Add Chromium-only Blender WebEngine parity work

blender-5.2.0/extern/audaspace/include/respec/ChannelMapper.h

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+/*******************************************************************************
+ * Copyright 2009-2016 Jörg Müller
+ *
+ * Licensed under the Apache License, Version 2.0 (the "License");
+ * you may not use this file except in compliance with the License.
+ * You may obtain a copy of the License at
+ *
+ *   http://www.apache.org/licenses/LICENSE-2.0
+ *
+ * Unless required by applicable law or agreed to in writing, software
+ * distributed under the License is distributed on an "AS IS" BASIS,
+ * WITHOUT WARRANTIES OR CONDITIONS OF ANY KIND, either express or implied.
+ * See the License for the specific language governing permissions and
+ * limitations under the License.
+ ******************************************************************************/
+
+#pragma once
+
+/**
+ * @file ChannelMapper.h
+ * @ingroup respec
+ * The ChannelMapper class.
+ */
+
+#include "respec/SpecsChanger.h"
+
+AUD_NAMESPACE_BEGIN
+
+/**
+ * This sound creates a reader that maps a sound source's channels to a
+ * specific output channel count.
+ */
+class AUD_API ChannelMapper : public SpecsChanger
+{
+private:
+	// delete copy constructor and operator=
+	ChannelMapper(const ChannelMapper&) = delete;
+	ChannelMapper& operator=(const ChannelMapper&) = delete;
+
+public:
+	/**
+	 * Creates a new sound.
+	 * \param sound The input sound.
+	 * \param specs The target specifications.
+	 */
+	ChannelMapper(std::shared_ptr<ISound> sound, DeviceSpecs specs);
+
+	virtual std::shared_ptr<IReader> createReader();
+};
+
+AUD_NAMESPACE_END

blender-5.2.0/extern/audaspace/include/respec/ChannelMapperReader.h

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+/*******************************************************************************
+ * Copyright 2009-2016 Jörg Müller
+ *
+ * Licensed under the Apache License, Version 2.0 (the "License");
+ * you may not use this file except in compliance with the License.
+ * You may obtain a copy of the License at
+ *
+ *   http://www.apache.org/licenses/LICENSE-2.0
+ *
+ * Unless required by applicable law or agreed to in writing, software
+ * distributed under the License is distributed on an "AS IS" BASIS,
+ * WITHOUT WARRANTIES OR CONDITIONS OF ANY KIND, either express or implied.
+ * See the License for the specific language governing permissions and
+ * limitations under the License.
+ ******************************************************************************/
+
+#pragma once
+
+/**
+ * @file ChannelMapperReader.h
+ * @ingroup respec
+ * The ChannelMapperReader class.
+ */
+
+#include "fx/EffectReader.h"
+#include "util/Buffer.h"
+
+AUD_NAMESPACE_BEGIN
+
+/**
+ * This class maps a sound source's channels to a specific output channel count.
+ * \note The input sample format must be float.
+ */
+class AUD_API ChannelMapperReader : public EffectReader
+{
+private:
+	/**
+	 * The sound reading buffer.
+	 */
+	Buffer m_buffer;
+
+	/**
+	 * The output specification.
+	 */
+	Channels m_target_channels;
+
+	/**
+	 * The channel count of the reader.
+	 */
+	Channels m_source_channels;
+
+	/**
+	 * The mapping specification.
+	 */
+	float* m_mapping;
+
+	/**
+	 * The size of the mapping.
+	 */
+	int m_map_size;
+
+	/**
+	 * The mono source angle.
+	 */
+	float m_mono_angle;
+
+	static const Channel MONO_MAP[];
+	static const Channel STEREO_MAP[];
+	static const Channel STEREO_LFE_MAP[];
+	static const Channel SURROUND4_MAP[];
+	static const Channel SURROUND5_MAP[];
+	static const Channel SURROUND51_MAP[];
+	static const Channel SURROUND61_MAP[];
+	static const Channel SURROUND71_MAP[];
+	static const Channel* CHANNEL_MAPS[];
+
+	static const float MONO_ANGLES[];
+	static const float STEREO_ANGLES[];
+	static const float STEREO_LFE_ANGLES[];
+	static const float SURROUND4_ANGLES[];
+	static const float SURROUND5_ANGLES[];
+	static const float SURROUND51_ANGLES[];
+	static const float SURROUND61_ANGLES[];
+	static const float SURROUND71_ANGLES[];
+	static const float* CHANNEL_ANGLES[];
+
+	// delete copy constructor and operator=
+	ChannelMapperReader(const ChannelMapperReader&) = delete;
+	ChannelMapperReader& operator=(const ChannelMapperReader&) = delete;
+
+	/**
+	 * Calculates the mapping matrix.
+	 */
+	void AUD_LOCAL calculateMapping();
+
+	/**
+	 * Calculates the distance between two angles.
+	 */
+	float AUD_LOCAL angleDistance(float alpha, float beta);
+
+public:
+	/**
+	 * Creates a channel mapper reader.
+	 * \param reader The reader to map.
+	 * \param channels The target channel count this reader should map to.
+	 */
+	ChannelMapperReader(std::shared_ptr<IReader> reader, Channels channels);
+
+	/**
+	 * Destroys the reader.
+	 */
+	~ChannelMapperReader();
+
+	/**
+	 * Returns the channel configuration of the source reader.
+	 * @return The channel configuration of the reader.
+	 */
+	Channels getSourceChannels() const;
+
+	/**
+	 * Returns the target channel configuration.
+	 * Equals getSpecs().channels.
+	 * @return The target channel configuration.
+	 */
+	Channels getChannels() const;
+
+	/**
+	 * Sets the requested channel output count.
+	 * \param channels The channel output count.
+	 */
+	void setChannels(Channels channels);
+
+	/**
+	 * Returns the mapping of the source channel to the target channel.
+	 * @param source The number of the source channel. Should be in the range [0, source channels).
+	 * @param target The number of the target channel. Should be in the range [0, target channels).
+	 * @return The mapping value which should be between 0.0 and 1.0. If source or target are out of range, NaN is returned.
+	 */
+	float getMapping(int source, int target);
+
+	/**
+	 * Sets the angle for mono sources.
+	 * \param angle The angle for mono sources.
+	 */
+	void setMonoAngle(float angle);
+
+	virtual Specs getSpecs() const;
+	virtual void read(int& length, bool& eos, sample_t* buffer);
+};
+
+AUD_NAMESPACE_END

