henrika@webrtc.org 474d1eb223 Adds C++/JNI/Java unit test for audio device module on Android.
This CL adds support for unittests of the AudioDeviceModule on Android using both Java and C++. The new framework uses ::testing::TesWithParam to support both Java-based audio and OpenSL ES based audio. However, given existing issues in our OpenSL ES implementation, the list of test parameters only contains Java in this first version. Open SL ES will be enabled as soon as the backend has been refactored.

It also:

- Removes the redundant JNIEnv* argument in webrtc::VoiceEngine::SetAndroidObjects().
- Modifies usage of enable_android_opensl and the WEBRTC_ANDROID_OPENSLES define.
- Adds kAndroidJavaAudio and kAndroidOpenSLESAudio to AudioLayer enumerator.
- Fixes some bugs which were discovered when running the tests.

BUG=NONE
R=phoglund@webrtc.org

Review URL: https://webrtc-codereview.appspot.com/40069004

Cr-Commit-Position: refs/heads/master@{#8651}
git-svn-id: http://webrtc.googlecode.com/svn/trunk@8651 4adac7df-926f-26a2-2b94-8c16560cd09d
2015-03-09 12:40:43 +00:00

215 lines
8.0 KiB
C++

/*
* 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 "webrtc/modules/audio_device/include/audio_device_defines.h"
#include "webrtc/modules/interface/module.h"
namespace webrtc {
class AudioDeviceModule : public RefCountedModule {
public:
enum ErrorCode {
kAdmErrNone = 0,
kAdmErrArgument = 1
};
enum AudioLayer {
kPlatformDefaultAudio = 0,
kWindowsWaveAudio = 1,
kWindowsCoreAudio = 2,
kLinuxAlsaAudio = 3,
kLinuxPulseAudio = 4,
kAndroidJavaAudio = 5,
kAndroidOpenSLESAudio = 6,
kDummyAudio = 7
};
enum WindowsDeviceType {
kDefaultCommunicationDevice = -1,
kDefaultDevice = -2
};
enum BufferType {
kFixedBufferSize = 0,
kAdaptiveBufferSize = 1
};
enum ChannelType {
kChannelLeft = 0,
kChannelRight = 1,
kChannelBoth = 2
};
public:
// Retrieve the currently utilized audio layer
virtual int32_t ActiveAudioLayer(AudioLayer* audioLayer) const = 0;
// Error handling
virtual ErrorCode LastError() const = 0;
virtual int32_t RegisterEventObserver(AudioDeviceObserver* eventCallback) = 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;
// Microphone Automatic Gain Control (AGC)
virtual int32_t SetAGC(bool enable) = 0;
virtual bool AGC() const = 0;
// Volume control based on the Windows Wave API (Windows only)
virtual int32_t SetWaveOutVolume(uint16_t volumeLeft,
uint16_t volumeRight) = 0;
virtual int32_t WaveOutVolume(uint16_t* volumeLeft,
uint16_t* volumeRight) 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;
virtual int32_t SpeakerVolumeStepSize(uint16_t* stepSize) 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;
virtual int32_t MicrophoneVolumeStepSize(uint16_t* stepSize) 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;
// Microphone boost control
virtual int32_t MicrophoneBoostIsAvailable(bool* available) = 0;
virtual int32_t SetMicrophoneBoost(bool enable) = 0;
virtual int32_t MicrophoneBoost(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;
virtual int32_t SetRecordingChannel(const ChannelType channel) = 0;
virtual int32_t RecordingChannel(ChannelType* channel) const = 0;
// Delay information and control
virtual int32_t SetPlayoutBuffer(const BufferType type,
uint16_t sizeMS = 0) = 0;
virtual int32_t PlayoutBuffer(BufferType* type, uint16_t* sizeMS) const = 0;
virtual int32_t PlayoutDelay(uint16_t* delayMS) const = 0;
virtual int32_t RecordingDelay(uint16_t* delayMS) const = 0;
// CPU load
virtual int32_t CPULoad(uint16_t* load) const = 0;
// Recording of raw PCM data
virtual int32_t StartRawOutputFileRecording(
const char pcmFileNameUTF8[kAdmMaxFileNameSize]) = 0;
virtual int32_t StopRawOutputFileRecording() = 0;
virtual int32_t StartRawInputFileRecording(
const char pcmFileNameUTF8[kAdmMaxFileNameSize]) = 0;
virtual int32_t StopRawInputFileRecording() = 0;
// Native sample rate controls (samples/sec)
virtual int32_t SetRecordingSampleRate(const uint32_t samplesPerSec) = 0;
virtual int32_t RecordingSampleRate(uint32_t* samplesPerSec) const = 0;
virtual int32_t SetPlayoutSampleRate(const uint32_t samplesPerSec) = 0;
virtual int32_t PlayoutSampleRate(uint32_t* samplesPerSec) const = 0;
// Mobile device specific functions
virtual int32_t ResetAudioDevice() = 0;
virtual int32_t SetLoudspeakerStatus(bool enable) = 0;
virtual int32_t GetLoudspeakerStatus(bool* enabled) const = 0;
// Only supported on Android.
// TODO(henrika): Make pure virtual after updating Chromium.
virtual bool BuiltInAECIsAvailable() const { return false; }
// Enables the built-in AEC. Only supported on Windows and Android.
//
// For usage on Windows (requires Core Audio):
// Must be called before InitRecording(). When enabled:
// 1. StartPlayout() must be called before StartRecording().
// 2. StopRecording() should be called before StopPlayout().
// The reverse order may cause garbage audio to be rendered or the
// capture side to halt until StopRecording() is called.
// TODO(henrika): Make pure virtual after updating Chromium.
virtual int32_t EnableBuiltInAEC(bool enable) { return -1; }
// Don't use.
virtual bool BuiltInAECIsEnabled() const { return false; }
protected:
virtual ~AudioDeviceModule() {};
};
AudioDeviceModule* CreateAudioDeviceModule(
int32_t id, AudioDeviceModule::AudioLayer audioLayer);
} // namespace webrtc
#endif // MODULES_AUDIO_DEVICE_INCLUDE_AUDIO_DEVICE_H_