philipel 2c9f9f2bc9 Only create H264 frames if there are no gaps in the packet sequence number. ...

In the case of H264 we can't know which packet that is the fist packet of a
frame. In order to avoid creating incomplete frames we keep track of which
packets that we haven't received, and if there is a gap in the packet sequence
number leading up to this frame then a frame wont be created.

BUG=chromium:716558

Review-Url: https://codereview.webrtc.org/2926083002
Cr-Commit-Position: refs/heads/master@{#18559}

2017-06-13 09:47:28 +00:00

..

audio_coding

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_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

Fix play block size mismatch in Win audio device.

2017-06-12 23:54:07 +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

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

bitrate_controller

Merge BitrateControllerImpl::RtcpBandwidthObserverImpl into BitrateControllerImpl

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

congestion_controller

Reland Refactored incoming bitrate estimator.

2017-06-12 06:57:17 +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

Add SafeClamp(), which accepts args of different types

2017-06-12 18:40:47 +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

Remove outdated warning suppressions.

2017-06-07 11:20:02 +00:00

video_coding

Only create H264 frames if there are no gaps in the packet sequence number.

2017-06-13 09:47:28 +00:00

video_processing

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

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