blender-5.2.0/extern/audaspace/include/respec/Converter.h

@@ -0,0 +1,51 @@
+/*******************************************************************************
+ * Copyright 2009-2016 Jörg Müller
+ *
+ * Licensed under the Apache License, Version 2.0 (the "License");
+ * you may not use this file except in compliance with the License.
+ * You may obtain a copy of the License at
+ *
+ *   http://www.apache.org/licenses/LICENSE-2.0
+ *
+ * Unless required by applicable law or agreed to in writing, software
+ * distributed under the License is distributed on an "AS IS" BASIS,
+ * WITHOUT WARRANTIES OR CONDITIONS OF ANY KIND, either express or implied.
+ * See the License for the specific language governing permissions and
+ * limitations under the License.
+ ******************************************************************************/
+
+#pragma once
+
+/**
+ * @file Converter.h
+ * @ingroup respec
+ * The Converter class.
+ */
+
+#include "respec/SpecsChanger.h"
+
+AUD_NAMESPACE_BEGIN
+
+/**
+ * This sound creates a converter reader that is able to convert from one
+ * audio format to another.
+ */
+class AUD_API Converter : public SpecsChanger
+{
+private:
+	// delete copy constructor and operator=
+	Converter(const Converter&) = delete;
+	Converter& operator=(const Converter&) = delete;
+
+public:
+	/**
+	 * Creates a new sound.
+	 * \param sound The input sound.
+	 * \param specs The target specifications.
+	 */
+	Converter(std::shared_ptr<ISound> sound, DeviceSpecs specs);
+
+	virtual std::shared_ptr<IReader> createReader();
+};
+
+AUD_NAMESPACE_END

