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Stefan Holmer c482eb3c84 Don't account for audio in the pacer budget. 
We should only account for audio packets in the pacer budget if we also
are allocating bandwidth for the audio streams.

BUG=chromium:567659,webrtc:5263
R=mflodman@webrtc.org

Review URL: https://codereview.webrtc.org/1524763002 .

Cr-Commit-Position: refs/heads/master@{#11053}
2015-12-16 15:55:09 +00:00
..
api
Update API for Objective-C RTCIceServer
2015-12-14 03:58:19 +00:00
audio
Support for unmixed remote audio into tracks.
2015-12-12 00:37:14 +00:00
base
New macro: RTC_DEPRECATED (for annotating deprecated functions)
2015-12-16 09:09:25 +00:00
build
Revert of Add APK targets to build libjingle tests for Android. (patchset #10 id:180001 of https://codereview.webrtc.org/1511633002/ )
2015-12-14 12:31:22 +00:00
call
Replace EventWrapper in video/, test/ and call/.
2015-12-10 12:03:00 +00:00
common_audio
Add aecdump support to audioproc_f
2015-11-20 08:11:58 +00:00
common_video
Enable cpplint for more webrtc subfolders and fix all uncovered cpplint errors.
2015-12-10 11:11:48 +00:00
examples
Reland of Made EglBase an abstract class and cleaned up. (patchset #1 id:1 of https://codereview.webrtc.org/1522073002/ )
2015-12-15 10:48:13 +00:00
libjingle
cleanup: get rid of basicdefs.h include
2015-11-13 22:37:43 +00:00
modules
Don't account for audio in the pacer budget.
2015-12-16 15:55:09 +00:00
p2p
Only try to pair protocol matching candidates for creating connections.
2015-12-15 20:20:22 +00:00
sound
Enable cpplint for more webrtc subfolders and fix all uncovered cpplint errors.
2015-12-10 11:11:48 +00:00
system_wrappers
Remove unused and rarely used LOG_ macros.
2015-12-14 16:22:21 +00:00
test
Reland of Base webrtc fuzzers on a template. (patchset #1 id:1 of https://codereview.webrtc.org/1528043002/ )
2015-12-15 20:58:00 +00:00
tools
Enable cpplint for more webrtc subfolders and fix all uncovered cpplint errors.
2015-12-10 11:11:48 +00:00
video
Wire up bandwidth limitation info to GetStats and adapt_reason.
2015-12-14 10:08:19 +00:00
voice_engine
AudioCodingModuleImpl: Stop failing artificially for non-Opus encoders
2015-12-16 14:24:09 +00:00
.gitignore
Flesh out webrtc/.gitignore
2015-11-16 19:02:02 +00:00
audio_receive_stream.h
Support for unmixed remote audio into tracks.
2015-12-12 00:37:14 +00:00
audio_send_stream.h
Prepare the AudioSendStream to be hooked up to send-side BWE.
2015-12-07 09:26:32 +00:00
audio_state.h
Add comments about the Audio parts of the public Call API being WIP.
2015-12-03 12:06:31 +00:00
BUILD.gn
Create fuzzer tests for audio decoders
2015-12-08 10:27:34 +00:00
call.h
Move VoiceEngineObserver into AudioSendStream so that we detect typing noises and return properly in GetStats().
2015-11-06 23:34:58 +00:00
codereview.settings
Improve informative message in codereview.settings.
2015-11-10 10:08:36 +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
Revert of Make overuse estimator one dimensional. (patchset #5 id:80001 of https://codereview.webrtc.org/1376423002/ )
2015-11-27 09:02:35 +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
Prepare the AudioSendStream to be hooked up to send-side BWE.
2015-12-07 09:26:32 +00:00
engine_configurations.h
Remove VIDEOCODEC_* from engine_configurations.h.
2015-11-24 12:56:05 +00:00
frame_callback.h
Unify FrameType and VideoFrameType.
2015-10-19 09:39:15 +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
Roll chromium_revision 68cf0b8..aa8e58a (361406:361601)
2015-11-25 21:20:11 +00:00
transport.h
Reland of Merge webrtc/video_engine/ into webrtc/video/ (patchset #2 id:300001 of https://codereview.webrtc.org/1507903005/ )
2015-12-09 11:13:40 +00:00
typedefs.h
Update WARN_UNUSED_RESULT macro to match Chromium's version.
2015-12-12 13:47:24 +00:00
video_decoder.h
Add VideoCodec::PreferDecodeLate
2015-12-10 17:27:45 +00:00
video_encoder.h
Remove redudant encoder rate calls.
2015-10-29 15:30:29 +00:00
video_engine_tests.isolate
…
video_frame.h
common_video: rename interface -> include
2015-11-16 21:52:31 +00:00
video_receive_stream.h
Add VideoCodec::PreferDecodeLate
2015-12-10 17:27:45 +00:00
video_renderer.h
Reland of Merge webrtc/video_engine/ into webrtc/video/ (patchset #2 id:300001 of https://codereview.webrtc.org/1507903005/ )
2015-12-09 11:13:40 +00:00
video_send_stream.h
Wire up bandwidth limitation info to GetStats and adapt_reason.
2015-12-14 10:08:19 +00:00
webrtc_examples.gyp
Fix Mac ObjC PeerConnection API compilation.
2015-12-03 00:42:41 +00:00
webrtc_nonparallel_tests.isolate
Create isolate files for nonparallel tests.
2015-10-29 14:29:15 +00:00
webrtc_perf_tests.isolate
Enable GoogleWifiTrace3Mbps simulations.
2015-05-04 12:26:26 +00:00
webrtc_tests.gypi
Update API for Objective-C RTCIceServer
2015-12-14 03:58:19 +00:00
webrtc.gyp
Remove 'video_engine_core_unittests' binary.
2015-12-04 09:59:02 +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.