// Copyright (c) 2014, 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/

#include "libaegisub/audio/provider.h"

#include "libaegisub/fs.h"
#include "libaegisub/io.h"
#include "libaegisub/log.h"
#include "libaegisub/util.h"

namespace {

template<typename Source>
class ConvertFloatToInt16 {
	Source* src;
public:
	ConvertFloatToInt16(Source* src) :src(src) {}
	int16_t operator[](size_t idx) const {
		Source expanded = src[idx] * 32768;
		return expanded < -32768 ? -32768 :
			expanded > 32767 ? 32767 :
			static_cast<int16_t>(expanded);
	}
};

// 8 bits per sample is assumed to be unsigned with a bias of 128,
// while everything else is assumed to be signed with zero bias
class ConvertIntToInt16 {
	void* src;
	int bytes_per_sample;
public:
	ConvertIntToInt16(void* src, int bytes_per_sample) :src(src), bytes_per_sample(bytes_per_sample) {}
	const int16_t& operator[](size_t idx) const {
		return *reinterpret_cast<int16_t*>(reinterpret_cast<char*>(src) + (idx + 1) * bytes_per_sample - sizeof(int16_t));
	}
};
class ConvertUInt8ToInt16 {
	uint8_t* src;
public:
	ConvertUInt8ToInt16(uint8_t* src) :src(src) {}
	int16_t operator[](size_t idx) const {
		return int16_t(src[idx]-128) << 8;
	}
};

template<typename Source>
class DownmixToMono {
	Source src;
	int channels;
public:
	DownmixToMono(Source src, int channels) :src(src), channels(channels) {}
	int16_t operator[](size_t idx) const {
		int ret = 0;
		// Just average the channels together
		for (int i = 0; i < channels; ++i)
			ret += src[idx * channels + i];
		return ret / channels;
	}
};
}

namespace agi {
void AudioProvider::FillBufferInt16Mono(int16_t* buf, int64_t start, int64_t count) const {
	if (!float_samples && bytes_per_sample == 2 && channels == 1) {
		FillBuffer(buf, start, count);
		return;
	}
	void* buff = malloc(bytes_per_sample * count * channels);
	FillBuffer(buff, start, count);
	if (channels == 1) {
		if (float_samples) {
			if (bytes_per_sample == sizeof(float))
				for (int64_t i = 0; i < count; ++i)
					buf[i] = ConvertFloatToInt16<float>(reinterpret_cast<float*>(buff))[i];
			else if (bytes_per_sample == sizeof(double))
				for (int64_t i = 0; i < count; ++i)
					buf[i] = ConvertFloatToInt16<double>(reinterpret_cast<double*>(buff))[i];
		}
		else {
			if (bytes_per_sample == sizeof(uint8_t))
				for (int64_t i = 0; i < count; ++i)
					buf[i] = ConvertUInt8ToInt16(reinterpret_cast<uint8_t*>(buff))[i];
			else
				for (int64_t i = 0; i < count; ++i)
					buf[i] = ConvertIntToInt16(buff, bytes_per_sample)[i];
		}
	}
	else {
		if (float_samples) {
			if (bytes_per_sample == sizeof(float))
				for (int64_t i = 0; i < count; ++i)
					buf[i] = DownmixToMono<ConvertFloatToInt16<float> >(ConvertFloatToInt16<float>(reinterpret_cast<float*>(buff)), channels)[i];
			else if (bytes_per_sample == sizeof(double))
				for (int64_t i = 0; i < count; ++i)
					buf[i] = DownmixToMono<ConvertFloatToInt16<double> >(ConvertFloatToInt16<double>(reinterpret_cast<double*>(buff)), channels)[i];
		}
		else {
			if (bytes_per_sample == sizeof(uint8_t))
				for (int64_t i = 0; i < count; ++i)
					buf[i] = DownmixToMono<ConvertUInt8ToInt16>(ConvertUInt8ToInt16(reinterpret_cast<uint8_t*>(buff)), channels)[i];
			else
				for (int64_t i = 0; i < count; ++i)
					buf[i] = DownmixToMono<ConvertIntToInt16>(ConvertIntToInt16(buff, bytes_per_sample), channels)[i];
		}
	}
	free(buff);
}

// This entire file has turned into a mess. For now I'm just following the pattern of the wangqr code, but
// this should really be restructured entirely again. The original type constructor-based system worked very well - it could
// just give downmix/conversion control to the players instead.
void AudioProvider::GetAudioWithVolume(void *buf, int64_t start, int64_t count, double volume) const {
	GetAudio(buf, start, count);
	if (volume == 1.0) return;

	int64_t n = count * GetChannels();

	if (float_samples) {
		if (bytes_per_sample == sizeof(float)) {
			float *buff = reinterpret_cast<float *>(buf);
			for (int64_t i = 0; i < n; ++i)
				buff[i] = static_cast<float>(buff[i] * volume);
		} else if (bytes_per_sample == sizeof(double)) {
			double *buff = reinterpret_cast<double *>(buf);
			for (int64_t i = 0; i < n; ++i)
				buff[i] = buff[i] * volume;
		}
	}
	else {
		if (bytes_per_sample == sizeof(uint8_t)) {
			uint8_t *buff = reinterpret_cast<uint8_t *>(buf);
			for (int64_t i = 0; i < n; ++i)
				buff[i] = util::mid(0, static_cast<int>(((int) buff[i] - 128) * volume + 128), 0xFF);
		} else if (bytes_per_sample == sizeof(int16_t)) {
			int16_t *buff = reinterpret_cast<int16_t *>(buf);
			for (int64_t i = 0; i < n; ++i)
				buff[i] = util::mid(-0x8000, static_cast<int>(buff[i] * volume), 0x7FFF);
		} else if (bytes_per_sample == sizeof(int32_t)) {
			int32_t *buff = reinterpret_cast<int32_t *>(buf);
			for (int64_t i = 0; i < n; ++i)
				buff[i] = static_cast<int32_t>(buff[i] * volume);
		} else if (bytes_per_sample == sizeof(int64_t)) {
			int64_t *buff = reinterpret_cast<int64_t *>(buf);
			for (int64_t i = 0; i < n; ++i)
				buff[i] = static_cast<int64_t>(buff[i] * volume);
		}
	}
}

