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henrik.lundin@webrtc.org 34509d9f33 Fix an issue with comfort noise in ACMGenericCodecWrapper 
In some cases it was not possible to set another payload type for CNG
than the default one. This CL fixes this. The problem was also
dependent on whether the comfort noise codec was registered before or
after the speech codec.

A test is implement to expose the bug, registering comfort noise at a
non-default payload type, and both before and after the speech codec.

BUG=4228
R=kwiberg@webrtc.org

Review URL: https://webrtc-codereview.appspot.com/35199004

Cr-Commit-Position: refs/heads/master@{#8380}
git-svn-id: http://webrtc.googlecode.com/svn/trunk@8380 4adac7df-926f-26a2-2b94-8c16560cd09d
2015-02-16 16:02:40 +00:00
..
base
Remove the incremental IP address behavior from virtualsocketserver
2015-02-14 00:44:20 +00:00
build
GN: Changes for vp9, opus and direct trace
2015-02-16 12:47:45 +00:00
common_audio
WebRtc_GetCPUFeaturesARM is only available on android
2015-02-11 17:03:24 +00:00
common_video
GN: Changes for vp9, opus and direct trace
2015-02-16 12:47:45 +00:00
examples
This is a code clean up. No functional change intended.
2015-02-13 14:32:13 +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
Fix an issue with comfort noise in ACMGenericCodecWrapper
2015-02-16 16:02:40 +00:00
overrides
Add arraysize to overrides to avoid macros redefinitions in Chromium
2015-02-14 02:46:13 +00:00
p2p
Use std::min and std::max instead of self-defined functions such as rtc::_min/_max.
2015-02-12 11:55:32 +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
GN: Changes for vp9, opus and direct trace
2015-02-16 12:47:45 +00:00
test
Remove EventWrapper::Reset().
2015-02-13 14:58:38 +00:00
tools
Re-land "Remove <(webrtc_root) from source file entries."
2015-01-29 14:30:41 +00:00
video
Remove EventWrapper::Reset().
2015-02-13 14:58:38 +00:00
video_engine
This is a code clean up. No functional change intended.
2015-02-13 14:32:13 +00:00
voice_engine
Remove EventWrapper::Reset().
2015-02-13 14:58:38 +00:00
.gitignore
…
BUILD.gn
Rename GYP and GN targets for video capture+render.
2015-02-11 07:47:47 +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 method for incrementing RtpPacketCounter. Removes duplicate code.
2015-02-04 08:35:21 +00:00
common.gyp
Make it easier to use external libyuv + cleanup GYP files.
2015-01-26 19:17:26 +00:00
common.h
…
config.cc
Revert 8028 "Support associated payload type when registering Rt..."
2015-01-09 20:22:46 +00:00
config.h
Revert 8028 "Support associated payload type when registering Rt..."
2015-01-09 20:22:46 +00:00
engine_configurations.h
Add VP9 codec to VCM and vie_auto_test.
2014-11-01 06:10:48 +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
…
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 289723:291647
2014-08-25 14:16:32 +00:00
transport.h
…
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
CVO capturer feature: allow unrotated frame flows through the capture pipeline.
2015-02-11 18:38:53 +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
Update isolate files for Android APK tests.
2014-11-13 08:35:05 +00:00
webrtc_tests.gypi
Rename GYP and GN targets for video capture+render.
2015-02-11 07:47:47 +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.