danilchap ac287ee8b5 VideoCaptureInput enforce VideoFrame::render_time to be generated by webrtc clock. ...

render_time time field (means capture time for sender side) is used by rtcp SenderReport to calculate offset since last frame and to estimate rtp timestamp for the time SenderReport should be send at.
mapping between rtp timestamp and ntp time in SenderReport is used for stream synchronization.

calculation of rtp_timestamp (using ntp_time of incoming video frame) for rtp packets is unchanged.

BUG=webrtc:5433, webrtc:5504, webrtc:5505

Review URL: https://codereview.webrtc.org/1693443002

Cr-Commit-Position: refs/heads/master@{#11820}

2016-02-29 20:17:10 +00:00

..

mock

Move congestion controller to a separate module.

2016-02-23 12:30:51 +00:00

bitrate_allocator_unittest.cc

Move BitrateAllocator from BitrateController logic to Call.

2015-11-13 05:02:46 +00:00

bitrate_allocator.cc

Switch use of CriticalSectionWrapper -> rtc::CriticalSection in call/

2016-01-20 10:32:58 +00:00

bitrate_allocator.h

Remove mutable from rtc::CriticalSection members.

2016-01-25 11:52:53 +00:00

bitrate_estimator_tests.cc

Simplify CongestionController.

2016-02-17 14:52:25 +00:00

BUILD.gn

Move congestion controller to a separate module.

2016-02-23 12:30:51 +00:00

call_perf_tests.cc

VideoCaptureInput enforce VideoFrame::render_time to be generated by webrtc clock.

2016-02-29 20:17:10 +00:00

call_unittest.cc

Move some send stream configuration into webrtc::AudioSendStream.

2015-11-16 15:34:59 +00:00

call.cc

Break out pacer thread from CongestionController to increase testability.

2016-02-24 15:03:08 +00:00

OWNERS

Split webrtc/video into webrtc/{audio,call,video}.

2015-09-25 11:58:39 +00:00

packet_injection_tests.cc

Add audio streams to CallTest and a first A/V call test.

2016-01-07 16:43:31 +00:00

rampup_tests.cc

Disabled flaky tests

2016-02-23 17:56:49 +00:00

rampup_tests.h

Add ramp-up tests for transport sequence number with and w/o audio.

2016-01-14 09:00:34 +00:00

rtc_event_log2rtp_dump.cc

Split webrtc/video into webrtc/{audio,call,video}.

2015-09-25 11:58:39 +00:00

rtc_event_log_unittest.cc

[rtp_rtcp] rtc::scoped_ptr<rtcp::RawPacket> replaced with rtc::Buffer

2016-02-17 11:11:50 +00:00

rtc_event_log.cc

Fixes a bug which incorrectly logs incoming RTCP as outgoing.

2016-01-21 13:42:10 +00:00

rtc_event_log.h

Fixes a bug which incorrectly logs incoming RTCP as outgoing.

2016-01-21 13:42:10 +00:00

rtc_event_log.proto

Removed cname and receiver_reference_time_report from proto and logging code. Changed logging of RTCP to omit messages of type SDES and APP.

2015-11-06 17:00:21 +00:00

transport_adapter.cc

Add a PacketOptions struct to webrtc::Transport.

2015-10-02 10:39:40 +00:00

transport_adapter.h

system_wrappers: rename interface -> include

2015-10-28 17:17:50 +00:00

webrtc_call.gypi

Move congestion controller to a separate module.

2016-02-23 12:30:51 +00:00