Stefan Holmer ff2a6351e0 Add ramp-up tests for transport sequence number with and w/o audio. ...

Also add a perf metric tracking the average network latency.

The audio stream test is disabled for now since audio isn't included in bitrate allocation.

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

Review URL: https://codereview.webrtc.org/1582833002 .

Cr-Commit-Position: refs/heads/master@{#11244}

2016-01-14 09:00:34 +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

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

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