webrtc_m130

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Niels Möller 90397d9aa9 Remove use of RTPFragmentationHeader from RTPSenderAudio

The RTPFragmentationHeader was used when sending audio using RED
for loss protection. This feature has been deprecated and
gradually removed. This cl removes remnants of support from
the RTP send path.

Bug: webrtc:6471
Change-Id: Ia1249047b09c16f79498827f74c2ce07aa38b8f7
Reviewed-on: https://webrtc-review.googlesource.com/16427
Commit-Queue: Niels Moller <nisse@webrtc.org>
Reviewed-by: Henrik Lundin <henrik.lundin@webrtc.org>
Cr-Commit-Position: refs/heads/master@{#20473}

2017-10-30 09:56:19 +00:00

api

Fix clang style warnings in api/candidate.h

2017-10-26 23:22:18 +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

Revert ObjC API changes for BWE allocation strategy

2017-10-27 12:33:24 +00:00

infra

Temporarily remove linux_ubsan from commit queue

2017-10-27 09:49:17 +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

Remove use of RTPFragmentationHeader from RTPSenderAudio

2017-10-30 09:56:19 +00:00

ortc

Disable flaky test OrtcFactoryIntegrationTest.BasicTwoWayAudioVideoRtpSendersAndReceivers.

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

p2p

Move inlined methods from p2p/base/packetsocketfactory.h

2017-10-26 23:14:37 +00:00

Reland: Reject the description with fewer m= sections.

2017-10-27 01:07:27 +00:00

resources

Fix of integer overflow in WebRtcAecm_ProcessBlock / ApmTest.Process

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

rtc_base

Adding RTC_ prefixed LOG macros.

2017-10-27 08:24:37 +00:00

rtc_tools

Estimate RTP clock frequency and plot capture-send delay.

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

sdk

Revert ObjC API changes for BWE allocation strategy

2017-10-27 12:33:24 +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 gradle to version 4.3-rc4.

2017-10-30 08:41:25 +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

.gn files in the API directories are part of the API too...

2017-10-29 03:04:33 +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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