platform-external-webrtc

Files

Jiawei Ou 4de6622bcc Fix a bug in computing audio delay on ios device. Converts seconds to

milliseconds by multiplying 1000 instead of dividing 1000.

BUG=
R=tkchin@webrtc.org

Review URL: https://codereview.webrtc.org/1265823003 .

Patch from Jiawei Ou <jiawei.ou@gmail.com>.

Cr-Commit-Position: refs/heads/master@{#9693}

2015-08-10 20:24:56 +00:00

base

Regenerate bind.h using pump.py

2015-08-06 10:41:04 +00:00

build

Suppress a race in libjingle_peerconnection_unittest

2015-08-06 11:34:39 +00:00

common_audio

Split MoveReadPosition into Forward and Backward versions.

2015-07-28 07:53:05 +00:00

common_video

Re-land "Convert native handles to buffers before encoding."

2015-06-05 09:08:12 +00:00

examples

Move all the examples from the talk directory into the webrtc examples directory.

2015-08-05 22:48:29 +00:00

libjingle

Remove webrtc/libjingle/{examples,session}.

2015-06-12 11:13:46 +00:00

modules

Fix a bug in computing audio delay on ios device. Converts seconds to

2015-08-10 20:24:56 +00:00

overrides

Attempt at fixing error on the Chrome Windows FYI bots.

2015-05-25 19:22:23 +00:00

p2p

Remove GICE (gone forever!) and PORTALLOCATOR_ENABLE_SHARED_UFRAG (enabled forever).

2015-08-10 19:22:59 +00:00

sound

Remove henrike@ from OWNERS

2015-04-01 15:08:49 +00:00

system_wrappers

When we trace to file, add eol of each trace message.

2015-07-28 02:46:01 +00:00

test

Add support for transport wide sequence numbers

2015-08-03 11:38:48 +00:00

tools

Configure AudioProcessing directly in agc_harness.

2015-07-23 01:27:16 +00:00

video

Selectable number of TL screenshare loopback test. Also contains some tweaks to make a single TL perform better.

2015-08-05 09:02:09 +00:00

video_engine

Add average rtt to CallStatsObserver and an average rtt histogram.

2015-08-07 11:27:56 +00:00

voice_engine

Removing AudioMixerStatusReceiver and ParticipantStatistics.

2015-07-28 09:07:35 +00:00

.gitignore

…

audio_receive_stream.h

Control combined_audio_video_bwe with config bool.

2015-07-24 14:10:25 +00:00

audio_send_stream.h

Define Stream base classes

2015-07-16 07:30:20 +00:00

BUILD.gn

Includes webrtc/build/protoc.gypi instead of build/protoc.gypi

2015-07-30 10:45:24 +00:00

call.h

Define Stream base classes

2015-07-16 07:30:20 +00:00

codereview.settings

Update deeper codereview.settings files to match the root.

2015-06-24 19:40:39 +00:00

common_types.cc

Separating voice activity flag from audio level in RtpHeaderExtension.

2015-08-10 13:08:46 +00:00

common_types.h

Separating voice activity flag from audio level in RtpHeaderExtension.

2015-08-10 13:08:46 +00:00

common.gyp

Fix style violations in common_types.h and config.h

2015-02-26 14:01:28 +00:00

common.h

Make Config::default_value leak instead of having an exit-time destructor.

2015-05-23 00:50:33 +00:00

config.cc

Add support for transport wide sequence numbers

2015-08-03 11:38:48 +00:00

config.h

Add support for transport wide sequence numbers

2015-08-03 11:38:48 +00:00

engine_configurations.h

Remove VideoEngine interfaces.

2015-05-12 14:51:08 +00:00

frame_callback.h

Rename I420VideoFrame to VideoFrame.

2015-05-30 00:21:56 +00:00

libjingle_examples.gyp

Move all the examples from the talk directory into the webrtc examples directory.

2015-08-05 22:48:29 +00:00

LICENSE

…

LICENSE_THIRD_PARTY

Add the Ooura FFT to RealFourier.

2015-03-19 20:07:43 +00:00

OWNERS

…

PATENTS

…

PRESUBMIT.py

…

README.chromium

…

rtc_unittests.isolate

Update all .isolate files for the new format.

2014-10-31 18:08:09 +00:00

stream.h

Define Stream base classes

2015-07-16 07:30:20 +00:00

supplement.gypi

Move target_subarch from gyp_webrtc to supplement.gypi

2015-04-16 07:24:23 +00:00

transport.h

…

typedefs.h

Clear ARM NEON flag

2015-05-20 05:20:04 +00:00

video_decoder.h

iOS HW H264 support.

2015-06-29 21:35:08 +00:00

video_encoder.h

Allow for framerate reduction for HW encoder.

2015-07-13 23:26:40 +00:00

video_engine_tests.isolate

Update isolate files for Android APK tests.

2014-11-13 08:35:05 +00:00

video_frame.h

Re-land "Convert native handles to buffers before encoding."

2015-06-05 09:08:12 +00:00

video_receive_stream.h

Define Stream base classes

2015-07-16 07:30:20 +00:00

video_renderer.h

Rename I420VideoFrame to VideoFrame.

2015-05-30 00:21:56 +00:00

video_send_stream.h

Define Stream base classes

2015-07-16 07:30:20 +00:00

webrtc_examples.gyp

Adding support for OpenSL ES output in native WebRTC

2015-05-18 14:49:04 +00:00

webrtc_perf_tests.isolate

Enable GoogleWifiTrace3Mbps simulations.

2015-05-04 12:26:26 +00:00

webrtc_tests.gypi

Includes webrtc/build/protoc.gypi instead of build/protoc.gypi

2015-07-30 10:45:24 +00:00

webrtc.gyp

Includes webrtc/build/protoc.gypi instead of build/protoc.gypi

2015-07-30 10:45:24 +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.