webrtc/modules/audio_device/test_audio_device_impl.h
Artem Titov 2cf8eb9f78 Reland "Migrate TestAudioDeviceModule on AudioDeviceModuleImpl"
This CL will add AudioDeviceBuffer into the SUT increasing test coverage
for audio quality regression detection.

This reverts commit b035dcc0a2.

Reason for revert: reland with a fix

Original change's description:
> Revert "Reland "Migrate TestAudioDeviceModule on AudioDeviceModuleImpl""
>
> This reverts commit eeae962997.
>
> 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 69c8d3c843.
> >
> > Reason for revert: Reland with a fix
> >
> > Original change's description:
> > > Revert "Migrate TestAudioDeviceModule on AudioDeviceModuleImpl"
> > >
> > > This reverts commit e42bf81486.
> > >
> > > 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 16:15:06 +00:00

198 lines
7.4 KiB
C++

/*
* Copyright (c) 2023 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_TEST_AUDIO_DEVICE_IMPL_H_
#define MODULES_AUDIO_DEVICE_TEST_AUDIO_DEVICE_IMPL_H_
#include <memory>
#include <vector>
#include "api/task_queue/task_queue_factory.h"
#include "modules/audio_device/audio_device_buffer.h"
#include "modules/audio_device/audio_device_generic.h"
#include "modules/audio_device/include/audio_device.h"
#include "modules/audio_device/include/audio_device_defines.h"
#include "modules/audio_device/include/test_audio_device.h"
#include "rtc_base/buffer.h"
#include "rtc_base/synchronization/mutex.h"
#include "rtc_base/task_queue.h"
namespace webrtc {
class TestAudioDevice : public AudioDeviceGeneric {
public:
// Creates a new TestAudioDevice. When capturing or playing, 10 ms audio
// frames will be processed every 10ms / `speed`.
// `capturer` is an object that produces audio data. Can be nullptr if this
// device is never used for recording.
// `renderer` is an object that receives audio data that would have been
// played out. Can be nullptr if this device is never used for playing.
TestAudioDevice(TaskQueueFactory* task_queue_factory,
std::unique_ptr<TestAudioDeviceModule::Capturer> capturer,
std::unique_ptr<TestAudioDeviceModule::Renderer> renderer,
float speed = 1);
TestAudioDevice(const TestAudioDevice&) = delete;
TestAudioDevice& operator=(const TestAudioDevice&) = delete;
~TestAudioDevice() override = default;
// Retrieve the currently utilized audio layer
int32_t ActiveAudioLayer(
AudioDeviceModule::AudioLayer& audioLayer) const override {
return 0;
}
// Main initializaton and termination
InitStatus Init() override;
int32_t Terminate() override { return 0; }
bool Initialized() const override { return true; }
// Device enumeration
int16_t PlayoutDevices() override { return 0; }
int16_t RecordingDevices() override { return 0; }
int32_t PlayoutDeviceName(uint16_t index,
char name[kAdmMaxDeviceNameSize],
char guid[kAdmMaxGuidSize]) override {
return 0;
}
int32_t RecordingDeviceName(uint16_t index,
char name[kAdmMaxDeviceNameSize],
char guid[kAdmMaxGuidSize]) override {
return 0;
}
// Device selection
int32_t SetPlayoutDevice(uint16_t index) override { return 0; }
int32_t SetPlayoutDevice(
AudioDeviceModule::WindowsDeviceType device) override {
return 0;
}
int32_t SetRecordingDevice(uint16_t index) override { return 0; }
int32_t SetRecordingDevice(
AudioDeviceModule::WindowsDeviceType device) override {
return 0;
}
// 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 { return 0; }
bool SpeakerIsInitialized() const override { return true; }
int32_t InitMicrophone() override { return 0; }
bool MicrophoneIsInitialized() const override { return true; }
