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webrtc_m130/webrtc/call
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stefan 32f81542c2 Support REMB in combination with send-side BWE. 
BUG=webrtc:4173

Review URL: https://codereview.webrtc.org/1581113006

Cr-Commit-Position: refs/heads/master@{#11322}
2016-01-20 15:14:03 +00:00
..
mock
Wire-up BWE feedback for audio receive streams.
2016-01-12 12:55:11 +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
Switch use of CriticalSectionWrapper -> rtc::CriticalSection in call/
2016-01-20 10:32:58 +00:00
bitrate_estimator_tests.cc
Add audio streams to CallTest and a first A/V call test.
2016-01-07 16:43:31 +00:00
BUILD.gn
Move BitrateAllocator from BitrateController logic to Call.
2015-11-13 05:02:46 +00:00
call_perf_tests.cc
Remove extra_options from VideoCodec.
2016-01-19 15:26:24 +00:00
call_unittest.cc
Move some send stream configuration into webrtc::AudioSendStream.
2015-11-16 15:34:59 +00:00
call.cc
Remove extra_options from VideoCodec.
2016-01-19 15:26:24 +00:00
congestion_controller.cc
Support REMB in combination with send-side BWE.
2016-01-20 15:14:03 +00:00
congestion_controller.h
Wire-up BWE feedback for audio receive streams.
2016-01-12 12:55:11 +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 ramp-up tests for transport sequence number with and w/o audio.
2016-01-14 09:00:34 +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] rtcp::SenderReport moved into own file and got Parse function
2016-01-18 10:43:38 +00:00
rtc_event_log.cc
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
rtc_event_log.h
Added protobuf message for loss-based BWE events, and wired it up to the send side bandwidth estimator.
2015-11-05 20:02:19 +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