webrtc

History

aluebs@webrtc.org d35a5c3506 Make ChannelBuffer aware of frequency bands

Now the ChannelBuffer has 2 separate arrays, one for the full-band data and one for the splitted one. The corresponding accessors are added to the ChannelBuffer.
This is done to avoid having to refresh the bands pointers in AudioBuffer. It will also allow us to have a general accessor like data()[band][channel][sample].
All the files using the ChannelBuffer needed to be re-factored.
Tested with modules_unittests, common_audio_unittests, audioproc, audioproc_f, voe_cmd_test.

R=andrew@webrtc.org, kwiberg@webrtc.org

Review URL: https://webrtc-codereview.appspot.com/36999004

Cr-Commit-Position: refs/heads/master@{#8318}
git-svn-id: http://webrtc.googlecode.com/svn/trunk@8318 4adac7df-926f-26a2-2b94-8c16560cd09d

2015-02-10 22:52:43 +00:00

base

base/arraysize.h: We use size_t, so need to include stddef.h

2015-02-10 22:47:14 +00:00

build

Restructure GYP for vp9, opus and direct trace

2015-02-09 10:01:40 +00:00

common_audio

Make ChannelBuffer aware of frequency bands

2015-02-10 22:52:43 +00:00

common_video

Cleanup: unify rotation to be enum based instead of int for degree.

2015-02-09 01:29:45 +00:00

examples

Add arraysize() macro from Chromium, and make use of it in a few places.

2015-01-28 21:37:13 +00:00

libjingle

Enable Clang warning implicit-fallthrough and annotate the code.

2015-01-28 18:38:13 +00:00

modules

Make ChannelBuffer aware of frequency bands

2015-02-10 22:52:43 +00:00

overrides

Clean kForever from basictypes and move it to the interfaces that actually have it.

2015-02-09 14:19:39 +00:00

p2p

Remove USE_WEBRTC_DEV_BRANCH.

2015-02-09 10:17:12 +00:00

sound

rtc_unittest: turned sound's test gyp into gypi to speed up GYP generation.

2014-10-01 16:33:03 +00:00

system_wrappers

Fix bug in EventPosix where we'd miss a set event.

2015-02-10 09:34:02 +00:00

test

Use VideoReceiveStream as an ExternalRenderer.

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

tools

Re-land "Remove <(webrtc_root) from source file entries."

2015-01-29 14:30:41 +00:00

video

Always use DeliverI420Frame in WebRtcVideoEngine.

2015-02-10 10:20:38 +00:00

video_engine

Always use DeliverI420Frame in WebRtcVideoEngine.

2015-02-10 10:20:38 +00:00

voice_engine

audio_processing: Now records mic volume level also when using new AGC

2015-02-06 19:44:46 +00:00

.gitignore

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

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

BUILD.gn

Make it easier to use external libyuv + cleanup GYP files.

2015-01-26 19:17:26 +00:00

call.h

Use int64_t more consistently for times, in particular for RTT values.

2015-01-12 21:51:21 +00:00

codereview.settings

Add codereview.settings to the /webrtc subdirectory

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

common_types.h

Add method for incrementing RtpPacketCounter. Removes duplicate code.

2015-02-04 08:35:21 +00:00

common.gyp

Make it easier to use external libyuv + cleanup GYP files.

2015-01-26 19:17:26 +00:00

common.h

Add a Config class interface to AudioProcessing for passing options.

2013-07-25 18:28:29 +00:00

config.cc

Revert 8028 "Support associated payload type when registering Rt..."

2015-01-09 20:22:46 +00:00

config.h

Revert 8028 "Support associated payload type when registering Rt..."

2015-01-09 20:22:46 +00:00

engine_configurations.h

Add VP9 codec to VCM and vie_auto_test.

2014-11-01 06:10:48 +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

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

rtc_unittests.isolate

Update all .isolate files for the new format.

2014-10-31 18:08:09 +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

Enable Clang warning implicit-fallthrough and annotate the code.

2015-01-28 18:38:13 +00:00

video_decoder.h

Add support for VP9 in webrtc::Call and video_loopback.

2014-11-04 19:41:15 +00:00

video_encoder.h

Remove default arguments in EncodedImageCallback.

2015-02-09 09:14:48 +00:00

video_engine_tests.isolate

Update isolate files for Android APK tests.

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

video_frame.h

Report encoded frame size in VideoSendStream.

2014-12-01 15:23:21 +00:00

video_receive_stream.h

Log configs when creating video streams in Call.

2015-01-15 10:09:39 +00:00

video_renderer.h

Use VideoReceiveStream as an ExternalRenderer.

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

video_send_stream.h

Log configs when creating video streams in Call.

2015-01-15 10:09:39 +00:00

webrtc_examples.gyp

Move internal capture+render to build_with_chromium==0 condition

2015-01-20 11:40:45 +00:00

webrtc_perf_tests.isolate

Update isolate files for Android APK tests.

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

webrtc_tests.gypi

Re-land "Remove <(webrtc_root) from source file entries."

2015-01-29 14:30:41 +00:00

webrtc.gyp

Make it easier to use external libyuv + cleanup GYP files.

2015-01-26 19:17:26 +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.