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webrtc_m130/webrtc
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Ivo Creusen adf89b7e33 Added SetBitRate function to VoE API to allow changing the audio bitrate. 
If the requested bitrate is not valid for the codec, the codec will decide on
an appropriate value.
Updated VoE command line tool to use new SetBitRate function.
Includes unittests for SetBitRate function.

BUG=
R=henrik.lundin@webrtc.org, henrika@webrtc.org, kwiberg@webrtc.org, pbos@webrtc.org

Review URL: https://webrtc-codereview.appspot.com/50789004

Cr-Commit-Position: refs/heads/master@{#9115}
2015-04-29 14:03:45 +00:00
..
base
Added a BitBufferWriter subclass that contains methods for writing bit and byte-sized data, along with exponential golomb encoded data.
2015-04-28 22:13:50 +00:00
build
GYP variables for building expat, icu, libsrtp, usrsctp
2015-04-27 12:38:37 +00:00
common_audio
Add 3-band filter-bank implementation
2015-04-27 18:34:16 +00:00
common_video
Remove i420_video_frame.h from common_video.gyp
2015-04-10 12:40:21 +00:00
examples/android
Use short include paths for icu headers.
2015-04-23 06:58:02 +00:00
libjingle
GYP variables for building expat, icu, libsrtp, usrsctp
2015-04-27 12:38:37 +00:00
modules
Added SetBitRate function to VoE API to allow changing the audio bitrate.
2015-04-29 14:03:45 +00:00
overrides
Cleanup: Remove DISALLOW_EVIL_CONSTRUCTORS macro.
2015-04-04 23:56:56 +00:00
p2p
Add safeguards against signalling peer-reflexive candidates.
2015-04-28 16:06:34 +00:00
sound
Remove henrike@ from OWNERS
2015-04-01 15:08:49 +00:00
system_wrappers
Set correct .type directive for asm functions.
2015-04-28 20:51:57 +00:00
test
Add AudioReceiveStream to Call API.
2015-04-29 13:24:10 +00:00
tools
Disables mic bump-up level if not built with chromium
2015-04-22 04:39:47 +00:00
video
Add AudioReceiveStream to Call API.
2015-04-29 13:24:10 +00:00
video_engine
Remove ViEImageProcess usage in VideoSendStream.
2015-04-29 12:08:49 +00:00
voice_engine
Added SetBitRate function to VoE API to allow changing the audio bitrate.
2015-04-29 14:03:45 +00:00
.gitignore
…
audio_receive_stream.h
Add AudioReceiveStream to Call API.
2015-04-29 13:24:10 +00:00
BUILD.gn
Let Chromium declare the mips_dsp_rev build variable.
2015-03-04 09:51:17 +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
…
config.cc
Add AudioReceiveStream to Call API.
2015-04-29 13:24:10 +00:00
config.h
Add AudioReceiveStream to Call API.
2015-04-29 13:24:10 +00:00
engine_configurations.h
Delete all codec-specific subclasses of ACMGenericCodec
2015-02-23 09:26:51 +00:00
experiments.h
…
frame_callback.h
Changing include guard in frame_callback.h.
2015-02-03 14:51:39 +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
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
…
typedefs.h
Roll chromium_revision e144d30..6fdb142 (318658:318841) + remove OVERRIDE macro
2015-03-04 13:04:54 +00:00
video_decoder.h
Revert "Changed argument occurences of const I420VideoFrame* to const I420VideoFrame& and non-const I420VideoFrame& to I420VideoFrame*."
2015-03-16 13:48:18 +00:00
video_encoder.h
Dynamic resolution change for VP8 HW encode.
2015-04-21 22:29:53 +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
GYP variables for building expat, icu, libsrtp, usrsctp
2015-04-27 12:38:37 +00:00
webrtc_perf_tests.isolate
Offline screenshare quality test, plus loopback.
2015-02-18 12:46:44 +00:00
webrtc_tests.gypi
Add a BWE fairness test.
2015-04-16 18:20:26 +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.