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asapersson da535c4055 Add histogram for percentage of sent frames that are limited in resolution due to bandwidth: 
- "WebRTC.Video.BandwidthLimitedResolutionInPercent"

If the frame is bandwidth limited, the average number of disabled resolutions is logged:
- "WebRTC.Video.BandwidthLimitedResolutionsDisabled"

BUG=

Review URL: https://codereview.webrtc.org/1311533012

Cr-Commit-Position: refs/heads/master@{#10333}
2015-10-20 06:32:48 +00:00
..
audio
Add webrtc::AudioSendStream and methods on webrtc::Call to create and delete AudioSendStreams.
2015-10-16 21:35:11 +00:00
base
Create network change notifier and pass the event to NetworkManager
2015-10-19 16:39:38 +00:00
build
Landmines support to ease clobbering builds
2015-10-15 12:24:29 +00:00
call
Remove the video channel id completely.
2015-10-19 15:12:19 +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
Add histogram for percentage of sent frames that are limited in resolution due to bandwidth:
2015-10-20 06:32:48 +00:00
p2p
Wire up packet_id / send time callbacks to webrtc via libjingle.
2015-10-15 14:26:17 +00:00
sound
Add RTC_ prefix to contructormagic macros.
2015-09-16 12:37:52 +00:00
system_wrappers
RefCountInterface: Make AddRef() and Release() const
2015-10-12 13:50:50 +00:00
test
Fix off-by-one error in PRNG.
2015-10-19 21:07:46 +00:00
tools
Fix minor GYP error in webrtc/tools/internal_tools.gyp
2015-10-07 12:51:40 +00:00
video
Add histogram for percentage of sent frames that are limited in resolution due to bandwidth:
2015-10-20 06:32:48 +00:00
video_engine
Remove the video channel id completely.
2015-10-19 15:12:19 +00:00
voice_engine
Fix RTPPayloadRegistry to correctly restore RTX, if a valid mapping exists.
2015-10-14 18:29:56 +00:00
.gitignore
…
audio_receive_stream.h
Align new VoE API with design.
2015-10-01 15:13:46 +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
Add the Ooura FFT to RealFourier.
2015-03-19 20:07:43 +00:00
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
Move target_subarch from gyp_webrtc to supplement.gypi
2015-04-16 07:24:23 +00:00
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
Unify FrameType and VideoFrameType.
2015-10-19 09:39:15 +00:00
video_engine_tests.isolate
…
video_frame.h
Add histogram for percentage of sent frames that are limited in resolution due to bandwidth:
2015-10-20 06:32:48 +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
Enable GoogleWifiTrace3Mbps simulations.
2015-05-04 12:26:26 +00:00
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.