blender-5.2.0/extern/audaspace/include/respec/ConverterFunctions.h

@@ -0,0 +1,377 @@
+/*******************************************************************************
+ * Copyright 2009-2016 Jörg Müller
+ *
+ * Licensed under the Apache License, Version 2.0 (the "License");
+ * you may not use this file except in compliance with the License.
+ * You may obtain a copy of the License at
+ *
+ *   http://www.apache.org/licenses/LICENSE-2.0
+ *
+ * Unless required by applicable law or agreed to in writing, software
+ * distributed under the License is distributed on an "AS IS" BASIS,
+ * WITHOUT WARRANTIES OR CONDITIONS OF ANY KIND, either express or implied.
+ * See the License for the specific language governing permissions and
+ * limitations under the License.
+ ******************************************************************************/
+
+#pragma once
+
+/**
+ * @file ConverterFunctions.h
+ * @ingroup respec
+ * Defines several conversion functions between different sample formats.
+ */
+
+#include "Audaspace.h"
+
+#include <cstring>
+
+AUD_NAMESPACE_BEGIN
+
+/**
+ * The function template for functions converting from one sample format
+ * to another, having the same parameter order as std::memcpy.
+ */
+typedef void (*convert_f)(data_t* target, data_t* source, int length);
+
+/**
+ * The copy conversion function simply calls std::memcpy.
+ * @param target The target buffer.
+ * @param source The source buffer.
+ * @param length The amount of samples to be converted.
+ */
+template <class T>
+void convert_copy(data_t* target, data_t* source, int length)
+{
+	std::memcpy(target, source, length*sizeof(T));
+}
+
+/**
+ * @brief Converts from FORMAT_U8 to FORMAT_S16.
+ * @param target The target buffer.
+ * @param source The source buffer.
+ * @param length The amount of samples to be converted.
+ */
+void AUD_API convert_u8_s16(data_t* target, data_t* source, int length);
+
+/**
+ * @brief Converts from FORMAT_U8 to FORMAT_S24 big endian.
+ * @param target The target buffer.
+ * @param source The source buffer.
+ * @param length The amount of samples to be converted.
+ */
+void AUD_API convert_u8_s24_be(data_t* target, data_t* source, int length);
+
+/**
+ * @brief Converts from FORMAT_U8 to FORMAT_S24 little endian.
+ * @param target The target buffer.
+ * @param source The source buffer.
+ * @param length The amount of samples to be converted.
+ */
+void AUD_API convert_u8_s24_le(data_t* target, data_t* source, int length);
+
+/**
+ * @brief Converts from FORMAT_U8 to FORMAT_S32.
+ * @param target The target buffer.
+ * @param source The source buffer.
+ * @param length The amount of samples to be converted.
+ */
+void AUD_API convert_u8_s32(data_t* target, data_t* source, int length);
+
+/**
+ * @brief Converts from FORMAT_U8 to FORMAT_FLOAT32.
+ * @param target The target buffer.
+ * @param source The source buffer.
+ * @param length The amount of samples to be converted.
+ */
+void AUD_API convert_u8_float(data_t* target, data_t* source, int length);
+
+/**
+ * @brief Converts from FORMAT_U8 to FORMAT_FLOAT64.
+ * @param target The target buffer.
+ * @param source The source buffer.
+ * @param length The amount of samples to be converted.
+ */
+void AUD_API convert_u8_double(data_t* target, data_t* source, int length);
+
+/**
+ * @brief Converts from FORMAT_S16 to FORMAT_U8.
+ * @param target The target buffer.
+ * @param source The source buffer.
+ * @param length The amount of samples to be converted.
+ */
+void AUD_API convert_s16_u8(data_t* target, data_t* source, int length);
+
+/**
+ * @brief Converts from FORMAT_S16 to FORMAT_S24 big endian.
+ * @param target The target buffer.
+ * @param source The source buffer.
+ * @param length The amount of samples to be converted.
+ */
+void AUD_API convert_s16_s24_be(data_t* target, data_t* source, int length);
+
+/**
+ * @brief Converts from FORMAT_S16 to FORMAT_S24 little endian.
+ * @param target The target buffer.
+ * @param source The source buffer.
+ * @param length The amount of samples to be converted.
+ */
+void AUD_API convert_s16_s24_le(data_t* target, data_t* source, int length);
+
+/**
+ * @brief Converts from FORMAT_S16 to FORMAT_S32.
+ * @param target The target buffer.
+ * @param source The source buffer.
+ * @param length The amount of samples to be converted.
+ */
+void AUD_API convert_s16_s32(data_t* target, data_t* source, int length);
+
+/**
+ * @brief Converts from FORMAT_S16 to FORMAT_FLOAT32.
+ * @param target The target buffer.
+ * @param source The source buffer.
+ * @param length The amount of samples to be converted.
+ */
+void AUD_API convert_s16_float(data_t* target, data_t* source, int length);
+
+/**
+ * @brief Converts from FORMAT_S16 to FORMAT_FLOAT64.
+ * @param target The target buffer.
+ * @param source The source buffer.
+ * @param length The amount of samples to be converted.
+ */
+void AUD_API convert_s16_double(data_t* target, data_t* source, int length);
+
+/**
+ * @brief Converts from FORMAT_S24 big endian to FORMAT_U8.
+ * @param target The target buffer.
+ * @param source The source buffer.
+ * @param length The amount of samples to be converted.
+ */
+void AUD_API convert_s24_u8_be(data_t* target, data_t* source, int length);
+
+/**
+ * @brief Converts from FORMAT_S24 little endian to FORMAT_U8.
+ * @param target The target buffer.
+ * @param source The source buffer.
+ * @param length The amount of samples to be converted.
+ */
+void AUD_API convert_s24_u8_le(data_t* target, data_t* source, int length);
+
+/**
+ * @brief Converts from FORMAT_S24 big endian to FORMAT_S16.
+ * @param target The target buffer.
+ * @param source The source buffer.
+ * @param length The amount of samples to be converted.
+ */
+void AUD_API convert_s24_s16_be(data_t* target, data_t* source, int length);
+
+/**
+ * @brief Converts from FORMAT_S24 little endian to FORMAT_S16.
+ * @param target The target buffer.
+ * @param source The source buffer.
+ * @param length The amount of samples to be converted.
+ */
+void AUD_API convert_s24_s16_le(data_t* target, data_t* source, int length);
+
+/**
+ * @brief Converts from FORMAT_S24 to FORMAT_S24 simply using std::memcpy.
+ * @param target The target buffer.
+ * @param source The source buffer.
+ * @param length The amount of samples to be converted.
+ */
+void AUD_API convert_s24_s24(data_t* target, data_t* source, int length);
+
+/**
+ * @brief Converts from FORMAT_S24 big endian to FORMAT_S32.
+ * @param target The target buffer.
+ * @param source The source buffer.
+ * @param length The amount of samples to be converted.
+ */
+void AUD_API convert_s24_s32_be(data_t* target, data_t* source, int length);
+
+/**
+ * @brief Converts from FORMAT_S24 little endian to FORMAT_S32.
+ * @param target The target buffer.
+ * @param source The source buffer.
+ * @param length The amount of samples to be converted.
+ */
+void AUD_API convert_s24_s32_le(data_t* target, data_t* source, int length);
+
+/**
+ * @brief Converts from FORMAT_S24 big endian to FORMAT_FLOAT32.
+ * @param target The target buffer.
+ * @param source The source buffer.
+ * @param length The amount of samples to be converted.
+ */
+void AUD_API convert_s24_float_be(data_t* target, data_t* source, int length);
+
+/**
+ * @brief Converts from FORMAT_S24 little endian to FORMAT_FLOAT32.
+ * @param target The target buffer.
+ * @param source The source buffer.
+ * @param length The amount of samples to be converted.
+ */
+void AUD_API convert_s24_float_le(data_t* target, data_t* source, int length);
+
+/**
+ * @brief Converts from FORMAT_S24 big endian to FORMAT_FLOAT64.
+ * @param target The target buffer.
+ * @param source The source buffer.
+ * @param length The amount of samples to be converted.
+ */
+void AUD_API convert_s24_double_be(data_t* target, data_t* source, int length);
+
+/**
+ * @brief Converts from FORMAT_S24 little endian to FORMAT_FLOAT64.
+ * @param target The target buffer.
+ * @param source The source buffer.
+ * @param length The amount of samples to be converted.
+ */
+void AUD_API convert_s24_double_le(data_t* target, data_t* source, int length);
+
+/**
+ * @brief Converts from FORMAT_S32 to FORMAT_U8.
+ * @param target The target buffer.
+ * @param source The source buffer.
+ * @param length The amount of samples to be converted.
+ */
+void AUD_API convert_s32_u8(data_t* target, data_t* source, int length);
+
+/**
+ * @brief Converts from FORMAT_S32 to FORMAT_S16.
+ * @param target The target buffer.
+ * @param source The source buffer.
+ * @param length The amount of samples to be converted.
+ */
+void AUD_API convert_s32_s16(data_t* target, data_t* source, int length);
+
+/**
+ * @brief Converts from FORMAT_S32 to FORMAT_S24 big endian.
+ * @param target The target buffer.
+ * @param source The source buffer.
+ * @param length The amount of samples to be converted.
+ */
+void AUD_API convert_s32_s24_be(data_t* target, data_t* source, int length);
+
+/**
+ * @brief Converts from FORMAT_S32 to FORMAT_S24 little endian.
+ * @param target The target buffer.
+ * @param source The source buffer.
+ * @param length The amount of samples to be converted.
+ */
+void AUD_API convert_s32_s24_le(data_t* target, data_t* source, int length);
+
+/**
+ * @brief Converts from FORMAT_S32 to FORMAT_FLOAT32.
+ * @param target The target buffer.
+ * @param source The source buffer.
+ * @param length The amount of samples to be converted.
+ */
+void AUD_API convert_s32_float(data_t* target, data_t* source, int length);
+
+/**
+ * @brief Converts from FORMAT_S32 to FORMAT_FLOAT64.
+ * @param target The target buffer.
+ * @param source The source buffer.
+ * @param length The amount of samples to be converted.
+ */
+void AUD_API convert_s32_double(data_t* target, data_t* source, int length);
+
+/**
+ * @brief Converts from FORMAT_FLOAT32 to FORMAT_U8.
+ * @param target The target buffer.
+ * @param source The source buffer.
+ * @param length The amount of samples to be converted.
+ */
+void AUD_API convert_float_u8(data_t* target, data_t* source, int length);
+
+/**
+ * @brief Converts from FORMAT_FLOAT32 to FORMAT_S16.
+ * @param target The target buffer.
+ * @param source The source buffer.
+ * @param length The amount of samples to be converted.
+ */
+void AUD_API convert_float_s16(data_t* target, data_t* source, int length);
+
+/**
+ * @brief Converts from FORMAT_FLOAT32 to FORMAT_S24 big endian.
+ * @param target The target buffer.
+ * @param source The source buffer.
+ * @param length The amount of samples to be converted.
+ */
+void AUD_API convert_float_s24_be(data_t* target, data_t* source, int length);
+
+/**
+ * @brief Converts from FORMAT_FLOAT32 to FORMAT_S24 little endian.
+ * @param target The target buffer.
+ * @param source The source buffer.
+ * @param length The amount of samples to be converted.
+ */
+void AUD_API convert_float_s24_le(data_t* target, data_t* source, int length);
+
+/**
+ * @brief Converts from FORMAT_FLOAT32 to FORMAT_S32.
+ * @param target The target buffer.
+ * @param source The source buffer.
+ * @param length The amount of samples to be converted.
+ */
+void AUD_API convert_float_s32(data_t* target, data_t* source, int length);
+
+/**
+ * @brief Converts from FORMAT_FLOAT32 to FORMAT_FLOAT64.
+ * @param target The target buffer.
+ * @param source The source buffer.
+ * @param length The amount of samples to be converted.
+ */
+void AUD_API convert_float_double(data_t* target, data_t* source, int length);
+
+/**
+ * @brief Converts from FORMAT_FLOAT64 to FORMAT_U8.
+ * @param target The target buffer.
+ * @param source The source buffer.
+ * @param length The amount of samples to be converted.
+ */
+void AUD_API convert_double_u8(data_t* target, data_t* source, int length);
+
+/**
+ * @brief Converts from FORMAT_FLOAT64 to FORMAT_S16.
+ * @param target The target buffer.
+ * @param source The source buffer.
+ * @param length The amount of samples to be converted.
+ */
+void AUD_API convert_double_s16(data_t* target, data_t* source, int length);
+
+/**
+ * @brief Converts from FORMAT_FLOAT64 to FORMAT_S24 big endian.
+ * @param target The target buffer.
+ * @param source The source buffer.
+ * @param length The amount of samples to be converted.
+ */
+void AUD_API convert_double_s24_be(data_t* target, data_t* source, int length);
+
+/**
+ * @brief Converts from FORMAT_FLOAT64 to FORMAT_S24 little endian.
+ * @param target The target buffer.
+ * @param source The source buffer.
+ * @param length The amount of samples to be converted.
+ */
+void AUD_API convert_double_s24_le(data_t* target, data_t* source, int length);
+
+/**
+ * @brief Converts from FORMAT_FLOAT64 to FORMAT_S32.
+ * @param target The target buffer.
+ * @param source The source buffer.
+ * @param length The amount of samples to be converted.
+ */
+void AUD_API convert_double_s32(data_t* target, data_t* source, int length);
+
+/**
+ * @brief Converts from FORMAT_FLOAT64 to FORMAT_FLOAT32.
+ * @param target The target buffer.
+ * @param source The source buffer.
+ * @param length The amount of samples to be converted.
+ */
+void AUD_API convert_double_float(data_t* target, data_t* source, int length);
+
+AUD_NAMESPACE_END

