ba35d05a49
TBR=tkchin BUG=webrtc:4789 TEST=modules_unittests --gtest_filter=AudioDeviceTest* and AppRTCDemo Review URL: https://codereview.webrtc.org/1206783002 . Cr-Commit-Position: refs/heads/master@{#9578}
237 lines
9.0 KiB
C++
237 lines
9.0 KiB
C++
/*
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* Copyright (c) 2012 The WebRTC project authors. All Rights Reserved.
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*
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* Use of this source code is governed by a BSD-style license
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* that can be found in the LICENSE file in the root of the source
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* tree. An additional intellectual property rights grant can be found
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* in the file PATENTS. All contributing project authors may
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* be found in the AUTHORS file in the root of the source tree.
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*/
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#ifndef WEBRTC_AUDIO_DEVICE_AUDIO_DEVICE_IOS_H
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#define WEBRTC_AUDIO_DEVICE_AUDIO_DEVICE_IOS_H
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#include <AudioUnit/AudioUnit.h>
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#include "webrtc/base/thread_checker.h"
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#include "webrtc/modules/audio_device/audio_device_generic.h"
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#include "webrtc/system_wrappers/interface/critical_section_wrapper.h"
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#include "webrtc/system_wrappers/interface/thread_wrapper.h"
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namespace webrtc {
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const uint32_t N_REC_SAMPLES_PER_SEC = 44100;
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const uint32_t N_PLAY_SAMPLES_PER_SEC = 44100;
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const uint32_t ENGINE_REC_BUF_SIZE_IN_SAMPLES = (N_REC_SAMPLES_PER_SEC / 100);
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const uint32_t ENGINE_PLAY_BUF_SIZE_IN_SAMPLES = (N_PLAY_SAMPLES_PER_SEC / 100);
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// Number of 10 ms recording blocks in recording buffer
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const uint16_t N_REC_BUFFERS = 20;
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class AudioDeviceIOS : public AudioDeviceGeneric {
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public:
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AudioDeviceIOS();
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~AudioDeviceIOS();
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void AttachAudioBuffer(AudioDeviceBuffer* audioBuffer) override;
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int32_t Init() override;
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int32_t Terminate() override;
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bool Initialized() const override { return _initialized; }
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int32_t InitPlayout() override;
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bool PlayoutIsInitialized() const override { return _playIsInitialized; }
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int32_t InitRecording() override;
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bool RecordingIsInitialized() const override { return _recIsInitialized; }
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int32_t StartPlayout() override;
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int32_t StopPlayout() override;
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bool Playing() const override { return _playing; }
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int32_t StartRecording() override;
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int32_t StopRecording() override;
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bool Recording() const override { return _recording; }
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int32_t SetLoudspeakerStatus(bool enable) override;
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int32_t GetLoudspeakerStatus(bool& enabled) const override;
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// TODO(henrika): investigate if we can reduce the complexity here.
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// Do we even need delay estimates?
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int32_t PlayoutDelay(uint16_t& delayMS) const override;
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int32_t RecordingDelay(uint16_t& delayMS) const override;
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int32_t PlayoutBuffer(AudioDeviceModule::BufferType& type,
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uint16_t& sizeMS) const override;
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// These methods are unique for the iOS implementation.
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// Native audio parameters stored during construction.
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int GetPlayoutAudioParameters(AudioParameters* params) const override;
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int GetRecordAudioParameters(AudioParameters* params) const override;
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// These methods are currently not implemented on iOS.
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// See audio_device_not_implemented_ios.mm for dummy implementations.
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int32_t ActiveAudioLayer(AudioDeviceModule::AudioLayer& audioLayer) const;
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int32_t ResetAudioDevice() override;
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int32_t PlayoutIsAvailable(bool& available) override;
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int32_t RecordingIsAvailable(bool& available) override;
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int32_t SetAGC(bool enable) override;
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bool AGC() const override;
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int16_t PlayoutDevices() override;
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int16_t RecordingDevices() override;
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int32_t PlayoutDeviceName(uint16_t index,
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char name[kAdmMaxDeviceNameSize],
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char guid[kAdmMaxGuidSize]) override;
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int32_t RecordingDeviceName(uint16_t index,
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char name[kAdmMaxDeviceNameSize],
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char guid[kAdmMaxGuidSize]) override;
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int32_t SetPlayoutDevice(uint16_t index) override;
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int32_t SetPlayoutDevice(
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AudioDeviceModule::WindowsDeviceType device) override;
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int32_t SetRecordingDevice(uint16_t index) override;
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int32_t SetRecordingDevice(
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AudioDeviceModule::WindowsDeviceType device) override;
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int32_t SetWaveOutVolume(uint16_t volumeLeft, uint16_t volumeRight) override;
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int32_t WaveOutVolume(uint16_t& volumeLeft,
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uint16_t& volumeRight) const override;
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int32_t InitSpeaker() override;
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bool SpeakerIsInitialized() const override;
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int32_t InitMicrophone() override;
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bool MicrophoneIsInitialized() const override;
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int32_t SpeakerVolumeIsAvailable(bool& available) override;
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int32_t SetSpeakerVolume(uint32_t volume) override;
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int32_t SpeakerVolume(uint32_t& volume) const override;
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int32_t MaxSpeakerVolume(uint32_t& maxVolume) const override;
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int32_t MinSpeakerVolume(uint32_t& minVolume) const override;
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int32_t SpeakerVolumeStepSize(uint16_t& stepSize) const override;
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int32_t MicrophoneVolumeIsAvailable(bool& available) override;
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int32_t SetMicrophoneVolume(uint32_t volume) override;
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int32_t MicrophoneVolume(uint32_t& volume) const override;
