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}
82 lines
2.2 KiB
C++
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
|