| /* |
| * 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 MODULES_AUDIO_DEVICE_ANDROID_AUDIO_DEVICE_TEMPLATE_H_ |
| #define MODULES_AUDIO_DEVICE_ANDROID_AUDIO_DEVICE_TEMPLATE_H_ |
| |
| #include "modules/audio_device/android/audio_manager.h" |
| #include "modules/audio_device/audio_device_generic.h" |
| #include "rtc_base/checks.h" |
| #include "rtc_base/logging.h" |
| #include "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) { |
| RTC_LOG(INFO) << __FUNCTION__; |
| RTC_CHECK(audio_manager); |
| audio_manager_->SetActiveAudioLayer(audio_layer); |
| } |
| |
| virtual ~AudioDeviceTemplate() { RTC_LOG(INFO) << __FUNCTION__; } |
| |
| int32_t ActiveAudioLayer( |
| AudioDeviceModule::AudioLayer& audioLayer) const override { |
| RTC_LOG(INFO) << __FUNCTION__; |
| audioLayer = audio_layer_; |
| return 0; |
| } |
| |
| InitStatus Init() override { |
| RTC_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 { |
| RTC_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 { |
| RTC_LOG(INFO) << __FUNCTION__; |
| RTC_DCHECK(thread_checker_.CalledOnValidThread()); |
| return initialized_; |
| } |
| |
| int16_t PlayoutDevices() override { |
| RTC_LOG(INFO) << __FUNCTION__; |
| return 1; |
| } |
| |
| int16_t RecordingDevices() override { |
| RTC_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. |
| RTC_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. |
| RTC_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 { |
| RTC_LOG(INFO) << __FUNCTION__; |
| available = true; |
| return 0; |
| } |
| |
| int32_t InitPlayout() override { |
| RTC_LOG(INFO) << __FUNCTION__; |
| return output_.InitPlayout(); |
| } |
| |
| bool PlayoutIsInitialized() const override { |
| RTC_LOG(INFO) << __FUNCTION__; |
| return output_.PlayoutIsInitialized(); |
| } |
| |
| int32_t RecordingIsAvailable(bool& available) override { |
| RTC_LOG(INFO) << __FUNCTION__; |
| available = true; |
| return 0; |
| } |
| |
| int32_t InitRecording() override { |
| RTC_LOG(INFO) << __FUNCTION__; |
| return input_.InitRecording(); |
| } |
| |
| bool RecordingIsInitialized() const override { |
| RTC_LOG(INFO) << __FUNCTION__; |
| return input_.RecordingIsInitialized(); |
| } |
| |
| int32_t StartPlayout() override { |
| RTC_LOG(INFO) << __FUNCTION__; |
| if (!audio_manager_->IsCommunicationModeEnabled()) { |
| RTC_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; |
| RTC_LOG(INFO) << __FUNCTION__; |
| int32_t err = output_.StopPlayout(); |
| return err; |
| } |
| |
| bool Playing() const override { |
| RTC_LOG(INFO) << __FUNCTION__; |
| return output_.Playing(); |
| } |
| |
| int32_t StartRecording() override { |
| RTC_LOG(INFO) << __FUNCTION__; |
| if (!audio_manager_->IsCommunicationModeEnabled()) { |
| RTC_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. |
| RTC_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 { |
| RTC_LOG(INFO) << __FUNCTION__; |
| return false; |
| } |
| |
| int32_t InitSpeaker() override { |
| RTC_LOG(INFO) << __FUNCTION__; |
| return 0; |
| } |
| |
| bool SpeakerIsInitialized() const override { |
| RTC_LOG(INFO) << __FUNCTION__; |
| return true; |
| } |
| |
| int32_t InitMicrophone() override { |
| RTC_LOG(INFO) << __FUNCTION__; |
| return 0; |
| } |
| |
| bool MicrophoneIsInitialized() const override { |
| RTC_LOG(INFO) << __FUNCTION__; |
| return true; |
| } |
| |
| int32_t SpeakerVolumeIsAvailable(bool& available) override { |
| RTC_LOG(INFO) << __FUNCTION__; |
| return output_.SpeakerVolumeIsAvailable(available); |
| } |
| |
| int32_t SetSpeakerVolume(uint32_t volume) override { |
