webrtc

History

Erik Språng 11beccd712 Remove external report blocks from RtcpSender and rtp_rtcp interface.

Feature does not seem to be used and complicates other refactoring of
the rtcp module.

BUG=
R=asapersson@webrtc.org, henrik.lundin@webrtc.org

Review URL: https://webrtc-codereview.appspot.com/54569004

Cr-Commit-Position: refs/heads/master@{#9304}

2015-05-28 09:10:34 +00:00

base

Prevent potential double-free if srtp_create fails.

2015-05-27 21:41:52 +00:00

build

PRESUBMIT: Improve PyLint check and add GN format check.

2015-05-25 10:55:50 +00:00

common_audio

Add GetChunkLength to LappedTransform.

2015-05-28 00:09:56 +00:00

common_video

PRESUBMIT: Improve PyLint check and add GN format check.

2015-05-25 10:55:50 +00:00

examples/android/media_demo

VoE: Remove unused interfaces

2015-05-26 08:25:00 +00:00

libjingle

Some cleanup for base/logging and base/stream.h

2015-05-25 09:26:08 +00:00

modules

Remove external report blocks from RtcpSender and rtp_rtcp interface.

2015-05-28 09:10:34 +00:00

overrides

Attempt at fixing error on the Chrome Windows FYI bots.

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

p2p

Revert "Fix sending wrong candidates down to transportchannel."

2015-05-22 22:32:51 +00:00

sound

Remove henrike@ from OWNERS

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

system_wrappers

PRESUBMIT: Improve PyLint check and add GN format check.

2015-05-25 10:55:50 +00:00

test

PRESUBMIT: Improve PyLint check and add GN format check.

2015-05-25 10:55:50 +00:00

tools

PRESUBMIT: Improve PyLint check and add GN format check.

2015-05-25 10:55:50 +00:00

video

Configure default render delay as 10 ms.

2015-05-27 15:59:21 +00:00

video_engine

Revert "Remove default encoder/decoders."

2015-05-26 09:44:14 +00:00

voice_engine

Remove remaining .mk files.

2015-05-28 09:05:11 +00:00

.gitignore

.gitignore: Add *.mk, created as part of ChromiumOS build

2013-01-04 21:25:42 +00:00

audio_receive_stream.h

Add AudioReceiveStream to Call API.

2015-04-29 13:24:10 +00:00

BUILD.gn

PRESUBMIT: Improve PyLint check and add GN format check.

2015-05-25 10:55:50 +00:00

call.h

Add AudioReceiveStream to Call API.

2015-04-29 13:24:10 +00:00

codereview.settings

Add codereview.settings to the /webrtc subdirectory

2014-12-05 13:43:35 +00:00

common_types.cc

Parsing of transport wide sequence number rtp extension header.

2015-03-17 14:33:46 +00:00

common_types.h

Parsing of transport wide sequence number rtp extension header.

2015-03-17 14:33:46 +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 AudioReceiveStream to Call API.

2015-04-29 13:24:10 +00:00

config.h

Reland r9159 "Adding a new constraint to set NetEq buffer capacity ..."

2015-05-11 10:44:20 +00:00

engine_configurations.h

Remove VideoEngine interfaces.

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

experiments.h

Remove no longer used SkipEncodingUnusedStreams.

2014-07-22 07:17:17 +00:00

frame_callback.h

Changing include guard in frame_callback.h.

2015-02-03 14:51:39 +00:00

LICENSE

Move src/ -> webrtc/

2012-10-22 18:19:23 +00:00

LICENSE_THIRD_PARTY

Add the Ooura FFT to RealFourier.

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

OWNERS

GN: Add BUILD.gn files + kjellander to OWNERS

2014-06-23 19:21:07 +00:00

PATENTS

Move src/ -> webrtc/

2012-10-22 18:19:23 +00:00

PRESUBMIT.py

PRESUBMIT.py: accept variants on the copyright message that are present in the codebase.

2014-05-23 17:27:18 +00:00

README.chromium

Move src/ -> webrtc/

2012-10-22 18:19:23 +00:00

rtc_unittests.isolate

Update all .isolate files for the new format.

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

supplement.gypi

Move target_subarch from gyp_webrtc to supplement.gypi

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

transport.h

Rename newapi::Transport::SendRTP()->SendRtp().

2013-11-20 12:17:04 +00:00

typedefs.h

Clear ARM NEON flag

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

video_decoder.h

Remove unused VideoDecoder methods.

2015-05-21 07:42:14 +00:00

video_encoder.h

Add HW fallback option to software encoding.

2015-05-19 21:09:17 +00:00

video_engine_tests.isolate

Update isolate files for Android APK tests.

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

video_frame.h

Reland "Remove usage of webrtc::NativeHandle since is just adds an extra level of indirection.""

2015-04-09 11:44:19 +00:00

video_receive_stream.h

Configure default render delay as 10 ms.

2015-05-27 15:59:21 +00:00

video_renderer.h

Use VideoReceiveStream as an ExternalRenderer.

2015-02-09 15:15:24 +00:00

video_send_stream.h

Reland "Make the entry point for VideoFrames to webrtc const ref I420VideoFrame.""

2015-03-18 09:51:44 +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

Add HW fallback option to software encoding.

2015-05-19 21:09:17 +00:00

webrtc.gyp

Rename GYP and GN targets for video capture+render.

2015-02-11 07:47:47 +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.