File: LibWebRTCMediaEndpoint.h

package info (click to toggle)
webkit2gtk 2.48.5-1
  • links: PTS, VCS
  • area: main
  • in suites: forky, sid
  • size: 429,764 kB
  • sloc: cpp: 3,697,587; javascript: 194,444; ansic: 169,997; python: 46,499; asm: 19,295; ruby: 18,528; perl: 16,602; xml: 4,650; yacc: 2,360; sh: 2,098; java: 1,993; lex: 1,327; pascal: 366; makefile: 298
file content (214 lines) | stat: -rw-r--r-- 9,404 bytes parent folder | download | duplicates (7)
1
2
3
4
5
6
7
8
9
10
11
12
13
14
15
16
17
18
19
20
21
22
23
24
25
26
27
28
29
30
31
32
33
34
35
36
37
38
39
40
41
42
43
44
45
46
47
48
49
50
51
52
53
54
55
56
57
58
59
60
61
62
63
64
65
66
67
68
69
70
71
72
73
74
75
76
77
78
79
80
81
82
83
84
85
86
87
88
89
90
91
92
93
94
95
96
97
98
99
100
101
102
103
104
105
106
107
108
109
110
111
112
113
114
115
116
117
118
119
120
121
122
123
124
125
126
127
128
129
130
131
132
133
134
135
136
137
138
139
140
141
142
143
144
145
146
147
148
149
150
151
152
153
154
155
156
157
158
159
160
161
162
163
164
165
166
167
168
169
170
171
172
173
174
175
176
177
178
179
180
181
182
183
184
185
186
187
188
189
190
191
192
193
194
195
196
197
198
199
200
201
202
203
204
205
206
207
208
209
210
211
212
213
214
/*
 * Copyright (C) 2017-2018 Apple 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:
 * 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.
 *
 * THIS SOFTWARE IS PROVIDED BY APPLE INC. AND ITS 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 APPLE INC. OR ITS 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.
 */

#pragma once

#if ENABLE(WEB_RTC) && USE(LIBWEBRTC)

#include "LibWebRTCObservers.h"
#include "LibWebRTCProvider.h"
#include "LibWebRTCRtpSenderBackend.h"
#include "RTCRtpReceiver.h"
#include "Timer.h"
#include <wtf/WeakRef.h>

WTF_IGNORE_WARNINGS_IN_THIRD_PARTY_CODE_BEGIN

#include <webrtc/api/jsep.h>
#include <webrtc/api/peer_connection_interface.h>
// See Bug 274508: Disable thread-safety-reference-return warnings in libwebrtc
IGNORE_CLANG_WARNINGS_BEGIN("thread-safety-reference-return")
#include <webrtc/pc/peer_connection_factory.h>
#include <webrtc/pc/rtc_stats_collector.h>
IGNORE_CLANG_WARNINGS_END

WTF_IGNORE_WARNINGS_IN_THIRD_PARTY_CODE_END

#include <wtf/LoggerHelper.h>
#include <wtf/RobinHoodHashMap.h>
#include <wtf/ThreadSafeRefCounted.h>

namespace webrtc {
class CreateSessionDescriptionObserver;
class DataChannelInterface;
class IceCandidateInterface;
class MediaStreamInterface;
class PeerConnectionObserver;
class SessionDescriptionInterface;
class SetSessionDescriptionObserver;
}

namespace WebCore {
class LibWebRTCProvider;
class LibWebRTCPeerConnectionBackend;
class LibWebRTCRtpReceiverBackend;
class LibWebRTCRtpTransceiverBackend;
class LibWebRTCStatsCollector;
class MediaStreamTrack;
class RTCSessionDescription;

class LibWebRTCMediaEndpoint
    : public ThreadSafeRefCounted<LibWebRTCMediaEndpoint, WTF::DestructionThread::Main>
    , private webrtc::PeerConnectionObserver
    , private webrtc::RTCStatsCollectorCallback
#if !RELEASE_LOG_DISABLED
    , private LoggerHelper
#endif
{
public:
    static Ref<LibWebRTCMediaEndpoint> create(LibWebRTCPeerConnectionBackend& peerConnection, LibWebRTCProvider& client) { return adoptRef(*new LibWebRTCMediaEndpoint(peerConnection, client)); }
    virtual ~LibWebRTCMediaEndpoint() = default;

