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henrik.lundin@webrtc.org cfe073539c Convert AcmReceiverTest to new AudioCoding interface 
In order to maintain test coverage for the old API (AudioCodingModule)
during the transition period, the old test was copied to
AcmReceiverTestOldApi.

Modified and extended AudioCoding and the implementation to make the
test compile and run.

Created a converter method from new to old config struct

BUG=3520
R=kwiberg@webrtc.org

Review URL: https://webrtc-codereview.appspot.com/31409004

git-svn-id: http://webrtc.googlecode.com/svn/trunk@7259 4adac7df-926f-26a2-2b94-8c16560cd09d
2014-09-22 12:10:44 +00:00
..
base
Disabled several rtc_unittests so the tests can be turned on in the waterfall
2014-09-22 07:30:48 +00:00
build
Split video_render_module implementation into default and internal implementation.
2014-09-18 08:58:15 +00:00
common_audio
gn: Fix cflags usage
2014-09-16 17:57:02 +00:00
common_video
Revert 7151 "Revert 7114 "Expose VideoEncoders with webrtc/video_encoder.h.""
2014-09-17 09:02:25 +00:00
examples
Split video engine android initialization into each internal module initialization.
2014-09-17 11:44:51 +00:00
libjingle/xmllite
Put base tests in webrtc_tests.gyp
2014-09-10 17:28:19 +00:00
modules
Convert AcmReceiverTest to new AudioCoding interface
2014-09-22 12:10:44 +00:00
overrides
webrtc/overrides: add OWNERS-file.
2014-09-17 08:04:28 +00:00
sound
Put base tests in webrtc_tests.gyp
2014-09-10 17:28:19 +00:00
system_wrappers
Add a target for the approved subset of rtc_base.
2014-09-16 01:03:29 +00:00
test
Config struct for VideoEncoder.
2014-09-19 12:30:25 +00:00
tools
Fix printing of error stack in rtcbot when a test fails via test.fail().
2014-09-10 14:35:35 +00:00
video
Set number of temporal layers for VideoSendStream.
2014-09-22 09:32:46 +00:00
video_engine
Split video_render_module implementation into default and internal implementation.
2014-09-18 08:58:15 +00:00
voice_engine
Calculating round-trip-time in send-only channel in VoE.
2014-09-11 07:51:53 +00:00
.gitignore
…
BUILD.gn
Fix the RTC+Chromium GN build.
2014-09-09 19:15:33 +00:00
call.h
Config struct for VideoEncoder.
2014-09-19 12:30:25 +00:00
common_types.h
Move pacer to fully use webrtc::Clock instead of webrtc::TickTime.
2014-07-11 13:44:02 +00:00
common.gyp
Add ToString() to VideoSendStream::Config.
2014-05-15 09:35:06 +00:00
common.h
…
config.cc
Add ToString() to VideoSendStream::Config.
2014-05-15 09:35:06 +00:00
config.h
Config struct for VideoEncoder.
2014-09-19 12:30:25 +00:00
engine_configurations.h
Add boilerplate code for H.264.
2014-07-04 12:42:07 +00:00
experiments.h
Remove no longer used SkipEncodingUnusedStreams.
2014-07-22 07:17:17 +00:00
frame_callback.h
Wire up statistics in video receive stream of new API
2014-02-07 12:06:29 +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
…
supplement.gypi
Roll chromium_revision 289723:291647
2014-08-25 14:16:32 +00:00
transport.h
Rename newapi::Transport::SendRTP()->SendRtp().
2013-11-20 12:17:04 +00:00
typedefs.h
Add CHECK and friends from Chromium.
2014-08-28 16:28:26 +00:00
video_encoder.h
Expose VP8/H264 defaults through video_encoder.h.
2014-09-18 12:42:28 +00:00
video_engine_tests.isolate
Pass GYP DEPTH variable to isolate.
2014-06-13 09:02:15 +00:00
video_frame.h
Revert 7151 "Revert 7114 "Expose VideoEncoders with webrtc/video_encoder.h.""
2014-09-17 09:02:25 +00:00
video_receive_stream.h
Change return value for number of discarded packets to be int.
2014-09-04 07:07:44 +00:00
video_renderer.h
Separate Call API/build files from video_engine/.
2013-10-28 16:32:01 +00:00
video_send_stream.h
Config struct for VideoEncoder.
2014-09-19 12:30:25 +00:00
webrtc_examples.gyp
Split video_render_module implementation into default and internal implementation.
2014-09-18 08:58:15 +00:00
webrtc_perf_tests.isolate
Pass GYP DEPTH variable to isolate.
2014-06-13 09:02:15 +00:00
webrtc_tests.gypi
Remove the 'webrtc_test_video_render_dependencies' target.
2014-09-18 17:22:18 +00:00
webrtc.gyp
Split video_render_module implementation into default and internal implementation.
2014-09-18 08:58:15 +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.