webrtc_m130

History

aluebs@webrtc.org 021e76fd39 Add support for WAV output in audioproc

The default output is a WAV file, except if the --pcm_output flag is set.

BUG=webrtc:3359
R=bjornv@webrtc.org, kwiberg@webrtc.org

Review URL: https://webrtc-codereview.appspot.com/18359004

git-svn-id: http://webrtc.googlecode.com/svn/trunk@7069 4adac7df-926f-26a2-2b94-8c16560cd09d

2014-09-04 18:12:00 +00:00

base

Change gflags and gmock includes to be full paths.

2014-09-04 09:55:40 +00:00

build

Partial revert of r7014 (Android APK refactor)

2014-09-03 14:34:46 +00:00

common_audio

Fix MSVC warnings about value truncations, webrtc/common_audio/ edition.

2014-09-04 13:21:44 +00:00

common_video

Partial revert of r7014 (Android APK refactor)

2014-09-03 14:34:46 +00:00

examples

Remove former team members from OWNERS and WATCHLISTS

2014-08-26 06:12:08 +00:00

libjingle/xmllite

Fix strange owners files with comments that crashs "git cl presubmit"

2014-09-04 14:25:07 +00:00

modules

Add support for WAV output in audioproc

2014-09-04 18:12:00 +00:00

overrides/webrtc/base

Rename webrtc/base's IS_ALIGNED macro to RTC_IS_ALIGNED to avoid conflict between webrtc/base/basictypes.h and third_party/.../vpx_codec.h.

2014-05-21 16:52:14 +00:00

sound

Add talk owners to migrated talk folders

2014-08-28 16:03:58 +00:00

system_wrappers

Partial revert of r7014 (Android APK refactor)

2014-09-03 14:34:46 +00:00

test

Fix audio/video sync when FEC is enabled.

2014-09-04 06:48:14 +00:00

tools

Moving the api.js and bot.js to /rtcbot/bot/ to be shared between

2014-09-04 13:44:47 +00:00

video

Change return value for number of discarded packets to be int.

2014-09-04 07:07:44 +00:00

video_engine

Change return value for number of discarded packets to be int.

2014-09-04 07:07:44 +00:00

voice_engine

Partial revert of r7014 (Android APK refactor)

2014-09-03 14:34:46 +00:00

.gitignore

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

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

BUILD.gn

[MIPS] Fix gn gen failure for MIPS in webrtc

2014-09-04 14:10:49 +00:00

call.h

Network up/down signaling in Call.

2014-09-03 16:17:12 +00:00

common_types.h

Move pacer to fully use webrtc::Clock instead of webrtc::TickTime.

2014-07-11 13:44:02 +00:00

common.gyp

Add ToString() to VideoSendStream::Config.

2014-05-15 09:35:06 +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 ToString() to VideoSendStream::Config.

2014-05-15 09:35:06 +00:00

config.h

Check before send/receive rtp header extensions.

2014-07-20 15:27:35 +00:00

engine_configurations.h

Add boilerplate code for H.264.

2014-07-04 12:42:07 +00:00

experiments.h

Remove no longer used SkipEncodingUnusedStreams.

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

frame_callback.h

Wire up statistics in video receive stream of new API

2014-02-07 12:06:29 +00:00

LICENSE

Move src/ -> webrtc/

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

LICENSE_THIRD_PARTY

Consolidate all third party licenses in LICENSE_THIRD_PARTY.

2013-05-05 18:54:10 +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

supplement.gypi

Roll chromium_revision 289723:291647

2014-08-25 14:16:32 +00:00

transport.h

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

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

typedefs.h

Add CHECK and friends from Chromium.

2014-08-28 16:28:26 +00:00

video_engine_tests.isolate

Pass GYP DEPTH variable to isolate.

2014-06-13 09:02:15 +00:00

video_receive_stream.h

Change return value for number of discarded packets to be int.

2014-09-04 07:07:44 +00:00

video_renderer.h

Separate Call API/build files from video_engine/.

2013-10-28 16:32:01 +00:00

video_send_stream.h

Move pacer to fully use webrtc::Clock instead of webrtc::TickTime.

2014-07-11 13:44:02 +00:00

webrtc_examples.gyp

WebRTCDemo: set local SSRC for loopback test, otherwise receiver would reset it due to ssrc clash, which would cause delayed remote rendering.

2014-07-03 05:59:22 +00:00

webrtc_perf_tests.isolate

Pass GYP DEPTH variable to isolate.

2014-06-13 09:02:15 +00:00

webrtc_tests.gypi

Partial revert of r7014 (Android APK refactor)

2014-09-03 14:34:46 +00:00

webrtc.gyp

Create a copy of talk/xmllite under webrtc/xmllite.

2014-09-02 15:41:12 +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.