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int32_t MaxMicrophoneVolume(uint32_t& maxVolume) const override;
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int32_t MinMicrophoneVolume(uint32_t& minVolume) const override;
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int32_t MicrophoneVolumeStepSize(uint16_t& stepSize) const override;
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int32_t MicrophoneMuteIsAvailable(bool& available) override;
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int32_t SetMicrophoneMute(bool enable) override;
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int32_t MicrophoneMute(bool& enabled) const override;
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int32_t SpeakerMuteIsAvailable(bool& available) override;
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int32_t SetSpeakerMute(bool enable) override;
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int32_t SpeakerMute(bool& enabled) const override;
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int32_t MicrophoneBoostIsAvailable(bool& available) override;
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int32_t SetMicrophoneBoost(bool enable) override;
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int32_t MicrophoneBoost(bool& enabled) const override;
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int32_t StereoPlayoutIsAvailable(bool& available) override;
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int32_t SetStereoPlayout(bool enable) override;
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int32_t StereoPlayout(bool& enabled) const override;
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int32_t StereoRecordingIsAvailable(bool& available) override;
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int32_t SetStereoRecording(bool enable) override;
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int32_t StereoRecording(bool& enabled) const override;
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int32_t SetPlayoutBuffer(const AudioDeviceModule::BufferType type,
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uint16_t sizeMS) override;
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int32_t CPULoad(uint16_t& load) const override;
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bool PlayoutWarning() const override;
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bool PlayoutError() const override;
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bool RecordingWarning() const override;
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bool RecordingError() const override;
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void ClearPlayoutWarning() override{};
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void ClearPlayoutError() override{};
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void ClearRecordingWarning() override{};
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void ClearRecordingError() override{};
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private:
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// TODO(henrika): try to remove these.
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void Lock() {
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_critSect.Enter();
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}
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void UnLock() {
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_critSect.Leave();
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}
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// Init and shutdown
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int32_t InitPlayOrRecord();
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int32_t ShutdownPlayOrRecord();
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void UpdateRecordingDelay();
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void UpdatePlayoutDelay();
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static OSStatus RecordProcess(void *inRefCon,
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AudioUnitRenderActionFlags *ioActionFlags,
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const AudioTimeStamp *timeStamp,
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UInt32 inBusNumber,
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UInt32 inNumberFrames,
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AudioBufferList *ioData);
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static OSStatus PlayoutProcess(void *inRefCon,
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AudioUnitRenderActionFlags *ioActionFlags,
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const AudioTimeStamp *timeStamp,
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UInt32 inBusNumber,
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UInt32 inNumberFrames,
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AudioBufferList *ioData);
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OSStatus RecordProcessImpl(AudioUnitRenderActionFlags *ioActionFlags,
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const AudioTimeStamp *timeStamp,
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uint32_t inBusNumber,
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uint32_t inNumberFrames);
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OSStatus PlayoutProcessImpl(uint32_t inNumberFrames,
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AudioBufferList *ioData);
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static bool RunCapture(void* ptrThis);
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bool CaptureWorkerThread();
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private:
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rtc::ThreadChecker thread_checker_;
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// Raw pointer handle provided to us in AttachAudioBuffer(). Owned by the
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// AudioDeviceModuleImpl class and called by AudioDeviceModuleImpl::Create().
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// The AudioDeviceBuffer is a member of the AudioDeviceModuleImpl instance
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// and therefore outlives this object.
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AudioDeviceBuffer* audio_device_buffer_;
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CriticalSectionWrapper& _critSect;
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AudioParameters playout_parameters_;
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AudioParameters record_parameters_;
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rtc::scoped_ptr<ThreadWrapper> _captureWorkerThread;
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AudioUnit _auVoiceProcessing;
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void* _audioInterruptionObserver;
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bool _initialized;
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bool _isShutDown;
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bool _recording;
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bool _playing;
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bool _recIsInitialized;
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bool _playIsInitialized;
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// The sampling rate to use with Audio Device Buffer
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int _adbSampFreq;
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// Delay calculation
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uint32_t _recordingDelay;
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uint32_t _playoutDelay;
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uint32_t _playoutDelayMeasurementCounter;
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uint32_t _recordingDelayHWAndOS;
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uint32_t _recordingDelayMeasurementCounter;
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// Playout buffer, needed for 44.0 / 44.1 kHz mismatch
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int16_t _playoutBuffer[ENGINE_PLAY_BUF_SIZE_IN_SAMPLES];
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uint32_t _playoutBufferUsed; // How much is filled
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// Recording buffers
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int16_t _recordingBuffer[N_REC_BUFFERS][ENGINE_REC_BUF_SIZE_IN_SAMPLES];
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uint32_t _recordingLength[N_REC_BUFFERS];
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uint32_t _recordingSeqNumber[N_REC_BUFFERS];
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uint32_t _recordingCurrentSeq;
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// Current total size all data in buffers, used for delay estimate
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uint32_t _recordingBufferTotalSize;
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};
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} // namespace webrtc
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#endif // WEBRTC_AUDIO_DEVICE_AUDIO_DEVICE_IOS_H
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