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mgraczyk@chromium.org d6e84d9d13 Always copy processed audio to output buffer in ProcessStream. 
In the old AudioFrame ProcessStream API, input and output buffers were shared.
Now that the buffers are distinct, the input must be copied to the
output even when no processing occurred.

R=andrew@webrtc.org

Committed: https://code.google.com/p/webrtc/source/detail?r=78de5010d167d1e375e05d26177aad43c2e2de08

Review URL: https://webrtc-codereview.appspot.com/41459004

git-svn-id: http://webrtc.googlecode.com/svn/trunk@8052 4adac7df-926f-26a2-2b94-8c16560cd09d
2015-01-14 01:33:54 +00:00
..
base
Unify the two copies of template_util.h
2015-01-13 20:32:04 +00:00
build
Use Java based audio as default for WebRTC.
2015-01-12 14:27:23 +00:00
common_audio
Optimize minimum delay in blocker
2015-01-13 22:28:35 +00:00
common_video
Use size_t more consistently for packet/payload lengths.
2014-11-20 22:28:14 +00:00
examples
Use int64_t more consistently for times, in particular for RTT values.
2015-01-12 21:51:21 +00:00
libjingle
Added support for an Origin header in STUN messages.
2015-01-10 00:47:02 +00:00
modules
Always copy processed audio to output buffer in ProcessStream.
2015-01-14 01:33:54 +00:00
overrides
Remove or rename typedefs with _t prefixes.
2014-12-17 13:43:55 +00:00
p2p
Add support for trying alternate server (STUN 300 error message) on TCP
2015-01-10 02:41:32 +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
Unify the two copies of template_util.h
2015-01-13 20:32:04 +00:00
test
Use int64_t more consistently for times, in particular for RTT values.
2015-01-12 21:51:21 +00:00
tools
Merge in AGC manager and AGC tools.
2014-12-15 16:33:16 +00:00
video
Use int64_t more consistently for times, in particular for RTT values.
2015-01-12 21:51:21 +00:00
video_engine
Only return Rtx mode in RTXSendStatus().
2015-01-13 14:15:15 +00:00
voice_engine
Remove useless AudioProcessing::Create() overload.
2015-01-13 06:48:06 +00:00
.gitignore
…
BUILD.gn
Enabling building with NEON on ARM64
2014-11-26 17:01:40 +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
Refactor some receive-side stats.
2014-12-18 13:50:16 +00:00
common.gyp
Add ToString() to VideoSendStream::Config.
2014-05-15 09:35:06 +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
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
…
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
Rename newapi::Transport::SendRTP()->SendRtp().
2013-11-20 12:17:04 +00:00
typedefs.h
Annotate COMPILE_ASSERT with __attribute__((unused)).
2014-11-17 13:47:38 +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
Use int64_t more consistently for times, in particular for RTT values.
2015-01-12 21:51:21 +00:00
video_engine_tests.isolate
Update isolate files for Android APK tests.
2014-11-13 08:35:05 +00:00
video_frame.h
Report encoded frame size in VideoSendStream.
2014-12-01 15:23:21 +00:00
video_receive_stream.h
Wire up bandwidth stats to the new API and webrtcvideoengine2.
2014-11-05 14:05:29 +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
Enable payload-based padding by default and remove the API.
2014-12-19 15:33:17 +00:00
webrtc_examples.gyp
Add macros and APIs for webrtc histograms.
2014-10-23 12:57:56 +00:00
webrtc_perf_tests.isolate
Update isolate files for Android APK tests.
2014-11-13 08:35:05 +00:00
webrtc_tests.gypi
Add AGC manager tests.
2014-12-16 14:48:47 +00:00
webrtc.gyp
Merge in AGC manager and AGC tools.
2014-12-15 16:33:16 +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.