1380 lines
55 KiB
C++
1380 lines
55 KiB
C++
/*
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==============================================================================
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This file is part of the JUCE library.
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Copyright (c) 2017 - ROLI Ltd.
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JUCE is an open source library subject to commercial or open-source
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licensing.
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The code included in this file is provided under the terms of the ISC license
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http://www.isc.org/downloads/software-support-policy/isc-license. Permission
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To use, copy, modify, and/or distribute this software for any purpose with or
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without fee is hereby granted provided that the above copyright notice and
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this permission notice appear in all copies.
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JUCE IS PROVIDED "AS IS" WITHOUT ANY WARRANTY, AND ALL WARRANTIES, WHETHER
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EXPRESSED OR IMPLIED, INCLUDING MERCHANTABILITY AND FITNESS FOR PURPOSE, ARE
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DISCLAIMED.
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==============================================================================
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*/
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namespace juce
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{
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//==============================================================================
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#ifndef SL_ANDROID_DATAFORMAT_PCM_EX
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#define SL_ANDROID_DATAFORMAT_PCM_EX ((SLuint32) 0x00000004)
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#endif
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#ifndef SL_ANDROID_PCM_REPRESENTATION_FLOAT
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#define SL_ANDROID_PCM_REPRESENTATION_FLOAT ((SLuint32) 0x00000003)
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#endif
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#ifndef SL_ANDROID_RECORDING_PRESET_UNPROCESSED
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#define SL_ANDROID_RECORDING_PRESET_UNPROCESSED ((SLuint32) 0x00000005)
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#endif
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//==============================================================================
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struct PCMDataFormatEx : SLDataFormat_PCM
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{
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SLuint32 representation;
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};
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//==============================================================================
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template <typename T> struct IntfIID;
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template <> struct IntfIID<SLObjectItf_> { static SLInterfaceID_ iid; };
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template <> struct IntfIID<SLEngineItf_> { static SLInterfaceID_ iid; };
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template <> struct IntfIID<SLOutputMixItf_> { static SLInterfaceID_ iid; };
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template <> struct IntfIID<SLPlayItf_> { static SLInterfaceID_ iid; };
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template <> struct IntfIID<SLRecordItf_> { static SLInterfaceID_ iid; };
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template <> struct IntfIID<SLAndroidSimpleBufferQueueItf_> { static SLInterfaceID_ iid; };
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template <> struct IntfIID<SLAndroidConfigurationItf_> { static SLInterfaceID_ iid; };
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SLInterfaceID_ IntfIID<SLObjectItf_>::iid = { 0x79216360, 0xddd7, 0x11db, 0xac16, {0x00, 0x02, 0xa5, 0xd5, 0xc5, 0x1b} };
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SLInterfaceID_ IntfIID<SLEngineItf_>::iid = { 0x8d97c260, 0xddd4, 0x11db, 0x958f, {0x00, 0x02, 0xa5, 0xd5, 0xc5, 0x1b} };
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SLInterfaceID_ IntfIID<SLOutputMixItf_>::iid = { 0x97750f60, 0xddd7, 0x11db, 0x92b1, {0x00, 0x02, 0xa5, 0xd5, 0xc5, 0x1b} };
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SLInterfaceID_ IntfIID<SLPlayItf_>::iid = { 0xef0bd9c0, 0xddd7, 0x11db, 0xbf49, {0x00, 0x02, 0xa5, 0xd5, 0xc5, 0x1b} };
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SLInterfaceID_ IntfIID<SLRecordItf_>::iid = { 0xc5657aa0, 0xdddb, 0x11db, 0x82f7, {0x00, 0x02, 0xa5, 0xd5, 0xc5, 0x1b} };
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SLInterfaceID_ IntfIID<SLAndroidSimpleBufferQueueItf_>::iid = { 0x198e4940, 0xc5d7, 0x11df, 0xa2a6, {0x00, 0x02, 0xa5, 0xd5, 0xc5, 0x1b} };
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SLInterfaceID_ IntfIID<SLAndroidConfigurationItf_>::iid = { 0x89f6a7e0, 0xbeac, 0x11df, 0x8b5c, {0x00, 0x02, 0xa5, 0xd5, 0xc5, 0x1b} };
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//==============================================================================
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// Some life-time and type management of OpenSL objects
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class SlObjectRef
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{
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public:
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//==============================================================================
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SlObjectRef() noexcept {}
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SlObjectRef (const SlObjectRef& obj) noexcept : cb (obj.cb) {}
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SlObjectRef (SlObjectRef&& obj) noexcept : cb (static_cast<ReferenceCountedObjectPtr<ControlBlock>&&> (obj.cb)) { obj.cb = nullptr; }
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explicit SlObjectRef (SLObjectItf o) : cb (new ControlBlock (o)) {}
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//==============================================================================
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SlObjectRef& operator=(const SlObjectRef& r) noexcept { cb = r.cb; return *this; }
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SlObjectRef& operator=(SlObjectRef&& r) noexcept { cb = static_cast<ReferenceCountedObjectPtr<ControlBlock>&&> (r.cb); r.cb = nullptr; return *this; }
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SlObjectRef& operator=(std::nullptr_t) noexcept { cb = nullptr; return *this; }
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//==============================================================================
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const SLObjectItf_* const operator*() noexcept { return *cb->ptr.get(); }
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SLObjectItf operator->() noexcept { return (cb == nullptr ? nullptr : cb->ptr.get()); }
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operator SLObjectItf() noexcept { return (cb == nullptr ? nullptr : cb->ptr.get()); }
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//==============================================================================
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bool operator== (nullptr_t) const noexcept { return (cb == nullptr || cb->ptr == nullptr); }
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bool operator!= (nullptr_t) const noexcept { return (cb != nullptr && cb->ptr != nullptr); }
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private:
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//==============================================================================
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struct ControlBlock : ReferenceCountedObject { std::unique_ptr<const SLObjectItf_* const> ptr; ControlBlock() {} ControlBlock (SLObjectItf o) : ptr (o) {} };
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ReferenceCountedObjectPtr<ControlBlock> cb;
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};
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template <typename T>
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class SlRef : public SlObjectRef
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{
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public:
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//==============================================================================
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SlRef() noexcept : type (nullptr) {}
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SlRef (SlRef& r) noexcept : SlObjectRef (r), type (r.type) {}
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SlRef (SlRef&& r) noexcept : SlObjectRef (static_cast<SlRef&&> (r)), type (r.type) { r.type = nullptr; }
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//==============================================================================
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SlRef& operator= (const SlRef& r) noexcept { SlObjectRef::operator= (r); type = r.type; return *this; }
