mirror of
https://github.com/mollyim/webrtc.git
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2efe87f8f6
This reverts commit 1882d8509a.
Reason for revert: Broke the internal projects.
Original change's description:
> Removes deprecated ADM APIs.
>
> Final stage since these APIs are no longer used in Chrome.
>
> Bug: webrtc:7306
> Change-Id: Ia116671bc888daa75c4105ad1ebeb21833f5d090
> Reviewed-on: https://webrtc-review.googlesource.com/25220
> Reviewed-by: Fredrik Solenberg <solenberg@webrtc.org>
> Commit-Queue: Henrik Andreassson <henrika@webrtc.org>
> Cr-Commit-Position: refs/heads/master@{#20836}
TBR=solenberg@webrtc.org,henrika@webrtc.org
Change-Id: If8dd4212fb3e8c6944643d3794f673837977bf4e
No-Presubmit: true
No-Tree-Checks: true
No-Try: true
Bug: webrtc:7306
Reviewed-on: https://webrtc-review.googlesource.com/25320
Reviewed-by: Zhi Huang <zhihuang@webrtc.org>
Commit-Queue: Zhi Huang <zhihuang@webrtc.org>
Cr-Commit-Position: refs/heads/master@{#20840}
192 lines
6.8 KiB
C++
192 lines
6.8 KiB
C++
/*
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* Copyright (c) 2012 The WebRTC project authors. All Rights Reserved.
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*
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* Use of this source code is governed by a BSD-style license
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* that can be found in the LICENSE file in the root of the source
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* tree. An additional intellectual property rights grant can be found
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* in the file PATENTS. All contributing project authors may
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* be found in the AUTHORS file in the root of the source tree.
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*/
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#ifndef MODULES_AUDIO_DEVICE_INCLUDE_AUDIO_DEVICE_H_
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#define MODULES_AUDIO_DEVICE_INCLUDE_AUDIO_DEVICE_H_
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#include "modules/audio_device/include/audio_device_defines.h"
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#include "rtc_base/scoped_ref_ptr.h"
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#include "rtc_base/refcount.h"
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namespace webrtc {
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class AudioDeviceModule : public rtc::RefCountInterface {
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public:
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// Deprecated.
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// TODO(henrika): to be removed.
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enum ErrorCode {
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kAdmErrNone = 0,
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kAdmErrArgument = 1
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};
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enum AudioLayer {
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kPlatformDefaultAudio = 0,
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kWindowsCoreAudio = 2,
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kLinuxAlsaAudio = 3,
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kLinuxPulseAudio = 4,
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kAndroidJavaAudio = 5,
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kAndroidOpenSLESAudio = 6,
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kAndroidJavaInputAndOpenSLESOutputAudio = 7,
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kDummyAudio = 8
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};
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enum WindowsDeviceType {
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kDefaultCommunicationDevice = -1,
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kDefaultDevice = -2
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};
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// TODO(bugs.webrtc.org/7306): deprecated.
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enum ChannelType {
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kChannelLeft = 0,
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kChannelRight = 1,
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kChannelBoth = 2
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};
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public:
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// Creates an ADM.
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static rtc::scoped_refptr<AudioDeviceModule> Create(
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const AudioLayer audio_layer);
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// TODO(bugs.webrtc.org/7306): deprecated (to be removed).
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static rtc::scoped_refptr<AudioDeviceModule> Create(
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const int32_t id,
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const AudioLayer audio_layer);
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// Retrieve the currently utilized audio layer
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virtual int32_t ActiveAudioLayer(AudioLayer* audioLayer) const = 0;
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// Full-duplex transportation of PCM audio
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virtual int32_t RegisterAudioCallback(AudioTransport* audioCallback) = 0;
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// Main initialization and termination
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virtual int32_t Init() = 0;
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virtual int32_t Terminate() = 0;
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virtual bool Initialized() const = 0;
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// Device enumeration
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virtual int16_t PlayoutDevices() = 0;
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virtual int16_t RecordingDevices() = 0;
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virtual int32_t PlayoutDeviceName(uint16_t index,
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char name[kAdmMaxDeviceNameSize],
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char guid[kAdmMaxGuidSize]) = 0;
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virtual int32_t RecordingDeviceName(uint16_t index,
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char name[kAdmMaxDeviceNameSize],
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char guid[kAdmMaxGuidSize]) = 0;
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// Device selection
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virtual int32_t SetPlayoutDevice(uint16_t index) = 0;
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virtual int32_t SetPlayoutDevice(WindowsDeviceType device) = 0;
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virtual int32_t SetRecordingDevice(uint16_t index) = 0;
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virtual int32_t SetRecordingDevice(WindowsDeviceType device) = 0;
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// Audio transport initialization
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virtual int32_t PlayoutIsAvailable(bool* available) = 0;
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virtual int32_t InitPlayout() = 0;
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virtual bool PlayoutIsInitialized() const = 0;
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virtual int32_t RecordingIsAvailable(bool* available) = 0;
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virtual int32_t InitRecording() = 0;
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virtual bool RecordingIsInitialized() const = 0;
