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henrik.lundin e9eca8f5ae Removing AudioCoding class, a.k.a the new ACM API 
We have decided not to do a switch from old (AudioCodingModule) to new
(AudioCoding) API. Instead, we will gradually evolve the old API to
meet the new design goals.

As a consequence of this decision, the AudioCoding and AudioCodingImpl
classes are deleted. Also removing associated unit test sources. No
test coverage is lost with this operation, since the tests for the
"old" API are testing more than the deleted tests did.

BUG=webrtc:3520

Review URL: https://codereview.webrtc.org/1415163002

Cr-Commit-Position: refs/heads/master@{#10406}
2015-10-26 12:26:45 +00:00
..
audio
Re-Land: Implement AudioReceiveStream::GetStats().
2015-10-22 08:49:39 +00:00
base
Update Bind to match its comments and always capture by value. Also update the generated count to 9 args.
2015-10-24 18:14:52 +00:00
build
Suppress libyuv::TestCpuFlag races.
2015-10-20 11:37:41 +00:00
call
Set send times in send time history via OnSentPacket.
2015-10-23 09:05:43 +00:00
common_audio
Add RTC_ prefix to (D)CHECKs and related macros.
2015-09-17 07:24:51 +00:00
common_video
Add RTC_ prefix to (D)CHECKs and related macros.
2015-09-17 07:24:51 +00:00
examples
Android: Replace EGL14 with EGL10
2015-10-22 23:52:45 +00:00
libjingle
Use suffixed {uint,int}{8,16,32,64}_t types.
2015-10-07 10:23:32 +00:00
modules
Removing AudioCoding class, a.k.a the new ACM API
2015-10-26 12:26:45 +00:00
p2p
Update Bind to match its comments and always capture by value. Also update the generated count to 9 args.
2015-10-24 18:14:52 +00:00
sound
Add GN Build file for rtc_sound target.
2015-10-26 12:15:33 +00:00
system_wrappers
clean up field_trial_default target, to be used by remoting_perftests.
2015-10-23 02:33:22 +00:00
test
Remove VideoFrameType aliases for FrameType.
2015-10-23 13:58:27 +00:00
tools
Fix minor GYP error in webrtc/tools/internal_tools.gyp
2015-10-07 12:51:40 +00:00
video
Use webrtc/base/logging.h for video.
2015-10-26 10:35:26 +00:00
video_engine
Use webrtc/base/logging.h for video.
2015-10-26 10:35:26 +00:00
voice_engine
Re-Land: Implement AudioReceiveStream::GetStats().
2015-10-22 08:49:39 +00:00
.gitignore
.gitignore: Add *.mk, created as part of ChromiumOS build
2013-01-04 21:25:42 +00:00
audio_receive_stream.h
Re-Land: Implement AudioReceiveStream::GetStats().
2015-10-22 08:49:39 +00:00
audio_send_stream.h
Add webrtc::AudioSendStream and methods on webrtc::Call to create and delete AudioSendStreams.
2015-10-16 21:35:11 +00:00
BUILD.gn
Add GN Build file for rtc_sound target.
2015-10-26 12:15:33 +00:00
call.h
Wire up packet_id / send time callbacks to webrtc via libjingle.
2015-10-15 14:26:17 +00:00
codereview.settings
Update lower-level codereview.settings files.
2015-10-06 21:19:34 +00:00
common_types.cc
Separating voice activity flag from audio level in RtpHeaderExtension.
2015-08-10 13:08:46 +00:00
common_types.h
Remove VideoFrameType aliases for FrameType.
2015-10-23 13:58:27 +00:00
common.gyp
Fix style violations in common_types.h and config.h
2015-02-26 14:01:28 +00:00
common.h
Add RTC_ prefix to contructormagic macros.
2015-09-16 12:37:52 +00:00
config.cc
Wire up packet_id / send time callbacks to webrtc via libjingle.
2015-10-15 14:26:17 +00:00
config.h
Revert of Adding support for simulcast and spatial layers into VideoQualityTest (patchset #10 id:180001 of https://codereview.webrtc.org/1353263005/ )
2015-10-12 13:33:24 +00:00
engine_configurations.h
Don't link with audio codecs that we don't use
2015-10-01 04:54:29 +00:00
frame_callback.h
Unify FrameType and VideoFrameType.
2015-10-19 09:39:15 +00:00
libjingle_examples.gyp
Wire up packet_id / send time callbacks to webrtc via libjingle.
2015-10-15 14:26:17 +00:00
LICENSE
…
LICENSE_THIRD_PARTY
Add the Ooura FFT to RealFourier.
2015-03-19 20:07:43 +00:00
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
stream.h
Wire up PacketTime to ReceiveStreams.
2015-09-08 12:36:23 +00:00
supplement.gypi
Move target_subarch from gyp_webrtc to supplement.gypi
2015-04-16 07:24:23 +00:00
transport.h
Wire up packet_id / send time callbacks to webrtc via libjingle.
2015-10-15 14:26:17 +00:00
typedefs.h
Use suffixed {uint,int}{8,16,32,64}_t types.
2015-10-07 10:23:32 +00:00
video_decoder.h
iOS HW H264 support.
2015-06-29 21:35:08 +00:00
video_encoder.h
Allow encoders to fall back dynamically to software.
2015-10-22 06:54:57 +00:00
video_engine_tests.isolate
Update isolate files for Android APK tests.
2014-11-13 08:35:05 +00:00
video_frame.h
Remove VideoFrameType aliases for FrameType.
2015-10-23 13:58:27 +00:00
video_receive_stream.h
Fix RTPPayloadRegistry to correctly restore RTX, if a valid mapping exists.
2015-10-14 18:29:56 +00:00
video_renderer.h
Rename I420VideoFrame to VideoFrame.
2015-05-30 00:21:56 +00:00
video_send_stream.h
Reland of Collecting encode_time_ms for each frame (patchset #1 id:1 of https://codereview.webrtc.org/1383283005/ )
2015-10-07 09:43:25 +00:00
webrtc_examples.gyp
Adding support for OpenSL ES output in native WebRTC
2015-05-18 14:49:04 +00:00
webrtc_perf_tests.isolate
Enable GoogleWifiTrace3Mbps simulations.
2015-05-04 12:26:26 +00:00
webrtc_tests.gypi
Add webrtc::AudioSendStream and methods on webrtc::Call to create and delete AudioSendStreams.
2015-10-16 21:35:11 +00:00
webrtc.gyp
Remove references to libpeerconnection.
2015-10-08 12:41:05 +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.