webrtc

History

bjornv@webrtc.org bac0012120 Extend delay estimation window in AEC to 500 ms on all platforms

On non-Android the delay estimator in audio_processing/aec has solely been used for logging purposes. The maximum possible observed delay has been 236 ms. We have seen longer delays for which the delay estimate at best ends up at 236 ms, but can also be 'random'. reported delays are clamped to 500 ms.
This cl extends the delay estimation window to match that.

BUG=4086, 3504, 4113
TESTED=locally on Linux and trybots
R=kwiberg@webrtc.org

Review URL: https://webrtc-codereview.appspot.com/36569004

git-svn-id: http://webrtc.googlecode.com/svn/trunk@7989 4adac7df-926f-26a2-2b94-8c16560cd09d

2015-01-02 09:23:49 +00:00

base

Support for DTLS in OpenSSLAdapter

2014-12-29 23:00:14 +00:00

build

Do not use openmax_dl for MIPS64 platform.

2014-12-30 18:19:56 +00:00

common_audio

Minor fixes regarding accumulator usage on MIPS platforms.

2014-12-29 17:08:44 +00:00

common_video

Use size_t more consistently for packet/payload lengths.

2014-11-20 22:28:14 +00:00

examples

Update Android projects to API level 21.

2014-10-31 23:26:10 +00:00

libjingle

Move the Jingle-specific network code into webrtc/libjingle.

2014-12-23 22:14:15 +00:00

modules

Extend delay estimation window in AEC to 500 ms on all platforms

2015-01-02 09:23:49 +00:00

overrides

Remove or rename typedefs with _t prefixes.

2014-12-17 13:43:55 +00:00

p2p

Accept incoming pings before remote answer is set to reduce connection latency.

2014-12-29 19:23:37 +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

Unify the two copies of move.h

2014-12-19 14:35:57 +00:00

test

Use int64_t for milliseconds more often, primarily for TimeUntilNextProcess.

2014-12-15 22:09:40 +00:00

tools

Merge in AGC manager and AGC tools.

2014-12-15 16:33:16 +00:00

video

Fix the ramp-up-down-up test which was using ts-offset extension with the abs-send-time estimator.

2014-12-19 18:00:21 +00:00

video_engine

Remove unneccessary lock causing a potential deadlock.

2014-12-19 17:55:20 +00:00

voice_engine

Rtp-Rtcp sender cleanup.

2014-12-19 13:49:55 +00:00

.gitignore

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

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

BUILD.gn

Enabling building with NEON on ARM64

2014-11-26 17:01:40 +00:00

call.h

Wire up RTT statistics to webrtc::Call.

2014-12-11 13:26:09 +00:00

codereview.settings

Add codereview.settings to the /webrtc subdirectory

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

common_types.h

Refactor some receive-side stats.

2014-12-18 13:50:16 +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

Log formatting fix for VideoEncoderConfig.

2014-11-06 09:35:08 +00:00

config.h

Refactor some receive-side stats.

2014-12-18 13:50:16 +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

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

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

Annotate COMPILE_ASSERT with __attribute__((unused)).

2014-11-17 13:47:38 +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

Report encoded frame size in VideoSendStream.

2014-12-01 15:23:21 +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

Wire up bandwidth stats to the new API and webrtcvideoengine2.

2014-11-05 14:05:29 +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

Enable payload-based padding by default and remove the API.

2014-12-19 15:33:17 +00:00

webrtc_examples.gyp

Add macros and APIs for webrtc histograms.

2014-10-23 12:57:56 +00:00

webrtc_perf_tests.isolate

Update isolate files for Android APK tests.

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

webrtc_tests.gypi

Add AGC manager tests.

2014-12-16 14:48:47 +00:00

webrtc.gyp

Merge in AGC manager and AGC tools.

2014-12-15 16:33:16 +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.