Stefan Holmer ea8c0f6fcb Fix capture ntp time issue introduced with r11187. ...

I think the problem was that I only introduced delay in one direction, and the estimation assumes that the RTT is evenly divided between the send direction and the receive direction, which was true for the old test.

BUG=chromium:576246
R=pbos@webrtc.org

Review URL: https://codereview.webrtc.org/1577853005 .

Cr-Commit-Position: refs/heads/master@{#11233}

2016-01-13 07:58:52 +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

Move BitrateAllocator from BitrateController logic to Call.

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

bitrate_allocator.h

Move BitrateAllocator from BitrateController logic to Call.

2015-11-13 05:02:46 +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

Fix capture ntp time issue introduced with r11187.

2016-01-13 07:58:52 +00:00

call_unittest.cc

Move some send stream configuration into webrtc::AudioSendStream.

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

call.cc

Wire-up BWE feedback for audio receive streams.

2016-01-12 12:55:11 +00:00

congestion_controller.cc

Reland of Merge webrtc/video_engine/ into webrtc/video/ (patchset #2 id:300001 of https://codereview.webrtc.org/1507903005/ )

2015-12-09 11:13:40 +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

Fix test bug introduced in r11101.

2016-01-12 14:51:28 +00:00

rampup_tests.h

Fix test bug introduced in r11101.

2016-01-12 14:51:28 +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

Remove DISABLED_ON_ macros.

2016-01-04 21:44:16 +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