| /* |
| * Copyright (C) 2018 Apple Inc. |
| * |
| * 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. |
| */ |
| |
| #include "config.h" |
| #include "LibWebRTCRtpReceiverBackend.h" |
| |
| #include "LibWebRTCUtils.h" |
| |
| #if ENABLE(WEB_RTC) && USE(LIBWEBRTC) |
| |
| namespace WebCore { |
| |
| RTCRtpParameters LibWebRTCRtpReceiverBackend::getParameters() |
| { |
| return toRTCRtpParameters(m_rtcReceiver->GetParameters()); |
| } |
| |
| static inline void fillRTCRtpContributingSource(RTCRtpContributingSource& source, const webrtc::RtpSource& rtcSource) |
| { |
| source.timestamp = rtcSource.timestamp_ms(); |
| source.source = rtcSource.source_id(); |
| if (rtcSource.audio_level()) |
| source.audioLevel = (*rtcSource.audio_level() == 127) ? 0 : pow(10, -*rtcSource.audio_level() / 20); |
| } |
| |
| static inline RTCRtpContributingSource toRTCRtpContributingSource(const webrtc::RtpSource& rtcSource) |
| { |
| RTCRtpContributingSource source; |
| fillRTCRtpContributingSource(source, rtcSource); |
| return source; |
| } |
| |
| static inline RTCRtpSynchronizationSource toRTCRtpSynchronizationSource(const webrtc::RtpSource& rtcSource) |
| { |
| RTCRtpSynchronizationSource source; |
| fillRTCRtpContributingSource(source, rtcSource); |
| return source; |
| } |
| |
| Vector<RTCRtpContributingSource> LibWebRTCRtpReceiverBackend::getContributingSources() const |
| { |
| Vector<RTCRtpContributingSource> sources; |
| for (auto& rtcSource : m_rtcReceiver->GetSources()) { |
| if (rtcSource.source_type() == webrtc::RtpSourceType::CSRC) |
| sources.append(toRTCRtpContributingSource(rtcSource)); |
| } |
| return sources; |
| } |
| |
| Vector<RTCRtpSynchronizationSource> LibWebRTCRtpReceiverBackend::getSynchronizationSources() const |
| { |
| Vector<RTCRtpSynchronizationSource> sources; |
| for (auto& rtcSource : m_rtcReceiver->GetSources()) { |
| if (rtcSource.source_type() == webrtc::RtpSourceType::SSRC) |
| sources.append(toRTCRtpSynchronizationSource(rtcSource)); |
| } |
| return sources; |
| } |
| |
| } // namespace WebCore |
| |
| #endif // ENABLE(WEB_RTC) && USE(LIBWEBRTC) |