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Revert of Implemented the GetSources() in native code. (patchset #11 id:510001 of https://codereview.webrtc.org/2770233003/ )
Reason for revert:
Suspected of WebRtcApprtcBrowserTest.MANUAL_WorksOnApprtc breakage, see
https://bugs.chromium.org/p/webrtc/issues/detail?id=7465
Original issue's description:
> Added the GetSources() to the RtpReceiverInterface and implemented
> it for the AudioRtpReceiver.
>
> This method returns a vector of RtpSource(both CSRC source and SSRC
> source) which contains the ID of a source, the timestamp, the source
> type (SSRC or CSRC) and the audio level.
>
> The RtpSource objects are buffered and maintained by the
> RtpReceiver in webrtc/modules/rtp_rtcp/. When the method is called,
> the info of the contributing source will be pulled along the object
> chain:
> AudioRtpReceiver -> VoiceChannel -> WebRtcVoiceMediaChannel ->
> AudioReceiveStream -> voe::Channel -> RtpRtcp module
>
> Spec:https://w3c.github.io/webrtc-pc/archives/20151006/webrtc.html#widl-RTCRtpReceiver-getContributingSources-sequence-RTCRtpContributingSource
>
> BUG= chromium:703122
> TBR=stefan@webrtc.org, danilchap@webrtc.org
>
> Review-Url: https://codereview.webrtc.org/2770233003
> Cr-Commit-Position: refs/heads/master@{#17591}
> Committed: https://chromium.googlesource.com/external/webrtc/+/292084c3765d9f3ee406ca2ec86eae206b540053
TBR=deadbeef@webrtc.org,solenberg@webrtc.org,hbos@webrtc.org,philipel@webrtc.org,stefan@webrtc.org,danilchap@webrtc.org,zhihuang@webrtc.org
# Not skipping CQ checks because original CL landed more than 1 days ago.
BUG= chromium:703122
Review-Url: https://codereview.webrtc.org/2809613002
Cr-Commit-Position: refs/heads/master@{#17616}
Committed: https://chromium.googlesource.com/external/webrtc/+/fbcc5cb3869d1370008e40f24fc03ac8fb69c675
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webrtc/api/rtpreceiverinterface.h
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webrtc/api/test/mock_rtpreceiver.h
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webrtc/audio/audio_receive_stream.h
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webrtc/audio/audio_receive_stream.cc
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webrtc/call/BUILD.gn
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webrtc/call/audio_receive_stream.h
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webrtc/media/engine/fakewebrtccall.h
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webrtc/media/engine/webrtcvoiceengine.h
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webrtc/media/engine/webrtcvoiceengine.cc
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webrtc/modules/rtp_rtcp/BUILD.gn
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webrtc/modules/rtp_rtcp/include/rtp_receiver.h
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webrtc/modules/rtp_rtcp/source/rtp_receiver_impl.h
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webrtc/modules/rtp_rtcp/source/rtp_receiver_impl.cc
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webrtc/modules/rtp_rtcp/source/rtp_receiver_unittest.cc
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webrtc/pc/channel.h
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webrtc/pc/channel.cc
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webrtc/pc/peerconnection_integrationtest.cc
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webrtc/pc/rtpreceiver.h
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webrtc/pc/rtpreceiver.cc
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webrtc/test/mock_voe_channel_proxy.h
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webrtc/video/vie_encoder.cc
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webrtc/voice_engine/BUILD.gn
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webrtc/voice_engine/channel.h
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webrtc/voice_engine/channel_proxy.h
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webrtc/voice_engine/channel_proxy.cc
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Total messages: 7 (3 generated)
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