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SlRef& operator= (SlRef&& r) noexcept { SlObjectRef::operator= (static_cast<SlObjectRef&&> (r)); type = r.type; r.type = nullptr; return *this; }
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SlRef& operator= (std::nullptr_t) noexcept { SlObjectRef::operator= (nullptr); type = nullptr; return *this; }
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//==============================================================================
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T* const operator*() noexcept { return *type; }
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T* const * operator->() noexcept { return type; }
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operator T* const *() noexcept { return type; }
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//==============================================================================
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static SlRef cast (SlObjectRef& base) { return SlRef (base); }
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static SlRef cast (SlObjectRef&& base) { return SlRef (static_cast<SlObjectRef&&> (base)); }
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private:
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//==============================================================================
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SlRef (SlObjectRef& base) : SlObjectRef (base)
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{
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SLObjectItf obj = SlObjectRef::operator->();
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SLresult err = (*obj)->GetInterface (obj, &IntfIID<T>::iid, &type);
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if (type == nullptr || err != SL_RESULT_SUCCESS)
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*this = nullptr;
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}
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SlRef (SlObjectRef&& base) : SlObjectRef (static_cast<SlObjectRef&&> (base))
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{
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SLObjectItf obj = SlObjectRef::operator->();
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SLresult err = (*obj)->GetInterface (obj, &IntfIID<T>::iid, &type);
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base = nullptr;
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if (type == nullptr || err != SL_RESULT_SUCCESS)
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*this = nullptr;
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}
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T* const * type;
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};
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template <>
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struct ContainerDeletePolicy<const SLObjectItf_* const>
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{
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static void destroy (SLObjectItf object)
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{
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if (object != nullptr)
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(*object)->Destroy (object);
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}
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};
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//==============================================================================
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template <typename T> struct BufferHelpers {};
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template <>
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struct BufferHelpers<int16>
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{
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enum { isFloatingPoint = 0 };
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static void initPCMDataFormat (PCMDataFormatEx& dataFormat, int numChannels, double sampleRate)
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{
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dataFormat.formatType = SL_DATAFORMAT_PCM;
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dataFormat.numChannels = (SLuint32) numChannels;
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dataFormat.samplesPerSec = (SLuint32) (sampleRate * 1000);
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dataFormat.bitsPerSample = SL_PCMSAMPLEFORMAT_FIXED_16;
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dataFormat.containerSize = SL_PCMSAMPLEFORMAT_FIXED_16;
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dataFormat.channelMask = (numChannels == 1) ? SL_SPEAKER_FRONT_CENTER :
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(SL_SPEAKER_FRONT_LEFT | SL_SPEAKER_FRONT_RIGHT);
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dataFormat.endianness = SL_BYTEORDER_LITTLEENDIAN;
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dataFormat.representation = 0;
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}
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static void prepareCallbackBuffer (AudioBuffer<float>&, int16*) {}
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static void convertFromOpenSL (const int16* srcInterleaved, AudioBuffer<float>& audioBuffer)
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{
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for (int i = 0; i < audioBuffer.getNumChannels(); ++i)
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{
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using DstSampleType = AudioData::Pointer<AudioData::Float32, AudioData::NativeEndian, AudioData::NonInterleaved, AudioData::NonConst>;
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using SrcSampleType = AudioData::Pointer<AudioData::Int16, AudioData::LittleEndian, AudioData::Interleaved, AudioData::Const>;
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DstSampleType dstData (audioBuffer.getWritePointer (i));
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SrcSampleType srcData (srcInterleaved + i, audioBuffer.getNumChannels());
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dstData.convertSamples (srcData, audioBuffer.getNumSamples());
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}
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}
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static void convertToOpenSL (const AudioBuffer<float>& audioBuffer, int16* dstInterleaved)
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{
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for (int i = 0; i < audioBuffer.getNumChannels(); ++i)
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{
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using DstSampleType = AudioData::Pointer<AudioData::Int16, AudioData::LittleEndian, AudioData::Interleaved, AudioData::NonConst>;
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using SrcSampleType = AudioData::Pointer<AudioData::Float32, AudioData::NativeEndian, AudioData::NonInterleaved, AudioData::Const>;
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DstSampleType dstData (dstInterleaved + i, audioBuffer.getNumChannels());
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SrcSampleType srcData (audioBuffer.getReadPointer (i));
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dstData.convertSamples (srcData, audioBuffer.getNumSamples());
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}
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}
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};
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template <>
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struct BufferHelpers<float>
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{
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enum { isFloatingPoint = 1 };
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static void initPCMDataFormat (PCMDataFormatEx& dataFormat, int numChannels, double sampleRate)
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{
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dataFormat.formatType = SL_ANDROID_DATAFORMAT_PCM_EX;
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dataFormat.numChannels = (SLuint32) numChannels;
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dataFormat.samplesPerSec = (SLuint32) (sampleRate * 1000);
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dataFormat.bitsPerSample = 32;
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dataFormat.containerSize = 32;
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dataFormat.channelMask = (numChannels == 1) ? SL_SPEAKER_FRONT_CENTER :
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(SL_SPEAKER_FRONT_LEFT | SL_SPEAKER_FRONT_RIGHT);
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dataFormat.endianness = SL_BYTEORDER_LITTLEENDIAN;
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dataFormat.representation = SL_ANDROID_PCM_REPRESENTATION_FLOAT;
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}
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static void prepareCallbackBuffer (AudioBuffer<float>& audioBuffer, float* native)
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{
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if (audioBuffer.getNumChannels() == 1)
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audioBuffer.setDataToReferTo (&native, 1, audioBuffer.getNumSamples());
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}
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static void convertFromOpenSL (const float* srcInterleaved, AudioBuffer<float>& audioBuffer)
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{
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if (audioBuffer.getNumChannels() == 1)
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{
