webrtc/modules/audio_device/linux/audio_mixer_manager_alsa_linux.h
Jonas Olsson a4d873786f Format almost everything.
This CL was generated by running

git ls-files | grep -P "(\.h|\.cc)$" | grep -v 'sdk/' | grep -v 'rtc_base/ssl_' | \
grep -v 'fake_rtc_certificate_generator.h' | grep -v 'modules/audio_device/win/' | \
grep -v 'system_wrappers/source/clock.cc' | grep -v 'rtc_base/trace_event.h' | \
grep -v 'modules/audio_coding/codecs/ilbc/' | grep -v 'screen_capturer_mac.h' | \
grep -v 'spl_inl_mips.h' | grep -v 'data_size_unittest.cc' | grep -v 'timestamp_unittest.cc' \
| xargs clang-format -i ; git cl format

Most of these changes are clang-format grouping and reordering includes
differently.

Bug: webrtc:9340
Change-Id: Ic83ddbc169bfacd21883e381b5181c3dd4fe8a63
Reviewed-on: https://webrtc-review.googlesource.com/c/src/+/144051
Commit-Queue: Jonas Olsson <jonasolsson@webrtc.org>
Reviewed-by: Karl Wiberg <kwiberg@webrtc.org>
Cr-Commit-Position: refs/heads/master@{#28505}
2019-07-08 13:45:15 +00:00

69 lines
2.4 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.
*/
#ifndef AUDIO_DEVICE_AUDIO_MIXER_MANAGER_ALSA_LINUX_H_
#define AUDIO_DEVICE_AUDIO_MIXER_MANAGER_ALSA_LINUX_H_
#include <alsa/asoundlib.h>
#include "modules/audio_device/include/audio_device.h"
#include "modules/audio_device/linux/alsasymboltable_linux.h"
#include "rtc_base/critical_section.h"
namespace webrtc {
class AudioMixerManagerLinuxALSA {
public:
int32_t OpenSpeaker(char* deviceName);
int32_t OpenMicrophone(char* deviceName);
int32_t SetSpeakerVolume(uint32_t volume);
int32_t SpeakerVolume(uint32_t& volume) const;
int32_t MaxSpeakerVolume(uint32_t& maxVolume) const;
int32_t MinSpeakerVolume(uint32_t& minVolume) const;
int32_t SpeakerVolumeIsAvailable(bool& available);
int32_t SpeakerMuteIsAvailable(bool& available);
int32_t SetSpeakerMute(bool enable);
int32_t SpeakerMute(bool& enabled) const;
int32_t MicrophoneMuteIsAvailable(bool& available);
int32_t SetMicrophoneMute(bool enable);
int32_t MicrophoneMute(bool& enabled) const;
int32_t MicrophoneVolumeIsAvailable(bool& available);
int32_t SetMicrophoneVolume(uint32_t volume);
int32_t MicrophoneVolume(uint32_t& volume) const;
int32_t MaxMicrophoneVolume(uint32_t& maxVolume) const;
int32_t MinMicrophoneVolume(uint32_t& minVolume) const;
int32_t Close();
int32_t CloseSpeaker();
int32_t CloseMicrophone();
bool SpeakerIsInitialized() const;
bool MicrophoneIsInitialized() const;
public:
AudioMixerManagerLinuxALSA();
~AudioMixerManagerLinuxALSA();
private:
int32_t LoadMicMixerElement() const;
int32_t LoadSpeakerMixerElement() const;
void GetControlName(char* controlName, char* deviceName) const;
private:
rtc::CriticalSection _critSect;
mutable snd_mixer_t* _outputMixerHandle;
char _outputMixerStr[kAdmMaxDeviceNameSize];
mutable snd_mixer_t* _inputMixerHandle;
char _inputMixerStr[kAdmMaxDeviceNameSize];
mutable snd_mixer_elem_t* _outputMixerElement;
mutable snd_mixer_elem_t* _inputMixerElement;
};
} // namespace webrtc
#endif // MODULES_AUDIO_DEVICE_MAIN_SOURCE_LINUX_AUDIO_MIXER_MANAGER_ALSA_LINUX_H_