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

..

audio_coding

Replace gflags usages with rtc_base/flags in all targets based on test_main

2017-08-29 12:51:57 +00:00

audio_conference_mixer

Remove remains of webrtc/base

2017-07-19 17:40:47 +00:00

audio_device

Removes unused WaveOut APIs from ADM.

2017-08-29 13:02:10 +00:00

audio_mixer

Remove remains of webrtc/base

2017-07-19 17:40:47 +00:00

audio_processing

Decrease the default reverberation in AEC3

2017-08-25 14:07:30 +00:00

bitrate_controller

Final version of BBR, with tweaks made for WebRTC, major changes:

2017-08-20 16:19:58 +00:00

congestion_controller

Move GetOutstandingBytes() call inside field trial check to avoid taking an unneccessary lock.

2017-08-26 13:33:43 +00:00

desktop_capture

Use windows' content rectangles rather than their borders to determine overlapping

2017-08-26 01:10:16 +00:00

include

Implement move constructor for RTPFragmentationHeader.

2017-08-28 10:22:39 +00:00

media_file

Replace WEBRTC_TRACE logging in modules/media_file/

2017-08-23 17:30:38 +00:00

pacing

Move PacedSender ownership to RtpTransportControllerSend.

2017-08-22 14:16:49 +00:00

remote_bitrate_estimator

Use small BWE period when there is a true network degradation.

2017-08-23 17:22:28 +00:00

rtp_rtcp

Reland of Verify sender ssrc when receiving rtcp target bitrate (patchset #1 id:1 of https://codereview.webrtc.org/3005633002/ )

2017-08-28 12:49:12 +00:00

utility

Now that https://codereview.webrtc.org/3003643002 is landed we can

2017-08-28 06:40:10 +00:00

video_capture

Replace old logging mechanism in webrtc/modules/video_capture/linux

2017-08-16 13:51:27 +00:00

video_coding

Verify parsed QP value when frame is encoded instead of storing parsed value and verifying later.

2017-08-29 12:39:36 +00:00

video_processing

Remove remains of webrtc/base

2017-07-19 17:40:47 +00:00

BUILD.gn

Remove remains of webrtc/base

2017-07-19 17:40:47 +00:00

module_common_types_unittest.cc

Add mute state field to AudioFrame and switch some callers to use it. Also make AudioFrame::data_ private and instead provide:

2017-06-12 19:45:32 +00:00

OWNERS

Remove all references to GYP

2016-11-16 19:11:38 +00:00