blender-5.2.0/extern/audaspace/include/respec/ConverterReader.h

@@ -0,0 +1,67 @@
+/*******************************************************************************
+ * Copyright 2009-2016 Jörg Müller
+ *
+ * Licensed under the Apache License, Version 2.0 (the "License");
+ * you may not use this file except in compliance with the License.
+ * You may obtain a copy of the License at
+ *
+ *   http://www.apache.org/licenses/LICENSE-2.0
+ *
+ * Unless required by applicable law or agreed to in writing, software
+ * distributed under the License is distributed on an "AS IS" BASIS,
+ * WITHOUT WARRANTIES OR CONDITIONS OF ANY KIND, either express or implied.
+ * See the License for the specific language governing permissions and
+ * limitations under the License.
+ ******************************************************************************/
+
+#pragma once
+
+/**
+ * @file ConverterReader.h
+ * @ingroup respec
+ * The ConverterReader class.
+ */
+
+#include "fx/EffectReader.h"
+#include "respec/ConverterFunctions.h"
+#include "util/Buffer.h"
+
+AUD_NAMESPACE_BEGIN
+
+/**
+ * This class converts a sound source from one to another format.
+ */
+class AUD_API ConverterReader : public EffectReader
+{
+private:
+	/**
+	 * The sound output buffer.
+	 */
+	Buffer m_buffer;
+
+	/**
+	 * The target specification.
+	 */
+	SampleFormat m_format;
+
+	/**
+	 * Converter function.
+	 */
+	convert_f m_convert;
+
+	// delete copy constructor and operator=
+	ConverterReader(const ConverterReader&) = delete;
+	ConverterReader& operator=(const ConverterReader&) = delete;
+
+public:
+	/**
+	 * Creates a converter reader.
+	 * \param reader The reader to convert.
+	 * \param specs The target specification.
+	 */
+	ConverterReader(std::shared_ptr<IReader> reader, DeviceSpecs specs);
+
+	virtual void read(int& length, bool& eos, sample_t* buffer);
+};
+
+AUD_NAMESPACE_END

