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henrik.lundin@webrtc.org 913f7b8d5e Fix for glitches in ACM when switching desired output sample rate 
The problem was that if the output sample rate is changed such from one
where no resampling is needed to a rate that requires resampling, the
first output from the resampler will contain an onset period. The
solution provided in this CL is to keep a copy of the last output frame
in ACM, and if the resampler is engaged, it will be primed with this
old frame before resampling the current frame.

BUG=3919
R=bjornv@webrtc.org, turaj@webrtc.org

Review URL: https://webrtc-codereview.appspot.com/27729004

git-svn-id: http://webrtc.googlecode.com/svn/trunk@7479 4adac7df-926f-26a2-2b94-8c16560cd09d
2014-10-21 06:54:23 +00:00
..
base
Fix data races in ThreadTest.ThreeThreadsInvoke.
2014-10-15 14:54:56 +00:00
build
Reverts r7459 "Create a copy of talk/xmpp and talk/p2p under webrtc/libjingle/xmpp and webrtc/p2p."
2014-10-17 22:03:39 +00:00
common_audio
common_audio: Replaced invalid operand in min_max_operations_neon.S"
2014-10-20 14:08:35 +00:00
common_video
GN: Add common configs to all targets.
2014-09-28 17:37:22 +00:00
examples
Split video engine android initialization into each internal module initialization.
2014-09-17 11:44:51 +00:00
libjingle/xmllite
Reverts r7459 "Create a copy of talk/xmpp and talk/p2p under webrtc/libjingle/xmpp and webrtc/p2p."
2014-10-17 22:03:39 +00:00
modules
Fix for glitches in ACM when switching desired output sample rate
2014-10-21 06:54:23 +00:00
overrides
webrtc/overrides: add OWNERS-file.
2014-09-17 08:04:28 +00:00
sound
rtc_unittest: turned sound's test gyp into gypi to speed up GYP generation.
2014-10-01 16:33:03 +00:00
system_wrappers
Import LappedTransform and friends.
2014-10-01 17:42:18 +00:00
test
GN: Add common configs to tools and test.
2014-09-30 19:07:58 +00:00
tools
Moving creating TURN configration to the host machine instead of the bots - rtcBot
2014-10-17 16:43:50 +00:00
video
Make avg_{psnr,ssim}_threshold_ const.
2014-10-20 09:14:38 +00:00
video_engine
Revert cls (original cl + fixes) 7422-7424 "Add VP9 codec to VCM..."
2014-10-17 18:54:46 +00:00
voice_engine
Mark all virtual overrides in the hierarchy of Transport as virtual + OVERRIDE.
2014-10-10 09:42:53 +00:00
.gitignore
.gitignore: Add *.mk, created as part of ChromiumOS build
2013-01-04 21:25:42 +00:00
BUILD.gn
GN: Enable libvpx, add link target and convert some test targets
2014-09-30 18:05:02 +00:00
call.h
Remove -1 from Call::Config::start_bitrate_bps.
2014-10-14 11:52:10 +00:00
common_types.h
Revert cls (original cl + fixes) 7422-7424 "Add VP9 codec to VCM..."
2014-10-17 18:54:46 +00:00
common.gyp
Add ToString() to VideoSendStream::Config.
2014-05-15 09:35:06 +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 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
Revert cls (original cl + fixes) 7422-7424 "Add VP9 codec to VCM..."
2014-10-17 18:54:46 +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
Move src/ -> webrtc/
2012-10-22 18:19:23 +00:00
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
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
Adds isolate for rtc_unittests and moves sound's unittests to rtc_unittest.
2014-09-30 14:21:10 +00:00
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
Revert cls (original cl + fixes) 7422-7424 "Add VP9 codec to VCM..."
2014-10-17 18:54:46 +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
Roll chromium_revision 6455c69..deaf2f7 (293954:295079)
2014-09-27 18:10:30 +00:00
webrtc_perf_tests.isolate
Pass GYP DEPTH variable to isolate.
2014-06-13 09:02:15 +00:00
webrtc_tests.gypi
Reverts r7459 "Create a copy of talk/xmpp and talk/p2p under webrtc/libjingle/xmpp and webrtc/p2p."
2014-10-17 22:03:39 +00:00
webrtc.gyp
Reverts r7459 "Create a copy of talk/xmpp and talk/p2p under webrtc/libjingle/xmpp and webrtc/p2p."
2014-10-17 22:03:39 +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.