Peter Boström 905f8e7e9d Make ReconfigureVideoEncoder void. ...

Also moves and simplifies SetSendCodec from VideoSendStream to mostly
inside ViEEncoder. This is necessary for making
ReconfigureVideoEncoder asynchronous as we don't post any result back.

BUG=webrtc:5494
R=stefan@webrtc.org
TBR=mflodman@webrtc.org

Review URL: https://codereview.webrtc.org/1754283002 .

Cr-Commit-Position: refs/heads/master@{#11847}

2016-03-02 16:00:07 +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

Make ReconfigureVideoEncoder void.

2016-03-02 16:00:07 +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