webrtc_m130/modules/audio_device/audio_device_impl.h

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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_AUDIO_DEVICE_IMPL_H_
#define MODULES_AUDIO_DEVICE_AUDIO_DEVICE_IMPL_H_
#if defined(WEBRTC_INCLUDE_INTERNAL_AUDIO_DEVICE)
#include <stdint.h>
#include <memory>
#include "api/audio/audio_device.h"
#include "api/task_queue/task_queue_factory.h"
#include "modules/audio_device/audio_device_buffer.h"
namespace webrtc {
class AudioDeviceGeneric;
class AudioDeviceModuleImpl : public AudioDeviceModuleForTest {
public:
enum PlatformType {
kPlatformNotSupported = 0,
kPlatformWin32 = 1,
kPlatformWinCe = 2,
kPlatformLinux = 3,
kPlatformMac = 4,
kPlatformAndroid = 5,
Reland "Migrate TestAudioDeviceModule on AudioDeviceModuleImpl" This CL will add AudioDeviceBuffer into the SUT increasing test coverage for audio quality regression detection. This reverts commit b035dcc0a274e6cdde3e0fc465244bc0e9e3d70e. Reason for revert: reland with a fix Original change's description: > Revert "Reland "Migrate TestAudioDeviceModule on AudioDeviceModuleImpl"" > > This reverts commit eeae96299784515f573379a64655eb07a5973a3a. > > Reason for revert: breaks WebRTC Chromium FYI ios-device > https://ci.chromium.org/ui/p/chromium/builders/webrtc.fyi/WebRTC%20Chromium%20FYI%20ios-device/14896/overview > > Original change's description: > > Reland "Migrate TestAudioDeviceModule on AudioDeviceModuleImpl" > > > > This reverts commit 69c8d3c843326aff9dee32cc639741c1cd7f8ae9. > > > > Reason for revert: Reland with a fix > > > > Original change's description: > > > Revert "Migrate TestAudioDeviceModule on AudioDeviceModuleImpl" > > > > > > This reverts commit e42bf81486d2f08b6dcbf1442287202e937ce52b. > > > > > > Reason for revert: Breaks iOS simulator bots and thus blocks chromium roll, https://chromium-review.googlesource.com/c/chromium/src/+/4433814 > > > > > > Original change's description: > > > > Migrate TestAudioDeviceModule on AudioDeviceModuleImpl > > > > > > > > Bug: b/272350185 > > > > Change-Id: Ia3d85d6fa3b0d4809e987a39d60d3eb022687132 > > > > Reviewed-on: https://webrtc-review.googlesource.com/c/src/+/300363 > > > > Commit-Queue: Artem Titov <titovartem@webrtc.org> > > > > Reviewed-by: Henrik Andreassson <henrika@webrtc.org> > > > > Cr-Commit-Position: refs/heads/main@{#39877} > > > > > > Bug: b/272350185 > > > Change-Id: I1e3b542fc1278797f283afedeae01cbb7412d353 > > > No-Presubmit: true > > > No-Tree-Checks: true > > > No-Try: true > > > Reviewed-on: https://webrtc-review.googlesource.com/c/src/+/301701 > > > Commit-Queue: Jeremy Leconte <jleconte@google.com> > > > Bot-Commit: rubber-stamper@appspot.gserviceaccount.com <rubber-stamper@appspot.gserviceaccount.com> > > > Reviewed-by: Jeremy Leconte <jleconte@google.com> > > > Auto-Submit: Christoffer Jansson <jansson@google.com> > > > Owners-Override: Christoffer Jansson <jansson@google.com> > > > Cr-Commit-Position: refs/heads/main@{#39881} > > > > Bug: b/272350185 > > Change-Id: I809466306b2e1fd54c44b90311059c98a53ef8ee > > Reviewed-on: https://webrtc-review.googlesource.com/c/src/+/301704 > > Reviewed-by: Henrik Andreassson <henrika@webrtc.org> > > Reviewed-by: Mirko Bonadei <mbonadei@webrtc.org> > > Commit-Queue: Artem Titov <titovartem@webrtc.org> > > Cr-Commit-Position: refs/heads/main@{#39936} > > Bug: b/272350185 > Change-Id: If0a10717bf14a0a618e52728fc3a61b9c55f3bd2 > No-Presubmit: true > No-Tree-Checks: true > No-Try: true > Reviewed-on: https://webrtc-review.googlesource.com/c/src/+/303460 > Commit-Queue: Jeremy Leconte <jleconte@google.com> > Owners-Override: Jeremy Leconte <jleconte@google.com> > Bot-Commit: rubber-stamper@appspot.gserviceaccount.com <rubber-stamper@appspot.gserviceaccount.com> > Cr-Commit-Position: refs/heads/main@{#39947} Bug: b/272350185 Change-Id: I7cf7c6bc25561f4eb722957f318c2af9ce20726d Reviewed-on: https://webrtc-review.googlesource.com/c/src/+/311101 Reviewed-by: Mirko Bonadei <mbonadei@webrtc.org> Reviewed-by: Tomas Gunnarsson <tommi@webrtc.org> Commit-Queue: Artem Titov <titovartem@webrtc.org> Cr-Commit-Position: refs/heads/main@{#40387}
2023-06-30 15:26:09 +02:00
kPlatformIOS = 6,
// Fuchsia isn't fully supported, as there is no implementation for
// AudioDeviceGeneric which will be created for Fuchsia, so
// `CreatePlatformSpecificObjects()` call will fail unless usable
// implementation will be provided by the user.
kPlatformFuchsia = 7,
};
int32_t CheckPlatform();
int32_t CreatePlatformSpecificObjects();
int32_t AttachAudioBuffer();
AudioDeviceModuleImpl(AudioLayer audio_layer,
TaskQueueFactory* task_queue_factory);
Reland "Migrate TestAudioDeviceModule on AudioDeviceModuleImpl" This CL will add AudioDeviceBuffer into the SUT increasing test coverage for audio quality regression detection. This reverts commit b035dcc0a274e6cdde3e0fc465244bc0e9e3d70e. Reason for revert: reland with a fix Original change's description: > Revert "Reland "Migrate TestAudioDeviceModule on AudioDeviceModuleImpl"" > > This reverts commit eeae96299784515f573379a64655eb07a5973a3a. > > Reason for revert: breaks WebRTC Chromium FYI ios-device > https://ci.chromium.org/ui/p/chromium/builders/webrtc.fyi/WebRTC%20Chromium%20FYI%20ios-device/14896/overview > > Original change's description: > > Reland "Migrate TestAudioDeviceModule on AudioDeviceModuleImpl" > > > > This reverts commit 69c8d3c843326aff9dee32cc639741c1cd7f8ae9. > > > > Reason for revert: Reland with a fix > > > > Original change's description: > > > Revert "Migrate TestAudioDeviceModule on AudioDeviceModuleImpl" > > > > > > This reverts commit e42bf81486d2f08b6dcbf1442287202e937ce52b. > > > > > > Reason for revert: Breaks iOS simulator bots and thus blocks chromium roll, https://chromium-review.googlesource.com/c/chromium/src/+/4433814 > > > > > > Original change's description: > > > > Migrate TestAudioDeviceModule on AudioDeviceModuleImpl > > > > > > > > Bug: b/272350185 > > > > Change-Id: Ia3d85d6fa3b0d4809e987a39d60d3eb022687132 > > > > Reviewed-on: https://webrtc-review.googlesource.com/c/src/+/300363 > > > > Commit-Queue: Artem Titov <titovartem@webrtc.org> > > > > Reviewed-by: Henrik Andreassson <henrika@webrtc.org> > > > > Cr-Commit-Position: refs/heads/main@{#39877} > > > > > > Bug: b/272350185 > > > Change-Id: I1e3b542fc1278797f283afedeae01cbb7412d353 > > > No-Presubmit: true > > > No-Tree-Checks: true > > > No-Try: