platform-external-webrtc

Files

zhihuang d06adf60c1 Make P2PTransportChannel inherit from IceTransportInternal.

Make P2PTransportChannel inherit from IceTransportInternal instead of
TransportChannelImpl and TransportChannel, so that the DTLS-related methods can
be separated from P2PTransportChannel.

BUG=webrtc:6951

Review-Url: https://codereview.webrtc.org/2608353003
Cr-Commit-Position: refs/heads/master@{#16041}

2017-01-12 23:58:31 +00:00

androidjunit

Format all Java in WebRTC.

2016-09-29 11:12:51 +00:00

api

Replace ASSERT by RTC_DCHECK in all non-test code.

2017-01-12 13:15:36 +00:00

audio

GN: Refactor webrtc_nonparallel_tests and audio_tests to avoid crossing package boundaries.

2017-01-05 14:03:24 +00:00

base

Update smoothed bitrate.

2017-01-12 18:17:38 +00:00

build

Remove lint ignores and add WrongCall ignore directly to the test.

2017-01-12 14:19:40 +00:00

call

Clean up ramp-up tests and make sure they all pass.

2017-01-09 12:19:24 +00:00

common_audio

Replace RTC_DCHECK(false) with RTC_NOTREACHED().

2017-01-12 10:24:27 +00:00

common_video

Move VideoFrame and related declarations to webrtc/api/video.

2017-01-10 15:44:26 +00:00

examples

Remove lint ignores and add WrongCall ignore directly to the test.

2017-01-12 14:19:40 +00:00

logging

Convert rtc_event_log from webrtc::Clock to rtc::TimeMicros.

2016-12-20 13:03:58 +00:00

media

Update smoothed bitrate.

2017-01-12 18:17:38 +00:00

modules

Update smoothed bitrate.

2017-01-12 18:17:38 +00:00

p2p

Make P2PTransportChannel inherit from IceTransportInternal.

2017-01-12 23:58:31 +00:00

Replace ASSERT by RTC_DCHECK in all non-test code.

2017-01-12 13:15:36 +00:00

sdk

Do not report camera disconnected event if Android camera is closing.

2017-01-12 21:05:47 +00:00

stats

Update smoothed bitrate.

2017-01-12 18:17:38 +00:00

system_wrappers

Reland of Delete unused code from systeminfo. (patchset #1 id:1 of https://codereview.webrtc.org/2584563004/ )

2017-01-09 09:22:16 +00:00

test

Don't detect a new frame if a previous packet is used in a previous frame.

2017-01-11 10:01:56 +00:00

tools

Add method needed to extract frame capture and arrival timestamps from rtc event logs.

2016-12-20 16:51:52 +00:00

video

Treat negative ntp time as unset.

2017-01-12 18:02:22 +00:00

voice_engine

Update smoothed bitrate.

2017-01-12 18:17:38 +00:00

.gitignore

…

BUILD.gn

Make P2PTransportChannel inherit from IceTransportInternal.

2017-01-12 23:58:31 +00:00

call.h

Moved call.h and most of api/call/* into call/

2016-12-07 12:53:04 +00:00

codereview.settings

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common_types.cc

Reject XR TargetBitrate items with unsupported layer indices

2016-12-06 14:09:00 +00:00

common_types.h

Move VideoFrame and related declarations to webrtc/api/video.

2017-01-10 15:44:26 +00:00

config.cc

Keep all codec parameters in VideoReceiveStream::Decoder

2016-11-25 11:56:41 +00:00

config.h

Keep all codec parameters in VideoReceiveStream::Decoder

2016-11-25 11:56:41 +00:00

DEPS

Move VideoFrame and related declarations to webrtc/api/video.

2017-01-10 15:44:26 +00:00

LICENSE

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LICENSE_THIRD_PARTY

Delete unused spreadsort implementation.

2016-12-05 11:03:26 +00:00

OWNERS

Cleanup unused rules in webrtc/DEPS + add kjellander to OWNERS for it

2016-11-29 10:52:22 +00:00

PATENTS

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PRESUBMIT.py

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README.chromium

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typedefs.h

Remove extra uses of basictypes.h.

2016-12-09 14:54:08 +00:00

video_decoder.h

Move VideoDecoder::Create() logic to separate internal video decoder factory

2016-12-01 08:27:35 +00:00

video_encoder.h

Move usage of QualityScaler to ViEEncoder.

2016-11-29 09:44:22 +00:00

video_frame.h

Move VideoFrame and related declarations to webrtc/api/video.

2017-01-10 15:44:26 +00:00

video_receive_stream.h

Moved transport.h from webrtc/ to webrtc/api, created build target and updated WebRTC dependencies.

2016-11-28 15:02:19 +00:00

video_send_stream.h

Implement periodic bandwidth probing in application-limited region.

2016-11-28 21:11:24 +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.