blender-5.2.0/extern/audaspace/include/respec/JOSResample.h

@@ -0,0 +1,52 @@
+/*******************************************************************************
+ * Copyright 2009-2016 Jörg Müller
+ *
+ * Licensed under the Apache License, Version 2.0 (the "License");
+ * you may not use this file except in compliance with the License.
+ * You may obtain a copy of the License at
+ *
+ *   http://www.apache.org/licenses/LICENSE-2.0
+ *
+ * Unless required by applicable law or agreed to in writing, software
+ * distributed under the License is distributed on an "AS IS" BASIS,
+ * WITHOUT WARRANTIES OR CONDITIONS OF ANY KIND, either express or implied.
+ * See the License for the specific language governing permissions and
+ * limitations under the License.
+ ******************************************************************************/
+
+#pragma once
+
+/**
+ * @file JOSResample.h
+ * @ingroup respec
+ * The JOSResample class.
+ */
+
+#include "respec/SpecsChanger.h"
+
+AUD_NAMESPACE_BEGIN
+
+/**
+ * This sound creates a resampling reader that does Julius O. Smith's resampling algorithm.
+ */
+class AUD_API JOSResample : public SpecsChanger
+{
+private:
+	// delete copy constructor and operator=
+	JOSResample(const JOSResample&) = delete;
+	JOSResample& operator=(const JOSResample&) = delete;
+
+	ResampleQuality m_quality;
+
+public:
+	/**
+	 * Creates a new sound.
+	 * \param sound The input sound.
+	 * \param specs The target specifications.
+	 */
+	JOSResample(std::shared_ptr<ISound> sound, DeviceSpecs specs, ResampleQuality quality = ResampleQuality::HIGH);
+
+	virtual std::shared_ptr<IReader> createReader();
+};
+
+AUD_NAMESPACE_END

blender-5.2.0/extern/audaspace/include/respec/JOSResampleReader.h

@@ -0,0 +1,170 @@
+/*******************************************************************************
+ * Copyright 2009-2016 Jörg Müller
+ *
+ * Licensed under the Apache License, Version 2.0 (the "License");
+ * you may not use this file except in compliance with the License.
+ * You may obtain a copy of the License at
+ *
+ *   http://www.apache.org/licenses/LICENSE-2.0
+ *
+ * Unless required by applicable law or agreed to in writing, software
+ * distributed under the License is distributed on an "AS IS" BASIS,
+ * WITHOUT WARRANTIES OR CONDITIONS OF ANY KIND, either express or implied.
+ * See the License for the specific language governing permissions and
+ * limitations under the License.
+ ******************************************************************************/
+
+#pragma once
+
+/**
+ * @file JOSResampleReader.h
+ * @ingroup respec
+ * The JOSResampleReader class.
+ */
+
+#include "respec/ResampleReader.h"
+#include "util/Buffer.h"
+
+AUD_NAMESPACE_BEGIN
+
+/**
+ * This resampling reader uses Julius O. Smith's resampling algorithm.
+ */
+class AUD_API JOSResampleReader : public ResampleReader
+{
+private:
+	typedef void (JOSResampleReader::*resample_f)(double target_factor, int length, sample_t* buffer);
+
+	/**
+	 * The half filter length for HIGH quality setting. 
+	 */
+	static const int m_len_high;
+	/**
+	 * The half filter length for MEDIUM quality setting.
+	 */
+	static const int m_len_medium;
+	/**
+	 * The half filter length for LOW quality setting.
+	 */
+	static const int m_len_low;
+	/**
+	 * The filter sample step size for HIGH quality setting.
+	 */
+	static const int m_L_high;
+	/**
+	 * The filter sample step size for MEDIUM quality setting.
+	 */
+	static const int m_L_medium;
+	/**
+	 * The filter sample step size for LOW quality setting.
+	 */
+	static const int m_L_low;
+	/**
+	 * The filter coefficients for HIGH quality setting.
+	 */
+	static const float m_coeff_high[];
+	/**
+	 * The filter coefficients for MEDIUM quality setting.
+	 */
+	static const float m_coeff_medium[];
+	/**
+	 * The filter coefficients for LOW quality setting.
+	 */
+	static const float m_coeff_low[];
+
+	/**
+	 * The half filter length.
+	 */
+	int m_len;
+
+	/**
+	 * The sample step size for the filter.
+	 */
+	int m_L;
+
+	/**
+	 * The filter coefficients.
+	 */
+	const float* m_coeff;
+
+	/**
+	 * The reader channels.
+	 */
+	Channels m_channels;
+
+	/**
+	 * The sample position in the cache.
+	 */
+	unsigned int m_n;
+
+	/**
+	 * The subsample position in the cache.
+	 */
+	double m_P;
+
+	/**
+	 * The input data buffer.
+	 */
+	Buffer m_buffer;
+
+	/**
+	 * Double buffer for the sums.
+	 */
+	Buffer m_sums;
+
+	/**
+	 * How many samples in the cache are valid.
+	 */
+	int m_cache_valid;
+
+	/**
+	 * Resample function.
+	 */
+	resample_f m_resample;
+
+	/**
+	 * Last resampling factor.
+	 */
+	double m_last_factor;
+
+	// delete copy constructor and operator=
+	JOSResampleReader(const JOSResampleReader&) = delete;
+	JOSResampleReader& operator=(const JOSResampleReader&) = delete;
+
+	/**
+	 * Resets the resampler to its initial state.
+	 */
+	void AUD_LOCAL reset();
+
+	/**
+	 * Updates the buffer to be as small as possible for the coming reading.
+	 * \param size The size of samples to be read.
+	 * \param factor The next resampling factor.
+	 * \param samplesize The size of a sample.
+	 */
+	void AUD_LOCAL updateBuffer(int size, double factor, int samplesize);
+
+	void AUD_LOCAL resample_generic(double target_factor, int length, sample_t* buffer);
+	void AUD_LOCAL resample_mono(double target_factor, int length, sample_t* buffer);
+	void AUD_LOCAL resample_stereo(double target_factor, int length, sample_t* buffer);
+
+	template <typename T>
+	void AUD_LOCAL resample(double target_factor, int length, sample_t* buffer);
+
+public:
+
+	/**
+	 * Creates a resampling reader.
+	 * \param reader The reader to mix.
+	 * \param rate The target sampling rate.
+	 */
+	JOSResampleReader(std::shared_ptr<IReader> reader, SampleRate rate, ResampleQuality quality = ResampleQuality::HIGH);
+
+	virtual void seek(int position);
+	virtual int getLength() const;
+	virtual int getPosition() const;
+	virtual Specs getSpecs() const;
+	virtual void read(int& length, bool& eos, sample_t* buffer);
+};
+
+AUD_NAMESPACE_END

