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deadbeef c80741f895 Fixing some issues with the direction attribute of m-lines in offers. 
By default, we'll now offer to receive if already receiving
(meaning that the last remote description contained a track).

Also, m-lines that are neither receiving nor sending are now correctly
marked "inactive".

Also moved some logic relating to default tracks out of webrtcsdp.cc,
such that now the direction seen by upper layers will always be
consistent with the consumed/produced SDP.

BUG=528089

Review URL: https://codereview.webrtc.org/1406803004

Cr-Commit-Position: refs/heads/master@{#10376}
2015-10-22 20:14:51 +00:00
..
audio
Re-Land: Implement AudioReceiveStream::GetStats().
2015-10-22 08:49:39 +00:00
base
Introduce rtc::Maybe<T>, which either contains a T or not.
2015-10-21 10:44:17 +00:00
build
Suppress libyuv::TestCpuFlag races.
2015-10-20 11:37:41 +00:00
call
Re-Land: Implement AudioReceiveStream::GetStats().
2015-10-22 08:49:39 +00:00
common_audio
Add RTC_ prefix to (D)CHECKs and related macros.
2015-09-17 07:24:51 +00:00
common_video
Add RTC_ prefix to (D)CHECKs and related macros.
2015-09-17 07:24:51 +00:00
examples
Create network change notifier and pass the event to NetworkManager
2015-10-19 16:39:38 +00:00
libjingle
Use suffixed {uint,int}{8,16,32,64}_t types.
2015-10-07 10:23:32 +00:00
modules
Prevent BWE rampdowns without new loss reports.
2015-10-22 15:52:28 +00:00
p2p
Fixing some issues with the direction attribute of m-lines in offers.
2015-10-22 20:14:51 +00:00
sound
Add RTC_ prefix to contructormagic macros.
2015-09-16 12:37:52 +00:00
system_wrappers
Add stats for average QP per frame for VP8 (for received video streams):
2015-10-21 06:55:32 +00:00
test
Re-Land: Implement AudioReceiveStream::GetStats().
2015-10-22 08:49:39 +00:00
tools
Fix minor GYP error in webrtc/tools/internal_tools.gyp
2015-10-07 12:51:40 +00:00
video
Allow encoders to fall back dynamically to software.
2015-10-22 06:54:57 +00:00
video_engine
Rename ChannelGroup to CongestionController and move to webrtc/call/.
2015-10-21 13:52:33 +00:00
voice_engine
Re-Land: Implement AudioReceiveStream::GetStats().
2015-10-22 08:49:39 +00:00
.gitignore
…
audio_receive_stream.h
Re-Land: Implement AudioReceiveStream::GetStats().
2015-10-22 08:49:39 +00:00
audio_send_stream.h
Add webrtc::AudioSendStream and methods on webrtc::Call to create and delete AudioSendStreams.
2015-10-16 21:35:11 +00:00
BUILD.gn
Update build files to use webrtc_overrides in Chromium instead of overrides.
2015-09-30 10:49:11 +00:00
call.h
Wire up packet_id / send time callbacks to webrtc via libjingle.
2015-10-15 14:26:17 +00:00
codereview.settings
Update lower-level codereview.settings files.
2015-10-06 21:19:34 +00:00
common_types.cc
Separating voice activity flag from audio level in RtpHeaderExtension.
2015-08-10 13:08:46 +00:00
common_types.h
Unify FrameType and VideoFrameType.
2015-10-19 09:39:15 +00:00
common.gyp
…
common.h
Add RTC_ prefix to contructormagic macros.
2015-09-16 12:37:52 +00:00
config.cc
Wire up packet_id / send time callbacks to webrtc via libjingle.
2015-10-15 14:26:17 +00:00
config.h
Revert of Adding support for simulcast and spatial layers into VideoQualityTest (patchset #10 id:180001 of https://codereview.webrtc.org/1353263005/ )
2015-10-12 13:33:24 +00:00
engine_configurations.h
Don't link with audio codecs that we don't use
2015-10-01 04:54:29 +00:00
frame_callback.h
Unify FrameType and VideoFrameType.
2015-10-19 09:39:15 +00:00
libjingle_examples.gyp
Wire up packet_id / send time callbacks to webrtc via libjingle.
2015-10-15 14:26:17 +00:00
LICENSE
…
LICENSE_THIRD_PARTY
…
OWNERS
…
PATENTS
…
PRESUBMIT.py
…
README.chromium
…
rtc_unittests.isolate
…
stream.h
Wire up PacketTime to ReceiveStreams.
2015-09-08 12:36:23 +00:00
supplement.gypi
…
transport.h
Wire up packet_id / send time callbacks to webrtc via libjingle.
2015-10-15 14:26:17 +00:00
typedefs.h
Use suffixed {uint,int}{8,16,32,64}_t types.
2015-10-07 10:23:32 +00:00
video_decoder.h
iOS HW H264 support.
2015-06-29 21:35:08 +00:00
video_encoder.h
Allow encoders to fall back dynamically to software.
2015-10-22 06:54:57 +00:00
video_engine_tests.isolate
…
video_frame.h
Add stats for average QP per frame for VP8 (for received video streams):
2015-10-21 06:55:32 +00:00
video_receive_stream.h
Fix RTPPayloadRegistry to correctly restore RTX, if a valid mapping exists.
2015-10-14 18:29:56 +00:00
video_renderer.h
Rename I420VideoFrame to VideoFrame.
2015-05-30 00:21:56 +00:00
video_send_stream.h
Reland of Collecting encode_time_ms for each frame (patchset #1 id:1 of https://codereview.webrtc.org/1383283005/ )
2015-10-07 09:43:25 +00:00
webrtc_examples.gyp
Adding support for OpenSL ES output in native WebRTC
2015-05-18 14:49:04 +00:00
webrtc_perf_tests.isolate
…
webrtc_tests.gypi
Add webrtc::AudioSendStream and methods on webrtc::Call to create and delete AudioSendStreams.
2015-10-16 21:35:11 +00:00
webrtc.gyp
Remove references to libpeerconnection.
2015-10-08 12:41:05 +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.