// Speaker volume controls
int32_t SpeakerVolumeIsAvailable(bool& available) override { return 0; }
int32_t SetSpeakerVolume(uint32_t volume) override { return 0; }
int32_t SpeakerVolume(uint32_t& volume) const override { return 0; }
int32_t MaxSpeakerVolume(uint32_t& maxVolume) const override { return 0; }
int32_t MinSpeakerVolume(uint32_t& minVolume) const override { return 0; }
// Microphone volume controls
int32_t MicrophoneVolumeIsAvailable(bool& available) override { return 0; }
int32_t SetMicrophoneVolume(uint32_t volume) override { return 0; }
int32_t MicrophoneVolume(uint32_t& volume) const override { return 0; }
int32_t MaxMicrophoneVolume(uint32_t& maxVolume) const override { return 0; }
int32_t MinMicrophoneVolume(uint32_t& minVolume) const override { return 0; }
// Speaker mute control
int32_t SpeakerMuteIsAvailable(bool& available) override { return 0; }
int32_t SetSpeakerMute(bool enable) override { return 0; }
int32_t SpeakerMute(bool& enabled) const override { return 0; }
// Microphone mute control
int32_t MicrophoneMuteIsAvailable(bool& available) override { return 0; }
int32_t SetMicrophoneMute(bool enable) override { return 0; }
int32_t MicrophoneMute(bool& enabled) const override { return 0; }
// Stereo support
int32_t StereoPlayoutIsAvailable(bool& available) override {
available = false;
return 0;
}
int32_t SetStereoPlayout(bool enable) override { return 0; }
int32_t StereoPlayout(bool& enabled) const override { return 0; }
int32_t StereoRecordingIsAvailable(bool& available) override {
available = false;
return 0;
}
int32_t SetStereoRecording(bool enable) override { return 0; }
int32_t StereoRecording(bool& enabled) const override { return 0; }
// Delay information and control
int32_t PlayoutDelay(uint16_t& delayMS) const override {
delayMS = 0;
return 0;
}
// Android only
bool BuiltInAECIsAvailable() const override { return false; }
bool BuiltInAGCIsAvailable() const override { return false; }
bool BuiltInNSIsAvailable() const override { return false; }
// Windows Core Audio and Android only.
int32_t EnableBuiltInAEC(bool enable) override { return -1; }
int32_t EnableBuiltInAGC(bool enable) override { return -1; }
int32_t EnableBuiltInNS(bool enable) override { return -1; }
// Play underrun count.
int32_t GetPlayoutUnderrunCount() const override { return -1; }
// iOS only.
// TODO(henrika): add Android support.
#if defined(WEBRTC_IOS)
int GetPlayoutAudioParameters(AudioParameters* params) const override {
return -1;
}
int GetRecordAudioParameters(AudioParameters* params) const override {
return -1;
}
#endif // WEBRTC_IOS
void AttachAudioBuffer(AudioDeviceBuffer* audio_buffer) override;
private:
void ProcessAudio();
TaskQueueFactory* const task_queue_factory_;
const std::unique_ptr<TestAudioDeviceModule::Capturer> capturer_
RTC_GUARDED_BY(lock_);
const std::unique_ptr<TestAudioDeviceModule::Renderer> renderer_
RTC_GUARDED_BY(lock_);
const int64_t process_interval_us_;
mutable Mutex lock_;
AudioDeviceBuffer* audio_buffer_ RTC_GUARDED_BY(lock_) = nullptr;
bool rendering_ RTC_GUARDED_BY(lock_) = false;
bool capturing_ RTC_GUARDED_BY(lock_) = false;
bool rendering_initialized_ RTC_GUARDED_BY(lock_) = false;
bool capturing_initialized_ RTC_GUARDED_BY(lock_) = false;
std::vector<int16_t> playout_buffer_ RTC_GUARDED_BY(lock_);
rtc::BufferT<int16_t> recording_buffer_ RTC_GUARDED_BY(lock_);
std::unique_ptr<rtc::TaskQueue> task_queue_;
};
} // namespace webrtc
#endif // MODULES_AUDIO_DEVICE_TEST_AUDIO_DEVICE_IMPL_H_