1 | /* |
2 | * Copyright (C) 2018 Apple Inc. |
3 | * |
4 | * Redistribution and use in source and binary forms, with or without |
5 | * modification, are permitted provided that the following conditions |
6 | * are met: |
7 | * 1. Redistributions of source code must retain the above copyright |
8 | * notice, this list of conditions and the following disclaimer. |
9 | * 2. Redistributions in binary form must reproduce the above copyright |
10 | * notice, this list of conditions and the following disclaimer in the |
11 | * documentation and/or other materials provided with the distribution. |
12 | * |
13 | * THIS SOFTWARE IS PROVIDED BY APPLE INC. AND ITS CONTRIBUTORS ``AS IS'' AND ANY |
14 | * EXPRESS OR IMPLIED WARRANTIES, INCLUDING, BUT NOT LIMITED TO, THE IMPLIED |
15 | * WARRANTIES OF MERCHANTABILITY AND FITNESS FOR A PARTICULAR PURPOSE ARE |
16 | * DISCLAIMED. IN NO EVENT SHALL APPLE INC. OR ITS CONTRIBUTORS BE LIABLE FOR ANY |
17 | * DIRECT, INDIRECT, INCIDENTAL, SPECIAL, EXEMPLARY, OR CONSEQUENTIAL DAMAGES |
18 | * (INCLUDING, BUT NOT LIMITED TO, PROCUREMENT OF SUBSTITUTE GOODS OR SERVICES; |
19 | * LOSS OF USE, DATA, OR PROFITS; OR BUSINESS INTERRUPTION) HOWEVER CAUSED AND ON |
20 | * ANY THEORY OF LIABILITY, WHETHER IN CONTRACT, STRICT LIABILITY, OR TORT |
21 | * (INCLUDING NEGLIGENCE OR OTHERWISE) ARISING IN ANY WAY OUT OF THE USE OF THIS |
22 | * SOFTWARE, EVEN IF ADVISED OF THE POSSIBILITY OF SUCH DAMAGE. |
23 | */ |
24 | |
25 | #pragma once |
26 | |
27 | #if ENABLE(WEB_RTC) |
28 | |
29 | #include "LibWebRTCMacros.h" |
30 | #include "LibWebRTCPeerConnectionBackend.h" |
31 | #include "RTCRtpSenderBackend.h" |
32 | #include "RealtimeOutgoingAudioSource.h" |
33 | #include "RealtimeOutgoingVideoSource.h" |
34 | #include <wtf/WeakPtr.h> |
35 | |
36 | ALLOW_UNUSED_PARAMETERS_BEGIN |
37 | |
38 | #include <webrtc/api/rtpsenderinterface.h> |
39 | #include <webrtc/rtc_base/scoped_ref_ptr.h> |
40 | |
41 | ALLOW_UNUSED_PARAMETERS_END |
42 | |
43 | namespace WebCore { |
44 | |
45 | class LibWebRTCPeerConnectionBackend; |
46 | |
47 | class LibWebRTCRtpSenderBackend final : public RTCRtpSenderBackend { |
48 | WTF_MAKE_FAST_ALLOCATED; |
49 | public: |
50 | LibWebRTCRtpSenderBackend(LibWebRTCPeerConnectionBackend& backend, rtc::scoped_refptr<webrtc::RtpSenderInterface>&& rtcSender) |
51 | : m_peerConnectionBackend(makeWeakPtr(&backend)) |
52 | , m_rtcSender(WTFMove(rtcSender)) |
53 | { |
54 | } |
55 | |
56 | using Source = Variant<std::nullptr_t, Ref<RealtimeOutgoingAudioSource>, Ref<RealtimeOutgoingVideoSource>>; |
57 | LibWebRTCRtpSenderBackend(LibWebRTCPeerConnectionBackend& backend, rtc::scoped_refptr<webrtc::RtpSenderInterface>&& rtcSender, Source&& source) |
58 | : m_peerConnectionBackend(makeWeakPtr(&backend)) |
59 | , m_rtcSender(WTFMove(rtcSender)) |
60 | , m_source(WTFMove(source)) |
61 | { |
62 | } |
63 | |
64 | void setRTCSender(rtc::scoped_refptr<webrtc::RtpSenderInterface>&& rtcSender) { m_rtcSender = WTFMove(rtcSender); } |
65 | webrtc::RtpSenderInterface* rtcSender() { return m_rtcSender.get(); } |
66 | |
67 | RealtimeOutgoingAudioSource* audioSource() |
68 | { |
69 | return WTF::switchOn(m_source, |
70 | [] (Ref<RealtimeOutgoingAudioSource>& source) { return source.ptr(); }, |
71 | [] (const auto&) -> RealtimeOutgoingAudioSource* { return nullptr; } |
72 | ); |
73 | } |
74 | |
75 | RealtimeOutgoingVideoSource* videoSource() |
76 | { |
77 | return WTF::switchOn(m_source, |
78 | [] (Ref<RealtimeOutgoingVideoSource>& source) { return source.ptr(); }, |
79 | [] (const auto&) -> RealtimeOutgoingVideoSource* { return nullptr; } |
80 | ); |
81 | } |
82 | |
83 | bool hasSource() const |
84 | { |
85 | return WTF::switchOn(m_source, |
86 | [] (const std::nullptr_t&) { return false; }, |
87 | [] (const auto&) { return true; } |
88 | ); |
89 | } |
90 | |
91 | void clearSource() |
92 | { |
93 | ASSERT(hasSource()); |
94 | m_source = nullptr; |
95 | } |
96 | |
97 | void setSource(Source&& source) |
98 | { |
99 | ASSERT(!hasSource()); |
100 | m_source = WTFMove(source); |
101 | ASSERT(hasSource()); |
102 | } |
103 | |
104 | void takeSource(LibWebRTCRtpSenderBackend& backend) |
105 | { |
106 | ASSERT(backend.hasSource()); |
107 | setSource(WTFMove(backend.m_source)); |
108 | } |
109 | |
110 | private: |
111 | void replaceTrack(ScriptExecutionContext&, RTCRtpSender&, RefPtr<MediaStreamTrack>&&, DOMPromiseDeferred<void>&&) final; |
112 | RTCRtpSendParameters getParameters() const final; |
113 | void setParameters(const RTCRtpSendParameters&, DOMPromiseDeferred<void>&&) final; |
114 | |
115 | WeakPtr<LibWebRTCPeerConnectionBackend> m_peerConnectionBackend; |
116 | rtc::scoped_refptr<webrtc::RtpSenderInterface> m_rtcSender; |
117 | Source m_source; |
118 | mutable Optional<webrtc::RtpParameters> m_currentParameters; |
119 | }; |
120 | |
121 | } // namespace WebCore |
122 | |
123 | #endif // ENABLE(WEB_RTC) |
124 | |