Jianjun Zhu 037f3e42f2 Replace absolute path with relative path for GN files. ...

Bug: webrtc:7952
Change-Id: I45d889bd976f58386f803d0dc27147ea00a52e56
Reviewed-on: https://chromium-review.googlesource.com/612786
Commit-Queue: Henrik Kjellander <kjellander@webrtc.org>
Reviewed-by: Henrik Kjellander <kjellander@webrtc.org>
Reviewed-by: Edward Lemur <ehmaldonado@webrtc.org>
Cr-Commit-Position: refs/heads/master@{#19357}

2017-08-15 15:57:36 +00:00

..

test

Disable flaky NoBandwidthDropAfterDtx test.

2017-07-19 14:52:47 +00:00

utility

Remove remains of webrtc/base

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

audio_receive_stream_unittest.cc

Adding stats that can be used to compute output audio levels as described here https://w3c.github.io/webrtc-stats/#dom-rtcmediastreamtrackstats-totalaudioenergy.

2017-07-14 19:17:49 +00:00

audio_receive_stream.cc

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

2017-07-26 09:09:44 +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

Reland "Allow AudioSendStream to reconfig AudioNetworkAdaptor"

2017-07-26 14:28:51 +00:00

audio_send_stream.cc

Reland "Allow AudioSendStream to reconfig AudioNetworkAdaptor"

2017-07-26 14:28:51 +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 absolute path with relative path for GN files.

2017-08-15 15:57:36 +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