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philipel aee3e0eb32 Only advance |first_seq_num_| if packets are explicitly cleared from the PacketBuffer. 
In this CL:
 - Don't insert a packet if we have explicitly cleared past it.
 - Added some logging to ExpandBufferSize.
 - Renamed IsContinuous to PotentialNewFrame.
 - Unittests updated/added for this new behavior.
 - Refactored TestPacketBuffer unittests.

BUG=webrtc:5514
R=danilchap@webrtc.org, stefan@webrtc.org

Review URL: https://codereview.webrtc.org/2399373002 .

Cr-Commit-Position: refs/heads/master@{#14871}
2016-11-01 10:45:43 +00:00
..
androidjunit
Format all Java in WebRTC.
2016-09-29 11:12:51 +00:00
api
RTCInboundRTPStreamStats[1] added.
2016-11-01 10:00:24 +00:00
audio
Clean up abs-send-time for audio.
2016-11-01 10:17:18 +00:00
base
Re-enable PostDelayed test for TaskQueue on Windows.
2016-10-31 09:17:17 +00:00
build
Setting PATH so that the 'plistbuddy' utility can be found, in a typical OS X environment
2016-10-31 06:09:50 +00:00
call
Remove voe::Channel::StopReceive() and associated logic.
2016-10-31 19:52:39 +00:00
common_audio
GN: Exclude suppressions of Chromium Clang warnings for Chromium builds.
2016-10-17 06:56:20 +00:00
common_video
Delete ShallowCopy, in favor of copy construction and assignment.
2016-10-31 15:05:58 +00:00
examples
Moving stun_prober target from webrtc/p2p to webrtc/examples
2016-11-01 10:09:19 +00:00
libjingle
GN: Exclude suppressions of Chromium Clang warnings for Chromium builds.
2016-10-17 06:56:20 +00:00
logging
GN: New conventions, default target and refactorings
2016-10-28 12:44:07 +00:00
media
Clean up abs-send-time for audio.
2016-11-01 10:17:18 +00:00
modules
Only advance |first_seq_num_| if packets are explicitly cleared from the PacketBuffer.
2016-11-01 10:45:43 +00:00
p2p
Moving stun_prober target from webrtc/p2p to webrtc/examples
2016-11-01 10:09:19 +00:00
pc
Introduce rtc::PacketTransportInterface and let cricket::TransportChannel inherit.
2016-10-25 17:50:41 +00:00
sdk
GN: New conventions, default target and refactorings
2016-10-28 12:44:07 +00:00
stats
RTCInboundRTPStreamStats[1] added.
2016-11-01 10:00:24 +00:00
system_wrappers
Delete DataLog abstraction, which was almost unused.
2016-10-21 08:53:01 +00:00
test
Clean up abs-send-time for audio.
2016-11-01 10:17:18 +00:00
tools
GN: New conventions, default target and refactorings
2016-10-28 12:44:07 +00:00
video
Add ability to handle data from multiple streams in RateAccCounter.
2016-11-01 07:21:41 +00:00
voice_engine
Clean up abs-send-time for audio.
2016-11-01 10:17:18 +00:00
.gitignore
…
BUILD.gn
GN: New conventions, default target and refactorings
2016-10-28 12:44:07 +00:00
call.h
Integrate FlexfecReceiveStream with Call.
2016-10-24 06:37:24 +00:00
codereview.settings
Give a more specific URL for creating WebRTC checkout
2016-04-01 07:36:53 +00:00
common_types.cc
Refactoring: Hide VideoCodec.codecSpecific as "private"
2016-10-25 16:05:15 +00:00
common_types.h
Add functionality for parsing H264 profile-level-id
2016-10-31 12:56:03 +00:00
common.gyp
Delete old video defines in engine config.
2016-10-07 05:07:36 +00:00
config.cc
Clean up logging in AudioSendStream::SetupSendCodec().
2016-10-25 18:19:11 +00:00
config.h
Clean up logging in AudioSendStream::SetupSendCodec().
2016-10-25 18:19:11 +00:00
DEPS
Delete old video defines in engine config.
2016-10-07 05:07:36 +00:00
LICENSE
…
LICENSE_THIRD_PARTY
Delete talk directory, and references from build_ios_libs.sh.
2016-09-02 10:07:13 +00:00
OWNERS
OWNERS: Make everyone able to change *.gn,*.gni files.
2016-09-09 12:51:48 +00:00
PATENTS
…
PRESUBMIT.py
…
README.chromium
…
supplement.gypi
Cleanup MIPS specific link configuration
2016-06-20 05:39:25 +00:00
transport.h
…
typedefs.h
[MIPS] Fix build issue for mips64el
2016-06-23 10:52:35 +00:00
video_decoder.h
Refactoring: Hide VideoCodec.codecSpecific as "private"
2016-10-25 16:05:15 +00:00
video_encoder.h
Refactoring: Hide VideoCodec.codecSpecific as "private"
2016-10-25 16:05:15 +00:00
video_frame.h
Delete ShallowCopy, in favor of copy construction and assignment.
2016-10-31 15:05:58 +00:00
video_receive_stream.h
Implement framesDecoded stat in video receive ssrc stats.
2016-10-26 14:09:29 +00:00
video_send_stream.h
Implement qpSum stat for video send ssrc stats.
2016-10-31 13:53:51 +00:00
voice_engine_configurations.h
Delete old video defines in engine config.
2016-10-07 05:07:36 +00:00
webrtc_examples.gyp
Moving stun_prober target from webrtc/p2p to webrtc/examples
2016-11-01 10:09:19 +00:00
webrtc_tests.gypi
GYP: Remove targets inside include_tests==1 that are converted to GN.
2016-09-15 11:57:39 +00:00
webrtc.gyp
Clean up logging in AudioSendStream::SetupSendCodec().
2016-10-25 18:19:11 +00:00

README.chromium 

Name: WebRTC
URL: http://www.webrtc.org
Version: 90
License: BSD
License File: LICENSE

Description:
WebRTC provides real time voice and video processing
functionality to enable the implementation of 
PeerConnection/MediaStream.

Third party code used in this project is described 
in the file LICENSE_THIRD_PARTY.