webrtc

History

Tommi a4463b2b8a Further updates to fix libjingle logging.

Since libjingle log constant values decrease as severety goes up while Chrome's increase, I decided to handle the verbosity level check explicitly and convert libjingle severity over to chrome constants only when we log.
This also requires updating the unittests on the Chrome side.

BUG=chromium:401963
TBR=magjed@webrtc.org

Review URL: https://webrtc-codereview.appspot.com/51839004

Cr-Commit-Position: refs/heads/master@{#9189}

2015-05-14 02:18:15 +00:00

base

Windows utility to setTheadName to help debugging.

2015-05-13 07:20:48 +00:00

build

Set msvs_error_on_missing_sources=1 in GYP_GENERATOR_FLAGS on Windows.

2015-05-13 13:00:11 +00:00

common_audio

Add 3-band filter-bank implementation

2015-04-27 18:34:16 +00:00

common_video

Move IncomingVideoFrames to common_video/.

2015-05-13 11:28:04 +00:00

examples/android

Roll chromium_revision ec5b768..62a5bb3 (328242:329063)

2015-05-11 09:34:26 +00:00

libjingle

GYP variables for building expat, icu, libsrtp, usrsctp

2015-04-27 12:38:37 +00:00

modules

Move IncomingVideoFrames to common_video/.

2015-05-13 11:28:04 +00:00

overrides

Further updates to fix libjingle logging.

2015-05-14 02:18:15 +00:00

p2p

Switch to use SHA-256 for certificates / fingerprints.

2015-05-12 01:32:22 +00:00

sound

Remove henrike@ from OWNERS

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

system_wrappers

Move rtc::AtomicOps to webrtc/base/atomicops.h.

2015-04-30 12:16:14 +00:00

test

Remove VideoEngine interfaces.

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

tools

Disables mic bump-up level if not built with chromium

2015-04-22 04:39:47 +00:00

video

Move IncomingVideoFrames to common_video/.

2015-05-13 11:28:04 +00:00

video_engine

Set msvs_error_on_missing_sources=1 in GYP_GENERATOR_FLAGS on Windows.

2015-05-13 13:00:11 +00:00

voice_engine

Propagating RTT from send-only channel to receive-only channel.

2015-05-13 12:14:36 +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

Let Chromium declare the mips_dsp_rev build variable.

2015-03-04 09:51:17 +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

Add a Config class interface to AudioProcessing for passing options.

2013-07-25 18:28:29 +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

Roll chromium_revision e144d30..6fdb142 (318658:318841) + remove OVERRIDE macro

2015-03-04 13:04:54 +00:00

video_decoder.h

Revert "Changed argument occurences of const I420VideoFrame* to const I420VideoFrame& and non-const I420VideoFrame& to I420VideoFrame*."

2015-03-16 13:48:18 +00:00

video_encoder.h

Dynamic resolution change for VP8 HW encode.

2015-04-21 22:29:53 +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

Use IncomingVideoStream in VideoReceiveStream.

2015-04-24 13:15:40 +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

GYP variables for building expat, icu, libsrtp, usrsctp

2015-04-27 12:38:37 +00:00

webrtc_perf_tests.isolate

Enable GoogleWifiTrace3Mbps simulations.

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

webrtc_tests.gypi

Add a BWE fairness test.

2015-04-16 18:20:26 +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.