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jassert (srcInterleaved == audioBuffer.getWritePointer (0));
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return;
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}
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for (int i = 0; i < audioBuffer.getNumChannels(); ++i)
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{
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using DstSampleType = AudioData::Pointer<AudioData::Float32, AudioData::NativeEndian, AudioData::NonInterleaved, AudioData::NonConst>;
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using SrcSampleType = AudioData::Pointer<AudioData::Float32, AudioData::LittleEndian, AudioData::Interleaved, AudioData::Const>;
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DstSampleType dstData (audioBuffer.getWritePointer (i));
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SrcSampleType srcData (srcInterleaved + i, audioBuffer.getNumChannels());
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dstData.convertSamples (srcData, audioBuffer.getNumSamples());
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}
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}
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static void convertToOpenSL (const AudioBuffer<float>& audioBuffer, float* dstInterleaved)
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{
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if (audioBuffer.getNumChannels() == 1)
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{
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jassert (dstInterleaved == audioBuffer.getReadPointer (0));
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return;
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}
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for (int i = 0; i < audioBuffer.getNumChannels(); ++i)
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{
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using DstSampleType = AudioData::Pointer<AudioData::Float32, AudioData::LittleEndian, AudioData::Interleaved, AudioData::NonConst>;
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using SrcSampleType = AudioData::Pointer<AudioData::Float32, AudioData::NativeEndian, AudioData::NonInterleaved, AudioData::Const>;
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DstSampleType dstData (dstInterleaved + i, audioBuffer.getNumChannels());
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SrcSampleType srcData (audioBuffer.getReadPointer (i));
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dstData.convertSamples (srcData, audioBuffer.getNumSamples());
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}
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}
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};
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class SLRealtimeThread;
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//==============================================================================
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class OpenSLAudioIODevice : public AudioIODevice
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{
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public:
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//==============================================================================
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template <typename T>
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class OpenSLSessionT;
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//==============================================================================
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// CRTP
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template <typename T, class Child, typename RunnerObjectType>
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struct OpenSLQueueRunner
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{
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OpenSLQueueRunner (OpenSLSessionT<T>& sessionToUse, int numChannelsToUse)
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: owner (sessionToUse),
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numChannels (numChannelsToUse),
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nativeBuffer (static_cast<size_t> (numChannels * owner.bufferSize * owner.numBuffers)),
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scratchBuffer (numChannelsToUse, owner.bufferSize),
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sampleBuffer (scratchBuffer.getArrayOfWritePointers(), numChannelsToUse, owner.bufferSize),
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nextBlock (0), numBlocksOut (0)
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{}
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~OpenSLQueueRunner()
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{
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if (config != nullptr && javaProxy != nullptr)
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{
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javaProxy.clear();
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(*config)->ReleaseJavaProxy (config, /*SL_ANDROID_JAVA_PROXY_ROUTING*/1);
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}
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}
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bool init()
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{
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runner = crtp().createPlayerOrRecorder();
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if (runner == nullptr)
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return false;
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const bool supportsJavaProxy = (getEnv()->GetStaticIntField (AndroidBuildVersion, AndroidBuildVersion.SDK_INT) >= 24);
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if (supportsJavaProxy)
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{
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// may return nullptr on some platforms - that's ok
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config = SlRef<SLAndroidConfigurationItf_>::cast (runner);
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if (config != nullptr)
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{
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jobject audioRoutingJni;
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auto status = (*config)->AcquireJavaProxy (config, /*SL_ANDROID_JAVA_PROXY_ROUTING*/1,
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&audioRoutingJni);
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if (status == SL_RESULT_SUCCESS && audioRoutingJni != 0)
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javaProxy = GlobalRef (audioRoutingJni);
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}
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}
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queue = SlRef<SLAndroidSimpleBufferQueueItf_>::cast (runner);
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if (queue == nullptr)
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return false;
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return ((*queue)->RegisterCallback (queue, staticFinished, this) == SL_RESULT_SUCCESS);
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}
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void clear()
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{
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nextBlock.set (0);
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numBlocksOut.set (0);
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zeromem (nativeBuffer.get(), static_cast<size_t> (owner.bufferSize * numChannels * owner.numBuffers) * sizeof (T));
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scratchBuffer.clear();
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(*queue)->Clear (queue);
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}
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void enqueueBuffer()
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{
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(*queue)->Enqueue (queue, getCurrentBuffer(), static_cast<SLuint32> (getBufferSizeInSamples() * sizeof (T)));
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++numBlocksOut;
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}
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bool isBufferAvailable() const { return (numBlocksOut.get() < owner.numBuffers); }
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T* getNextBuffer() { nextBlock.set((nextBlock.get() + 1) % owner.numBuffers); return getCurrentBuffer(); }
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T* getCurrentBuffer() { return nativeBuffer.get() + (static_cast<size_t> (nextBlock.get()) * getBufferSizeInSamples()); }
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size_t getBufferSizeInSamples() const { return static_cast<size_t> (owner.bufferSize * numChannels); }
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void finished (SLAndroidSimpleBufferQueueItf)
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{
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attachAndroidJNI();
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--numBlocksOut;
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owner.doSomeWorkOnAudioThread();
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}
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static void staticFinished (SLAndroidSimpleBufferQueueItf caller, void *pContext)
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{
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reinterpret_cast<OpenSLQueueRunner*> (pContext)->finished (caller);
