platform-external-webrtc

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Sami Kalliomäki f6515cd0e3 Fix and optimize input buffer filling in HardwareVideoEncoder.

Previously input buffers would be filled incorrectly for sparsely
packed buffers where stride is not equal to the plane width.

Bug: webrtc:8478
Change-Id: I080fa3c354a27982bb996be8c1e41b103384e4bc
Reviewed-on: https://webrtc-review.googlesource.com/17321
Reviewed-by: Magnus Jedvert <magjed@webrtc.org>
Commit-Queue: Sami Kalliomäki <sakal@webrtc.org>
Cr-Commit-Position: refs/heads/master@{#20550}

2017-11-02 12:22:07 +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

Adds new DisableAndEnableAudioRecording integration test to Peerconnection.

2017-11-02 11:43:17 +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

Fix and optimize input buffer filling in HardwareVideoEncoder.

2017-11-02 12:22:07 +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

Create .git-blame-ignore-revs and add Java format CL to it.

2016-10-20 09:20:39 +00:00

.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

Update template to follow chromium copyright style

2013-04-24 01:01:28 +00:00

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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