webrtc_m130

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Sami Kalliomäki ef5df1ae52 Fix WebSocketObserver getting garbage collected.

Apparently WebSocketObserver gets garbage collected if it is not stored
by us. This caused some external tests to break.

Bug: None
Change-Id: If62786e84f84a5a63172d67962bb4de8ae3e8479
Reviewed-on: https://webrtc-review.googlesource.com/16100
Reviewed-by: Magnus Jedvert <magjed@webrtc.org>
Commit-Queue: Sami Kalliomäki <sakal@webrtc.org>
Cr-Commit-Position: refs/heads/master@{#20449}

2017-10-26 14:24:06 +00:00

api

Hide the internal AudioEncoderOpus class by giving it an "Impl" suffix

2017-10-25 10:19:06 +00:00

audio

New classes RefCounter and RefCountedBase.

2017-10-23 11:46:47 +00:00

build_overrides

Add phoglund@ to various OWNERS and remove kjellander@

2017-10-19 09:21:12 +00:00

call

Reland "Add fine grained dropped video frames counters on sending side"

2017-10-25 09:32:15 +00:00

common_audio

Removing undefined left shifts in AudioProcessing

2017-10-25 13:35:36 +00:00

common_video

Use $rtc_libyuv_dir in common_video/BUILD.gn, not hard-encoded "libyuv"

2017-10-16 16:05:37 +00:00

data

WebRTC: Replace ProjectRootPath by ResourcePath

2016-11-22 18:43:05 +00:00

examples

Fix WebSocketObserver getting garbage collected.

2017-10-26 14:24:06 +00:00

infra

Adding win_more_configs to CQ

2017-10-25 11:51:36 +00:00

logging

Prevent unbounded memory consumption through RtcEventLogImpl::config_history_

2017-10-13 10:47:26 +00:00

media

Remove FlexFEC feedback params from internal encoder factory

2017-10-26 11:33:27 +00:00

modules

Improves native Android audio implementations.

2017-10-26 13:45:36 +00:00

ortc

Disable flaky test OrtcFactoryIntegrationTest.BasicTwoWayAudioVideoRtpSendersAndReceivers.

2017-09-27 09:14:28 +00:00

p2p

Enable the clang style plugin for the stunprober target

2017-10-26 04:40:25 +00:00

Fix flaky DataChannel integration test

2017-10-24 20:41:48 +00:00

resources

Fix of integer overflow in WebRtcAecm_ProcessBlock / ApmTest.Process

2017-10-23 14:25:37 +00:00

rtc_base

Move sequence_number_utils.h to rtc_base/

2017-10-25 12:33:57 +00:00

rtc_tools

Estimate RTP clock frequency and plot capture-send delay.

2017-10-26 08:42:54 +00:00

sdk

Revert "Simple Default ObjC video codec factories."

2017-10-26 11:55:57 +00:00

stats

Added RTCMediaStreamTrackStats.jitterBufferDelay for audio

2017-10-02 10:47:00 +00:00

system_wrappers

Switch metrics_default and field_trial_default to source_set.

2017-10-26 11:47:46 +00:00

test

Allow injection of custom network models in place of FakeNetworkPipe.

2017-10-26 11:11:25 +00:00

tools_webrtc

MB: Add Android Perf (swarming)

2017-10-23 13:02:07 +00:00

video

Delete VCMSendStatisticsCallback and corresponding use of ProcessThread

2017-10-26 08:13:55 +00:00

voice_engine

Let ChanneOwner use scoped_refptr and RefCountedBase.

2017-10-23 14:22:17 +00:00

.clang-format

Tune ObjC clang-format configuration

2017-05-11 09:14:18 +00:00

.git-blame-ignore-revs

…

.gitignore

MB: Add support for isolating scripts + isolate low_bandwidth_audio_test.py.

2017-10-02 16:57:09 +00:00

.gn

Remove remaining mentions of gflags

2017-09-25 15:34:41 +00:00

.vpython

Add psutil to vpython dependencies (used on builder bots)

2017-09-04 08:04:16 +00:00

AUTHORS

Reland of Fix the video buffer size should take rtt into consideration (patchset #2 id:160001 of https://codereview.chromium.org/3002033002/ )

2017-09-25 13:37:12 +00:00

BUILD.gn

Revert "Reland of Add full stack tests for MediaCodec encoder (moved from Rietveld)."

2017-09-29 13:48:29 +00:00

CODE_OF_CONDUCT.md

Add code of conduct to WebRTC repo

2017-05-16 12:09:13 +00:00

codereview.settings

Make Gerrit the default for WebRTC changes

2017-09-29 01:38:07 +00:00

common_types.cc

Adding NOLINT for typedefs.h and common_types.h

2017-09-15 13:03:51 +00:00

common_types.h

Set RTPVideoHeader picture id in PayloadRouter if forced fallback for VP8 is enabled.

2017-10-06 13:41:14 +00:00

DEPS

Roll chromium_revision 65e1b24a78..f4ecd4bed3 (508708:508787)

2017-10-16 11:00:27 +00:00

LICENSE

Moving src/webrtc into src/.

2017-09-15 04:25:06 +00:00

license_template.txt

…

LICENSE_THIRD_PARTY

Moving src/webrtc into src/.

2017-09-15 04:25:06 +00:00

OWNERS

Add phoglund@ to various OWNERS and remove kjellander@

2017-10-19 09:21:12 +00:00

PATENTS

Moving src/webrtc into src/.

2017-09-15 04:25:06 +00:00

presubmit_test_mocks.py

Using Change.BugsFromDescription to read CL bugs in PRESUBMIT checks.

2017-10-13 03:48:26 +00:00

presubmit_test.py

Using Change.BugsFromDescription to read CL bugs in PRESUBMIT checks.

2017-10-13 03:48:26 +00:00

PRESUBMIT.py

Presubmit: Don't forget to warn when changing headers in subdirs of api/

2017-10-25 11:53:16 +00:00

pylintrc

Removing invalid-name from disabled pylint checks.

2017-10-11 08:06:49 +00:00

README.chromium

Moving src/webrtc into src/.

2017-09-15 04:25:06 +00:00

README.md

Update README.md and codereview.settings for new source location

2017-09-13 19:54:59 +00:00

style-guide.md

Style guide: Attempt to make the L2 and L3 headings more visually distinct

2017-09-09 03:52:23 +00:00

typedefs.h

Fixing WebRTC after moving from src/webrtc to src/

2017-09-15 05:02:56 +00:00

WATCHLISTS

Add myself to the watchlist for webrtc/api/ and webrtc/base/

2017-05-04 13:22:46 +00:00

webrtc.gni

Remove QUIC transport/data channel

2017-10-24 16:14:18 +00:00

whitespace.txt

Moving src/webrtc into src/.

2017-09-15 04:25:06 +00:00

README.md

WebRTC is a free, open software project that provides browsers and mobile applications with Real-Time Communications (RTC) capabilities via simple APIs. The WebRTC components have been optimized to best serve this purpose.

Our mission: To enable rich, high-quality RTC applications to be developed for the browser, mobile platforms, and IoT devices, and allow them all to communicate via a common set of protocols.

The WebRTC initiative is a project supported by Google, Mozilla and Opera, amongst others.

Development

See http://www.webrtc.org/native-code/development for instructions on how to get started developing with the native code.

More info

Official web site: http://www.webrtc.org
Master source code repo: https://webrtc.googlesource.com/src
Samples and reference apps: https://github.com/webrtc
Mailing list: http://groups.google.com/group/discuss-webrtc
Continuous build: http://build.chromium.org/p/client.webrtc
Coding style guide
Code of conduct

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