Fredrik Solenberg 73276ad7ed - Removes voe_conference_test. ...

- Adds a new AudioStatsTest, with better coverage of the same features, based on call_test.
- Adds an AudioEndToEndTest utility, which AudioStatsTest and LowBandwidthAudioTest uses.

BUG=webrtc:4690
R=kwiberg@webrtc.org

Review-Url: https://codereview.webrtc.org/3008273002 .
Cr-Commit-Position: refs/heads/master@{#19833}

2017-09-14 12:46:50 +00:00

..

test

- Removes voe_conference_test.

2017-09-14 12:46:50 +00:00

utility

Remove remains of webrtc/base

2017-07-19 17:40:47 +00:00

audio_receive_stream_unittest.cc

Wiring discard rate of audio FEC/RED packets up to StatsReport.

2017-08-28 13:17:55 +00:00

audio_receive_stream.cc

Wiring discard rate of audio FEC/RED packets up to StatsReport.

2017-08-28 13:17:55 +00:00

audio_receive_stream.h

Update thread annotiation macros to use RTC_ prefix

2017-09-09 11:17:22 +00:00

audio_send_stream_unittest.cc

Add new ANA stats to the old GetStats() to count the number of actions taken by each controller.

2017-09-08 15:13:19 +00:00

audio_send_stream.cc

Add new ANA stats to the old GetStats() to count the number of actions taken by each controller.

2017-09-08 15:13:19 +00:00

audio_send_stream.h

Update thread annotiation macros to use RTC_ prefix

2017-09-09 11:17:22 +00:00

audio_state_unittest.cc

Allow an external audio processing module to be used in WebRTC

2017-06-29 15:32:09 +00:00

audio_state.cc

External APM usage downstream dependency support cleanup

2017-07-07 11:25:11 +00:00

audio_state.h

Update thread annotiation macros to use RTC_ prefix

2017-09-09 11:17:22 +00:00

audio_transport_proxy.cc

Allow an external audio processing module to be used in WebRTC

2017-06-29 15:32:09 +00:00

audio_transport_proxy.h

Rename webrtc/base -> webrtc/rtc_base

2017-07-06 19:11:40 +00:00

BUILD.gn

- Removes voe_conference_test.

2017-09-14 12:46:50 +00:00

conversion.h

Implement AudioSendStream::GetStats().

2015-10-27 10:35:30 +00:00

DEPS

Recently we moved webrtc/base to webrtc/rtc_base, so these

2017-08-30 11:45:58 +00:00

OWNERS

Add ossu@ to OWNERS of audio/ and modules/audio_coding/

2016-12-15 15:52:14 +00:00

scoped_voe_interface.h

Rename webrtc/base -> webrtc/rtc_base

2017-07-06 19:11:40 +00:00

time_interval_unittest.cc

Adds a histogram metric tracking for how long audio RTP packets are sent

2017-07-19 07:39:19 +00:00

time_interval.cc

Adds a histogram metric tracking for how long audio RTP packets are sent

2017-07-19 07:39:19 +00:00

time_interval.h

Move optional.h to webrtc/api/

2017-09-05 15:43:13 +00:00