blender-5.2.0/extern/audaspace/include/respec/LinearResample.h

@@ -0,0 +1,50 @@
+/*******************************************************************************
+ * Copyright 2009-2016 Jörg Müller
+ *
+ * Licensed under the Apache License, Version 2.0 (the "License");
+ * you may not use this file except in compliance with the License.
+ * You may obtain a copy of the License at
+ *
+ *   http://www.apache.org/licenses/LICENSE-2.0
+ *
+ * Unless required by applicable law or agreed to in writing, software
+ * distributed under the License is distributed on an "AS IS" BASIS,
+ * WITHOUT WARRANTIES OR CONDITIONS OF ANY KIND, either express or implied.
+ * See the License for the specific language governing permissions and
+ * limitations under the License.
+ ******************************************************************************/
+
+#pragma once
+
+/**
+ * @file LinearResample.h
+ * @ingroup respec
+ * The LinearResample class.
+ */
+
+#include "respec/SpecsChanger.h"
+
+AUD_NAMESPACE_BEGIN
+
+/**
+ * This sound creates a resampling reader that does simple linear resampling.
+ */
+class AUD_API LinearResample : public SpecsChanger
+{
+private:
+	// delete copy constructor and operator=
+	LinearResample(const LinearResample&) = delete;
+	LinearResample& operator=(const LinearResample&) = delete;
+
+public:
+	/**
+	 * Creates a new sound.
+	 * \param sound The input sound.
+	 * \param specs The target specifications.
+	 */
+	LinearResample(std::shared_ptr<ISound> sound, DeviceSpecs specs);
+
+	virtual std::shared_ptr<IReader> createReader();
+};
+
+AUD_NAMESPACE_END

blender-5.2.0/extern/audaspace/include/respec/LinearResampleReader.h

@@ -0,0 +1,80 @@
+/*******************************************************************************
+ * Copyright 2009-2016 Jörg Müller
+ *
+ * Licensed under the Apache License, Version 2.0 (the "License");
+ * you may not use this file except in compliance with the License.
+ * You may obtain a copy of the License at
+ *
+ *   http://www.apache.org/licenses/LICENSE-2.0
+ *
+ * Unless required by applicable law or agreed to in writing, software
+ * distributed under the License is distributed on an "AS IS" BASIS,
+ * WITHOUT WARRANTIES OR CONDITIONS OF ANY KIND, either express or implied.
+ * See the License for the specific language governing permissions and
+ * limitations under the License.
+ ******************************************************************************/
+
+#pragma once
+
+/**
+ * @file LinearResampleReader.h
+ * @ingroup respec
+ * The LinearResampleReader class.
+ */
+
+#include "respec/ResampleReader.h"
+#include "util/Buffer.h"
+
+AUD_NAMESPACE_BEGIN
+
+/**
+ * This resampling reader does simple first-order hold resampling.
+ */
+class AUD_API LinearResampleReader : public ResampleReader
+{
+private:
+	/**
+	 * The reader channels.
+	 */
+	Channels m_channels;
+
+	/**
+	 * The position in the cache.
+	 */
+	float m_cache_pos;
+
+	/**
+	 * The sound output buffer.
+	 */
+	Buffer m_buffer;
+
+	/**
+	 * The input caching buffer.
+	 */
+	Buffer m_cache;
+
+	/**
+	 * Whether the cache contains valid data.
+	 */
+	bool m_cache_ok;
+
+	// delete copy constructor and operator=
+	LinearResampleReader(const LinearResampleReader&) = delete;
+	LinearResampleReader& operator=(const LinearResampleReader&) = delete;
+
+public:
+	/**
+	 * Creates a resampling reader.
+	 * \param reader The reader to mix.
+	 * \param rate The target sampling rate.
+	 */
+	LinearResampleReader(std::shared_ptr<IReader> reader, SampleRate rate);
+
+	virtual void seek(int position);
+	virtual int getLength() const;
+	virtual int getPosition() const;
+	virtual Specs getSpecs() const;
+	virtual void read(int& length, bool& eos, sample_t* buffer);
+};
+
+AUD_NAMESPACE_END

