webrtc_m130

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Magnus Jedvert 56231d07b3 Reland "Android: Generate JNI code for VideoSink and VideoEncoder"

This is a reland of ba78b5a905bffa05933a135673996df02328f2a4
Original change's description:
> Android: Generate JNI code for VideoSink and VideoEncoder
> 
> This is the first CL to start generating JNI code. It has updated two of
> the most recent classes to use JNI code generation.
> 
> Bug: webrtc:8278
> Change-Id: I1b19ee78c273346ceeaa0401dbdf8696803f16c7
> Reviewed-on: https://webrtc-review.googlesource.com/3820
> Reviewed-by: Sami Kalliomäki <sakal@webrtc.org>
> Commit-Queue: Magnus Jedvert <magjed@webrtc.org>
> Cr-Commit-Position: refs/heads/master@{#19994}

Bug: webrtc:8278
Change-Id: Id3e6513736eb87d7c234be3b0d13c5d30435201c
Reviewed-on: https://webrtc-review.googlesource.com/4500
Reviewed-by: Magnus Jedvert <magjed@webrtc.org>
Reviewed-by: Sami Kalliomäki <sakal@webrtc.org>
Commit-Queue: Magnus Jedvert <magjed@webrtc.org>
Cr-Commit-Position: refs/heads/master@{#20548}

2017-11-02 10:27:47 +00:00

api

Don't select audio codecs depending on GN vars build_with_{chromium|mozilla}

2017-11-01 18:59:27 +00:00

audio

Add SetAudioPlayout and SetAudioRecording methods to the PeerConnection API (II)

2017-11-01 11:04:26 +00:00

build_overrides

Add phoglund@ to various OWNERS and remove kjellander@

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

call

Add SetAudioPlayout and SetAudioRecording methods to the PeerConnection API (II)

2017-11-01 11:04:26 +00:00

common_audio

Replacing undefined left shifts with multiplication.

2017-10-31 09:43:02 +00:00

common_video

Remove pbos@webrtc.org from all OWNERS.

2017-11-01 08:03:46 +00:00

data

WebRTC: Replace ProjectRootPath by ResourcePath

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

examples

Fix/suppress new warnings introduced in Chromium roll.

2017-10-30 16:10:29 +00:00

infra

Link win_rel with LUCI win_rel.

2017-10-31 21:08:43 +00:00

logging

Remove redundant unit tests for RtcEventLog.

2017-11-01 16:09:16 +00:00

media

Add SetAudioPlayout and SetAudioRecording methods to the PeerConnection API (II)

2017-11-01 11:04:26 +00:00

modules

Don't select audio codecs depending on GN vars build_with_{chromium|mozilla}

2017-11-01 18:59:27 +00:00

ortc

Disable flaky test OrtcFactoryIntegrationTest.BasicTwoWayAudioVideoRtpSendersAndReceivers.

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

p2p

Enable the clang style plugin in primary p2p/ target

2017-11-01 00:19:05 +00:00

Add SetAudioPlayout and SetAudioRecording methods to the PeerConnection API (II)

2017-11-01 11:04:26 +00:00

resources

Fix of integer overflow in WebRtcAecm_ProcessBlock / ApmTest.Process

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

rtc_base

Add MovingMedianFilter to rtc_base/numerics

2017-11-01 13:56:16 +00:00

rtc_tools

Estimate RTP clock frequency and plot capture-send delay.

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

sdk

Reland "Android: Generate JNI code for VideoSink and VideoEncoder"

2017-11-02 10:27:47 +00:00

stats

Added RTCMediaStreamTrackStats.jitterBufferDelay for audio

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

system_wrappers

Remove references to and implementation of GetHistogramName().

2017-10-30 19:20:49 +00:00

test

Remove pbos@webrtc.org from all OWNERS.

2017-11-01 08:03:46 +00:00

tools_webrtc

Use urllib2 in roll_deps, which raises on non-200 HTTP status

2017-11-02 00:47:46 +00:00

video

Don't use old RTCP SR reports for remote clock estimation

2017-11-01 12:34:26 +00:00

voice_engine

Add SetAudioPlayout and SetAudioRecording methods to the PeerConnection API (II)

2017-11-01 11:04:26 +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 baae088436..f93b8b19f2 (513340:513366)

2017-11-02 02:24:36 +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

Enable cpplint in media/

2017-10-31 17:46:42 +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

Don't select audio codecs depending on GN vars build_with_{chromium|mozilla}

2017-11-01 18:59:27 +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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