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}
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// get the "this" pointer for CRTP
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Child& crtp() { return * ((Child*) this); }
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const Child& crtp() const { return * ((Child*) this); }
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OpenSLSessionT<T>& owner;
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SlRef<RunnerObjectType> runner;
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SlRef<SLAndroidSimpleBufferQueueItf_> queue;
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SlRef<SLAndroidConfigurationItf_> config;
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GlobalRef javaProxy;
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int numChannels;
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HeapBlock<T> nativeBuffer;
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AudioBuffer<float> scratchBuffer, sampleBuffer;
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Atomic<int> nextBlock, numBlocksOut;
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};
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//==============================================================================
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template <typename T>
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struct OpenSLQueueRunnerPlayer : OpenSLQueueRunner<T, OpenSLQueueRunnerPlayer<T>, SLPlayItf_>
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{
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using Base = OpenSLQueueRunner<T, OpenSLQueueRunnerPlayer<T>, SLPlayItf_>;
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enum { isPlayer = 1 };
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OpenSLQueueRunnerPlayer (OpenSLSessionT<T>& sessionToUse, int numChannelsToUse)
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: Base (sessionToUse, numChannelsToUse)
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{}
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SlRef<SLPlayItf_> createPlayerOrRecorder()
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{
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SLDataLocator_AndroidSimpleBufferQueue queueLocator = {SL_DATALOCATOR_ANDROIDSIMPLEBUFFERQUEUE, static_cast<SLuint32> (Base::owner.numBuffers)};
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SLDataLocator_OutputMix outputMix = {SL_DATALOCATOR_OUTPUTMIX, Base::owner.outputMix};
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PCMDataFormatEx dataFormat;
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BufferHelpers<T>::initPCMDataFormat (dataFormat, Base::numChannels, Base::owner.sampleRate);
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SLDataSource source = {&queueLocator, &dataFormat};
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SLDataSink sink = {&outputMix, nullptr};
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SLInterfaceID queueInterfaces[] = { &IntfIID<SLAndroidSimpleBufferQueueItf_>::iid, &IntfIID<SLAndroidConfigurationItf_>::iid };
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SLboolean interfaceRequired[] = {SL_BOOLEAN_TRUE, SL_BOOLEAN_FALSE};
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SLObjectItf obj = nullptr;
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SLresult status = (*Base::owner.engine)->CreateAudioPlayer (Base::owner.engine, &obj, &source, &sink, 2, queueInterfaces, interfaceRequired);
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if (status != SL_RESULT_SUCCESS || obj == nullptr || (*obj)->Realize (obj, 0) != SL_RESULT_SUCCESS)
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{
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if (obj != nullptr)
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(*obj)->Destroy (obj);
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return SlRef<SLPlayItf_>();
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}
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return SlRef<SLPlayItf_>::cast (SlObjectRef (obj));
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}
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void setState (bool running) { (*Base::runner)->SetPlayState (Base::runner, running ? SL_PLAYSTATE_PLAYING : SL_PLAYSTATE_STOPPED); }
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};
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template <typename T>
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struct OpenSLQueueRunnerRecorder : OpenSLQueueRunner<T, OpenSLQueueRunnerRecorder<T>, SLRecordItf_>
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{
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using Base = OpenSLQueueRunner<T, OpenSLQueueRunnerRecorder<T>, SLRecordItf_>;
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enum { isPlayer = 0 };
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OpenSLQueueRunnerRecorder (OpenSLSessionT<T>& sessionToUse, int numChannelsToUse)
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: Base (sessionToUse, numChannelsToUse)
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{}
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SlRef<SLRecordItf_> createPlayerOrRecorder()
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{
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SLDataLocator_IODevice ioDeviceLocator = {SL_DATALOCATOR_IODEVICE, SL_IODEVICE_AUDIOINPUT, SL_DEFAULTDEVICEID_AUDIOINPUT, nullptr};
|
|
SLDataLocator_AndroidSimpleBufferQueue queueLocator = {SL_DATALOCATOR_ANDROIDSIMPLEBUFFERQUEUE, static_cast<SLuint32> (Base::owner.numBuffers)};
|
|
|
|
PCMDataFormatEx dataFormat;
|
|
BufferHelpers<T>::initPCMDataFormat (dataFormat, Base::numChannels, Base::owner.sampleRate);
|
|
|
|
SLDataSource source = {&ioDeviceLocator, nullptr};
|
|
SLDataSink sink = {&queueLocator, &dataFormat};
|
|
|
|
SLInterfaceID queueInterfaces[] = { &IntfIID<SLAndroidSimpleBufferQueueItf_>::iid, &IntfIID<SLAndroidConfigurationItf_>::iid };
|
|
SLboolean interfaceRequired[] = {SL_BOOLEAN_TRUE, SL_BOOLEAN_FALSE};
|
|
|
|
SLObjectItf obj = nullptr;
|
|
|
|
SLresult status = (*Base::owner.engine)->CreateAudioRecorder (Base::owner.engine, &obj, &source, &sink, 2, queueInterfaces, interfaceRequired);
|
|
if (status != SL_RESULT_SUCCESS || obj == nullptr || (*obj)->Realize (obj, 0) != SL_RESULT_SUCCESS)
|
|
{
|
|
if (obj != nullptr)
|
|
(*obj)->Destroy (obj);
|
|
|
|
return SlRef<SLRecordItf_>();
|
|
}
|
|
|
|
SlRef<SLRecordItf_> recorder = SlRef<SLRecordItf_>::cast (SlObjectRef (obj));
|
|
|
|
return recorder;
|
|
}
|
|
|
|
bool setAudioPreprocessingEnabled (bool shouldEnable)
|
|
{
|
|
if (Base::config != nullptr)
|
|
{
|
|
const bool supportsUnprocessed = (getEnv()->GetStaticIntField (AndroidBuildVersion, AndroidBuildVersion.SDK_INT) >= 25);
|
|
const SLuint32 recordingPresetValue
|
|
= (shouldEnable ? SL_ANDROID_RECORDING_PRESET_GENERIC
|
|
: (supportsUnprocessed ? SL_ANDROID_RECORDING_PRESET_UNPROCESSED
|
|
: SL_ANDROID_RECORDING_PRESET_VOICE_RECOGNITION));
|
|
|
|
SLresult status = (*Base::config)->SetConfiguration (Base::config, SL_ANDROID_KEY_RECORDING_PRESET,
|
|
&recordingPresetValue, sizeof (recordingPresetValue));
|
|
|
|
return (status == SL_RESULT_SUCCESS);
|
|
}
|
|
|
|
return false;
|
|
}
|
|
|
|
void setState (bool running) { (*Base::runner)->SetRecordState (Base::runner, running ? SL_RECORDSTATE_RECORDING : SL_RECORDSTATE_STOPPED); }
|
|
};
|
|
|
|
//==============================================================================
|
|
class OpenSLSession
|
|
{
|
|
public:
|
|
OpenSLSession (DynamicLibrary& slLibraryToUse,
|
|
int numInputChannels, int numOutputChannels,
|
|
double samleRateToUse, int bufferSizeToUse,
|
|
int numBuffersToUse)
|
|
: inputChannels (numInputChannels), outputChannels (numOutputChannels),
|
|
sampleRate (samleRateToUse), bufferSize (bufferSizeToUse), numBuffers (numBuffersToUse),
|
|
running (false), audioProcessingEnabled (true), callback (nullptr)
|
|
{
|
|
jassert (numInputChannels > 0 || numOutputChannels > 0);
|
|
|
|
if (CreateEngineFunc createEngine = (CreateEngineFunc) slLibraryToUse.getFunction ("slCreateEngine"))
|
|
{
|
|
SLObjectItf obj = nullptr;
|
|
|
|
SLresult err = createEngine (&obj, 0, nullptr, 0, nullptr, nullptr);
|
|
if (err != SL_RESULT_SUCCESS || obj == nullptr || (*obj)->Realize (obj, 0) != SL_RESULT_SUCCESS)
|
|
{
|
|
if (obj != nullptr)
|
|
(*obj)->Destroy (obj);
|
|
|
|
return;
|
|
}
|
|
|
|
engine = SlRef<SLEngineItf_>::cast (SlObjectRef (obj));
|
|
}
|
|
|
|
if (outputChannels > 0)
|
|
{
|
|
SLObjectItf obj = nullptr;
|
|
|
|
SLresult err = (*engine)->CreateOutputMix (engine, &obj, 0, nullptr, nullptr);
|
|
if (err != SL_RESULT_SUCCESS || obj == nullptr || (*obj)->Realize (obj, 0) != SL_RESULT_SUCCESS)
|
|
{
|
|
if (obj != nullptr)
|
|
(*obj)->Destroy (obj);
|
|
|
|
return;
|
|
}
|
|
|
|
outputMix = SlRef<SLOutputMixItf_>::cast (SlObjectRef (obj));
|
|
}
|
|
}
|
|
|
|
virtual ~OpenSLSession() {}
|
|
|
|
virtual bool openedOK() const { return (engine != nullptr && (outputChannels == 0 || (outputMix != nullptr))); }
|
|
virtual void start() { stop(); jassert (callback.get() != nullptr); running = true; }
|
|
virtual void stop() { running = false; }
|
|
virtual bool setAudioPreprocessingEnabled (bool shouldEnable) = 0;
|
|
virtual bool supportsFloatingPoint() const noexcept = 0;
|
|
virtual int getXRunCount() const noexcept = 0;
|
|
|
|
void setCallback (AudioIODeviceCallback* callbackToUse)
|
|
{
|
|
if (! running)
|
|
{
|
|
callback.set (callbackToUse);
|
|
return;
|
|
}
|
|
|
|
// don't set callback to null! stop the playback instead!
|
|
jassert (callbackToUse != nullptr);
|
|
|
|
// spin-lock until we can set the callback
|
|
while (true)
|
|
{
|
|
AudioIODeviceCallback* old = callback.get();
|
|
if (old == callbackToUse)