    void restartIce();
    bool setConfiguration(LibWebRTCProvider&, webrtc::PeerConnectionInterface::RTCConfiguration&&);

    webrtc::PeerConnectionInterface& backend() const { ASSERT(m_backend); return *m_backend.get(); }
    void doSetLocalDescription(const RTCSessionDescription*);
    void doSetRemoteDescription(const RTCSessionDescription&);
    void doCreateOffer(const RTCOfferOptions&);
    void doCreateAnswer();
    void gatherDecoderImplementationName(Function<void(String&&)>&&);
    void getStats(Ref<DeferredPromise>&&);
    void getStats(webrtc::RtpReceiverInterface&, Ref<DeferredPromise>&&);
    void getStats(webrtc::RtpSenderInterface&, Ref<DeferredPromise>&&);
    std::unique_ptr<RTCDataChannelHandler> createDataChannel(const String&, const RTCDataChannelInit&);
    void addIceCandidate(std::unique_ptr<webrtc::IceCandidateInterface>&&, PeerConnectionBackend::AddIceCandidateCallback&&);

    void close();
    void stop();
    bool isStopped() const { return !m_backend; }

    bool addTrack(LibWebRTCRtpSenderBackend&, MediaStreamTrack&, const FixedVector<String>&);
    void removeTrack(LibWebRTCRtpSenderBackend&);

    struct Backends {
        std::unique_ptr<LibWebRTCRtpSenderBackend> senderBackend;
        std::unique_ptr<LibWebRTCRtpReceiverBackend> receiverBackend;
        std::unique_ptr<LibWebRTCRtpTransceiverBackend> transceiverBackend;
    };
    ExceptionOr<Backends> addTransceiver(const String& trackKind, const RTCRtpTransceiverInit&, PeerConnectionBackend::IgnoreNegotiationNeededFlag);
    ExceptionOr<Backends> addTransceiver(MediaStreamTrack&, const RTCRtpTransceiverInit&, PeerConnectionBackend::IgnoreNegotiationNeededFlag);
    std::unique_ptr<LibWebRTCRtpTransceiverBackend> transceiverBackendFromSender(LibWebRTCRtpSenderBackend&);

    void setSenderSourceFromTrack(LibWebRTCRtpSenderBackend&, MediaStreamTrack&);
    void collectTransceivers();

    std::optional<bool> canTrickleIceCandidates() const;

    void suspend();
    void resume();
    LibWebRTCProvider::SuspendableSocketFactory* rtcSocketFactory() { return m_rtcSocketFactory.get(); }

    bool isNegotiationNeeded(uint32_t) const;

    void startRTCLogs();
    void stopRTCLogs();

private:
    LibWebRTCMediaEndpoint(LibWebRTCPeerConnectionBackend&, LibWebRTCProvider&);

    // webrtc::PeerConnectionObserver API
    void OnSignalingChange(webrtc::PeerConnectionInterface::SignalingState) final;
    void OnDataChannel(rtc::scoped_refptr<webrtc::DataChannelInterface>) final;

    void OnNegotiationNeededEvent(uint32_t) final;
    void OnStandardizedIceConnectionChange(webrtc::PeerConnectionInterface::IceConnectionState) final;
    void OnIceGatheringChange(webrtc::PeerConnectionInterface::IceGatheringState) final;
    void OnIceCandidate(const webrtc::IceCandidateInterface*) final;
    void OnIceCandidatesRemoved(const std::vector<cricket::Candidate>&) final;

