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 | #include "config.h" |
26 | #include "LibWebRTCRtpReceiverBackend.h" |
27 | |
28 | #include "LibWebRTCUtils.h" |
29 | |
30 | #if ENABLE(WEB_RTC) && USE(LIBWEBRTC) |
31 | |
32 | namespace WebCore { |
33 | |
34 | RTCRtpParameters LibWebRTCRtpReceiverBackend::getParameters() |
35 | { |
36 | return toRTCRtpParameters(m_rtcReceiver->GetParameters()); |
37 | } |
38 | |
39 | static inline void fillRTCRtpContributingSource(RTCRtpContributingSource& source, const webrtc::RtpSource& rtcSource) |
40 | { |
41 | source.timestamp = rtcSource.timestamp_ms(); |
42 | source.source = rtcSource.source_id(); |
43 | if (rtcSource.audio_level()) |
44 | source.audioLevel = (*rtcSource.audio_level() == 127) ? 0 : pow(10, -*rtcSource.audio_level() / 20); |
45 | } |
46 | |
47 | static inline RTCRtpContributingSource toRTCRtpContributingSource(const webrtc::RtpSource& rtcSource) |
48 | { |
49 | RTCRtpContributingSource source; |
50 | fillRTCRtpContributingSource(source, rtcSource); |
51 | return source; |
52 | } |
53 | |
54 | static inline RTCRtpSynchronizationSource toRTCRtpSynchronizationSource(const webrtc::RtpSource& rtcSource) |
55 | { |
56 | RTCRtpSynchronizationSource source; |
57 | fillRTCRtpContributingSource(source, rtcSource); |
58 | return source; |
59 | } |
60 | |
61 | Vector<RTCRtpContributingSource> LibWebRTCRtpReceiverBackend::getContributingSources() const |
62 | { |
63 | Vector<RTCRtpContributingSource> sources; |
64 | for (auto& rtcSource : m_rtcReceiver->GetSources()) { |
65 | if (rtcSource.source_type() == webrtc::RtpSourceType::CSRC) |
66 | sources.append(toRTCRtpContributingSource(rtcSource)); |
67 | } |
68 | return sources; |
69 | } |
70 | |
71 | Vector<RTCRtpSynchronizationSource> LibWebRTCRtpReceiverBackend::getSynchronizationSources() const |
72 | { |
73 | Vector<RTCRtpSynchronizationSource> sources; |
74 | for (auto& rtcSource : m_rtcReceiver->GetSources()) { |
75 | if (rtcSource.source_type() == webrtc::RtpSourceType::SSRC) |
76 | sources.append(toRTCRtpSynchronizationSource(rtcSource)); |
77 | } |
78 | return sources; |
79 | } |
80 | |
81 | } // namespace WebCore |
82 | |
83 | #endif // ENABLE(WEB_RTC) && USE(LIBWEBRTC) |
84 | |