nisse d44ce0563f Reland of Always call RemoteBitrateEstimator::IncomingPacket from Call. (patchset #1 id:1 of https://codereview.webrtc.org/2668973003/ ) ...

Reason for revert:
Intending to fix issues and reland.

Original issue's description:
> Revert of Always call RemoteBitrateEstimator::IncomingPacket from Call. (patchset #9 id:160001 of https://codereview.webrtc.org/2659563002/ )
>
> Reason for revert:
> This change causes excessive logging when running tests, and possibly also broke perf tests, see https://build.chromium.org/p/client.webrtc.perf/builders/Linux%20Trusty/builds/1040/steps/webrtc_perf_tests/logs/stdio
>
>
> Original issue's description:
> > Always call RemoteBitrateEstimator::IncomingPacket from Call.
> >
> > Delete the calls from RtpStreamReceiver (for video) and
> > AudioReceiveStream.
> >
> > BUG=webrtc:6847
> >
> > Review-Url: https://codereview.webrtc.org/2659563002
> > Cr-Commit-Position: refs/heads/master@{#16393}
> > Committed: 6d4dd593a8
>
> TBR=stefan@webrtc.org,brandtr@webrtc.org
> # Skipping CQ checks because original CL landed less than 1 days ago.
> NOPRESUBMIT=true
> NOTREECHECKS=true
> NOTRY=true
> BUG=webrtc:6847
>
> Review-Url: https://codereview.webrtc.org/2668973003
> Cr-Commit-Position: refs/heads/master@{#16400}
> Committed: 14245cc939

TBR=stefan@webrtc.org,brandtr@webrtc.org
BUG=webrtc:6847

Review-Url: https://codereview.webrtc.org/2673523003
Cr-Commit-Position: refs/heads/master@{#16440}

2017-02-06 10:23:00 +00:00

..

audio_receive_stream.h

Pass SdpAudioFormat through Channel, without converting to CodecInst

2017-01-19 15:03:59 +00:00

audio_send_stream.cc

Moved call.h and most of api/call/* into call/

2016-12-07 12:53:04 +00:00

audio_send_stream.h

Added a new echo likelihood stat that reports the maximum value from a previous time period.

2017-01-15 16:29:46 +00:00

audio_state.h

Moved call.h and most of api/call/* into call/

2016-12-07 12:53:04 +00:00

bitrate_allocator_unittest.cc

Relanding "Pass time constant to bwe smoothing filter."

2016-11-30 12:47:47 +00:00

bitrate_allocator.cc

Relanding "Pass time constant to bwe smoothing filter."

2016-11-30 12:47:47 +00:00

bitrate_allocator.h

Relanding "Pass time constant to bwe smoothing filter."

2016-11-30 12:47:47 +00:00

bitrate_estimator_tests.cc

Moved call.h and most of api/call/* into call/

2016-12-07 12:53:04 +00:00

BUILD.gn

Stop using VoEVideoSync in Call/VideoReceiveStream.

2017-01-31 11:58:40 +00:00

call_perf_tests.cc

Refactor FakeAudioDevice to have separate methods for starting recording and playout.

2017-01-31 21:32:49 +00:00

call_unittest.cc

Make |rtcp_send_transport| mandatory in FlexfecReceiveStream::Config.

2017-01-13 15:41:19 +00:00

call.cc

Reland of Always call RemoteBitrateEstimator::IncomingPacket from Call. (patchset #1 id:1 of https://codereview.webrtc.org/2668973003/ )

2017-02-06 10:23:00 +00:00

call.h

Inform jitter buffer about FlexFEC protection.

2017-01-27 14:47:55 +00:00

DEPS

Passed AudioMixer to AudioState::Config.

2016-11-17 14:48:56 +00:00

flexfec_receive_stream_impl.cc

Let FlexfecReceiveStreamImpl send RTCP RRs.

2017-01-17 09:33:54 +00:00

flexfec_receive_stream_impl.h

Let FlexfecReceiveStreamImpl send RTCP RRs.

2017-01-17 09:33:54 +00:00

flexfec_receive_stream_unittest.cc

Let FlexfecReceiveStreamImpl send RTCP RRs.

2017-01-17 09:33:54 +00:00

flexfec_receive_stream.h

Make |rtcp_send_transport| mandatory in FlexfecReceiveStream::Config.

2017-01-13 15:41:19 +00:00

OWNERS

Remove all references to GYP

2016-11-16 19:11:38 +00:00

packet_injection_tests.cc

Add FlexFEC to CallTest.

2016-11-15 15:11:00 +00:00

rampup_tests.cc

Set the start bitrate to the delay-based BWE.

2017-01-27 14:43:18 +00:00

rampup_tests.h

Set the start bitrate to the delay-based BWE.

2017-01-27 14:43:18 +00:00

syncable.cc

Stop using VoEVideoSync in Call/VideoReceiveStream.

2017-01-31 11:58:40 +00:00

syncable.h

Stop using VoEVideoSync in Call/VideoReceiveStream.

2017-01-31 11:58:40 +00:00