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webrtc_m130/webrtc/call
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stefan ba4c0e45ff Add send-side BWE to WebRtcVoiceEngine under a finch experiment. 
This adds negotiation of both transport sequence number and transport
feedback. Only offers transport seq num if the
WebRTC-Audio-SendSideBwe finch experiment is enabled.

TBR=mflodman@webrtc.org
BUG=webrtc:5263

Review URL: https://codereview.webrtc.org/1604563002

Cr-Commit-Position: refs/heads/master@{#11487}
2016-02-04 12:12:31 +00:00
..
mock
Fixes a bug which incorrectly logs incoming RTCP as outgoing.
2016-01-21 13:42:10 +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
Add send-side BWE to WebRtcVoiceEngine under a finch experiment.
2016-02-04 12:12:31 +00:00
BUILD.gn
Move BitrateAllocator from BitrateController logic to Call.
2015-11-13 05:02:46 +00:00
call_perf_tests.cc
Remove mutable from rtc::CriticalSection members.
2016-01-25 11:52:53 +00:00
call_unittest.cc
Move some send stream configuration into webrtc::AudioSendStream.
2015-11-16 15:34:59 +00:00
call.cc
Switch to use new implementation in metrics.h.
2016-01-25 13:58:27 +00:00
congestion_controller.cc
Use separate rtp module lists for send and receive in PacketRouter.
2016-02-01 12:40:04 +00:00
congestion_controller.h
Remove mutable from rtc::CriticalSection members.
2016-01-25 11:52:53 +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