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https://github.com/mollyim/webrtc.git
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57858b3be0
Original change's description: > Update RTCStatsCollector to work with RtpTransceivers > > Bug: webrtc:8764 > Change-Id: I8b442345869eb6d8b65fd12241ed7cb6e7d7ce3d > Reviewed-on: https://webrtc-review.googlesource.com/49580 > Commit-Queue: Steve Anton <steveanton@webrtc.org> > Reviewed-by: Taylor Brandstetter <deadbeef@webrtc.org> > Reviewed-by: Henrik Boström <hbos@webrtc.org> > Cr-Commit-Position: refs/heads/master@{#22026} Bug: webrtc:8764 Change-Id: I6a682824febf3f4f41397fc1a8dd7396c4ffa8e3 Reviewed-on: https://webrtc-review.googlesource.com/54160 Reviewed-by: Taylor Brandstetter <deadbeef@webrtc.org> Commit-Queue: Steve Anton <steveanton@webrtc.org> Cr-Commit-Position: refs/heads/master@{#22064}
44 lines
1.3 KiB
C++
44 lines
1.3 KiB
C++
/*
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* Copyright 2018 The WebRTC Project Authors. All rights reserved.
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*
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* Use of this source code is governed by a BSD-style license
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* that can be found in the LICENSE file in the root of the source
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* tree. An additional intellectual property rights grant can be found
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* in the file PATENTS. All contributing project authors may
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* be found in the AUTHORS file in the root of the source tree.
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*/
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#include "api/rtpreceiverinterface.h"
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namespace webrtc {
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RtpSource::RtpSource(int64_t timestamp_ms,
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uint32_t source_id,
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RtpSourceType source_type)
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: timestamp_ms_(timestamp_ms),
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source_id_(source_id),
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source_type_(source_type) {}
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RtpSource::RtpSource(int64_t timestamp_ms,
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uint32_t source_id,
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RtpSourceType source_type,
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uint8_t audio_level)
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: timestamp_ms_(timestamp_ms),
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source_id_(source_id),
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source_type_(source_type),
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audio_level_(audio_level) {}
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RtpSource::RtpSource(const RtpSource&) = default;
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RtpSource& RtpSource::operator=(const RtpSource&) = default;
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RtpSource::~RtpSource() = default;
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std::vector<rtc::scoped_refptr<MediaStreamInterface>>
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RtpReceiverInterface::streams() const {
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return {};
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}
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std::vector<RtpSource> RtpReceiverInterface::GetSources() const {
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return {};
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}
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} // namespace webrtc
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