webrtc

History

magjed@webrtc.org 73d763e71f Add I420 buffer pool to avoid unnecessary allocations

Now when we don't use SwapFrame consistently anymore, we need to recycle allocations with a buffer pool instead. This CL adds a buffer pool class, and updates the vp8 decoder to use it. If this CL lands successfully I will update the other video producers as well.

BUG=1128
R=stefan@webrtc.org, tommi@webrtc.org

Review URL: https://webrtc-codereview.appspot.com/41189004

Cr-Commit-Position: refs/heads/master@{#8748}
git-svn-id: http://webrtc.googlecode.com/svn/trunk@8748 4adac7df-926f-26a2-2b94-8c16560cd09d

2015-03-17 11:41:15 +00:00

base

Chromium build fix: Include new .cc files in rtc_base

2015-03-10 01:00:17 +00:00

build

Remove libvpx suppression due to fixed bug.

2015-03-16 20:43:47 +00:00

common_audio

Clean up LappedTransform and Blocker.

2015-03-12 23:24:19 +00:00

common_video

Add I420 buffer pool to avoid unnecessary allocations

2015-03-17 11:41:15 +00:00

examples

Roll chromium_revision 00e438c..8d51d96 (320241:320682)

2015-03-16 09:00:41 +00:00

libjingle

Use base/scoped_ptr.h; system_wrappers/interface/scoped_ptr.h is going away

2015-02-26 14:43:50 +00:00

modules

Add I420 buffer pool to avoid unnecessary allocations

2015-03-17 11:41:15 +00:00

overrides

Re-landing perf improvement for libjingle logging after reverting the general change.

2015-03-07 12:18:14 +00:00

p2p

Remove a dependency of BaseChannel on WebRtcSession by having WebRtcSession push down new media descriptions to BaseChannel rather than having BaseChannel listen to the description changes from WebRtcSession.

2015-03-16 21:16:23 +00:00

sound

Use std::min and std::max instead of self-defined functions such as rtc::_min/_max.

2015-02-12 11:55:32 +00:00

system_wrappers

Change ThreadPosix to use an auto-reset event instead of manual reset now that we know the problem we had with EventWrapper::Wait was simply a bug in the EventWrapper. Also removing |started_| since we can just check the thread_ instead.

2015-03-16 16:06:16 +00:00

test

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

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

tools

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

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

video

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_engine

2015-03-16 16:06:16 +00:00

voice_engine

Supporting Opus DTX in Voice Engine.

2015-03-13 09:38:55 +00:00

.gitignore

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

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

BUILD.gn

Let Chromium declare the mips_dsp_rev build variable.

2015-03-04 09:51:17 +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.cc

Remove DCHECK from common_types.cc

2015-03-04 20:10:38 +00:00

common_types.h

Add CVO support to RTP sender side.

2015-03-04 22:55:43 +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

Fix style violations in common_types.h and config.h

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

config.h

Add CVO support to Vie layer.

2015-03-12 20:51:50 +00:00

engine_configurations.h

Delete all codec-specific subclasses of ACMGenericCodec

2015-02-23 09:26:51 +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

Remove webrtc/system_wrappers/interface/scoped_ptr.h

2015-03-03 10:17:49 +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 601e6f3..b0c3ed3 (315263:316737)

2015-02-18 10:38:11 +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

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

I420VideoFrame.CreateFrame: Removed unnecessary buffer size arguments.

2015-03-16 13:26:41 +00:00

video_receive_stream.h

Add decoder-timing stats to VideoReceiveStream.

2015-02-25 10:42:45 +00:00

video_renderer.h

Use VideoReceiveStream as an ExternalRenderer.

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

video_send_stream.h

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

2015-03-10 15:13:13 +00:00

webrtc_examples.gyp

Improve cleaning for Android demo applications

2015-02-28 11:18:20 +00:00

webrtc_perf_tests.isolate

Offline screenshare quality test, plus loopback.

2015-02-18 12:46:44 +00:00

webrtc_tests.gypi

rtc_unittests on Android

2015-03-04 08:48:17 +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.