blob: 52107cc8b70636927fcf790708936c7f54aabe7f [file] [log] [blame]
/*
* Copyright (c) 2013 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 WEBRTC_MODULES_AUDIO_DEVICE_ANDROID_AUDIO_DEVICE_TEMPLATE_H_
#define WEBRTC_MODULES_AUDIO_DEVICE_ANDROID_AUDIO_DEVICE_TEMPLATE_H_
#include "webrtc/modules/audio_device/android/audio_manager.h"
#include "webrtc/modules/audio_device/audio_device_generic.h"
#include "webrtc/rtc_base/checks.h"
#include "webrtc/rtc_base/logging.h"
#include "webrtc/rtc_base/thread_checker.h"
namespace webrtc {
// InputType/OutputType can be any class that implements the capturing/rendering
// part of the AudioDeviceGeneric API.
// Construction and destruction must be done on one and the same thread. Each
// internal implementation of InputType and OutputType will RTC_DCHECK if that
// is not the case. All implemented methods must also be called on the same
// thread. See comments in each InputType/OutputType class for more info.
// It is possible to call the two static methods (SetAndroidAudioDeviceObjects
// and ClearAndroidAudioDeviceObjects) from a different thread but both will
// RTC_CHECK that the calling thread is attached to a Java VM.
template <class InputType, class OutputType>
class AudioDeviceTemplate : public AudioDeviceGeneric {
public:
AudioDeviceTemplate(AudioDeviceModule::AudioLayer audio_layer,
AudioManager* audio_manager)
: audio_layer_(audio_layer),
audio_manager_(audio_manager),
output_(audio_manager_),
input_(audio_manager_),
initialized_(false) {
LOG(INFO) << __FUNCTION__;
RTC_CHECK(audio_manager);
audio_manager_->SetActiveAudioLayer(audio_layer);
}
virtual ~AudioDeviceTemplate() { LOG(INFO) << __FUNCTION__; }
int32_t ActiveAudioLayer(
AudioDeviceModule::AudioLayer& audioLayer) const override {
LOG(INFO) << __FUNCTION__;
audioLayer = audio_layer_;
return 0;
}
InitStatus Init() override {
LOG(INFO) << __FUNCTION__;
RTC_DCHECK(thread_checker_.CalledOnValidThread());
RTC_DCHECK(!initialized_);
if (!audio_manager_->Init()) {
return InitStatus::OTHER_ERROR;
}
if (output_.Init() != 0) {
audio_manager_->Close();
return InitStatus::PLAYOUT_ERROR;
}
if (input_.Init() != 0) {
output_.Terminate();
audio_manager_->Close();
return InitStatus::RECORDING_ERROR;
}
initialized_ = true;
return InitStatus::OK;
}
int32_t Terminate() override {
LOG(INFO) << __FUNCTION__;
RTC_DCHECK(thread_checker_.CalledOnValidThread());
int32_t err = input_.Terminate();
err |= output_.Terminate();
err |= !audio_manager_->Close();
initialized_ = false;
RTC_DCHECK_EQ(err, 0);
return err;
}
bool Initialized() const override {
LOG(INFO) << __FUNCTION__;
RTC_DCHECK(thread_checker_.CalledOnValidThread());
return initialized_;
}
int16_t PlayoutDevices() override {
LOG(INFO) << __FUNCTION__;
return 1;
}
int16_t RecordingDevices() override {
LOG(INFO) << __FUNCTION__;
return 1;
}
int32_t PlayoutDeviceName(
uint16_t index,
char name[kAdmMaxDeviceNameSize],
char guid[kAdmMaxGuidSize]) override {
FATAL() << "Should never be called";
return -1;
}
int32_t RecordingDeviceName(
uint16_t index,
char name[kAdmMaxDeviceNameSize],
char guid[kAdmMaxGuidSize]) override {
FATAL() << "Should never be called";
return -1;
}
int32_t SetPlayoutDevice(uint16_t index) override {
// OK to use but it has no effect currently since device selection is
// done using Andoid APIs instead.
LOG(INFO) << __FUNCTION__;
return 0;
}
int32_t SetPlayoutDevice(
AudioDeviceModule::WindowsDeviceType device) override {
FATAL() << "Should never be called";
return -1;
}
int32_t SetRecordingDevice(uint16_t index) override {
// OK to use but it has no effect currently since device selection is
// done using Andoid APIs instead.
