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/*
* Copyright (C) 2012 Google Inc. All rights reserved.
*
* Redistribution and use in source and binary forms, with or without
* modification, are permitted provided that the following conditions are
* met:
*
* * Redistributions of source code must retain the above copyright
* notice, this list of conditions and the following disclaimer.
* * 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.
* * Neither the name of Google Inc. nor the names of its
* contributors may be used to endorse or promote products derived from
* this software without specific prior written permission.
*
* THIS SOFTWARE IS PROVIDED BY THE COPYRIGHT HOLDERS AND CONTRIBUTORS
* "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 COPYRIGHT
* OWNER OR CONTRIBUTORS 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 THIRD_PARTY_BLINK_PUBLIC_PLATFORM_WEB_RTC_PEER_CONNECTION_HANDLER_H_
#define THIRD_PARTY_BLINK_PUBLIC_PLATFORM_WEB_RTC_PEER_CONNECTION_HANDLER_H_
#include "third_party/blink/public/platform/web_rtc_configuration.h"
#include "third_party/blink/public/platform/web_rtc_ice_candidate.h"
#include "third_party/blink/public/platform/web_rtc_rtp_transceiver.h"
#include "third_party/blink/public/platform/web_rtc_stats.h"
#include "third_party/blink/public/platform/web_vector.h"
#include "third_party/webrtc/api/rtcerror.h"
#include "third_party/webrtc/api/rtptransceiverinterface.h"
namespace webrtc {
enum class RTCErrorType;
}
namespace blink {
class WebMediaConstraints;
class WebMediaStream;
class WebMediaStreamTrack;
class WebRTCAnswerOptions;
class WebRTCDataChannelHandler;
class WebRTCOfferOptions;
class WebRTCRtpSender;
class WebRTCSessionDescription;
class WebRTCSessionDescriptionRequest;
class WebRTCStatsRequest;
class WebRTCVoidRequest;
class WebString;
struct WebRTCConfiguration;
struct WebRTCDataChannelInit;
class WebRTCPeerConnectionHandler {
public:
virtual ~WebRTCPeerConnectionHandler() = default;
virtual bool Initialize(const WebRTCConfiguration&,
const WebMediaConstraints&,
WebRTCSdpSemantics original_sdp_semantics_value) = 0;
virtual void CreateOffer(const WebRTCSessionDescriptionRequest&,
const WebMediaConstraints&) = 0;
virtual void CreateOffer(const WebRTCSessionDescriptionRequest&,
const WebRTCOfferOptions&) = 0;
virtual void CreateAnswer(const WebRTCSessionDescriptionRequest&,
const WebMediaConstraints&) = 0;
virtual void CreateAnswer(const WebRTCSessionDescriptionRequest&,
const WebRTCAnswerOptions&) = 0;
virtual void SetLocalDescription(const WebRTCVoidRequest&,
const WebRTCSessionDescription&) = 0;
virtual void SetRemoteDescription(const WebRTCVoidRequest&,
const WebRTCSessionDescription&) = 0;
virtual WebRTCSessionDescription LocalDescription() = 0;
virtual WebRTCSessionDescription RemoteDescription() = 0;
virtual webrtc::RTCErrorType SetConfiguration(const WebRTCConfiguration&) = 0;
// DEPRECATED
virtual bool AddICECandidate(scoped_refptr<WebRTCICECandidate>) {
return false;
}
virtual bool AddICECandidate(const WebRTCVoidRequest&,
scoped_refptr<WebRTCICECandidate>) {
return false;
}
virtual void GetStats(const WebRTCStatsRequest&) = 0;
// Gets stats using the new stats collection API, see
// third_party/webrtc/api/stats/. These will replace the old stats collection
// API when the new API has matured enough.
virtual void GetStats(std::unique_ptr<WebRTCStatsReportCallback>) = 0;
virtual WebRTCDataChannelHandler* CreateDataChannel(
const WebString& label,
const WebRTCDataChannelInit&) = 0;
virtual webrtc::RTCErrorOr<std::unique_ptr<WebRTCRtpTransceiver>>
AddTransceiverWithTrack(const WebMediaStreamTrack&,
const webrtc::RtpTransceiverInit&) = 0;
virtual webrtc::RTCErrorOr<std::unique_ptr<WebRTCRtpTransceiver>>
AddTransceiverWithKind(
// webrtc::MediaStreamTrackInterface::kAudioKind or kVideoKind
std::string kind,
const webrtc::RtpTransceiverInit&) = 0;
// Adds the track to the peer connection, returning the resulting transceiver
// or error.
virtual webrtc::RTCErrorOr<std::unique_ptr<WebRTCRtpTransceiver>> AddTrack(
const WebMediaStreamTrack&,
const WebVector<WebMediaStream>&) = 0;
// Removes the sender.
// In Plan B: Returns OK() with value nullptr on success. The sender's track
// must be nulled by the caller.
// In Unified Plan: Returns OK() with the updated transceiver state.
virtual webrtc::RTCErrorOr<std::unique_ptr<WebRTCRtpTransceiver>> RemoveTrack(
WebRTCRtpSender*) = 0;
virtual void Stop() = 0;
// Origin Trial - RtcPeerConnectionId
virtual WebString Id() const = 0;
};
} // namespace blink
#endif // THIRD_PARTY_BLINK_PUBLIC_PLATFORM_WEB_RTC_PEER_CONNECTION_HANDLER_H_
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