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henrika ecf312e603 Removes unused WaveOut APIs from ADM. 
Will remove default implementations as well once landed and removed
in Chrome as well.

These two AudioDeviceModule APIs are removed:

int32_t SetWaveOutVolume(uint16_t volumeLeft, uint16_t volumeRight)
int32_t WaveOutVolume(uint16_t* volumeLeft, uint16_t* volumeRight) const

BUG=webrtc:7306

Review-Url: https://codereview.webrtc.org/3006793002
Cr-Commit-Position: refs/heads/master@{#19581}
2017-08-29 13:02:10 +00:00
..
api
Wiring discard rate of audio FEC/RED packets up to StatsReport.
2017-08-28 13:17:55 +00:00
audio
Replace gflags usages with rtc_base/flags in all targets based on test_main
2017-08-29 12:51:57 +00:00
call
Wiring discard rate of audio FEC/RED packets up to StatsReport.
2017-08-28 13:17:55 +00:00
common_audio
Replace absolute path with relative path for GN files.
2017-08-15 15:57:36 +00:00
common_video
Change logging method from WEBRTC_TRACE macro to LOG macro.
2017-08-22 07:29:13 +00:00
examples
Android: Replace webrtc_jni namespace with nested jni namespace
2017-08-28 23:24:06 +00:00
logging
Revert of Add logging of host lookups made by TurnPort to the RtcEventLog. (patchset #11 id:200001 of https://codereview.webrtc.org/2996933003/ )
2017-08-29 11:49:00 +00:00
media
Let VideoEncoderSoftwareFallbackWrapper own the wrapped encoder
2017-08-29 07:58:52 +00:00
modules
Removes unused WaveOut APIs from ADM.
2017-08-29 13:02:10 +00:00
ortc
Wire up RTP keep-alive in ortc api.
2017-08-09 13:42:32 +00:00
p2p
Revert of Add logging of host lookups made by TurnPort to the RtcEventLog. (patchset #11 id:200001 of https://codereview.webrtc.org/2996933003/ )
2017-08-29 11:49:00 +00:00
pc
Removes unused WaveOut APIs from ADM.
2017-08-29 13:02:10 +00:00
rtc_base
Revert of Add logging of host lookups made by TurnPort to the RtcEventLog. (patchset #11 id:200001 of https://codereview.webrtc.org/2996933003/ )
2017-08-29 11:49:00 +00:00
rtc_tools
Revert of Add logging of host lookups made by TurnPort to the RtcEventLog. (patchset #11 id:200001 of https://codereview.webrtc.org/2996933003/ )
2017-08-29 11:49:00 +00:00
sdk
ObjC: Add support for injectable native audio and video codecs
2017-08-29 10:57:22 +00:00
stats
Add stats totalSamplesReceived and concealedSamples
2017-08-25 00:50:42 +00:00
system_wrappers
Replace absolute path with relative path for GN files.
2017-08-15 15:57:36 +00:00
test
Replace gflags usages with rtc_base/flags in all targets based on test_main
2017-08-29 12:51:57 +00:00
video
Replace gflags usages with rtc_base/flags in all targets based on test_main
2017-08-29 12:51:57 +00:00
voice_engine
Replace gflags usages with rtc_base/flags in all targets based on test_main
2017-08-29 12:51:57 +00:00
.gitignore
…
BUILD.gn
Removing EXPAT_RELATIVE_PATH and rtc_relative_path.
2017-08-25 15:14:04 +00:00
codereview.settings
…
common_types.cc
Revert of Remove deprecated CodecType methods. (patchset #1 id:1 of https://codereview.webrtc.org/3009583002/ )
2017-08-28 14:16:43 +00:00
common_types.h
Revert of Remove deprecated CodecType methods. (patchset #1 id:1 of https://codereview.webrtc.org/3009583002/ )
2017-08-28 14:16:43 +00:00
config_unittest.cc
Rename webrtc/base -> webrtc/rtc_base
2017-07-06 19:11:40 +00:00
config.cc
Audit of kConstants missing the const qualifier
2017-08-08 17:48:15 +00:00
config.h
Audit of kConstants missing the const qualifier
2017-08-08 17:48:15 +00:00
DEPS
Removing video_{receive,send}_stream.h stub headers.
2017-08-29 10:35:02 +00:00
LICENSE
…
LICENSE_THIRD_PARTY
Delete unused spreadsort implementation.
2016-12-05 11:03:26 +00:00
OWNERS
Don't add or rename files in webrtc/ and webrtc/api/ without a proper review
2017-05-08 13:01:54 +00:00
PATENTS
…
PRESUBMIT.py
…
README.chromium
…
typedefs.h
Make WARN_UNUSED_RESULT a no-op on gcc
2017-04-06 11:32:27 +00:00
webrtc.gni
Removing EXPAT_RELATIVE_PATH and rtc_relative_path.
2017-08-25 15:14:04 +00:00
whitespace.txt
Moving whitespace file up by one folder
2017-01-24 13:19:42 +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.