LOG(INFO) << __FUNCTION__;
return 0;
}
int32_t SetRecordingDevice(
AudioDeviceModule::WindowsDeviceType device) override {
FATAL() << "Should never be called";
return -1;
}
int32_t PlayoutIsAvailable(bool& available) override {
LOG(INFO) << __FUNCTION__;
available = true;
return 0;
}
int32_t InitPlayout() override {
LOG(INFO) << __FUNCTION__;
return output_.InitPlayout();
}
bool PlayoutIsInitialized() const override {
LOG(INFO) << __FUNCTION__;
return output_.PlayoutIsInitialized();
}
int32_t RecordingIsAvailable(bool& available) override {
LOG(INFO) << __FUNCTION__;
available = true;
return 0;
}
int32_t InitRecording() override {
LOG(INFO) << __FUNCTION__;
return input_.InitRecording();
}
bool RecordingIsInitialized() const override {
LOG(INFO) << __FUNCTION__;
return input_.RecordingIsInitialized();
}
int32_t StartPlayout() override {
LOG(INFO) << __FUNCTION__;
if (!audio_manager_->IsCommunicationModeEnabled()) {
LOG(WARNING)
<< "The application should use MODE_IN_COMMUNICATION audio mode!";
}
return output_.StartPlayout();
}
int32_t StopPlayout() override {
// Avoid using audio manger (JNI/Java cost) if playout was inactive.
if (!Playing())
return 0;
LOG(INFO) << __FUNCTION__;
int32_t err = output_.StopPlayout();
return err;
}
bool Playing() const override {
LOG(INFO) << __FUNCTION__;
return output_.Playing();
}
int32_t StartRecording() override {
LOG(INFO) << __FUNCTION__;
if (!audio_manager_->IsCommunicationModeEnabled()) {
LOG(WARNING)
<< "The application should use MODE_IN_COMMUNICATION audio mode!";
}
return input_.StartRecording();
}
int32_t StopRecording() override {
// Avoid using audio manger (JNI/Java cost) if recording was inactive.
LOG(INFO) << __FUNCTION__;
if (!Recording())
return 0;
int32_t err = input_.StopRecording();
return err;
}
bool Recording() const override {
return input_.Recording() ;
}
int32_t SetAGC(bool enable) override {
if (enable) {
FATAL() << "Should never be called";
}
return -1;
}
bool AGC() const override {
LOG(INFO) << __FUNCTION__;
return false;
}
int32_t SetWaveOutVolume(
uint16_t volumeLeft, uint16_t volumeRight) override {
FATAL() << "Should never be called";
return -1;
}
int32_t WaveOutVolume(
uint16_t& volumeLeft, uint16_t& volumeRight) const override {
FATAL() << "Should never be called";
return -1;
}
int32_t InitSpeaker() override {
LOG(INFO) << __FUNCTION__;
return 0;
}
bool SpeakerIsInitialized() const override {
LOG(INFO) << __FUNCTION__;
return true;
}
int32_t InitMicrophone() override {
LOG(INFO) << __FUNCTION__;
return 0;
}
bool MicrophoneIsInitialized() const override {
LOG(INFO) << __FUNCTION__;
return true;
}
int32_t SpeakerVolumeIsAvailable(bool& available) override {
LOG(INFO) << __FUNCTION__;
return output_.SpeakerVolumeIsAvailable(available);
}
int32_t SetSpeakerVolume(uint32_t volume) override {
LOG(INFO) << __FUNCTION__;
return output_.SetSpeakerVolume(volume);
}
int32_t SpeakerVolume(uint32_t& volume) const override {
LOG(INFO) << __FUNCTION__;
return output_.SpeakerVolume(volume);
}
int32_t MaxSpeakerVolume(uint32_t& maxVolume) const override {
LOG(INFO) << __FUNCTION__;
return output_.MaxSpeakerVolume(maxVolume);
}
int32_t MinSpeakerVolume(uint32_t& minVolume) const override {
