Aegisub/aegisub/src/audio_provider_convert.cpp

// Copyright (c) 2011, Thomas Goyne <plorkyeran@aegisub.org>
//
// Permission to use, copy, modify, and distribute this software for any
// purpose with or without fee is hereby granted, provided that the above
// copyright notice and this permission notice appear in all copies.
//
// THE SOFTWARE IS PROVIDED "AS IS" AND THE AUTHOR DISCLAIMS ALL WARRANTIES
// WITH REGARD TO THIS SOFTWARE INCLUDING ALL IMPLIED WARRANTIES OF
// MERCHANTABILITY AND FITNESS. IN NO EVENT SHALL THE AUTHOR BE LIABLE FOR
// ANY SPECIAL, DIRECT, INDIRECT, OR CONSEQUENTIAL DAMAGES OR ANY DAMAGES
// WHATSOEVER RESULTING FROM LOSS OF USE, DATA OR PROFITS, WHETHER IN AN
// ACTION OF CONTRACT, NEGLIGENCE OR OTHER TORTIOUS ACTION, ARISING OUT OF
// OR IN CONNECTION WITH THE USE OR PERFORMANCE OF THIS SOFTWARE.
//
// Aegisub Project http://www.aegisub.org/

/// @file audio_provider_convert.cpp
/// @brief Intermediate sample format-converting audio provider
/// @ingroup audio_input
///

#include "config.h"

#include "audio_provider_convert.h"

#include "aegisub_endian.h"
#include "audio_controller.h"
#include "include/aegisub/audio_provider.h"

#include <libaegisub/scoped_ptr.h>
#include <libaegisub/log.h>

#include <limits>

/// Base class for all wrapping converters
class AudioProviderConverter : public AudioProvider {
protected:
	agi::scoped_ptr<AudioProvider> source;
public:
	AudioProviderConverter(AudioProvider *src) : source(src) {
		channels = source->GetChannels();
		num_samples = source->GetNumSamples();
		sample_rate = source->GetSampleRate();
		bytes_per_sample = source->GetBytesPerSample();
		float_samples = source->AreSamplesFloat();
	}

	bool AreSamplesNativeEndian() const { return true; }
	agi::fs::path GetFilename() const { return source->GetFilename(); }
};

/// Anything integral -> 16 bit signed machine-endian audio converter
template<class Target>
class BitdepthConvertAudioProvider : public AudioProviderConverter {
	int src_bytes_per_sample;
	bool src_is_native_endian;
public:
	BitdepthConvertAudioProvider(AudioProvider *src) : AudioProviderConverter(src) {
		if (bytes_per_sample > 8)
			throw agi::AudioProviderOpenError("Audio format converter: audio with bitdepths greater than 64 bits/sample is currently unsupported", 0);

		src_is_native_endian = src->AreSamplesNativeEndian();
		src_bytes_per_sample = bytes_per_sample;
		bytes_per_sample = sizeof(Target);
	}

	void FillBuffer(void *buf, int64_t start, int64_t count) const {
		std::vector<char> src_buf(count * src_bytes_per_sample * channels);
		source->GetAudio(&src_buf[0], start, count);

		int16_t *dest = reinterpret_cast<int16_t*>(buf);

		for (int64_t i = 0; i < count * channels; ++i) {
			int64_t sample = 0;
			char *sample_ptr = (char*)&sample;
			char *src = &src_buf[i * src_bytes_per_sample];

			// 8 bits per sample is assumed to be unsigned with a bias of 127,
			// while everything else is assumed to be signed with zero bias
			if (src_bytes_per_sample == 1)
				*sample_ptr = static_cast<uint8_t>(*src) - 127;
			else if (src_is_native_endian) {
#ifdef HAVE_LITTLE_ENDIAN
				memcpy(sample_ptr, src, src_bytes_per_sample);
#else
				memcpy(sample_ptr + sizeof(int64_t) - src_bytes_per_sample, src, src_bytes_per_sample);
#endif
			}
			else {
				for (int byte_index = 0; i < src_bytes_per_sample; ++i) {
#ifdef HAVE_LITTLE_ENDIAN
					sample_ptr[byte_index] = src[src_bytes_per_sample - byte_index - 1];
#else
					sample_ptr[sizeof(int64_t) - byte_index - 1] = src[byte_index];
#endif
				}
			}

			if (static_cast<size_t>(src_bytes_per_sample) > sizeof(Target))
				sample >>= (src_bytes_per_sample - sizeof(Target)) * 8;
			else if (static_cast<size_t>(src_bytes_per_sample) < sizeof(Target))
				sample <<= (sizeof(Target) - src_bytes_per_sample ) * 8;

			dest[i] = (Target)sample;
		}
	}
};