| RTC_LOG(INFO) << __FUNCTION__; |
| return output_.SetSpeakerVolume(volume); |
| } |
| |
| int32_t SpeakerVolume(uint32_t& volume) const override { |
| RTC_LOG(INFO) << __FUNCTION__; |
| return output_.SpeakerVolume(volume); |
| } |
| |
| int32_t MaxSpeakerVolume(uint32_t& maxVolume) const override { |
| RTC_LOG(INFO) << __FUNCTION__; |
| return output_.MaxSpeakerVolume(maxVolume); |
| } |
| |
| int32_t MinSpeakerVolume(uint32_t& minVolume) const override { |
| RTC_LOG(INFO) << __FUNCTION__; |
| return output_.MinSpeakerVolume(minVolume); |
| } |
| |
| 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 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; |
| } |
| |
| // Returns true if the audio manager has been configured to support stereo |
| // and false otherwised. Default is mono. |
| int32_t StereoPlayoutIsAvailable(bool& available) override { |
| RTC_LOG(INFO) << __FUNCTION__; |
| available = audio_manager_->IsStereoPlayoutSupported(); |
| return 0; |
| } |
| |
| int32_t SetStereoPlayout(bool enable) override { |
| RTC_LOG(INFO) << __FUNCTION__; |
| bool available = audio_manager_->IsStereoPlayoutSupported(); |
| // Android does not support changes between mono and stero on the fly. |
| // Instead, the native audio layer is configured via the audio manager |
| // to either support mono or stereo. It is allowed to call this method |
| // if that same state is not modified. |
| return (enable == available) ? 0 : -1; |
| } |
| |
| int32_t StereoPlayout(bool& enabled) const override { |
| enabled = audio_manager_->IsStereoPlayoutSupported(); |
| return 0; |
| } |
| |
| int32_t StereoRecordingIsAvailable(bool& available) override { |
| RTC_LOG(INFO) << __FUNCTION__; |
| available = audio_manager_->IsStereoRecordSupported(); |
| return 0; |
| } |
| |
| int32_t SetStereoRecording(bool enable) override { |
| RTC_LOG(INFO) << __FUNCTION__; |
| bool available = audio_manager_->IsStereoRecordSupported(); |
| // Android does not support changes between mono and stero on the fly. |
| // Instead, the native audio layer is configured via the audio manager |
| // to either support mono or stereo. It is allowed to call this method |
| // if that same state is not modified. |
| return (enable == available) ? 0 : -1; |
| } |
| |
| int32_t StereoRecording(bool& enabled) const override { |
| RTC_LOG(INFO) << __FUNCTION__; |
| enabled = audio_manager_->IsStereoRecordSupported(); |
| return 0; |
| } |
| |
| 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; |
| } |
| |
| void AttachAudioBuffer(AudioDeviceBuffer* audioBuffer) override { |
| RTC_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 { |
| RTC_LOG(INFO) << __FUNCTION__; |
| return audio_manager_->IsAcousticEchoCancelerSupported(); |
| } |
| |
| // TODO(henrika): add implementation for OpenSL ES based audio as well. |
| int32_t EnableBuiltInAEC(bool enable) override { |
| RTC_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 { |
| RTC_LOG(INFO) << __FUNCTION__; |
| return audio_manager_->IsAutomaticGainControlSupported(); |
| } |
| |
| // TODO(henrika): add implementation for OpenSL ES based audio as well. |
| int32_t EnableBuiltInAGC(bool enable) override { |
| RTC_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 { |
| RTC_LOG(INFO) << __FUNCTION__; |
| return audio_manager_->IsNoiseSuppressorSupported(); |
| } |
| |
| // TODO(henrika): add implementation for OpenSL ES based audio as well. |
| int32_t EnableBuiltInNS(bool enable) override { |
| RTC_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 // MODULES_AUDIO_DEVICE_ANDROID_AUDIO_DEVICE_TEMPLATE_H_ |