blender-5.2.0/extern/audaspace/include/respec/Mixer.h

@@ -0,0 +1,129 @@
+/*******************************************************************************
+ * Copyright 2009-2016 Jörg Müller
+ *
+ * Licensed under the Apache License, Version 2.0 (the "License");
+ * you may not use this file except in compliance with the License.
+ * You may obtain a copy of the License at
+ *
+ *   http://www.apache.org/licenses/LICENSE-2.0
+ *
+ * Unless required by applicable law or agreed to in writing, software
+ * distributed under the License is distributed on an "AS IS" BASIS,
+ * WITHOUT WARRANTIES OR CONDITIONS OF ANY KIND, either express or implied.
+ * See the License for the specific language governing permissions and
+ * limitations under the License.
+ ******************************************************************************/
+
+#pragma once
+
+/**
+ * @file Mixer.h
+ * @ingroup respec
+ * The Mixer class.
+ */
+
+#include "respec/Specification.h"
+#include "respec/ConverterFunctions.h"
+#include "util/Buffer.h"
+
+#include <memory>
+
+AUD_NAMESPACE_BEGIN
+
+class IReader;
+
+/**
+ * This abstract class is able to mix audiosignals with same channel count
+ * and sample rate and convert it to a specific output format.
+ */
+class AUD_API Mixer
+{
+private:
+	// delete copy constructor and operator=
+	Mixer(const Mixer&) = delete;
+	Mixer& operator=(const Mixer&) = delete;
+
+protected:
+	/**
+	 * The output specification.
+	 */
+	DeviceSpecs m_specs;
+
+	/**
+	 * The length of the mixing buffer.
+	 */
+	int m_length;
+
+	/**
+	 * The mixing buffer.
+	 */
+	Buffer m_buffer;
+
+	/**
+	 * Converter function.
+	 */
+	convert_f m_convert;
+
+public:
+	/**
+	 * Creates the mixer.
+	 */
+	Mixer(DeviceSpecs specs);
+
+	/**
+	 * Destroys the mixer.
+	 */
+	virtual ~Mixer() {}
+
+	/**
+	 * Returns the target specification for superposing.
+	 * \return The target specification.
+	 */
+	DeviceSpecs getSpecs() const;
+
+	/**
+	 * Sets the target specification for superposing.
+	 * \param specs The target specification.
+	 */
+	void setSpecs(Specs specs);
+
+	/**
+	 * Sets the target specification for superposing.
+	 * \param specs The target specification.
+	 */
+	void setSpecs(DeviceSpecs specs);
+
+	/**
+	 * Mixes a buffer.
+	 * \param buffer The buffer to superpose.
+	 * \param start The start sample of the buffer.
+	 * \param length The length of the buffer in samples.
+	 * \param volume The mixing volume. Must be a value between 0.0 and 1.0.
+	 */
+	void mix(sample_t* buffer, int start, int length, float volume);
+
+	/**
+	 * Mixes a buffer with linear volume interpolation.
+	 * \param buffer The buffer to superpose.
+	 * \param start The start sample of the buffer.
+	 * \param length The length of the buffer in samples.
+	 * \param volume_to The target mixing volume. Must be a value between 0.0 and 1.0.
+	 * \param volume_from The start mixing volume. Must be a value between 0.0 and 1.0.
+	 */
+	void mix(sample_t* buffer, int start, int length, float volume_to, float volume_from);
+
+	/**
+	 * Writes the mixing buffer into an output buffer.
+	 * \param buffer The target buffer for superposing.
+	 * \param volume The mixing volume. Must be a value between 0.0 and 1.0.
+	 */
+	void read(data_t* buffer, float volume);
+
+	/**
+	 * Clears the mixing buffer.
+	 * \param length The length of the buffer in samples.
+	 */
+	void clear(int length);
+};
+
+AUD_NAMESPACE_END

blender-5.2.0/extern/audaspace/include/respec/ResampleReader.h

@@ -0,0 +1,61 @@
+/*******************************************************************************
+ * Copyright 2009-2016 Jörg Müller
+ *
+ * Licensed under the Apache License, Version 2.0 (the "License");
+ * you may not use this file except in compliance with the License.
+ * You may obtain a copy of the License at
+ *
+ *   http://www.apache.org/licenses/LICENSE-2.0
+ *
+ * Unless required by applicable law or agreed to in writing, software
+ * distributed under the License is distributed on an "AS IS" BASIS,
+ * WITHOUT WARRANTIES OR CONDITIONS OF ANY KIND, either express or implied.
+ * See the License for the specific language governing permissions and
+ * limitations under the License.
+ ******************************************************************************/
+
+#pragma once
+
+/**
+ * @file ResampleReader.h
+ * @ingroup respec
+ * The ResampleReader class.
+ */
+
+#include "fx/EffectReader.h"
+
+AUD_NAMESPACE_BEGIN
+
+/**
+ * This is the base class for all resampling readers.
+ */
+class AUD_API ResampleReader : public EffectReader
+{
+protected:
+	/**
+	 * The target sampling rate.
+	 */
+	SampleRate m_rate;
+
+	/**
+	 * Creates a resampling reader.
+	 * \param reader The reader to mix.
+	 * \param rate The target sampling rate.
+	 */
+	ResampleReader(std::shared_ptr<IReader> reader, SampleRate rate);
+
+public:
+	/**
+	 * Sets the sample rate.
+	 * \param rate The target sampling rate.
+	 */
+	virtual void setRate(SampleRate rate);
+
+	/**
+	 * Retrieves the sample rate.
+	 * \return The target sampling rate.
+	 */
+	virtual SampleRate getRate();
+};
+
+AUD_NAMESPACE_END

