ee2388f3f0
This reverts commit 56bae8ded39c3fab2635b7d2d1d17a87d5d2818b. Reason for revert: Speculative revert. This CL is suspect of making Chrome trybots fail the following test, preventing rolls: external/wpt/webrtc/RTCPeerConnection-track-stats.https.html Some failed roll attempts: https://chromium-review.googlesource.com/c/chromium/src/+/921421 https://chromium-review.googlesource.com/c/chromium/src/+/921422 https://chromium-review.googlesource.com/c/chromium/src/+/921781 Some failed bot runs: https://ci.chromium.org/buildbot/tryserver.chromium.linux/linux_chromium_rel_ng/647669 https://ci.chromium.org/buildbot/tryserver.chromium.win/win7_chromium_rel_ng/103786 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} TBR=steveanton@webrtc.org,deadbeef@webrtc.org,hbos@webrtc.org,pthatcher@webrtc.org Change-Id: I21ce2109087d7b2d9470471ee9a6757f904296d2 No-Presubmit: true No-Tree-Checks: true No-Try: true Bug: webrtc:8764 Reviewed-on: https://webrtc-review.googlesource.com/54000 Reviewed-by: Guido Urdaneta <guidou@webrtc.org> Commit-Queue: Guido Urdaneta <guidou@webrtc.org> Cr-Commit-Position: refs/heads/master@{#22036}
49 lines
1.4 KiB
C++
49 lines
1.4 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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int RtpReceiverInterface::AttachmentId() const {
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return 0;
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}
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} // namespace webrtc
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