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jmarusic@webrtc.org 13ca5f6db2 AudioEncoderOpus: CHECK that encode call doesn't fail 
WebRtcOpus_Encode will only ever fail if fed bad input, and since we don't do that, we can CHECK that it doesn't fail instead of having code that tries to handle failure.

R=kwiberg@webrtc.org

Review URL: https://webrtc-codereview.appspot.com/40899004

Cr-Commit-Position: refs/heads/master@{#8469}
git-svn-id: http://webrtc.googlecode.com/svn/trunk@8469 4adac7df-926f-26a2-2b94-8c16560cd09d
2015-02-24 09:57:18 +00:00
..
base
Roll chromium_revision b0c3ed3..2c3ffb2 (316737:317530)
2015-02-23 19:09:22 +00:00
build
Roll chromium_revision b0c3ed3..2c3ffb2 (316737:317530)
2015-02-23 19:09:22 +00:00
common_audio
Roll chromium_revision b0c3ed3..2c3ffb2 (316737:317530)
2015-02-23 19:09:22 +00:00
common_video
I420VideoFrame: Remove functions set_width, set_height, and ResetSize
2015-02-19 15:35:50 +00:00
examples
Refactoring WebRTC Java/JNI audio track in C++ and Java.
2015-02-23 11:54:41 +00:00
libjingle
Use std::min and std::max instead of self-defined functions such as rtc::_min/_max.
2015-02-12 11:55:32 +00:00
modules
AudioEncoderOpus: CHECK that encode call doesn't fail
2015-02-24 09:57:18 +00:00
overrides
Add arraysize to overrides to avoid macros redefinitions in Chromium
2015-02-14 02:46:13 +00:00
p2p
Change default from GICE to ICE5245 for SDP offers
2015-02-20 02:20:19 +00:00
sound
Use std::min and std::max instead of self-defined functions such as rtc::_min/_max.
2015-02-12 11:55:32 +00:00
system_wrappers
Copy SetThreadName from webrtc/base/thread.cc into thread_win.cc
2015-02-19 19:18:54 +00:00
test
Misc. cleanup split out of https://webrtc-codereview.appspot.com/37699004/ :
2015-02-23 21:29:45 +00:00
tools
Re-land "Remove <(webrtc_root) from source file entries."
2015-01-29 14:30:41 +00:00
video
Enable NACK under SendsAndReceivesH264.
2015-02-23 11:12:18 +00:00
video_engine
Misc. cleanup split out of https://webrtc-codereview.appspot.com/37699004/ :
2015-02-23 21:29:45 +00:00
voice_engine
Misc. cleanup split out of https://webrtc-codereview.appspot.com/37699004/ :
2015-02-23 21:29:45 +00:00
.gitignore
…
BUILD.gn
Roll chromium_revision b0c3ed3..2c3ffb2 (316737:317530)
2015-02-23 19:09:22 +00:00
call.h
Use int64_t more consistently for times, in particular for RTT values.
2015-01-12 21:51:21 +00:00
codereview.settings
Add codereview.settings to the /webrtc subdirectory
2014-12-05 13:43:35 +00:00
common_types.h
Add RtcpPacketTypeCounter stats to new API.
2015-02-19 12:47:45 +00:00
common.gyp
Make it easier to use external libyuv + cleanup GYP files.
2015-01-26 19:17:26 +00:00
common.h
Add a Config class interface to AudioProcessing for passing options.
2013-07-25 18:28:29 +00:00
config.cc
Revert 8028 "Support associated payload type when registering Rt..."
2015-01-09 20:22:46 +00:00
config.h
Add RtcpPacketTypeCounter stats to new API.
2015-02-19 12:47:45 +00:00
engine_configurations.h
Delete all codec-specific subclasses of ACMGenericCodec
2015-02-23 09:26:51 +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
…
LICENSE_THIRD_PARTY
Consolidate all third party licenses in LICENSE_THIRD_PARTY.
2013-05-05 18:54:10 +00:00
OWNERS
GN: Add BUILD.gn files + kjellander to OWNERS
2014-06-23 19:21:07 +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.isolate
Update all .isolate files for the new format.
2014-10-31 18:08:09 +00:00
supplement.gypi
Roll chromium_revision 601e6f3..b0c3ed3 (315263:316737)
2015-02-18 10:38:11 +00:00
transport.h
Rename newapi::Transport::SendRTP()->SendRtp().
2013-11-20 12:17:04 +00:00
typedefs.h
Enable Clang warning implicit-fallthrough and annotate the code.
2015-01-28 18:38:13 +00:00
video_decoder.h
Add support for VP9 in webrtc::Call and video_loopback.
2014-11-04 19:41:15 +00:00
video_encoder.h
Remove default arguments in EncodedImageCallback.
2015-02-09 09:14:48 +00:00
video_engine_tests.isolate
Update isolate files for Android APK tests.
2014-11-13 08:35:05 +00:00
video_frame.h
I420VideoFrame: Remove functions set_width, set_height, and ResetSize
2015-02-19 15:35:50 +00:00
video_receive_stream.h
Log configs when creating video streams in Call.
2015-01-15 10:09:39 +00:00
video_renderer.h
Use VideoReceiveStream as an ExternalRenderer.
2015-02-09 15:15:24 +00:00
video_send_stream.h
Log configs when creating video streams in Call.
2015-01-15 10:09:39 +00:00
webrtc_examples.gyp
Move internal capture+render to build_with_chromium==0 condition
2015-01-20 11:40:45 +00:00
webrtc_perf_tests.isolate
Offline screenshare quality test, plus loopback.
2015-02-18 12:46:44 +00:00
webrtc_tests.gypi
Offline screenshare quality test, plus loopback.
2015-02-18 12:46:44 +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.