/// Floating point -> 16 bit signed machine-endian audio converter
template<class Source, class Target>
class FloatConvertAudioProvider : public AudioProviderConverter {
public:
	FloatConvertAudioProvider(AudioProvider *src) : AudioProviderConverter(src) {
		if (!src->AreSamplesNativeEndian())
			throw agi::AudioProviderOpenError("Audio format converter: Float audio with non-native endianness is currently unsupported.", 0);
		bytes_per_sample = sizeof(Target);
		float_samples = false;
	}

	void FillBuffer(void *buf, int64_t start, int64_t count) const {
		std::vector<Source> src_buf(count * channels);
		source->GetAudio(&src_buf[0], start, count);

		Target *dest = reinterpret_cast<Target*>(buf);

		for (size_t i = 0; i < static_cast<size_t>(count * channels); ++i) {
			Source expanded;
			if (src_buf[i] < 0)
				expanded = static_cast<Target>(-src_buf[i] * std::numeric_limits<Target>::min());
			else
				expanded = static_cast<Target>(src_buf[i] * std::numeric_limits<Target>::max());

			if (expanded < std::numeric_limits<Target>::min())
				dest[i] = std::numeric_limits<Target>::min();
			else if (expanded > std::numeric_limits<Target>::max())
				dest[i] = std::numeric_limits<Target>::max();
			else
				dest[i] = static_cast<Target>(expanded);
		}
	}
};

/// Non-mono 16-bit signed machine-endian -> mono 16-bit signed machine endian converter
class DownmixAudioProvider : public AudioProviderConverter {
	int src_channels;
public:
	DownmixAudioProvider(AudioProvider *src) : AudioProviderConverter(src) {
		if (bytes_per_sample != 2)
			throw agi::InternalError("DownmixAudioProvider requires 16-bit input", 0);
		if (channels == 1)
			throw agi::InternalError("DownmixAudioProvider requires multi-channel input", 0);
		src_channels = channels;
		channels = 1;
	}

	void FillBuffer(void *buf, int64_t start, int64_t count) const {
		if (count == 0) return;

		std::vector<int16_t> src_buf(count * src_channels);
		source->GetAudio(&src_buf[0], start, count);

		int16_t *dst = reinterpret_cast<int16_t*>(buf);
		// Just average the channels together
		while (count-- > 0) {
			int sum = 0;
			for (int c = 0; c < src_channels; ++c)
				sum += src_buf[count * src_channels + c];
			dst[count] = static_cast<int16_t>(sum / src_channels);
		}
	}
};

/// Sample doubler with linear interpolation for the new samples
/// Requires 16-bit mono input
class SampleDoublingAudioProvider : public AudioProviderConverter {
public:
	SampleDoublingAudioProvider(AudioProvider *src) : AudioProviderConverter(src) {
		if (src->GetBytesPerSample() != 2)
			throw agi::InternalError("UpsampleAudioProvider requires 16-bit input", 0);
		if (src->GetChannels() != 1)
			throw agi::InternalError("UpsampleAudioProvider requires mono input", 0);

		sample_rate *= 2;
		num_samples *= 2;
	}

	void FillBuffer(void *buf, int64_t start, int64_t count) const {
		if (count == 0) return;

		int not_end = start + count < num_samples;
		int64_t src_count = count / 2;
		source->GetAudio(buf, start / 2, src_count + not_end);

		int16_t *buf16 = reinterpret_cast<int16_t*>(buf);

		if (!not_end) {
			// We weren't able to request a sample past the end so just
			// duplicate the last sample
			buf16[src_count] = buf16[src_count + 1];
		}

		if (count % 2)
			buf16[count - 1] = buf16[src_count];

		// walking backwards so that the conversion can be done in place
		for (int64_t i = src_count - 1; i >= 0; --i) {
			buf16[i * 2] = buf16[i];
			buf16[i * 2 + 1] = (int16_t)(((int32_t)buf16[i] + buf16[i + 1]) / 2);
		}
	}
};

AudioProvider *CreateConvertAudioProvider(AudioProvider *source_provider) {
	AudioProvider *provider = source_provider;

	// Ensure 16-bit audio with proper endianness
	if (provider->AreSamplesFloat()) {
		LOG_D("audio_provider") << "Converting float to S16";
		if (provider->GetBytesPerSample() == sizeof(float))
			provider = new FloatConvertAudioProvider<float, int16_t>(provider);
		else
			provider = new FloatConvertAudioProvider<double, int16_t>(provider);
	}
	if (provider->GetBytesPerSample() != 2 || !provider->AreSamplesNativeEndian()) {
		LOG_D("audio_provider") << "Converting " << provider->GetBytesPerSample() << " bytes per sample or wrong endian to S16";
		provider = new BitdepthConvertAudioProvider<int16_t>(provider);
	}

	// We currently only support mono audio
	if (provider->GetChannels() != 1) {
		LOG_D("audio_provider") << "Downmixing to mono from " << provider->GetChannels() << " channels";
		provider = new DownmixAudioProvider(provider);
	}

	// Some players don't like low sample rate audio
	while (provider->GetSampleRate() < 32000) {
		LOG_D("audio_provider") << "Doubling sample rate";
		provider = new SampleDoublingAudioProvider(provider);
	}

	return provider;
}