|
|
break;
|
|
|
|
if (callback.compareAndSetBool (callbackToUse, old))
|
|
break;
|
|
|
|
Thread::sleep (1);
|
|
}
|
|
}
|
|
|
|
void process (const float** inputChannelData, float** outputChannelData)
|
|
{
|
|
if (AudioIODeviceCallback* cb = callback.exchange(nullptr))
|
|
{
|
|
cb->audioDeviceIOCallback (inputChannelData, inputChannels, outputChannelData, outputChannels, bufferSize);
|
|
callback.set (cb);
|
|
}
|
|
else
|
|
{
|
|
for (int i = 0; i < outputChannels; ++i)
|
|
zeromem (outputChannelData[i], sizeof(float) * static_cast<size_t> (bufferSize));
|
|
}
|
|
}
|
|
|
|
static OpenSLSession* create (DynamicLibrary& slLibrary,
|
|
int numInputChannels, int numOutputChannels,
|
|
double samleRateToUse, int bufferSizeToUse,
|
|
int numBuffersToUse);
|
|
|
|
//==============================================================================
|
|
typedef SLresult (*CreateEngineFunc)(SLObjectItf*, SLuint32, const SLEngineOption*, SLuint32, const SLInterfaceID*, const SLboolean*);
|
|
|
|
//==============================================================================
|
|
int inputChannels, outputChannels;
|
|
double sampleRate;
|
|
int bufferSize, numBuffers;
|
|
|
|
bool running, audioProcessingEnabled;
|
|
|
|
SlRef<SLEngineItf_> engine;
|
|
SlRef<SLOutputMixItf_> outputMix;
|
|
|
|
Atomic<AudioIODeviceCallback*> callback;
|
|
};
|
|
|
|
template <typename T>
|
|
class OpenSLSessionT : public OpenSLSession
|
|
{
|
|
public:
|
|
OpenSLSessionT (DynamicLibrary& slLibraryToUse,
|
|
int numInputChannels, int numOutputChannels,
|
|
double samleRateToUse, int bufferSizeToUse,
|
|
int numBuffersToUse)
|
|
: OpenSLSession (slLibraryToUse, numInputChannels, numOutputChannels, samleRateToUse, bufferSizeToUse, numBuffersToUse)
|
|
{
|
|
jassert (numInputChannels > 0 || numOutputChannels > 0);
|
|
|
|
if (OpenSLSession::openedOK())
|
|
{
|
|
if (inputChannels > 0)
|
|
{
|
|
recorder.reset (new OpenSLQueueRunnerRecorder<T> (*this, inputChannels));
|
|
|
|
if (! recorder->init())
|
|
{
|
|
recorder = nullptr;
|
|
return;
|
|
}
|
|
}
|
|
|
|
if (outputChannels > 0)
|
|
{
|
|
player.reset (new OpenSLQueueRunnerPlayer<T> (*this, outputChannels));
|
|
|
|
if (! player->init())
|
|
{
|
|
player = nullptr;
|
|
return;
|
|
}
|
|
|
|
const bool supportsUnderrunCount = (getEnv()->GetStaticIntField (AndroidBuildVersion, AndroidBuildVersion.SDK_INT) >= 24);
|
|
getUnderrunCount = supportsUnderrunCount ? getEnv()->GetMethodID (AudioTrack, "getUnderrunCount", "()I") : 0;
|
|
}
|
|
}
|
|
}
|
|
|
|
bool openedOK() const override
|
|
{
|
|
return (OpenSLSession::openedOK() && (inputChannels == 0 || recorder != nullptr)
|
|
&& (outputChannels == 0 || player != nullptr));
|
|
}
|
|
|
|
void start() override
|
|
{
|
|
OpenSLSession::start();
|
|
|
|
guard.set (0);
|
|
|
|
if (inputChannels > 0)
|
|
recorder->clear();
|
|
|
|
if (outputChannels > 0)
|
|
player->clear();
|
|
|
|
// first enqueue all buffers
|
|
for (int i = 0; i < numBuffers; ++i)
|
|
doSomeWorkOnAudioThread();
|
|
|
|
if (inputChannels > 0)
|
|
recorder->setState (true);
|
|
|
|
if (outputChannels > 0)
|
|
player->setState (true);
|
|
}
|
|
|
|
void stop() override
|
|
{
|
|
OpenSLSession::stop();
|
|
|
|
while (! guard.compareAndSetBool (1, 0))
|
|
Thread::sleep (1);
|
|
|
|
if (inputChannels > 0)
|
|
recorder->setState (false);
|
|
|
|
if (outputChannels > 0)
|
|
player->setState (false);
|
|
|
|
guard.set (0);
|
|
}
|
|
|
|
bool setAudioPreprocessingEnabled (bool shouldEnable) override
|
|
{
|
|
if (shouldEnable != audioProcessingEnabled)
|
|
{
|
|
audioProcessingEnabled = shouldEnable;
|
|
|
|
if (recorder != nullptr)
|
|
return recorder->setAudioPreprocessingEnabled (audioProcessingEnabled);
|
|
}
|
|
|
|
return true;
|
|
}
|
|
|
|
int getXRunCount() const noexcept override
|
|
{
|
|
if (player != nullptr && player->javaProxy != nullptr && getUnderrunCount != 0)
|
|
return getEnv()->CallIntMethod (player->javaProxy, getUnderrunCount);
|
|
|
|
return -1;
|
|
}
|
|
|
|
bool supportsFloatingPoint() const noexcept override { return (BufferHelpers<T>::isFloatingPoint != 0); }
|
|
|
|
void doSomeWorkOnAudioThread()
|
|
{
|
|
// only the player or the recorder should enter this section at any time
|
|
if (guard.compareAndSetBool (1, 0))
|
|
{
|
|
// are there enough buffers avaialable to process some audio
|
|
if ((inputChannels == 0 || recorder->isBufferAvailable()) && (outputChannels == 0 || player->isBufferAvailable()))
|
|
{
|
|
T* recorderBuffer = (inputChannels > 0 ? recorder->getNextBuffer() : nullptr);
|
|
T* playerBuffer = (outputChannels > 0 ? player->getNextBuffer() : nullptr);
|
|
|
|
const float** inputChannelData = nullptr;
|
|
float** outputChannelData = nullptr;
|
|
|
|
if (recorderBuffer != nullptr)
|
|
{
|
|
BufferHelpers<T>::prepareCallbackBuffer (recorder->sampleBuffer, recorderBuffer);
|
|
BufferHelpers<T>::convertFromOpenSL (recorderBuffer, recorder->sampleBuffer);
|
|
|
|
inputChannelData = recorder->sampleBuffer.getArrayOfReadPointers();
|
|
}
|
|
|
|
if (playerBuffer != nullptr)
|
|
{
|
|
BufferHelpers<T>::prepareCallbackBuffer (player->sampleBuffer, playerBuffer);
|
|
outputChannelData = player->sampleBuffer.getArrayOfWritePointers();
|
|
}
|
|
|
|
process (inputChannelData, outputChannelData);
|
|
|
|
if (recorderBuffer != nullptr)
|
|
recorder->enqueueBuffer();
|
|
|
|
if (playerBuffer != nullptr)
|
|
{
|
|
BufferHelpers<T>::convertToOpenSL (player->sampleBuffer, playerBuffer);
|
|
player->enqueueBuffer();
|
|
}
|
|
}
|
|
|
|
guard.set (0);
|
|
}
|
|
}
|
|
|
|
//==============================================================================
|
|
std::unique_ptr<OpenSLQueueRunnerPlayer<T>> player;
|
|
std::unique_ptr<OpenSLQueueRunnerRecorder<T>> recorder;
|
|
Atomic<int> guard;
|
|
jmethodID getUnderrunCount = 0;
|
|
};
|
|
|
|
//==============================================================================
|
|
OpenSLAudioIODevice (const String& deviceName)
|
|
: AudioIODevice (deviceName, openSLTypeName),
|
|
actualBufferSize (0), sampleRate (0), audioBuffersToEnqueue (0),
|
|
audioProcessingEnabled (true),
|
|
callback (nullptr)
|
|
{
|
|
// OpenSL has piss-poor support for determining latency, so the only way I can find to
|
|
// get a number for this is by asking the AudioTrack/AudioRecord classes..
|
|
AndroidAudioIODevice javaDevice (deviceName);
|
|
|
|
// this is a total guess about how to calculate the latency, but seems to vaguely agree
|
|
// with the devices I've tested.. YMMV
|
|
inputLatency = (javaDevice.minBufferSizeIn * 2) / 3;
|
|
outputLatency = (javaDevice.minBufferSizeOut * 2) / 3;
|
|
|
|
const int64 longestLatency = jmax (inputLatency, outputLatency);
|
|
const int64 totalLatency = inputLatency + outputLatency;
|
|
inputLatency = (int) ((longestLatency * inputLatency) / totalLatency) & ~15;
|
|
outputLatency = (int) ((longestLatency * outputLatency) / totalLatency) & ~15;
|
|
|
|
bool success = slLibrary.open ("libOpenSLES.so");
|
|
|
|
// You can only create this class if you are sure that your hardware supports OpenSL
|
|
jassert (success);
|
|
ignoreUnused (success);
|
|
}
|
|
|
|
~OpenSLAudioIODevice()
|
|
{
|
|
close();
|
|
}
|
|
|
|
bool openedOk() const { return session != nullptr; }
|
|
|
|
StringArray getOutputChannelNames() override
|
|
{
|
|
StringArray s;
|
|
s.add ("Left");
|
|
s.add ("Right");
|
|
return s;
|
|
}
|
|
|
|
StringArray getInputChannelNames() override
|
|
{
|
|
StringArray s;
|
|
s.add ("Audio Input");
|
|
return s;
|
|
}
|
|
|
|
Array<double> getAvailableSampleRates() override
|
|
{
|
|
//see https://developer.android.com/ndk/guides/audio/opensl-for-android.html
|
|
|
|
static const double rates[] = { 8000.0, 11025.0, 12000.0, 16000.0,
|
|
22050.0, 24000.0, 32000.0, 44100.0, 48000.0 };
|
|
Array<double> retval (rates, numElementsInArray (rates));
|
|
|
|
// make sure the native sample rate is pafrt of the list
|
|
double native = getNativeSampleRate();
|
|
if (native != 0.0 && ! retval.contains (native))
|
|
retval.add (native);
|
|
|
|
return retval;
|
|
}
|
|
|
|
Array<int> getAvailableBufferSizes() override
|
|
{
|
|
// we need to offer the lowest possible buffer size which
|
|
// is the native buffer size
|
|
auto nativeBufferSize = getNativeBufferSize();
|
|
auto minBuffersToQueue = getMinimumBuffersToEnqueue();
|
|
auto maxBuffersToQueue = getMaximumBuffersToEnqueue();
|
|
|
|
Array<int> retval;
|
|
for (int i = minBuffersToQueue; i <= maxBuffersToQueue; ++i)
|
|
retval.add (i * nativeBufferSize);
|
|
|
|
return retval;
|
|
}
|
|
|
|
String open (const BigInteger& inputChannels,
|
|
const BigInteger& outputChannels,
|
|
double requestedSampleRate,
|
|
int bufferSize) override
|
|
{
|
|
close();
|
|
|
|
lastError.clear();
|
|
sampleRate = (int) requestedSampleRate;
|
|
|
|
auto totalPreferredBufferSize = (bufferSize <= 0) ? getDefaultBufferSize() : bufferSize;
|
|
auto nativeBufferSize = getNativeBufferSize();
|
|
bool useHighPerformanceAudioPath = canUseHighPerformanceAudioPath (totalPreferredBufferSize, sampleRate);
|
|
|
|
audioBuffersToEnqueue = useHighPerformanceAudioPath ? (totalPreferredBufferSize / nativeBufferSize) : 1;
|
|
actualBufferSize = totalPreferredBufferSize / audioBuffersToEnqueue;
|
|
jassert ((actualBufferSize * audioBuffersToEnqueue) == totalPreferredBufferSize);
|
|
|
|
activeOutputChans = outputChannels;
|
|
activeOutputChans.setRange (2, activeOutputChans.getHighestBit(), false);
|
|
int numOutputChannels = activeOutputChans.countNumberOfSetBits();
|
|
|
|
activeInputChans = inputChannels;
|
|
activeInputChans.setRange (1, activeInputChans.getHighestBit(), false);
|
|
int numInputChannels = activeInputChans.countNumberOfSetBits();
|
|
|
|
if (numInputChannels > 0 && (! RuntimePermissions::isGranted (RuntimePermissions::recordAudio)))
|
|
{
|
|
// If you hit this assert, you probably forgot to get RuntimePermissions::recordAudio
|
|
// before trying to open an audio input device. This is not going to work!
