philipel 112adf9ca9 Validate references of frames inserted into FrameBuffer2. ...

BUG=chromium:730603

Review-Url: https://codereview.webrtc.org/2937243002
Cr-Commit-Position: refs/heads/master@{#18614}

2017-06-15 16:06:21 +00:00

..

audio_coding

ACM: Make AcmReceiver's ownership of NetEq more obvious

2017-06-15 10:11:07 +00:00

audio_conference_mixer

Add mute state field to AudioFrame and switch some callers to use it. Also make AudioFrame::data_ private and instead provide:

2017-06-12 19:45:32 +00:00

audio_device

Adds detection of audio glitches for playout on iOS (reland)

2017-06-13 16:00:18 +00:00

audio_mixer

Add mute state field to AudioFrame and switch some callers to use it. Also make AudioFrame::data_ private and instead provide:

2017-06-12 19:45:32 +00:00

audio_processing

This CL finalizes the Conversational Speech tool.

2017-06-15 10:49:57 +00:00

bitrate_controller

Merge BitrateControllerImpl::RtcpBandwidthObserverImpl into BitrateControllerImpl

2017-06-12 15:21:59 +00:00

congestion_controller

Plot acknowledged bitrate when compiled with rtc_enable_bwe_test_logging.

2017-06-14 09:41:59 +00:00

desktop_capture

desktopCapture: scale the cursor image according to screen scale factor on OSX

2017-06-01 21:27:41 +00:00

include

Add mute state field to AudioFrame and switch some callers to use it. Also make AudioFrame::data_ private and instead provide:

2017-06-12 19:45:32 +00:00

media_file

Reland of Creating webrtc/modules:module_api (patchset #1 id:1 of https://codereview.webrtc.org/2839963005/ )

2017-04-26 10:38:35 +00:00

pacing

Address some violations of chromium-style.

2017-05-31 09:24:52 +00:00

remote_bitrate_estimator

Plot acknowledged bitrate when compiled with rtc_enable_bwe_test_logging.

2017-06-14 09:41:59 +00:00

rtp_rtcp

Change rtp header extension AbsoluteSendTime::Write to take time in 24bit format

2017-06-13 09:08:14 +00:00

utility

Remove deprecation warning from JVM::Initialize with the context parameter.

2017-06-01 11:59:52 +00:00

video_capture

Update webrtc/media and webrtc/modules to new VideoFrameBuffer interface

2017-06-14 13:46:38 +00:00

video_coding

Validate references of frames inserted into FrameBuffer2.

2017-06-15 16:06:21 +00:00

video_processing

Update webrtc/media and webrtc/modules to new VideoFrameBuffer interface

2017-06-14 13:46:38 +00:00

BUILD.gn

This is an initial cl, which contains small amount of implemented functions, and large amount of unimplemented ones.

2017-06-05 13:01:26 +00:00

module_common_types_unittest.cc

Add mute state field to AudioFrame and switch some callers to use it. Also make AudioFrame::data_ private and instead provide:

2017-06-12 19:45:32 +00:00

OWNERS

Remove all references to GYP

2016-11-16 19:11:38 +00:00