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webrtc/audio/test
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Jakob Ivarsson 47a03e8743 Default enable sending transport sequence numbers on audio packets. 
This enables send side bandwidth estimation for audio and removes field
trial "WebRTC-Audio-SendSideBwe" which this was controlled through.

Transport-cc extension still needs to be negotiated.

Bug: webrtc:12222
Change-Id: Ie2268fad13703eeb0f0d38fcf484baaa29715b7c
Reviewed-on: https://webrtc-review.googlesource.com/c/src/+/194142
Reviewed-by: Stefan Holmer <stefan@webrtc.org>
Commit-Queue: Jakob Ivarsson <jakobi@webrtc.org>
Cr-Commit-Position: refs/heads/master@{#32681}
2020-11-24 09:19:54 +00:00
..
unittests Reformat python files checked by pylint (part 1/2). 2020-10-30 10:13:11 +00:00
audio_bwe_integration_test.cc Default enable sending transport sequence numbers on audio packets. 2020-11-24 09:19:54 +00:00
audio_bwe_integration_test.h Propagate task queue to create test::DirectTransport by TaskQueueBase interface 2019-09-30 03:23:07 +00:00
audio_end_to_end_test.cc Propagate task queue to create test::DirectTransport by TaskQueueBase interface 2019-09-30 03:23:07 +00:00
audio_end_to_end_test.h Propagate task queue to create test::DirectTransport by TaskQueueBase interface 2019-09-30 03:23:07 +00:00
audio_stats_test.cc Fix standard GetStats to not modify NetEq state. 2020-09-14 09:51:21 +00:00
low_bandwidth_audio_test.cc Add missing headers to fix chromium roll 2020-06-05 17:49:04 +00:00
low_bandwidth_audio_test.py Reformat python files checked by pylint (part 1/2). 2020-10-30 10:13:11 +00:00
low_bandwidth_audio_test_flags.cc Migrate WebRTC test infra to ABSL_FLAG. 2019-07-19 06:54:04 +00:00
pc_low_bandwidth_audio_test.cc Add TimeController to the CreatePeerConnectionE2EQualityTestFixture API 2020-07-01 15:18:34 +00:00