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// Audio transport control
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virtual int32_t StartPlayout() = 0;
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virtual int32_t StopPlayout() = 0;
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virtual bool Playing() const = 0;
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virtual int32_t StartRecording() = 0;
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virtual int32_t StopRecording() = 0;
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virtual bool Recording() const = 0;
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// Microphone Automatic Gain Control (AGC)
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virtual int32_t SetAGC(bool enable) = 0;
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virtual bool AGC() const = 0;
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// Audio mixer initialization
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virtual int32_t InitSpeaker() = 0;
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virtual bool SpeakerIsInitialized() const = 0;
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virtual int32_t InitMicrophone() = 0;
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virtual bool MicrophoneIsInitialized() const = 0;
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// Speaker volume controls
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virtual int32_t SpeakerVolumeIsAvailable(bool* available) = 0;
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virtual int32_t SetSpeakerVolume(uint32_t volume) = 0;
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virtual int32_t SpeakerVolume(uint32_t* volume) const = 0;
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virtual int32_t MaxSpeakerVolume(uint32_t* maxVolume) const = 0;
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virtual int32_t MinSpeakerVolume(uint32_t* minVolume) const = 0;
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// Microphone volume controls
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virtual int32_t MicrophoneVolumeIsAvailable(bool* available) = 0;
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virtual int32_t SetMicrophoneVolume(uint32_t volume) = 0;
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virtual int32_t MicrophoneVolume(uint32_t* volume) const = 0;
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virtual int32_t MaxMicrophoneVolume(uint32_t* maxVolume) const = 0;
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virtual int32_t MinMicrophoneVolume(uint32_t* minVolume) const = 0;
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// Speaker mute control
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virtual int32_t SpeakerMuteIsAvailable(bool* available) = 0;
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virtual int32_t SetSpeakerMute(bool enable) = 0;
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virtual int32_t SpeakerMute(bool* enabled) const = 0;
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// Microphone mute control
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virtual int32_t MicrophoneMuteIsAvailable(bool* available) = 0;
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virtual int32_t SetMicrophoneMute(bool enable) = 0;
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virtual int32_t MicrophoneMute(bool* enabled) const = 0;
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// Stereo support
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virtual int32_t StereoPlayoutIsAvailable(bool* available) const = 0;
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virtual int32_t SetStereoPlayout(bool enable) = 0;
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virtual int32_t StereoPlayout(bool* enabled) const = 0;
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virtual int32_t StereoRecordingIsAvailable(bool* available) const = 0;
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virtual int32_t SetStereoRecording(bool enable) = 0;
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virtual int32_t StereoRecording(bool* enabled) const = 0;
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// TODO(bugs.webrtc.org/7306): deprecated.
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virtual int32_t SetRecordingChannel(const ChannelType channel) { return -1; }
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virtual int32_t RecordingChannel(ChannelType* channel) const { return -1; }
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// Playout delay
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virtual int32_t PlayoutDelay(uint16_t* delayMS) const = 0;
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// TODO(bugs.webrtc.org/7306): deprecated (to be removed).
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virtual int32_t SetRecordingSampleRate(const uint32_t samplesPerSec) {
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return -1;
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}
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virtual int32_t RecordingSampleRate(uint32_t* samplesPerSec) const {
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return -1;
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}
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virtual int32_t SetPlayoutSampleRate(const uint32_t samplesPerSec) {
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return -1;
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}
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virtual int32_t PlayoutSampleRate(uint32_t* samplesPerSec) const {
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return -1;
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}
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// TODO(bugs.webrtc.org/7306): deprecated (to be removed).
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virtual int32_t SetLoudspeakerStatus(bool enable) { return -1; }
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virtual int32_t GetLoudspeakerStatus(bool* enabled) const { return -1; }
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// Only supported on Android.
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virtual bool BuiltInAECIsAvailable() const = 0;
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virtual bool BuiltInAGCIsAvailable() const = 0;
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virtual bool BuiltInNSIsAvailable() const = 0;
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// Enables the built-in audio effects. Only supported on Android.
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virtual int32_t EnableBuiltInAEC(bool enable) = 0;
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virtual int32_t EnableBuiltInAGC(bool enable) = 0;
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virtual int32_t EnableBuiltInNS(bool enable) = 0;
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// Only supported on iOS.
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#if defined(WEBRTC_IOS)
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virtual int GetPlayoutAudioParameters(AudioParameters* params) const = 0;
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virtual int GetRecordAudioParameters(AudioParameters* params) const = 0;
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#endif // WEBRTC_IOS
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protected:
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~AudioDeviceModule() override {}
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};
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} // namespace webrtc
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#endif // MODULES_AUDIO_DEVICE_INCLUDE_AUDIO_DEVICE_H_
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