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webrtc_m130/modules/audio_device/include/audio_device.h
henrika 5b6afc0ce6 Adds stream-switch support in new Windows ADM. 
Second round of the new Windows ADM is now ready for review. Main
changes are:

Supports internal (automatic) restart of audio streams when an active
audio stream disconnects (happens when a device is removed).

Adds support for IAudioClient3 and IAudioClient2 for platforms which
supports it (>Win8 and >Win10).

Modifies the threading model to support restart "from the inside" on
the native audio thread.

Adds two new test methods for the ADM to emulate restart events or
stream-switch events.

Adds two new test methods to support rate conversion to ensure that
audio can be tested in loopback even if devices runs at different
sample rates.

Added initial components for low-latency support. Verified that it works
but disabled it with a flag for now.

Bug: webrtc:9265
Change-Id: Ia8e577daabea6b433f2c2eabab4e46ce8added6a
Reviewed-on: https://webrtc-review.googlesource.com/86020
Reviewed-by: Oskar Sundbom <ossu@webrtc.org>
Commit-Queue: Henrik Andreassson <henrika@webrtc.org>
Cr-Commit-Position: refs/heads/master@{#24578}
2018-09-05 13:04:01 +00:00

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/*
* Copyright (c) 2012 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 MODULES_AUDIO_DEVICE_INCLUDE_AUDIO_DEVICE_H_
#define MODULES_AUDIO_DEVICE_INCLUDE_AUDIO_DEVICE_H_
#include "modules/audio_device/include/audio_device_defines.h"
#include "rtc_base/refcount.h"
#include "rtc_base/scoped_ref_ptr.h"
namespace webrtc {
class AudioDeviceModuleForTest;
class AudioDeviceModule : public rtc::RefCountInterface {
public:
// Deprecated.
// TODO(henrika): to be removed.
enum ErrorCode { kAdmErrNone = 0, kAdmErrArgument = 1 };
enum AudioLayer {
kPlatformDefaultAudio = 0,
kWindowsCoreAudio,
kWindowsCoreAudio2, // experimental
kLinuxAlsaAudio,
kLinuxPulseAudio,
kAndroidJavaAudio,
kAndroidOpenSLESAudio,
kAndroidJavaInputAndOpenSLESOutputAudio,
kAndroidAAudioAudio,
kAndroidJavaInputAndAAudioOutputAudio,
kDummyAudio,
};
enum WindowsDeviceType {
kDefaultCommunicationDevice = -1,
kDefaultDevice = -2
};
// TODO(bugs.webrtc.org/7306): deprecated.
enum ChannelType { kChannelLeft = 0, kChannelRight = 1, kChannelBoth = 2 };
public:
// Creates a default ADM for usage in production code.
static rtc::scoped_refptr<AudioDeviceModule> Create(
const AudioLayer audio_layer);
// Creates an ADM with support for extra test methods. Don't use this factory
// in production code.
static rtc::scoped_refptr<AudioDeviceModuleForTest> CreateForTest(
const AudioLayer audio_layer);
// TODO(bugs.webrtc.org/7306): deprecated (to be removed).
static rtc::scoped_refptr<AudioDeviceModule> Create(
const int32_t id,
const AudioLayer audio_layer);
// Retrieve the currently utilized audio layer
virtual int32_t ActiveAudioLayer(AudioLayer* audioLayer) const = 0;
// Full-duplex transportation of PCM audio
virtual int32_t RegisterAudioCallback(AudioTransport* audioCallback) = 0;
// Main initialization and termination
virtual int32_t Init() = 0;
virtual int32_t Terminate() = 0;
virtual bool Initialized() const = 0;
// Device enumeration
virtual int16_t PlayoutDevices() = 0;
virtual int16_t RecordingDevices() = 0;
virtual int32_t PlayoutDeviceName(uint16_t index,
char name[kAdmMaxDeviceNameSize],
char guid[kAdmMaxGuidSize]) = 0;
virtual int32_t RecordingDeviceName(uint16_t index,
char name[kAdmMaxDeviceNameSize],
char guid[kAdmMaxGuidSize]) = 0;
// Device selection
virtual int32_t SetPlayoutDevice(uint16_t index) = 0;
virtual int32_t SetPlayoutDevice(WindowsDeviceType device) = 0;
virtual int32_t SetRecordingDevice(uint16_t index) = 0;