blender-5.2.0/extern/audaspace/include/respec/Specification.h

@@ -0,0 +1,147 @@
+/*******************************************************************************
+ * Copyright 2009-2016 Jörg Müller
+ *
+ * Licensed under the Apache License, Version 2.0 (the "License");
+ * you may not use this file except in compliance with the License.
+ * You may obtain a copy of the License at
+ *
+ *   http://www.apache.org/licenses/LICENSE-2.0
+ *
+ * Unless required by applicable law or agreed to in writing, software
+ * distributed under the License is distributed on an "AS IS" BASIS,
+ * WITHOUT WARRANTIES OR CONDITIONS OF ANY KIND, either express or implied.
+ * See the License for the specific language governing permissions and
+ * limitations under the License.
+ ******************************************************************************/
+
+#pragma once
+
+/**
+ * @file Specification.h
+ * @ingroup respec
+ * Defines all important macros and basic data structures for stream format descriptions.
+ */
+
+#include "Audaspace.h"
+
+/// The size of a format in bytes.
+#define AUD_FORMAT_SIZE(format) (format & 0x0F)
+/// The size of a sample in the specified device format in bytes.
+#define AUD_DEVICE_SAMPLE_SIZE(specs) (specs.channels * (specs.format & 0x0F))
+/// The size of a sample in the specified format in bytes.
+#define AUD_SAMPLE_SIZE(specs) (specs.channels * sizeof(sample_t))
+
+/// Compares two audio data specifications.
+#define AUD_COMPARE_SPECS(s1, s2) ((s1.rate == s2.rate) && (s1.channels == s2.channels))
+
+/// Returns the bit for a channel mask.
+#define AUD_CHANNEL_BIT(channel) (0x01 << channel)
+
+AUD_NAMESPACE_BEGIN
+
+/**
+ * The format of a sample.
+ * The last 4 bit save the byte count of the format.
+ */
+enum SampleFormat
+{
+	FORMAT_INVALID = 0x00,		/// Invalid sample format.
+	FORMAT_U8      = 0x01,		/// 1 byte unsigned byte.
+	FORMAT_S16     = 0x12,		/// 2 byte signed integer.
+	FORMAT_S24     = 0x13,		/// 3 byte signed integer.
+	FORMAT_S32     = 0x14,		/// 4 byte signed integer.
+	FORMAT_FLOAT32 = 0x24,		/// 4 byte float.
+	FORMAT_FLOAT64 = 0x28		/// 8 byte float.
+};
+
+/// The channel count.
+enum Channels
+{
+	CHANNELS_INVALID    = 0,	/// Invalid channel count.
+	CHANNELS_MONO       = 1,	/// Mono.
+	CHANNELS_STEREO     = 2,	/// Stereo.
+	CHANNELS_STEREO_LFE = 3,	/// Stereo with LFE channel.
+	CHANNELS_SURROUND4  = 4,	/// 4 channel surround sound.
+	CHANNELS_SURROUND5  = 5,	/// 5 channel surround sound.
+	CHANNELS_SURROUND51 = 6,	/// 5.1 surround sound.
+	CHANNELS_SURROUND61 = 7,	/// 6.1 surround sound.
+	CHANNELS_SURROUND71 = 8	/// 7.1 surround sound.
+};
+
+/// The channel names.
+enum Channel
+{
+	CHANNEL_FRONT_LEFT = 0,
+	CHANNEL_FRONT_RIGHT,
+	CHANNEL_FRONT_CENTER,
+	CHANNEL_LFE,
+	CHANNEL_REAR_LEFT,
+	CHANNEL_REAR_RIGHT,
+	CHANNEL_REAR_CENTER,
+	CHANNEL_SIDE_LEFT,
+	CHANNEL_SIDE_RIGHT,
+	CHANNEL_MAX
+};
+
+/// Resampling algorithm and quality.
+enum class ResampleQuality
+{
+	FASTEST = 0, /// Linear resample, very fast but lowest quality.
+	LOW,         /// JOS resample at low quality preset.
+	MEDIUM,      /// JOS resample at medium quality preset.
+	HIGH         /// JOS resample at high quality preset.
+};
+
+/**
+ * The sample rate tells how many samples are played back within one second.
+ * Some exotic formats may use other sample rates than provided here.
+ */
+enum DefaultSampleRate
+{
+	RATE_INVALID = 0,			/// Invalid sample rate.
+	RATE_8000    = 8000,		/// 8000 Hz.
+	RATE_16000   = 16000,		/// 16000 Hz.
+	RATE_11025   = 11025,		/// 11025 Hz.
+	RATE_22050   = 22050,		/// 22050 Hz.
+	RATE_32000   = 32000,		/// 32000 Hz.
+	RATE_44100   = 44100,		/// 44100 Hz.
+	RATE_48000   = 48000,		/// 48000 Hz.
+	RATE_88200   = 88200,		/// 88200 Hz.
+	RATE_96000   = 96000,		/// 96000 Hz.
+	RATE_192000  = 192000		/// 192000 Hz.
+};
+
+/// Sample rate type.
+typedef double SampleRate;
+
+/// Specification of a sound source.
+struct Specs
+{
+	/// Sample rate in Hz.
+	SampleRate rate;
+
+	/// Channel count.
+	Channels channels;
+};
+
+/// Specification of a sound device.
+struct DeviceSpecs
+{
+	/// Sample format.
+	SampleFormat format;
+
+	union
+	{
+		struct
+		{
+			/// Sample rate in Hz.
+			SampleRate rate;
+
+			/// Channel count.
+			Channels channels;
+		};
+		Specs specs;
+	};
+};
+
+AUD_NAMESPACE_END

blender-5.2.0/extern/audaspace/include/respec/SpecsChanger.h

@@ -0,0 +1,74 @@
+/*******************************************************************************
+ * Copyright 2009-2016 Jörg Müller
+ *
+ * Licensed under the Apache License, Version 2.0 (the "License");
+ * you may not use this file except in compliance with the License.
+ * You may obtain a copy of the License at
+ *
+ *   http://www.apache.org/licenses/LICENSE-2.0
+ *
+ * Unless required by applicable law or agreed to in writing, software
+ * distributed under the License is distributed on an "AS IS" BASIS,
+ * WITHOUT WARRANTIES OR CONDITIONS OF ANY KIND, either express or implied.
+ * See the License for the specific language governing permissions and
+ * limitations under the License.
+ ******************************************************************************/
+
+#pragma once
+
+/**
+ * @file SpecsChanger.h
+ * @ingroup respec
+ * The SpecsChanger class.
+ */
+
+#include "ISound.h"
+#include "respec/Specification.h"
+
+AUD_NAMESPACE_BEGIN
+
+/**
+ * This sound is a base class for all mixer factories.
+ */
+class AUD_API SpecsChanger : public ISound
+{
+protected:
+	/**
+	 * The target specification for resampling.
+	 */
+	const DeviceSpecs m_specs;
+
+	/**
+	 * If there is no reader it is created out of this sound.
+	 */
+	std::shared_ptr<ISound> m_sound;
+
+	/**
+	 * Returns the reader created out of the sound.
+	 * This method can be used for the createReader function of the implementing
+	 * classes.
+	 * \return The reader to mix.
+	 */
+	std::shared_ptr<IReader> getReader() const;
+
+public:
+	/**
+	 * Creates a new sound.
+	 * \param sound The sound to create the readers to mix out of.
+	 * \param specs The target specification.
+	 */
+	SpecsChanger(std::shared_ptr<ISound> sound, DeviceSpecs specs);
+
+	/**
+	 * Returns the target specification for resampling.
+	 */
+	DeviceSpecs getSpecs() const;
+
+	/**
+	 * Returns the saved sound.
+	 * \return The sound.
+	 */
+	std::shared_ptr<ISound> getSound() const;
+};
+
+AUD_NAMESPACE_END