oprypin 9b2f20c618 Replace gflags usages with rtc_base/flags in all targets based on test_main ...

BUG=webrtc:7644

Review-Url: https://codereview.webrtc.org/2995363002
Cr-Commit-Position: refs/heads/master@{#19580}

2017-08-29 12:51:57 +00:00

..

test

Replace gflags usages with rtc_base/flags in all targets based on test_main

2017-08-29 12:51:57 +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

Rename webrtc/base -> webrtc/rtc_base

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

audio_send_stream_unittest.cc

Move PacedSender ownership to RtpTransportControllerSend.

2017-08-22 14:16:49 +00:00

audio_send_stream.cc

Uncomment thread-check in AudioSendStream::OnPacketFeedbackVector()

2017-08-24 14:26:25 +00:00

audio_send_stream.h

Make ~webrtc::AudioSendStream public, and s/config()/GetConfig(), as well as make public.

2017-07-26 09:09:44 +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

External APM usage downstream dependency support cleanup

2017-07-07 11:25:11 +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

Replace gflags usages with rtc_base/flags in all targets based on test_main

2017-08-29 12:51:57 +00:00

conversion.h

Implement AudioSendStream::GetStats().

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

DEPS

Injectable audio encoders: WebRtcVoiceEngine and company

2017-04-27 09:08:52 +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

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

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