webrtc_m130/webrtc/modules/audio_device/audio_device_generic.cc
henrika 9868042b05 Removes unused APIs from the ADM (part II).
Removes:

int32_t SpeakerVolumeStepSize(uint16_t* stepSize)
int32_t MicrophoneVolumeStepSize(uint16_t* stepSize)
int32_t MicrophoneBoostIsAvailable(bool* available)
int32_t SetMicrophoneBoost(bool enable)
int32_t MicrophoneBoost(bool* enabled)
int32_t SetPlayoutBuffer(const BufferType type, uint16_t sizeMS = 0)
int32_t PlayoutBuffer(BufferType* type, uint16_t* sizeMS)
int32_t CPULoad(uint16_t* load)
int32_t StartRawOutputFileRecording(const char pcmFileNameUTF8[kAdmMaxFileNameSize])
int32_t StopRawOutputFileRecording()
int32_t StartRawInputFileRecording(const char pcmFileNameUTF8[kAdmMaxFileNameSize])
int32_t StopRawInputFileRecording()
int32_t ResetAudioDevice()

BUG=webrtc:7306

Review-Url: https://codereview.webrtc.org/3006803002
Cr-Commit-Position: refs/heads/master@{#19632}
2017-08-31 13:47:32 +00:00

82 lines
2.2 KiB
C++

/*
* Copyright (c) 2011 The WebRTC project authors. All Rights Reserved.
*
* Use of this source code is governed by a BSD-style license
* that can be found in the LICENSE file in the root of the source
* tree. An additional intellectual property rights grant can be found
* in the file PATENTS. All contributing project authors may
* be found in the AUTHORS file in the root of the source tree.
*/
#include "webrtc/modules/audio_device/audio_device_generic.h"
#include "webrtc/rtc_base/logging.h"
namespace webrtc {
int32_t AudioDeviceGeneric::SetRecordingSampleRate(
const uint32_t samplesPerSec) {
LOG_F(LS_ERROR) << "Not supported on this platform";
return -1;
}
int32_t AudioDeviceGeneric::SetPlayoutSampleRate(const uint32_t samplesPerSec) {
LOG_F(LS_ERROR) << "Not supported on this platform";
return -1;
}
int32_t AudioDeviceGeneric::SetLoudspeakerStatus(bool enable) {
LOG_F(LS_ERROR) << "Not supported on this platform";
return -1;
}
int32_t AudioDeviceGeneric::GetLoudspeakerStatus(bool& enable) const {
LOG_F(LS_ERROR) << "Not supported on this platform";
return -1;
}
bool AudioDeviceGeneric::BuiltInAECIsAvailable() const {
LOG_F(LS_ERROR) << "Not supported on this platform";
return false;
}
int32_t AudioDeviceGeneric::EnableBuiltInAEC(bool enable) {
LOG_F(LS_ERROR) << "Not supported on this platform";
return -1;
}
bool AudioDeviceGeneric::BuiltInAGCIsAvailable() const {
LOG_F(LS_ERROR) << "Not supported on this platform";
return false;
}
int32_t AudioDeviceGeneric::EnableBuiltInAGC(bool enable) {
LOG_F(LS_ERROR) << "Not supported on this platform";
return -1;
}
bool AudioDeviceGeneric::BuiltInNSIsAvailable() const {
LOG_F(LS_ERROR) << "Not supported on this platform";
return false;
}
int32_t AudioDeviceGeneric::EnableBuiltInNS(bool enable) {
LOG_F(LS_ERROR) << "Not supported on this platform";
return -1;
}
#if defined(WEBRTC_IOS)
int AudioDeviceGeneric::GetPlayoutAudioParameters(
AudioParameters* params) const {
LOG_F(LS_ERROR) << "Not supported on this platform";
return -1;
}
int AudioDeviceGeneric::GetRecordAudioParameters(
AudioParameters* params) const {
LOG_F(LS_ERROR) << "Not supported on this platform";
return -1;
}
#endif // WEBRTC_IOS
} // namespace webrtc