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Peter Boström 4d71edef45 Add HW fallback option to software encoding. 
Permits falling back to software encoding for unsupported resolutions.

BUG=chromium:475116, chromium:487934
R=mflodman@webrtc.org, stefan@webrtc.org

Review URL: https://webrtc-codereview.appspot.com/46279004

Cr-Commit-Position: refs/heads/master@{#9227}
2015-05-19 21:09:17 +00:00
..
base
Disable the EXPECT_DEATH check in bitbuffer on Android
2015-05-19 20:17:40 +00:00
build
Adding support for OpenSL ES output in native WebRTC
2015-05-18 14:49:04 +00:00
common_audio
Add 3-band filter-bank implementation
2015-04-27 18:34:16 +00:00
common_video
Move IncomingVideoFrames to common_video/.
2015-05-13 11:28:04 +00:00
examples/android/media_demo
Adding support for OpenSL ES output in native WebRTC
2015-05-18 14:49:04 +00:00
libjingle
GYP variables for building expat, icu, libsrtp, usrsctp
2015-04-27 12:38:37 +00:00
modules
Remove AudioDeviceUtility.
2015-05-19 15:27:50 +00:00
overrides
Further updates to fix libjingle logging.
2015-05-14 02:18:15 +00:00
p2p
Address the corner cases
2015-05-19 04:36:06 +00:00
sound
Remove henrike@ from OWNERS
2015-04-01 15:08:49 +00:00
system_wrappers
Move SetCurrentThreadName to platform_thread.* in rtc_base_approved,
2015-05-18 11:50:31 +00:00
test
Remove VideoEngine interfaces.
2015-05-12 14:51:08 +00:00
tools
Disables mic bump-up level if not built with chromium
2015-04-22 04:39:47 +00:00
video
Add HW fallback option to software encoding.
2015-05-19 21:09:17 +00:00
video_engine
Remove some dead code in ViEChannel.
2015-05-18 09:48:17 +00:00
voice_engine
Adding support for OpenSL ES output in native WebRTC
2015-05-18 14:49:04 +00:00
.gitignore
.gitignore: Add *.mk, created as part of ChromiumOS build
2013-01-04 21:25:42 +00:00
audio_receive_stream.h
Add AudioReceiveStream to Call API.
2015-04-29 13:24:10 +00:00
BUILD.gn
Adding support for OpenSL ES output in native WebRTC
2015-05-18 14:49:04 +00:00
call.h
Add AudioReceiveStream to Call API.
2015-04-29 13:24:10 +00:00
codereview.settings
Add codereview.settings to the /webrtc subdirectory
2014-12-05 13:43:35 +00:00
common_types.cc
Parsing of transport wide sequence number rtp extension header.
2015-03-17 14:33:46 +00:00
common_types.h
Parsing of transport wide sequence number rtp extension header.
2015-03-17 14:33:46 +00:00
common.gyp
Fix style violations in common_types.h and config.h
2015-02-26 14:01:28 +00:00
common.h
Add a Config class interface to AudioProcessing for passing options.
2013-07-25 18:28:29 +00:00
config.cc
Add AudioReceiveStream to Call API.
2015-04-29 13:24:10 +00:00
config.h
Reland r9159 "Adding a new constraint to set NetEq buffer capacity ..."
2015-05-11 10:44:20 +00:00
engine_configurations.h
Remove VideoEngine interfaces.
2015-05-12 14:51:08 +00:00
experiments.h
Remove no longer used SkipEncodingUnusedStreams.
2014-07-22 07:17:17 +00:00
frame_callback.h
Changing include guard in frame_callback.h.
2015-02-03 14:51:39 +00:00
LICENSE
Move src/ -> webrtc/
2012-10-22 18:19:23 +00:00
LICENSE_THIRD_PARTY
Add the Ooura FFT to RealFourier.
2015-03-19 20:07:43 +00:00
OWNERS
GN: Add BUILD.gn files + kjellander to OWNERS
2014-06-23 19:21:07 +00:00
PATENTS
Move src/ -> webrtc/
2012-10-22 18:19:23 +00:00
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
Move src/ -> webrtc/
2012-10-22 18:19:23 +00:00
rtc_unittests.isolate
Update all .isolate files for the new format.
2014-10-31 18:08:09 +00:00
supplement.gypi
Move target_subarch from gyp_webrtc to supplement.gypi
2015-04-16 07:24:23 +00:00
transport.h
Rename newapi::Transport::SendRTP()->SendRtp().
2013-11-20 12:17:04 +00:00
typedefs.h
Roll chromium_revision e144d30..6fdb142 (318658:318841) + remove OVERRIDE macro
2015-03-04 13:04:54 +00:00
video_decoder.h
Add HW fallback option to software decoding.
2015-05-18 17:41:50 +00:00
video_encoder.h
Add HW fallback option to software encoding.
2015-05-19 21:09:17 +00:00
video_engine_tests.isolate
Update isolate files for Android APK tests.
2014-11-13 08:35:05 +00:00
video_frame.h
Reland "Remove usage of webrtc::NativeHandle since is just adds an extra level of indirection.""
2015-04-09 11:44:19 +00:00
video_receive_stream.h
Use IncomingVideoStream in VideoReceiveStream.
2015-04-24 13:15:40 +00:00
video_renderer.h
Use VideoReceiveStream as an ExternalRenderer.
2015-02-09 15:15:24 +00:00
video_send_stream.h
Reland "Make the entry point for VideoFrames to webrtc const ref I420VideoFrame.""
2015-03-18 09:51:44 +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 HW fallback option to software encoding.
2015-05-19 21:09:17 +00:00
webrtc.gyp
Rename GYP and GN targets for video capture+render.
2015-02-11 07:47:47 +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.