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Alex Glaznev 2cc8baa144 Adjust the amount of VP8 encoder threads for Android builds. 
Current number of threads selection code does not work well
for Android builds - middle and low end devices are having hard time
encoding VGA and QVGA with just one thread.

Increase the amount of vp8 encoder threads for 180p and above resolution.

Also limit maximum number of thread to 3, since for 8 core devices
most of time 4 cores are idle when thermal throttling kicks in.

BUG=b/27946721
R=marpan@webrtc.org

Review URL: https://codereview.webrtc.org/2058753003 .

Cr-Commit-Position: refs/heads/master@{#13142}
2016-06-14 21:28:42 +00:00
..
androidjunit
Reland of Android: Change camera fps range selection (patchset #1 id:1 of https://codereview.webrtc.org/2021233002/ )
2016-06-01 07:44:07 +00:00
api
Forward the SignalFirstPacketReceived to RtpReceiver.
2016-06-14 18:47:20 +00:00
audio
Configure VoE NACK through AudioReceiveStream::Config, for receive streams. Also minor refactoring of WVoE unit test.
2016-06-14 19:13:07 +00:00
base
Add SigslotTester0 for testing signals without argument.
2016-06-14 19:49:54 +00:00
build
GN: Add modules_unittests
2016-06-13 07:19:53 +00:00
call
GN: Add video_engine_tests
2016-06-14 10:53:09 +00:00
common_audio
GN: Add common_audio_unittests and common_video_unittests
2016-06-08 08:28:15 +00:00
common_video
Split IncomingVideoStream into two implementations, with smoothing and without.
2016-06-14 11:38:43 +00:00
examples
A missing path separator caused aecdump recordings
2016-06-14 08:34:55 +00:00
libjingle
Adding more detail to MessageQueue::Dispatch logging.
2016-06-10 21:17:33 +00:00
media
Configure VoE NACK through AudioReceiveStream::Config, for receive streams. Also minor refactoring of WVoE unit test.
2016-06-14 19:13:07 +00:00
modules
Adjust the amount of VP8 encoder threads for Android builds.
2016-06-14 21:28:42 +00:00
p2p
Delete left-over files.
2016-06-14 08:54:52 +00:00
pc
Forward the SignalFirstPacketReceived to RtpReceiver.
2016-06-14 18:47:20 +00:00
sdk
Adding more detail to MessageQueue::Dispatch logging.
2016-06-10 21:17:33 +00:00
system_wrappers
GN: Fix errors in rtc_include_tests conditions
2016-06-06 21:04:16 +00:00
test
Configure VoE NACK through AudioSendStream::Config, for send streams.
2016-06-14 17:02:46 +00:00
tools
Delete some unused header files.
2016-06-10 11:41:13 +00:00
video
Split IncomingVideoStream into two implementations, with smoothing and without.
2016-06-14 11:38:43 +00:00
voice_engine
Voice Engine: Remove RED support
2016-06-14 18:21:51 +00:00
.gitignore
Flesh out webrtc/.gitignore
2015-11-16 19:02:02 +00:00
audio_receive_stream.h
Configure VoE NACK through AudioReceiveStream::Config, for receive streams. Also minor refactoring of WVoE unit test.
2016-06-14 19:13:07 +00:00
audio_send_stream.h
Configure VoE NACK through AudioSendStream::Config, for send streams.
2016-06-14 17:02:46 +00:00
audio_sink.h
Reland "Move webrtc/audio/audio_sink.h to webrtc/ and fix some dependencies."
2016-02-26 21:46:22 +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
GN: Add video_engine_tests
2016-06-14 10:53:09 +00:00
call.h
Movable support for VideoReceiveStream::Config and avoid copies.
2016-06-10 15:58:12 +00:00
codereview.settings
Give a more specific URL for creating WebRTC checkout
2016-04-01 07:36:53 +00:00
common_types.cc
Add sender controlled playout delay limits
2016-06-08 07:24:30 +00:00
common_types.h
Add sender controlled playout delay limits
2016-06-08 07:24:30 +00:00
common.gyp
Revert of Add missing headers and fix some missing dependencies (patchset #1 id:20001 of https://codereview.webrtc.org/1990593002/ )
2016-05-18 12:00:55 +00:00
common.h
#include "webrtc/base/constructormagic.h" where appropriate
2016-04-26 15:14:48 +00:00
config.cc
Configure VoE NACK through AudioSendStream::Config, for send streams.
2016-06-14 17:02:46 +00:00
config.h
Configure VoE NACK through AudioSendStream::Config, for send streams.
2016-06-14 17:02:46 +00:00
DEPS
Moved creation of AudioDecoderFactory to inside PeerConnectionFactory.
2016-06-13 14:35:01 +00:00
engine_configurations.h
Remove the VoEDtmf interface.
2016-03-08 12:11:00 +00:00
LICENSE
…
LICENSE_THIRD_PARTY
Add the Ooura FFT to RealFourier.
2015-03-19 20:07:43 +00:00
OWNERS
Add kwiberg@webrtc.org as root owner.
2016-06-14 09:39:40 +00:00
PATENTS
…
PRESUBMIT.py
PRESUBMIT.py: accept variants on the copyright message that are present in the codebase.
2014-05-23 17:27:18 +00:00
README.chromium
…
rtc_unittests_apk.isolate
Add isolate files for Android tests
2016-04-18 03:08:28 +00:00
rtc_unittests.isolate
Update all .isolate files for the new format.
2014-10-31 18:08:09 +00:00
supplement.gypi
Roll chromium_revision 086802955f..7fa6701bc5 (396351:398458)
2016-06-08 11:05:27 +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
Remove runtime NEON detection
2016-05-17 17:56:48 +00:00
video_decoder.h
Replace scoped_ptr with unique_ptr in webrtc/common_video/
2016-03-02 11:41:40 +00:00
video_encoder.h
Remove QualityScaler framerate reduction.
2016-04-26 11:37:20 +00:00
video_engine_tests_apk.isolate
Add isolate files for Android tests
2016-04-18 03:08:28 +00:00
video_engine_tests.isolate
Add audio streams to CallTest and a first A/V call test.
2016-01-07 16:43:31 +00:00
video_frame.h
Add sender controlled playout delay limits
2016-06-08 07:24:30 +00:00
video_receive_stream.h
Movable support for VideoReceiveStream::Config and avoid copies.
2016-06-10 15:58:12 +00:00
video_send_stream.h
Remove webrtc/stream.h and unutilized inheritance.
2016-05-02 03:18:36 +00:00
webrtc_examples.gyp
Remove metrics_default from rtc_media dependencies.
2016-06-01 12:28:57 +00:00
webrtc_nonparallel_tests_apk.isolate
Add isolate files for Android tests
2016-04-18 03:08:28 +00:00
webrtc_nonparallel_tests.isolate
Create isolate files for nonparallel tests.
2015-10-29 14:29:15 +00:00
webrtc_perf_tests_apk.isolate
Add isolate files for Android tests
2016-04-18 03:08:28 +00:00
webrtc_perf_tests.isolate
Enable GoogleWifiTrace3Mbps simulations.
2015-05-04 12:26:26 +00:00
webrtc_tests.gypi
GN: Add SDK tests to rtc_unittests.
2016-06-08 08:14:22 +00:00
webrtc.gyp
Remove webrtc_all target
2016-06-14 09:09:24 +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.