    void createSessionDescriptionSucceeded(std::unique_ptr<webrtc::SessionDescriptionInterface>&&);
    void createSessionDescriptionFailed(ExceptionCode, const char*);
    void setLocalSessionDescriptionSucceeded();
    void setLocalSessionDescriptionFailed(ExceptionCode, const char*);
    void setRemoteSessionDescriptionSucceeded();
    void setRemoteSessionDescriptionFailed(ExceptionCode, const char*);

    template<typename T>
    ExceptionOr<Backends> createTransceiverBackends(T&&, webrtc::RtpTransceiverInit&&, LibWebRTCRtpSenderBackend::Source&&, PeerConnectionBackend::IgnoreNegotiationNeededFlag);

    void OnStatsDelivered(const rtc::scoped_refptr<const webrtc::RTCStatsReport>&) final;
    void gatherStatsForLogging();
    void startLoggingStats();
    void stopLoggingStats();

    rtc::scoped_refptr<LibWebRTCStatsCollector> createStatsCollector(Ref<DeferredPromise>&&);

    MediaStream& mediaStreamFromRTCStreamId(const String&);

    void AddRef() const { ref(); }
    webrtc::RefCountReleaseStatus Release() const
    {
        auto result = refCount() - 1;
        deref();
        return result ? webrtc::RefCountReleaseStatus::kOtherRefsRemained : webrtc::RefCountReleaseStatus::kDroppedLastRef;
    }

    std::pair<LibWebRTCRtpSenderBackend::Source, rtc::scoped_refptr<webrtc::MediaStreamTrackInterface>> createSourceAndRTCTrack(MediaStreamTrack&);
    RefPtr<RealtimeMediaSource> sourceFromNewReceiver(webrtc::RtpReceiverInterface&);

#if !RELEASE_LOG_DISABLED
    const Logger& logger() const final { return m_logger.get(); }
    uint64_t logIdentifier() const final { return m_logIdentifier; }
    ASCIILiteral logClassName() const final { return "LibWebRTCMediaEndpoint"_s; }
    WTFLogChannel& logChannel() const final;

    Seconds statsLogInterval(int64_t) const;
#endif

    Ref<LibWebRTCPeerConnectionBackend> protectedPeerConnectionBackend() const;

    WeakRef<LibWebRTCPeerConnectionBackend> m_peerConnectionBackend;
    rtc::scoped_refptr<webrtc::PeerConnectionFactoryInterface> m_peerConnectionFactory;
    rtc::scoped_refptr<webrtc::PeerConnectionInterface> m_backend;

    friend CreateSessionDescriptionObserver<LibWebRTCMediaEndpoint>;
    friend SetLocalSessionDescriptionObserver<LibWebRTCMediaEndpoint>;
    friend SetRemoteSessionDescriptionObserver<LibWebRTCMediaEndpoint>;

    CreateSessionDescriptionObserver<LibWebRTCMediaEndpoint> m_createSessionDescriptionObserver;
    SetLocalSessionDescriptionObserver<LibWebRTCMediaEndpoint> m_setLocalSessionDescriptionObserver;
    SetRemoteSessionDescriptionObserver<LibWebRTCMediaEndpoint> m_setRemoteSessionDescriptionObserver;

    MemoryCompactRobinHoodHashMap<String, RefPtr<MediaStream>> m_remoteStreamsById;
    HashMap<MediaStreamTrack*, Vector<String>> m_remoteStreamsFromRemoteTrack;

    bool m_isInitiator { false };
    Timer m_statsLogTimer;

    MemoryCompactRobinHoodHashMap<String, rtc::scoped_refptr<webrtc::MediaStreamInterface>> m_localStreams;

    std::unique_ptr<LibWebRTCProvider::SuspendableSocketFactory> m_rtcSocketFactory;
#if !RELEASE_LOG_DISABLED
    int64_t m_statsFirstDeliveredTimestamp { 0 };
    Ref<const Logger> m_logger;
    const uint64_t m_logIdentifier;
#endif
    bool m_isGatheringRTCLogs { false };
    bool m_shouldIgnoreNegotiationNeededSignal { false };
};

} // namespace WebCore

#endif // ENABLE(WEB_RTC) && USE(LIBWEBRTC)