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2b706343de
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Add audio_level member to RtpSource and set it from RtpReceiverImpl::IncomingRtpPacket.
BUG=webrtc:7987
Review-Url: https://codereview.webrtc.org/3000713002
Cr-Commit-Position: refs/heads/master@{#19503}
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2017-08-24 21:52:17 +00:00 |
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7fcdb6d7ca
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Delete class NullRtpData and function NullObjectRtpData.
BUG=webrtc:5565
Review-Url: https://codereview.webrtc.org/2885823002
Cr-Commit-Position: refs/heads/master@{#18366}
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2017-06-01 07:30:55 +00:00 |
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0426222f4c
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Modified the rtp_receiver_unittests.
Implemented operator == in RtpSource and use the gmock EXPECT_THAT to make the test cleaner.
Related CL: https://codereview.webrtc.org/2770233003/
BUG=chromium:703122
Review-Url: https://codereview.webrtc.org/2813753002
Cr-Commit-Position: refs/heads/master@{#17659}
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2017-04-11 18:28:10 +00:00 |
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8d609f6b6d
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Reland of Implemented the GetSources() in native code. (patchset #1 id:1 of https://codereview.webrtc.org/2809613002/ )
Reason for revert:
Re-land, reverting did not fix bug.
https://bugs.chromium.org/p/webrtc/issues/detail?id=7465
Original issue's description:
> 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: 292084c376
>
> 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: fbcc5cb386
TBR=deadbeef@webrtc.org,solenberg@webrtc.org,philipel@webrtc.org,stefan@webrtc.org,danilchap@webrtc.org,zhihuang@webrtc.org,olka@webrtc.org
# Skipping CQ checks because original CL landed less than 1 days ago.
NOPRESUBMIT=true
NOTREECHECKS=true
NOTRY=true
BUG=chromium:703122
Review-Url: https://codereview.webrtc.org/2810623003
Cr-Commit-Position: refs/heads/master@{#17621}
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2017-04-10 14:39:05 +00:00 |
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fbcc5cb386
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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: 292084c376
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}
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2017-04-10 11:38:13 +00:00 |
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292084c376
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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}
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2017-04-07 17:57:22 +00:00 |
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