KiB
Objective-C

/*
* 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.
*/
#include "webrtc/api/peerconnectioninterface.h"
#import "RTCPeerConnection.h"
#import "RTCPeerConnectionDelegate.h"
// These objects are created by RTCPeerConnectionFactory to wrap an
// id<RTCPeerConnectionDelegate> and call methods on that interface.
namespace webrtc {
class RTCPeerConnectionObserver : public PeerConnectionObserver {
public:
RTCPeerConnectionObserver(RTCPeerConnection* peerConnection);
virtual ~RTCPeerConnectionObserver();
// Triggered when the SignalingState changed.
void OnSignalingChange(
PeerConnectionInterface::SignalingState new_state) override;
// Triggered when media is received on a new stream from remote peer.
void OnAddStream(MediaStreamInterface* stream) override;
// Triggered when a remote peer close a stream.
void OnRemoveStream(MediaStreamInterface* stream) override;
// Triggered when a remote peer open a data channel.
void OnDataChannel(DataChannelInterface* data_channel) override;
// Triggered when renegotiation is needed, for example the ICE has restarted.
void OnRenegotiationNeeded() override;
// Called any time the ICEConnectionState changes
void OnIceConnectionChange(
PeerConnectionInterface::IceConnectionState new_state) override;
// Called any time the ICEGatheringState changes
void OnIceGatheringChange(
PeerConnectionInterface::IceGatheringState new_state) override;
// New Ice candidate have been found.
void OnIceCandidate(const IceCandidateInterface* candidate) override;
private:
__weak RTCPeerConnection* _peerConnection;
};
} // namespace webrtc