|
|
jassertfalse;
|
|
lastError = "Error opening OpenSL input device: the app was not granted android.permission.RECORD_AUDIO";
|
|
}
|
|
|
|
session.reset (OpenSLSession::create (slLibrary, numInputChannels, numOutputChannels,
|
|
sampleRate, actualBufferSize, audioBuffersToEnqueue));
|
|
if (session != nullptr)
|
|
session->setAudioPreprocessingEnabled (audioProcessingEnabled);
|
|
else
|
|
{
|
|
if (numInputChannels > 0 && numOutputChannels > 0 && RuntimePermissions::isGranted (RuntimePermissions::recordAudio))
|
|
{
|
|
// New versions of the Android emulator do not seem to support audio input anymore on OS X
|
|
activeInputChans = BigInteger(0);
|
|
numInputChannels = 0;
|
|
|
|
session.reset (OpenSLSession::create (slLibrary, numInputChannels, numOutputChannels,
|
|
sampleRate, actualBufferSize, audioBuffersToEnqueue));
|
|
}
|
|
}
|
|
|
|
DBG ("OpenSL: numInputChannels = " << numInputChannels
|
|
<< ", numOutputChannels = " << numOutputChannels
|
|
<< ", nativeBufferSize = " << getNativeBufferSize()
|
|
<< ", nativeSampleRate = " << getNativeSampleRate()
|
|
<< ", actualBufferSize = " << actualBufferSize
|
|
<< ", audioBuffersToEnqueue = " << audioBuffersToEnqueue
|
|
<< ", sampleRate = " << sampleRate
|
|
<< ", supportsFloatingPoint = " << (session != nullptr && session->supportsFloatingPoint() ? "true" : "false"));
|
|
|
|
if (session == nullptr)
|
|
lastError = "Unknown error initializing opensl session";
|
|
|
|
deviceOpen = (session != nullptr);
|
|
return lastError;
|
|
}
|
|
|
|
void close() override
|
|
{
|
|
stop();
|
|
session = nullptr;
|
|
callback = nullptr;
|
|
}
|
|
|
|
int getOutputLatencyInSamples() override { return outputLatency; }
|
|
int getInputLatencyInSamples() override { return inputLatency; }
|
|
bool isOpen() override { return deviceOpen; }
|
|
int getCurrentBufferSizeSamples() override { return actualBufferSize * audioBuffersToEnqueue; }
|
|
int getCurrentBitDepth() override { return (session != nullptr && session->supportsFloatingPoint() ? 32 : 16); }
|
|
BigInteger getActiveOutputChannels() const override { return activeOutputChans; }
|
|
BigInteger getActiveInputChannels() const override { return activeInputChans; }
|
|
String getLastError() override { return lastError; }
|
|
bool isPlaying() override { return callback != nullptr; }
|
|
int getXRunCount() const noexcept override { return (session != nullptr ? session->getXRunCount() : -1); }
|
|
|
|
int getDefaultBufferSize() override
|
|
{
|
|
auto defaultBufferLength = (hasLowLatencyAudioPath() ? defaultBufferSizeForLowLatencyDeviceMs
|
|
: defaultBufferSizeForStandardLatencyDeviceMs);
|
|
|
|
auto defaultBuffersToEnqueue = buffersToQueueForBufferDuration (defaultBufferLength, getCurrentSampleRate());
|
|
return defaultBuffersToEnqueue * getNativeBufferSize();
|
|
}
|
|
|
|
double getCurrentSampleRate() override
|
|
{
|
|
return (sampleRate == 0.0 ? getNativeSampleRate() : sampleRate);
|
|
}
|
|
|
|
void start (AudioIODeviceCallback* newCallback) override
|
|
{
|
|
if (session != nullptr && callback != newCallback)
|
|
{
|
|
AudioIODeviceCallback* oldCallback = callback;
|
|
|
|
if (newCallback != nullptr)
|
|
newCallback->audioDeviceAboutToStart (this);
|
|
|
|
if (oldCallback != nullptr)
|
|
{
|
|
// already running
|
|
if (newCallback == nullptr)
|
|
stop();
|
|
else
|
|
session->setCallback (newCallback);
|
|
|
|
oldCallback->audioDeviceStopped();
|
|
}
|
|
else
|
|
{
|
|
jassert (newCallback != nullptr);
|
|
|
|
// session hasn't started yet
|
|
session->setCallback (newCallback);
|
|
session->start();
|
|
}
|
|
|
|
callback = newCallback;
|
|
}
|
|
}
|
|
|
|
void stop() override
|
|
{
|
|
if (session != nullptr && callback != nullptr)
|
|
{
|
|
callback = nullptr;
|
|
session->stop();
|
|
session->setCallback (nullptr);
|
|
}
|
|
}
|
|
|
|
bool setAudioPreprocessingEnabled (bool shouldAudioProcessingBeEnabled) override
|
|
{
|
|
audioProcessingEnabled = shouldAudioProcessingBeEnabled;
|
|
|
|
if (session != nullptr)
|
|
session->setAudioPreprocessingEnabled (audioProcessingEnabled);
|
|
|
|
return true;
|
|
}
|
|
|
|
static const char* const openSLTypeName;
|
|
|
|
private:
|
|
//==============================================================================
|
|
friend class SLRealtimeThread;
|
|
|
|
//==============================================================================
|
|
DynamicLibrary slLibrary;
|
|
int actualBufferSize, sampleRate, audioBuffersToEnqueue;
|
|
int inputLatency, outputLatency;
|
|
bool deviceOpen, audioProcessingEnabled;
|
|
String lastError;
|
|
BigInteger activeOutputChans, activeInputChans;
|
|
AudioIODeviceCallback* callback;
|
|
|
|
std::unique_ptr<OpenSLSession> session;
|
|
|
|
enum
|
|
{
|
|
defaultBufferSizeForLowLatencyDeviceMs = 40,
|
|
defaultBufferSizeForStandardLatencyDeviceMs = 100
|
|
};
|
|
|
|
static int getMinimumBuffersToEnqueue (double sampleRateToCheck = getNativeSampleRate())
|
|
{
|
|
if (canUseHighPerformanceAudioPath (getNativeBufferSize(), (int) sampleRateToCheck))
|
|
{
|
|
// see https://developer.android.com/ndk/guides/audio/opensl/opensl-prog-notes.html#sandp
|
|
// "For Android 4.2 (API level 17) and earlier, a buffer count of two or more is required
|
|
// for lower latency. Beginning with Android 4.3 (API level 18), a buffer count of one
|
|
// is sufficient for lower latency."
