Chromium Code Reviews| Index: webrtc/api/objc/RTCPeerConnection+Private.h |
| diff --git a/webrtc/api/objc/RTCPeerConnection+Private.h b/webrtc/api/objc/RTCPeerConnection+Private.h |
| new file mode 100644 |
| index 0000000000000000000000000000000000000000..c54d42cf0272cba76c90fa699c1bb993efc2b8d1 |
| --- /dev/null |
| +++ b/webrtc/api/objc/RTCPeerConnection+Private.h |
| @@ -0,0 +1,98 @@ |
| +/* |
| + * Copyright 2015 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 "RTCPeerConnection.h" |
| + |
| +#include "talk/app/webrtc/peerconnectioninterface.h" |
| + |
| +NS_ASSUME_NONNULL_BEGIN |
| + |
| +/** |
| + * These objects are created by RTCPeerConnectionFactory to wrap an |
| + * id<RTCPeerConnectionDelegate> and call methods on that interface. |
| + */ |
| +namespace webrtc { |
| + |
| +class PeerConnectionDelegateAdapter : public PeerConnectionObserver { |
| + |
| + public: |
| + PeerConnectionDelegateAdapter(RTCPeerConnection *peerConnection); |
| + virtual ~PeerConnectionDelegateAdapter(); |
| + |
| + /** Triggered when the SignalingState changed. */ |
|
tkchin_webrtc
2016/02/05 16:15:15
You can remove all the comments here because they'
hjon_webrtc
2016/02/09 00:59:53
Done.
|
| + 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 closes a stream. */ |
| + void OnRemoveStream(MediaStreamInterface *stream) override; |
| + |
| + /** Triggered when a remote peer opens a data channel. */ |
| + void OnDataChannel(DataChannelInterface *data_channel) override; |
| + |
| + /** Triggered when renegotiation is needed, for example 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 has been found. |
| + void OnIceCandidate(const IceCandidateInterface *candidate) override; |
| + |
| + private: |
| + __weak RTCPeerConnection *peer_connection_; |
| +}; |
| + |
| +} // namespace webrtc |
| + |
| + |
| +@interface RTCPeerConnection () |
| + |
| +/** The native PeerConnectionInterface created during construction. */ |
| +@property(nonatomic, readonly) |
| + rtc::scoped_refptr<webrtc::PeerConnectionInterface> nativePeerConnection; |
| + |
| ++ (webrtc::PeerConnectionInterface::SignalingState)nativeSignalingStateForState: |
| + (RTCSignalingState)state; |
| + |
| ++ (RTCSignalingState)signalingStateForNativeState: |
| + (webrtc::PeerConnectionInterface::SignalingState)nativeState; |
| + |
| ++ (NSString *)stringForSignalingState:(RTCSignalingState)state; |
| + |
| ++ (webrtc::PeerConnectionInterface::IceConnectionState) |
| + nativeIceConnectionStateForState:(RTCIceConnectionState)state; |
| + |
| ++ (RTCIceConnectionState)iceConnectionStateForNativeState: |
| + (webrtc::PeerConnectionInterface::IceConnectionState)nativeState; |
| + |
| ++ (NSString *)stringForIceConnectionState:(RTCIceConnectionState)state; |
| + |
| ++ (webrtc::PeerConnectionInterface::IceGatheringState) |
| + nativeIceGatheringStateForState:(RTCIceGatheringState)state; |
| + |
| ++ (RTCIceGatheringState)iceGatheringStateForNativeState: |
| + (webrtc::PeerConnectionInterface::IceGatheringState)nativeState; |
| + |
| ++ (NSString *)stringForIceGatheringState:(RTCIceGatheringState)state; |
| + |
| ++ (webrtc::PeerConnectionInterface::StatsOutputLevel) |
| + nativeStatsOutputLevelForLevel:(RTCStatsOutputLevel)level; |
| + |
| +@end |
| + |
| +NS_ASSUME_NONNULL_END |