LOG(INFO) << __FUNCTION__;
return output_.MinSpeakerVolume(minVolume);
}
int32_t SpeakerVolumeStepSize(uint16_t& stepSize) const override {
FATAL() << "Should never be called";
return -1;
}
int32_t MicrophoneVolumeIsAvailable(bool& available) override{
available = false;
return -1;
}
int32_t SetMicrophoneVolume(uint32_t volume) override {
FATAL() << "Should never be called";
return -1;
}
int32_t MicrophoneVolume(uint32_t& volume) const override {
FATAL() << "Should never be called";
return -1;
}
int32_t MaxMicrophoneVolume(uint32_t& maxVolume) const override {
FATAL() << "Should never be called";
return -1;
}
int32_t MinMicrophoneVolume(uint32_t& minVolume) const override {
FATAL() << "Should never be called";
return -1;
}
int32_t MicrophoneVolumeStepSize(uint16_t& stepSize) const override {
FATAL() << "Should never be called";
return -1;
}
int32_t SpeakerMuteIsAvailable(bool& available) override {
FATAL() << "Should never be called";
return -1;
}
int32_t SetSpeakerMute(bool enable) override {
FATAL() << "Should never be called";
return -1;
}
int32_t SpeakerMute(bool& enabled) const override {
FATAL() << "Should never be called";
return -1;
}
int32_t MicrophoneMuteIsAvailable(bool& available) override {
FATAL() << "Not implemented";
return -1;
}
int32_t SetMicrophoneMute(bool enable) override {
FATAL() << "Not implemented";
return -1;
}
int32_t MicrophoneMute(bool& enabled) const override {
FATAL() << "Not implemented";
return -1;
}
int32_t MicrophoneBoostIsAvailable(bool& available) override {
FATAL() << "Should never be called";
return -1;
}
int32_t SetMicrophoneBoost(bool enable) override {
FATAL() << "Should never be called";
return -1;
}
int32_t MicrophoneBoost(bool& enabled) const override {
FATAL() << "Should never be called";
return -1;
}
int32_t StereoPlayoutIsAvailable(bool& available) override {
LOG(INFO) << __FUNCTION__;
available = false;
return 0;
}
// TODO(henrika): add support.
int32_t SetStereoPlayout(bool enable) override {
LOG(INFO) << __FUNCTION__;
return -1;
}
// TODO(henrika): add support.
int32_t StereoPlayout(bool& enabled) const override {
enabled = false;
FATAL() << "Should never be called";
return -1;
}
int32_t StereoRecordingIsAvailable(bool& available) override {
LOG(INFO) << __FUNCTION__;
available = false;
return 0;
}
int32_t SetStereoRecording(bool enable) override {
LOG(INFO) << __FUNCTION__;
return -1;
}
int32_t StereoRecording(bool& enabled) const override {
LOG(INFO) << __FUNCTION__;
enabled = false;
return 0;
}
int32_t SetPlayoutBuffer(
const AudioDeviceModule::BufferType type, uint16_t sizeMS) override {
FATAL() << "Should never be called";
return -1;
}
int32_t PlayoutBuffer(
AudioDeviceModule::BufferType& type, uint16_t& sizeMS) const override {
FATAL() << "Should never be called";
return -1;
}
int32_t PlayoutDelay(uint16_t& delay_ms) const override {
// Best guess we can do is to use half of the estimated total delay.
delay_ms = audio_manager_->GetDelayEstimateInMilliseconds() / 2;
RTC_DCHECK_GT(delay_ms, 0);
return 0;
}
int32_t RecordingDelay(uint16_t& delay_ms) const override {
// Best guess we can do is to use half of the estimated total delay.