|
|
|
|
auto sdkVersion = getEnv()->GetStaticIntField (AndroidBuildVersion, AndroidBuildVersion.SDK_INT);
|
|
return (sdkVersion >= 18 ? 1 : 2);
|
|
}
|
|
|
|
// we will not use the low-latency path so we can use the absolute minimum number of buffers
|
|
// to queue
|
|
return 1;
|
|
}
|
|
|
|
int getMaximumBuffersToEnqueue() noexcept
|
|
{
|
|
constexpr auto maxBufferSizeMs = 200;
|
|
|
|
auto availableSampleRates = getAvailableSampleRates();
|
|
auto maximumSampleRate = findMaximum(availableSampleRates.getRawDataPointer(), availableSampleRates.size());
|
|
|
|
// ensure we don't return something crazy small
|
|
return jmax (8, buffersToQueueForBufferDuration (maxBufferSizeMs, maximumSampleRate));
|
|
}
|
|
|
|
static int buffersToQueueForBufferDuration (int bufferDurationInMs, double sampleRate) noexcept
|
|
{
|
|
auto maxBufferFrames = static_cast<int> (std::ceil (bufferDurationInMs * sampleRate / 1000.0));
|
|
auto maxNumBuffers = static_cast<int> (std::ceil (static_cast<double> (maxBufferFrames)
|
|
/ static_cast<double> (getNativeBufferSize())));
|
|
|
|
return jmax (getMinimumBuffersToEnqueue (sampleRate), maxNumBuffers);
|
|
}
|
|
|
|
//==============================================================================
|
|
static String audioManagerGetProperty (const String& property)
|
|
{
|
|
const LocalRef<jstring> jProperty (javaString (property));
|
|
const LocalRef<jstring> text ((jstring) android.activity.callObjectMethod (JuceAppActivity.audioManagerGetProperty,
|
|
jProperty.get()));
|
|
if (text.get() != 0)
|
|
return juceString (text);
|
|
|
|
return {};
|
|
}
|
|
|
|
static bool androidHasSystemFeature (const String& property)
|
|
{
|
|
const LocalRef<jstring> jProperty (javaString (property));
|
|
return android.activity.callBooleanMethod (JuceAppActivity.hasSystemFeature, jProperty.get());
|
|
}
|
|
|
|
static double getNativeSampleRate()
|
|
{
|
|
return audioManagerGetProperty ("android.media.property.OUTPUT_SAMPLE_RATE").getDoubleValue();
|
|
}
|
|
|
|
static int getNativeBufferSize()
|
|
{
|
|
const int val = audioManagerGetProperty ("android.media.property.OUTPUT_FRAMES_PER_BUFFER").getIntValue();
|
|
return val > 0 ? val : 512;
|
|
}
|
|
|
|
static bool isProAudioDevice()
|
|
{
|
|
return androidHasSystemFeature ("android.hardware.audio.pro") || isSapaSupported();
|
|
}
|
|
|
|
static bool hasLowLatencyAudioPath()
|
|
{
|
|
return androidHasSystemFeature ("android.hardware.audio.low_latency");
|
|
}
|
|
|
|
static bool canUseHighPerformanceAudioPath (int requestedBufferSize, int requestedSampleRate)
|
|
{
|
|
return ((requestedBufferSize % getNativeBufferSize()) == 0)
|
|
&& (requestedSampleRate == getNativeSampleRate())
|
|
&& isProAudioDevice();
|
|
}
|
|
|
|
//==============================================================================
|
|
// Some minimum Sapa support to check if this device supports pro audio
|
|
static bool isSamsungDevice()
|
|
{
|
|
return SystemStats::getDeviceManufacturer().containsIgnoreCase ("SAMSUNG");
|
|
}
|
|
|
|
static bool isSapaSupported()
|
|
{
|
|
static bool supported = isSamsungDevice() && DynamicLibrary().open ("libapa_jni.so");
|
|
|
|
return supported;
|
|
}
|
|
|
|
JUCE_DECLARE_NON_COPYABLE_WITH_LEAK_DETECTOR (OpenSLAudioIODevice)
|
|
};
|
|
|
|
OpenSLAudioIODevice::OpenSLSession* OpenSLAudioIODevice::OpenSLSession::create (DynamicLibrary& slLibrary,
|
|
int numInputChannels, int numOutputChannels,
|
|
double samleRateToUse, int bufferSizeToUse,
|
|
int numBuffersToUse)
|
|
{
|
|
std::unique_ptr<OpenSLSession> retval;
|
|
auto sdkVersion = getEnv()->GetStaticIntField (AndroidBuildVersion, AndroidBuildVersion.SDK_INT);
|
|
|
|
// SDK versions 21 and higher should natively support floating point...
|
|
if (sdkVersion >= 21)
|
|
{
|
|
retval.reset (new OpenSLSessionT<float> (slLibrary, numInputChannels, numOutputChannels, samleRateToUse,
|
|
bufferSizeToUse, numBuffersToUse));
|
|
|
|
// ...however, some devices lie so re-try without floating point
|
|
if (retval != nullptr && (! retval->openedOK()))
|
|
retval = nullptr;
|
|
}
|
|
|
|
if (retval == nullptr)
|
|
{
|
|
retval.reset (new OpenSLSessionT<int16> (slLibrary, numInputChannels, numOutputChannels, samleRateToUse,
|
|
bufferSizeToUse, numBuffersToUse));
|
|
|
|
if (retval != nullptr && (! retval->openedOK()))
|
|
retval = nullptr;
|
|
}
|
|
|
|
return retval.release();
|
|
}
|
|
|
|
//==============================================================================
|
|
class OpenSLAudioDeviceType : public AudioIODeviceType
|
|
{
|
|
public:
|
|
OpenSLAudioDeviceType() : AudioIODeviceType (OpenSLAudioIODevice::openSLTypeName) {}
|
|
|
|
//==============================================================================
|
|
void scanForDevices() override {}
|
|
|
|
StringArray getDeviceNames (bool) const override { return StringArray (OpenSLAudioIODevice::openSLTypeName); }
|
|
int getDefaultDeviceIndex (bool) const override { return 0; }
|
|
int getIndexOfDevice (AudioIODevice* device, bool) const override { return device != nullptr ? 0 : -1; }
|
|
bool hasSeparateInputsAndOutputs() const override { return false; }
|
|
|
|
AudioIODevice* createDevice (const String& outputDeviceName,
|
|
const String& inputDeviceName) override
|
|
{
|
|
std::unique_ptr<OpenSLAudioIODevice> dev;
|
|
|
|
if (outputDeviceName.isNotEmpty() || inputDeviceName.isNotEmpty())
|
|
dev.reset (new OpenSLAudioIODevice (outputDeviceName.isNotEmpty() ? outputDeviceName
|
|
: inputDeviceName));
|
|
|
|
return dev.release();
|
|
}
|
|
|
|
static bool isOpenSLAvailable()
|
|
{
|
|
DynamicLibrary library;
|
|
return library.open ("libOpenSLES.so");
|
|
}
|
|
|
|
