platform-external-webrtc

Files

henrika 49085ef280 Improves audio-routing in combination with BT in AppRTCMobile on Android.

This CL improves (speeds up) audio routing for BT devices in AppRTCMobile.

NOTRY=TRUE
BUG=webrtc:7888

Review-Url: https://codereview.webrtc.org/2961403003
Cr-Commit-Position: refs/heads/master@{#18858}

2017-06-30 13:25:25 +00:00

api

Opus implementation of the AudioDecoderFactoryTemplate API

2017-06-30 12:24:56 +00:00

audio

Always ResetSenderCongestionControlObjects before RegisterEtc...

2017-06-30 08:38:56 +00:00

base

Delete SignalThread class.

2017-06-29 13:21:20 +00:00

call

Allow an external audio processing module to be used in WebRTC

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

common_audio

Put attribute before function name instead of after, as required by GCC

2017-06-26 08:32:40 +00:00

common_video

Revert "VideoFrameBuffer: Remove deprecated functions"

2017-06-29 13:39:35 +00:00

examples

Improves audio-routing in combination with BT in AppRTCMobile on Android.

2017-06-30 13:25:25 +00:00

logging

Limit the number of simultaneous event logs.

2017-06-28 13:40:51 +00:00

media

Support encrypted RTP extensions (RFC 6904)

2017-06-29 19:31:36 +00:00

modules

Opus implementation of the AudioDecoderFactoryTemplate API

2017-06-30 12:24:56 +00:00

ortc

Allow an external audio processing module to be used in WebRTC

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

p2p

Support encrypted RTP extensions (RFC 6904)

2017-06-29 19:31:36 +00:00

Fix -Wcomment warning in webrtcsdp.cc

2017-06-30 06:19:31 +00:00

rtc_base

Revert "Update includes for webrtc/{base => rtc_base} rename (3/3)"

2017-06-30 13:14:47 +00:00

rtc_tools

Fix android video_quality_loopback_test

2017-06-30 12:12:09 +00:00

sdk

Properly export the symbols of video frame-buffer classes for link-time

2017-06-29 16:11:10 +00:00

stats

Move RTCOutboundRTPStreamStats.roundTripTime to inbound, don't collect.

2017-03-01 09:02:45 +00:00

system_wrappers

Prevent data race in GetStaticInstance

2017-05-29 07:01:14 +00:00

test

Opus implementation of the AudioDecoderFactoryTemplate API

2017-06-30 12:24:56 +00:00

tools

Move webrtc/{tools => rtc_tools}

2017-06-30 10:04:59 +00:00

video

Fix receiving FlexFEC in video_loopback.

2017-06-30 09:34:20 +00:00

voice_engine

Allow an external audio processing module to be used in WebRTC

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

.gitignore

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BUILD.gn

Support encrypted RTP extensions (RFC 6904)

2017-06-29 19:31:36 +00:00

codereview.settings

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common_types.cc

Allow parsing empty RTCP TargetBitrate messages, but stop sending them.

2017-06-22 12:40:25 +00:00

common_types.h

Allow parsing empty RTCP TargetBitrate messages, but stop sending them.

2017-06-22 12:40:25 +00:00

config_unittest.cc

Support encrypted RTP extensions (RFC 6904)

2017-06-29 19:31:36 +00:00

config.cc

Support encrypted RTP extensions (RFC 6904)

2017-06-29 19:31:36 +00:00

config.h

Support encrypted RTP extensions (RFC 6904)

2017-06-29 19:31:36 +00:00

DEPS

Move webrtc/{tools => rtc_tools}

2017-06-30 10:04:59 +00:00

LICENSE

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LICENSE_THIRD_PARTY

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OWNERS

Don't add or rename files in webrtc/ and webrtc/api/ without a proper review

2017-05-08 13:01:54 +00:00

PATENTS

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PRESUBMIT.py

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README.chromium

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typedefs.h

Make WARN_UNUSED_RESULT a no-op on gcc

2017-04-06 11:32:27 +00:00

video_receive_stream.h

Delete VieRemb class, move functionality to PacketRouter.

2017-04-19 06:38:35 +00:00

video_send_stream.h

Add field trial for balanced degradation preference.

2017-06-14 12:52:21 +00:00

webrtc.gni

Adds detection of audio glitches for playout on iOS (reland)

2017-06-13 16:00:18 +00:00

whitespace.txt

Moving whitespace file up by one folder

2017-01-24 13:19:42 +00:00

README.chromium

Name: WebRTC
URL: http://www.webrtc.org
Version: 90
License: BSD
License File: LICENSE

Description:
WebRTC provides real time voice and video processing
functionality to enable the implementation of 
PeerConnection/MediaStream.

Third party code used in this project is described 
in the file LICENSE_THIRD_PARTY.