Stefan Holmer 62a5ccdb53 Update bitrate only when we have incoming packet. ...

Also cleans up some unused code and makes sure the min bitrate of the BWE can't be set to anything lower than 10 kbps.

BUG=webrtc:5474
R=pbos@webrtc.org

Review URL: https://codereview.webrtc.org/1699903003 .

Cr-Commit-Position: refs/heads/master@{#11636}

2016-02-16 16:07:31 +00:00

..

mock

Clean up of CongestionController.

2016-02-08 13:31:53 +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

Initialize VideoSendStream members in constructor.

2016-02-11 12:51:18 +00:00

BUILD.gn

Move BitrateAllocator from BitrateController logic to Call.

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

call_perf_tests.cc

removed five redundant avsync tests to make webrtc_perf_test faster

2016-02-15 10:15:10 +00:00

call_unittest.cc

Move some send stream configuration into webrtc::AudioSendStream.

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

call.cc

Clean up of CongestionController.

2016-02-08 13:31:53 +00:00

congestion_controller.cc

Update bitrate only when we have incoming packet.

2016-02-16 16:07:31 +00:00

congestion_controller.h

Clean up of CongestionController.

2016-02-08 13:31:53 +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

Add send-side BWE to WebRtcVoiceEngine under a finch experiment.

2016-02-04 12:12:31 +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

Fixes a bug which incorrectly logs incoming RTCP as outgoing.

2016-01-21 13:42:10 +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 BitrateAllocator from BitrateController logic to Call.

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