virtual int32_t SetRecordingDevice(WindowsDeviceType device) = 0;
// Audio transport initialization
virtual int32_t PlayoutIsAvailable(bool* available) = 0;
virtual int32_t InitPlayout() = 0;
virtual bool PlayoutIsInitialized() const = 0;
virtual int32_t RecordingIsAvailable(bool* available) = 0;
virtual int32_t InitRecording() = 0;
virtual bool RecordingIsInitialized() const = 0;
// Audio transport control
virtual int32_t StartPlayout() = 0;
virtual int32_t StopPlayout() = 0;
virtual bool Playing() const = 0;
virtual int32_t StartRecording() = 0;
virtual int32_t StopRecording() = 0;
virtual bool Recording() const = 0;
// Audio mixer initialization
virtual int32_t InitSpeaker() = 0;
virtual bool SpeakerIsInitialized() const = 0;
virtual int32_t InitMicrophone() = 0;
virtual bool MicrophoneIsInitialized() const = 0;
// Speaker volume controls
virtual int32_t SpeakerVolumeIsAvailable(bool* available) = 0;
virtual int32_t SetSpeakerVolume(uint32_t volume) = 0;
virtual int32_t SpeakerVolume(uint32_t* volume) const = 0;
virtual int32_t MaxSpeakerVolume(uint32_t* maxVolume) const = 0;
virtual int32_t MinSpeakerVolume(uint32_t* minVolume) const = 0;
// Microphone volume controls
virtual int32_t MicrophoneVolumeIsAvailable(bool* available) = 0;
virtual int32_t SetMicrophoneVolume(uint32_t volume) = 0;
virtual int32_t MicrophoneVolume(uint32_t* volume) const = 0;
virtual int32_t MaxMicrophoneVolume(uint32_t* maxVolume) const = 0;
virtual int32_t MinMicrophoneVolume(uint32_t* minVolume) const = 0;
// Speaker mute control
virtual int32_t SpeakerMuteIsAvailable(bool* available) = 0;
virtual int32_t SetSpeakerMute(bool enable) = 0;
virtual int32_t SpeakerMute(bool* enabled) const = 0;
// Microphone mute control
virtual int32_t MicrophoneMuteIsAvailable(bool* available) = 0;
virtual int32_t SetMicrophoneMute(bool enable) = 0;
virtual int32_t MicrophoneMute(bool* enabled) const = 0;
// Stereo support
virtual int32_t StereoPlayoutIsAvailable(bool* available) const = 0;
virtual int32_t SetStereoPlayout(bool enable) = 0;
virtual int32_t StereoPlayout(bool* enabled) const = 0;
virtual int32_t StereoRecordingIsAvailable(bool* available) const = 0;
virtual int32_t SetStereoRecording(bool enable) = 0;
virtual int32_t StereoRecording(bool* enabled) const = 0;
// Playout delay
virtual int32_t PlayoutDelay(uint16_t* delayMS) const = 0;
// Only supported on Android.
virtual bool BuiltInAECIsAvailable() const = 0;
virtual bool BuiltInAGCIsAvailable() const = 0;
virtual bool BuiltInNSIsAvailable() const = 0;
// Enables the built-in audio effects. Only supported on Android.
virtual int32_t EnableBuiltInAEC(bool enable) = 0;
virtual int32_t EnableBuiltInAGC(bool enable) = 0;
virtual int32_t EnableBuiltInNS(bool enable) = 0;
// Only supported on iOS.
#if defined(WEBRTC_IOS)
virtual int GetPlayoutAudioParameters(AudioParameters* params) const = 0;
virtual int GetRecordAudioParameters(AudioParameters* params) const = 0;
#endif // WEBRTC_IOS
protected:
~AudioDeviceModule() override {}
};
// Extends the default ADM interface with some extra test methods.
// Intended for usage in tests only and requires a unique factory method.
class AudioDeviceModuleForTest : public AudioDeviceModule {
public:
// Triggers internal restart sequences of audio streaming. Can be used by
// tests to emulate events corresponding to e.g. removal of an active audio
// device or other actions which causes the stream to be disconnected.
virtual int RestartPlayoutInternally() = 0;
virtual int RestartRecordingInternally() = 0;
virtual int SetPlayoutSampleRate(uint32_t sample_rate) = 0;
virtual int SetRecordingSampleRate(uint32_t sample_rate) = 0;
};
} // namespace webrtc
#endif // MODULES_AUDIO_DEVICE_INCLUDE_AUDIO_DEVICE_H_
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