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andrew@webrtc.org 8328e7c44d Revert "Revert part of r7561, "Refactor audio conversion functions."" 
This restores the conversion changes to AudioProcessing originally
added in r7561, with minor alterations to ensure it passes all tests.

TBR=kwiberg

Review URL: https://webrtc-codereview.appspot.com/28899004

git-svn-id: http://webrtc.googlecode.com/svn/trunk@7574 4adac7df-926f-26a2-2b94-8c16560cd09d
2014-10-31 04:58:14 +00:00
..
base
Only configure the SSL library in one place.
2014-10-27 18:13:40 +00:00
build
arm64 iOS build.
2014-10-31 00:14:39 +00:00
common_audio
Revert "Revert part of r7561, "Refactor audio conversion functions.""
2014-10-31 04:58:14 +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
move xmpp and p2p to webrtc
2014-10-28 22:20:11 +00:00
modules
Revert "Revert part of r7561, "Refactor audio conversion functions.""
2014-10-31 04:58:14 +00:00
overrides
webrtc/overrides: add OWNERS-file.
2014-09-17 08:04:28 +00:00
p2p
move xmpp and p2p to webrtc
2014-10-28 22:20:11 +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
scoped_ptr.h: Renames function and change namespace scope to fix conflicts with Chromium not detected by the FYI bots.
2014-10-28 18:06:42 +00:00
test
Use external VideoDecoders in VideoReceiveStream.
2014-10-29 15:28:39 +00:00
tools
Adding the subtool rtcBot report visualizer
2014-10-24 09:26:16 +00:00
video
Use external VideoDecoders in VideoReceiveStream.
2014-10-29 15:28:39 +00:00
video_engine
Remove unused code in overuse detector.
2014-10-29 10:05:21 +00:00
voice_engine
Add a simple AudioConverter class.
2014-10-27 18:18:17 +00:00
.gitignore
…
BUILD.gn
arm64 iOS build.
2014-10-31 00:14:39 +00:00
call.h
Remove -1 from Call::Config::start_bitrate_bps.
2014-10-14 11:52:10 +00:00
common_types.h
Add stats for duplicate sent and received NACK requests.
2014-10-29 12:42:30 +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
Move min transmit bitrate to VideoEncoderConfig.
2014-10-24 09:23:21 +00:00
config.h
Move min transmit bitrate to VideoEncoderConfig.
2014-10-24 09:23:21 +00:00
engine_configurations.h
Enable G.722 for Chromium builds
2014-10-29 08:32:44 +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
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_decoder.h
Use external VideoDecoders in VideoReceiveStream.
2014-10-29 15:28:39 +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
Use external VideoDecoders in VideoReceiveStream.
2014-10-29 15:28:39 +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
Move min transmit bitrate to VideoEncoderConfig.
2014-10-24 09:23:21 +00:00
webrtc_examples.gyp
Add macros and APIs for webrtc histograms.
2014-10-23 12:57:56 +00:00
webrtc_perf_tests.isolate
Pass GYP DEPTH variable to isolate.
2014-06-13 09:02:15 +00:00
webrtc_tests.gypi
move xmpp and p2p to webrtc
2014-10-28 22:20:11 +00:00
webrtc.gyp
move xmpp and p2p to webrtc
2014-10-28 22:20:11 +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.