true > > > Reviewed-on: https://webrtc-review.googlesource.com/c/src/+/301701 > > > Commit-Queue: Jeremy Leconte <jleconte@google.com> > > > Bot-Commit: rubber-stamper@appspot.gserviceaccount.com <rubber-stamper@appspot.gserviceaccount.com> > > > Reviewed-by: Jeremy Leconte <jleconte@google.com> > > > Auto-Submit: Christoffer Jansson <jansson@google.com> > > > Owners-Override: Christoffer Jansson <jansson@google.com> > > > Cr-Commit-Position: refs/heads/main@{#39881} > > > > Bug: b/272350185 > > Change-Id: I809466306b2e1fd54c44b90311059c98a53ef8ee > > Reviewed-on: https://webrtc-review.googlesource.com/c/src/+/301704 > > Reviewed-by: Henrik Andreassson <henrika@webrtc.org> > > Reviewed-by: Mirko Bonadei <mbonadei@webrtc.org> > > Commit-Queue: Artem Titov <titovartem@webrtc.org> > > Cr-Commit-Position: refs/heads/main@{#39936} > > Bug: b/272350185 > Change-Id: If0a10717bf14a0a618e52728fc3a61b9c55f3bd2 > No-Presubmit: true > No-Tree-Checks: true > No-Try: true > Reviewed-on: https://webrtc-review.googlesource.com/c/src/+/303460 > Commit-Queue: Jeremy Leconte <jleconte@google.com> > Owners-Override: Jeremy Leconte <jleconte@google.com> > Bot-Commit: rubber-stamper@appspot.gserviceaccount.com <rubber-stamper@appspot.gserviceaccount.com> > Cr-Commit-Position: refs/heads/main@{#39947} Bug: b/272350185 Change-Id: I7cf7c6bc25561f4eb722957f318c2af9ce20726d Reviewed-on: https://webrtc-review.googlesource.com/c/src/+/311101 Reviewed-by: Mirko Bonadei <mbonadei@webrtc.org> Reviewed-by: Tomas Gunnarsson <tommi@webrtc.org> Commit-Queue: Artem Titov <titovartem@webrtc.org> Cr-Commit-Position: refs/heads/main@{#40387}
2023-06-30 15:26:09 +02:00
// If `create_detached` is true, created ADM can be used on another thread
// compared to the one on which it was created. It's useful for testing.
AudioDeviceModuleImpl(AudioLayer audio_layer,
std::unique_ptr<AudioDeviceGeneric> audio_device,
TaskQueueFactory* task_queue_factory,
bool create_detached);
~AudioDeviceModuleImpl() override;
// Retrieve the currently utilized audio layer
int32_t ActiveAudioLayer(AudioLayer* audioLayer) const override;
// Full-duplex transportation of PCM audio
int32_t RegisterAudioCallback(AudioTransport* audioCallback) override;
// Main initializaton and termination
int32_t Init() override;
int32_t Terminate() override;
bool Initialized() const override;
// Device enumeration
int16_t PlayoutDevices() override;
int16_t RecordingDevices() override;
int32_t PlayoutDeviceName(uint16_t index,
char name[kAdmMaxDeviceNameSize],
char guid[kAdmMaxGuidSize]) override;
int32_t RecordingDeviceName(uint16_t index,
char name[kAdmMaxDeviceNameSize],
char guid[kAdmMaxGuidSize]) override;
// Device selection
int32_t SetPlayoutDevice(uint16_t index) override;
int32_t SetPlayoutDevice(WindowsDeviceType device) override;
int32_t SetRecordingDevice(uint16_t index) override;
int32_t SetRecordingDevice(WindowsDeviceType device) override;
// Audio transport initialization
int32_t PlayoutIsAvailable(bool* available) override;
int32_t InitPlayout() override;
bool PlayoutIsInitialized() const override;
int32_t RecordingIsAvailable(bool* available) override;
int32_t InitRecording() override;
