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webrtc_m130/webrtc/call
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Stefan Holmer c379fcb248 Break out pacer thread from CongestionController to increase testability. 
R=pbos@webrtc.org
TBR=mflodman@webrtc.org

Review URL: https://codereview.webrtc.org/1732863002 .

Cr-Commit-Position: refs/heads/master@{#11745}
2016-02-24 15:03:08 +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
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
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