Removes deprecated ADM APIs (reland)

Usage should now be removed and this change can be relanded.
It was reverted here: https://webrtc-review.googlesource.com/c/src/+/27200

NOTRY=TRUE
TBR=solenberg

Bug: webrtc:7306
Change-Id: I5191263e6cfd48952b59ff8f9af2e59c3e9eadef
Reviewed-on: https://webrtc-review.googlesource.com/29682
Commit-Queue: Henrik Andreassson <henrika@webrtc.org>
Reviewed-by: Henrik Andreassson <henrika@webrtc.org>
Cr-Commit-Position: refs/heads/master@{#21080}
This commit is contained in:
henrika 2017-12-05 11:23:00 +01:00 committed by Commit Bot
parent c3da1e61bc
commit f1978e5d1a

View File

@ -1,4 +1,4 @@
/*
/*
* Copyright (c) 2012 The WebRTC project authors. All Rights Reserved.
*
* Use of this source code is governed by a BSD-style license
@ -142,31 +142,10 @@ class AudioDeviceModule : public rtc::RefCountInterface {
virtual int32_t StereoRecordingIsAvailable(bool* available) const = 0;
virtual int32_t SetStereoRecording(bool enable) = 0;
virtual int32_t StereoRecording(bool* enabled) const = 0;
// TODO(bugs.webrtc.org/7306): deprecated.
virtual int32_t SetRecordingChannel(const ChannelType channel) { return -1; }
virtual int32_t RecordingChannel(ChannelType* channel) const { return -1; }
// Playout delay
virtual int32_t PlayoutDelay(uint16_t* delayMS) const = 0;
// TODO(bugs.webrtc.org/7306): deprecated (to be removed).
virtual int32_t SetRecordingSampleRate(const uint32_t samplesPerSec) {
return -1;
}
virtual int32_t RecordingSampleRate(uint32_t* samplesPerSec) const {
return -1;
}
virtual int32_t SetPlayoutSampleRate(const uint32_t samplesPerSec) {
return -1;
}
virtual int32_t PlayoutSampleRate(uint32_t* samplesPerSec) const {
return -1;
}
// TODO(bugs.webrtc.org/7306): deprecated (to be removed).
virtual int32_t SetLoudspeakerStatus(bool enable) { return -1; }
virtual int32_t GetLoudspeakerStatus(bool* enabled) const { return -1; }
// Only supported on Android.
virtual bool BuiltInAECIsAvailable() const = 0;
virtual bool BuiltInAGCIsAvailable() const = 0;