private:
|
|
|
|
|
|
JUCE_DECLARE_NON_COPYABLE_WITH_LEAK_DETECTOR (OpenSLAudioDeviceType)
|
|
};
|
|
|
|
const char* const OpenSLAudioIODevice::openSLTypeName = "Android OpenSL";
|
|
|
|
|
|
//==============================================================================
|
|
bool isOpenSLAvailable() { return OpenSLAudioDeviceType::isOpenSLAvailable(); }
|
|
|
|
AudioIODeviceType* AudioIODeviceType::createAudioIODeviceType_OpenSLES()
|
|
{
|
|
return isOpenSLAvailable() ? new OpenSLAudioDeviceType() : nullptr;
|
|
}
|
|
|
|
//==============================================================================
|
|
class SLRealtimeThread
|
|
{
|
|
public:
|
|
static constexpr int numBuffers = 4;
|
|
|
|
SLRealtimeThread()
|
|
{
|
|
if (auto createEngine = (OpenSLAudioIODevice::OpenSLSession::CreateEngineFunc) slLibrary.getFunction ("slCreateEngine"))
|
|
{
|
|
SLObjectItf obj = nullptr;
|
|
auto err = createEngine (&obj, 0, nullptr, 0, nullptr, nullptr);
|
|
|
|
if (err != SL_RESULT_SUCCESS || obj == nullptr)
|
|
return;
|
|
|
|
if ((*obj)->Realize (obj, 0) != SL_RESULT_SUCCESS)
|
|
{
|
|
(*obj)->Destroy (obj);
|
|
return;
|
|
}
|
|
|
|
engine = SlRef<SLEngineItf_>::cast (SlObjectRef (obj));
|
|
|
|
if (engine == nullptr)
|
|
{
|
|
(*obj)->Destroy (obj);
|
|
return;
|
|
}
|
|
|
|
obj = nullptr;
|
|
err = (*engine)->CreateOutputMix (engine, &obj, 0, nullptr, nullptr);
|
|
|
|
if (err != SL_RESULT_SUCCESS || obj == nullptr || (*obj)->Realize (obj, 0) != SL_RESULT_SUCCESS)
|
|
{
|
|
(*obj)->Destroy (obj);
|
|
return;
|
|
}
|
|
|
|
outputMix = SlRef<SLOutputMixItf_>::cast (SlObjectRef (obj));
|
|
|
|
if (outputMix == nullptr)
|
|
{
|
|
(*obj)->Destroy (obj);
|
|
return;
|
|
}
|
|
|
|
SLDataLocator_AndroidSimpleBufferQueue queueLocator = {SL_DATALOCATOR_ANDROIDSIMPLEBUFFERQUEUE, static_cast<SLuint32> (numBuffers)};
|
|
SLDataLocator_OutputMix outputMixLocator = {SL_DATALOCATOR_OUTPUTMIX, outputMix};
|
|
|
|
PCMDataFormatEx dataFormat;
|
|
BufferHelpers<int16>::initPCMDataFormat (dataFormat, 1, OpenSLAudioIODevice::getNativeSampleRate());
|
|
|
|
SLDataSource source = { &queueLocator, &dataFormat };
|
|
SLDataSink sink = { &outputMixLocator, nullptr };
|
|
|
|
SLInterfaceID queueInterfaces[] = { &IntfIID<SLAndroidSimpleBufferQueueItf_>::iid };
|
|
SLboolean trueFlag = SL_BOOLEAN_TRUE;
|
|
|
|
obj = nullptr;
|
|
err = (*engine)->CreateAudioPlayer (engine, &obj, &source, &sink, 1, queueInterfaces, &trueFlag);
|
|
|
|
if (err != SL_RESULT_SUCCESS || obj == nullptr)
|
|
return;
|
|
|
|
if ((*obj)->Realize (obj, 0) != SL_RESULT_SUCCESS)
|
|
{
|
|
(*obj)->Destroy (obj);
|
|
return;
|
|
}
|
|
|
|
player = SlRef<SLPlayItf_>::cast (SlObjectRef (obj));
|
|
|
|
if (player == nullptr)
|
|
{
|
|
(*obj)->Destroy (obj);
|
|
return;
|
|
}
|
|
|
|
queue = SlRef<SLAndroidSimpleBufferQueueItf_>::cast (player);
|
|
if (queue == nullptr)
|
|
return;
|
|
|
|
if ((*queue)->RegisterCallback (queue, staticFinished, this) != SL_RESULT_SUCCESS)
|
|
{
|
|
queue = nullptr;
|
|
return;
|
|
}
|
|
|
|
pthread_cond_init (&threadReady, nullptr);
|
|
pthread_mutex_init (&threadReadyMutex, nullptr);
|
|
}
|
|
}
|
|
|
|
bool isOK() const { return queue != nullptr; }
|
|
|
|
pthread_t startThread (void* (*entry) (void*), void* userPtr)
|
|
{
|
|
memset (buffer.get(), 0, static_cast<size_t> (sizeof (int16) * static_cast<size_t> (bufferSize * numBuffers)));
|
|
|
|
for (int i = 0; i < numBuffers; ++i)
|
|
{
|
|
int16* dst = buffer.get() + (bufferSize * i);
|
|
(*queue)->Enqueue (queue, dst, static_cast<SLuint32> (static_cast<size_t> (bufferSize) * sizeof (int16)));
|
|
}
|
|
|
|
pthread_mutex_lock (&threadReadyMutex);
|
|
|
|
threadEntryProc = entry;
|
|
threadUserPtr = userPtr;
|
|
|
|
(*player)->SetPlayState (player, SL_PLAYSTATE_PLAYING);
|
|
|
|
pthread_cond_wait (&threadReady, &threadReadyMutex);
|
|
pthread_mutex_unlock (&threadReadyMutex);
|
|
|
|
return threadID;
|
|
}
|
|
|
|
void finished()
|
|
{
|
|
if (threadEntryProc != nullptr)
|
|
{
|
|
pthread_mutex_lock (&threadReadyMutex);
|
|
|
|
threadID = pthread_self();
|
|
|
|
pthread_cond_signal (&threadReady);
|
|
pthread_mutex_unlock (&threadReadyMutex);
|
|
|
|
threadEntryProc (threadUserPtr);
|
|
threadEntryProc = nullptr;
|
|
|
|
(*player)->SetPlayState (player, SL_PLAYSTATE_STOPPED);
|
|
MessageManager::callAsync ([this] () { delete this; });
|
|
}
|
|
}
|
|
|
|
private:
|
|
//=============================================================================
|
|
static void staticFinished (SLAndroidSimpleBufferQueueItf, void* context)
|
|
{
|
|
static_cast<SLRealtimeThread*> (context)->finished();
|
|
}
|
|
|
|
//=============================================================================
|
|
DynamicLibrary slLibrary { "libOpenSLES.so" };
|
|
|
|
SlRef<SLEngineItf_> engine;
|
|
SlRef<SLOutputMixItf_> outputMix;
|
|
SlRef<SLPlayItf_> player;
|
|
SlRef<SLAndroidSimpleBufferQueueItf_> queue;
|
|
|
|
int bufferSize = OpenSLAudioIODevice::getNativeBufferSize();
|
|
HeapBlock<int16> buffer { HeapBlock<int16> (static_cast<size_t> (1 * bufferSize * numBuffers)) };
|
|
|
|
void* (*threadEntryProc) (void*) = nullptr;
|
|
void* threadUserPtr = nullptr;
|
|
|
|
pthread_cond_t threadReady;
|
|
pthread_mutex_t threadReadyMutex;
|
|
pthread_t threadID;
|
|
};
|
|
|
|
pthread_t juce_createRealtimeAudioThread (void* (*entry) (void*), void* userPtr)
|
|
{
|
|
std::unique_ptr<SLRealtimeThread> thread (new SLRealtimeThread);
|
|
|
|
if (! thread->isOK())
|
|
return 0;
|
|
|
|
pthread_t threadID = thread->startThread (entry, userPtr);
|
|
|
|
// the thread will de-allocate itself
|
|
thread.release();
|
|
|
|
return threadID;
|
|
}
|
|
|
|
} // namespace juce
|