LOG(INFO) << __FUNCTION__;
delay_ms = audio_manager_->GetDelayEstimateInMilliseconds() / 2;
RTC_DCHECK_GT(delay_ms, 0);
return 0;
}
int32_t CPULoad(uint16_t& load) const override {
FATAL() << "Should never be called";
return -1;
}
bool PlayoutWarning() const override {
return false;
}
bool PlayoutError() const override {
return false;
}
bool RecordingWarning() const override {
return false;
}
bool RecordingError() const override {
return false;
}
void ClearPlayoutWarning() override { LOG(INFO) << __FUNCTION__; }
void ClearPlayoutError() override { LOG(INFO) << __FUNCTION__; }
void ClearRecordingWarning() override { LOG(INFO) << __FUNCTION__; }
void ClearRecordingError() override { LOG(INFO) << __FUNCTION__; }
void AttachAudioBuffer(AudioDeviceBuffer* audioBuffer) override {
LOG(INFO) << __FUNCTION__;
output_.AttachAudioBuffer(audioBuffer);
input_.AttachAudioBuffer(audioBuffer);
}
// TODO(henrika): remove
int32_t SetPlayoutSampleRate(const uint32_t samplesPerSec) override {
FATAL() << "Should never be called";
return -1;
}
int32_t SetLoudspeakerStatus(bool enable) override {
FATAL() << "Should never be called";
return -1;
}
int32_t GetLoudspeakerStatus(bool& enable) const override {
FATAL() << "Should never be called";
return -1;
}
// Returns true if the device both supports built in AEC and the device
// is not blacklisted.
// Currently, if OpenSL ES is used in both directions, this method will still
// report the correct value and it has the correct effect. As an example:
// a device supports built in AEC and this method returns true. Libjingle
// will then disable the WebRTC based AEC and that will work for all devices
// (mainly Nexus) even when OpenSL ES is used for input since our current
// implementation will enable built-in AEC by default also for OpenSL ES.
// The only "bad" thing that happens today is that when Libjingle calls
// OpenSLESRecorder::EnableBuiltInAEC() it will not have any real effect and
// a "Not Implemented" log will be filed. This non-perfect state will remain
// until I have added full support for audio effects based on OpenSL ES APIs.
bool BuiltInAECIsAvailable() const override {
LOG(INFO) << __FUNCTION__;
return audio_manager_->IsAcousticEchoCancelerSupported();
}
// TODO(henrika): add implementation for OpenSL ES based audio as well.
int32_t EnableBuiltInAEC(bool enable) override {
LOG(INFO) << __FUNCTION__ << "(" << enable << ")";
RTC_CHECK(BuiltInAECIsAvailable()) << "HW AEC is not available";
return input_.EnableBuiltInAEC(enable);
}
// Returns true if the device both supports built in AGC and the device
// is not blacklisted.
// TODO(henrika): add implementation for OpenSL ES based audio as well.
// In addition, see comments for BuiltInAECIsAvailable().
bool BuiltInAGCIsAvailable() const override {
LOG(INFO) << __FUNCTION__;
return audio_manager_->IsAutomaticGainControlSupported();
}
// TODO(henrika): add implementation for OpenSL ES based audio as well.
int32_t EnableBuiltInAGC(bool enable) override {
LOG(INFO) << __FUNCTION__ << "(" << enable << ")";
RTC_CHECK(BuiltInAGCIsAvailable()) << "HW AGC is not available";
return input_.EnableBuiltInAGC(enable);
}
// Returns true if the device both supports built in NS and the device
// is not blacklisted.
// TODO(henrika): add implementation for OpenSL ES based audio as well.
// In addition, see comments for BuiltInAECIsAvailable().
bool BuiltInNSIsAvailable() const override {
LOG(INFO) << __FUNCTION__;
return audio_manager_->IsNoiseSuppressorSupported();
}
// TODO(henrika): add implementation for OpenSL ES based audio as well.
int32_t EnableBuiltInNS(bool enable) override {
LOG(INFO) << __FUNCTION__ << "(" << enable << ")";
RTC_CHECK(BuiltInNSIsAvailable()) << "HW NS is not available";
return input_.EnableBuiltInNS(enable);
}
private:
rtc::ThreadChecker thread_checker_;
// Local copy of the audio layer set during construction of the
// AudioDeviceModuleImpl instance. Read only value.
const AudioDeviceModule::AudioLayer audio_layer_;
// Non-owning raw pointer to AudioManager instance given to use at
// construction. The real object is owned by AudioDeviceModuleImpl and the
// life time is the same as that of the AudioDeviceModuleImpl, hence there
// is no risk of reading a NULL pointer at any time in this class.
AudioManager* const audio_manager_;
OutputType output_;
InputType input_;
bool initialized_;
};
} // namespace webrtc
#endif // WEBRTC_MODULES_AUDIO_DEVICE_ANDROID_AUDIO_DEVICE_TEMPLATE_H_