bool RecordingIsInitialized() const override;
// Audio transport control
int32_t StartPlayout() override;
int32_t StopPlayout() override;
bool Playing() const override;
int32_t StartRecording() override;
int32_t StopRecording() override;
bool Recording() const override;
// Audio mixer initialization
int32_t InitSpeaker() override;
bool SpeakerIsInitialized() const override;
int32_t InitMicrophone() override;
bool MicrophoneIsInitialized() const override;
// Speaker volume controls
int32_t SpeakerVolumeIsAvailable(bool* available) override;
int32_t SetSpeakerVolume(uint32_t volume) override;
int32_t SpeakerVolume(uint32_t* volume) const override;
int32_t MaxSpeakerVolume(uint32_t* maxVolume) const override;
int32_t MinSpeakerVolume(uint32_t* minVolume) const override;
// Microphone volume controls
int32_t MicrophoneVolumeIsAvailable(bool* available) override;
int32_t SetMicrophoneVolume(uint32_t volume) override;
int32_t MicrophoneVolume(uint32_t* volume) const override;
int32_t MaxMicrophoneVolume(uint32_t* maxVolume) const override;
int32_t MinMicrophoneVolume(uint32_t* minVolume) const override;
// Speaker mute control
int32_t SpeakerMuteIsAvailable(bool* available) override;
int32_t SetSpeakerMute(bool enable) override;
int32_t SpeakerMute(bool* enabled) const override;
// Microphone mute control
int32_t MicrophoneMuteIsAvailable(bool* available) override;
int32_t SetMicrophoneMute(bool enable) override;
int32_t MicrophoneMute(bool* enabled) const override;
// Stereo support
int32_t StereoPlayoutIsAvailable(bool* available) const override;
int32_t SetStereoPlayout(bool enable) override;
int32_t StereoPlayout(bool* enabled) const override;
int32_t StereoRecordingIsAvailable(bool* available) const override;
int32_t SetStereoRecording(bool enable) override;
int32_t StereoRecording(bool* enabled) const override;
// Delay information and control
int32_t PlayoutDelay(uint16_t* delayMS) const override;
bool BuiltInAECIsAvailable() const override;
int32_t EnableBuiltInAEC(bool enable) override;
bool BuiltInAGCIsAvailable() const override;
int32_t EnableBuiltInAGC(bool enable) override;
bool BuiltInNSIsAvailable() const override;
int32_t EnableBuiltInNS(bool enable) override;
// Play underrun count.
int32_t GetPlayoutUnderrunCount() const override;
#if defined(WEBRTC_IOS)
int GetPlayoutAudioParameters(AudioParameters* params) const override;
int GetRecordAudioParameters(AudioParameters* params) const override;
#endif // WEBRTC_IOS
AudioDeviceBuffer* GetAudioDeviceBuffer() { return &audio_device_buffer_; }
int RestartPlayoutInternally() override { return -1; }
int RestartRecordingInternally() override { return -1; }
int SetPlayoutSampleRate(uint32_t /* sample_rate */) override { return -1; }
int SetRecordingSampleRate(uint32_t /* sample_rate */) override { return -1; }
private:
PlatformType Platform() const;
AudioLayer PlatformAudioLayer() const;
AudioLayer audio_layer_;
PlatformType platform_type_ = kPlatformNotSupported;
bool initialized_ = false;
AudioDeviceBuffer audio_device_buffer_;
std::unique_ptr<AudioDeviceGeneric> audio_device_;
};
} // namespace webrtc
#endif // defined(WEBRTC_INCLUDE_INTERNAL_AUDIO_DEVICE)
#endif // MODULES_AUDIO_DEVICE_AUDIO_DEVICE_IMPL_H_