void AudioProvider::GetInt16MonoAudioWithVolume(int16_t *buf, int64_t start, int64_t count, double volume) const {
	GetInt16MonoAudio(buf, start, count);
	if (volume == 1.0) return;

	auto buffer = static_cast<int16_t *>(buf);
	for (size_t i = 0; i < (size_t)count; ++i)
		buffer[i] = util::mid(-0x8000, static_cast<int>(buffer[i] * volume + 0.5), 0x7FFF);
}

void AudioProvider::ZeroFill(void *buf, int64_t count) const {
	if (bytes_per_sample == 1)
		// 8 bit formats are usually unsigned with bias 128
		memset(buf, 128, count * channels);
	else // While everything else is signed
		memset(buf, 0, count * bytes_per_sample * channels);
}

void AudioProvider::GetAudio(void *buf, int64_t start, int64_t count) const {
	if (start < 0) {
		ZeroFill(buf, std::min(-start, count));
		buf = static_cast<char *>(buf) + -start * bytes_per_sample * channels;
		count += start;
		start = 0;
	}

	if (start + count > num_samples) {
		int64_t zero_count = std::min(count, start + count - num_samples);
		count -= zero_count;
		ZeroFill(static_cast<char *>(buf) + count * bytes_per_sample * channels, zero_count);
	}

	if (count <= 0) return;

	try {
		FillBuffer(buf, start, count);
	}
	catch (AudioDecodeError const& e) {
		// We don't have any good way to report errors here, so just log the
		// failure and return silence
		LOG_E("audio_provider") << e.GetMessage();
		ZeroFill(buf, count);
		return;
	}
	catch (...) {
		LOG_E("audio_provider") << "Unknown audio decoding error";
		ZeroFill(buf, count);
		return;
	}
}

void AudioProvider::GetInt16MonoAudio(int16_t* buf, int64_t start, int64_t count) const {
	if (start < 0) {
		memset(buf, 0, sizeof(int16_t) * std::min(-start, count));
		buf -= start;
		count += start;
		start = 0;
	}

	if (start + count > num_samples) {
		int64_t zero_count = std::min(count, start + count - num_samples);
		count -= zero_count;
		memset(buf + count, 0, sizeof(int16_t) * zero_count);
	}

	if (count <= 0) return;

	try {
		FillBufferInt16Mono(buf, start, count);
	}
	catch (AudioDecodeError const& e) {
		// We don't have any good way to report errors here, so just log the
		// failure and return silence
		LOG_E("audio_provider") << e.GetMessage();
		memset(buf, 0, sizeof(int16_t) * count);
		return;
	}
	catch (...) {
		LOG_E("audio_provider") << "Unknown audio decoding error";
		memset(buf, 0, sizeof(int16_t) * count);
		return;
	}
}

namespace {
class writer {
	io::Save outfile;
	std::ostream& out;

public:
	writer(agi::fs::path const& filename) : outfile(filename, true), out(outfile.Get()) { }

	template<int N>
	void write(const char(&str)[N]) {
		out.write(str, N - 1);
	}

	void write(std::vector<char> const& data) {
		out.write(data.data(), data.size());
	}

	template<typename Dest, typename Src>
	void write(Src v) {
		auto converted = static_cast<Dest>(v);
		out.write(reinterpret_cast<char *>(&converted), sizeof(Dest));
	}
};
}

void SaveAudioClip(AudioProvider const& provider, fs::path const& path, int start_time, int end_time) {
	const auto max_samples = provider.GetNumSamples();
	const auto start_sample = std::min(max_samples, ((int64_t)start_time * provider.GetSampleRate() + 999) / 1000);
	const auto end_sample = util::mid(start_sample, ((int64_t)end_time * provider.GetSampleRate() + 999) / 1000, max_samples);

	const size_t bytes_per_sample = provider.GetBytesPerSample() * provider.GetChannels();
	const size_t bufsize = (end_sample - start_sample) * bytes_per_sample;

	writer out{path};
	out.write("RIFF");
	out.write<int32_t>(bufsize + 36);

	out.write("WAVEfmt ");
	out.write<int32_t>(16); // Size of chunk
	out.write<int16_t>(provider.AreSamplesFloat() ? 3 : 1);  // compression format (1: WAVE_FORMAT_PCM, 3: WAVE_FORMAT_IEEE_FLOAT)
	out.write<int16_t>(provider.GetChannels());
	out.write<int32_t>(provider.GetSampleRate());
	out.write<int32_t>(provider.GetSampleRate() * provider.GetChannels() * provider.GetBytesPerSample());
	out.write<int16_t>(provider.GetChannels() * provider.GetBytesPerSample());
	out.write<int16_t>(provider.GetBytesPerSample() * 8);

	out.write("data");
	out.write<int32_t>(bufsize);

	// samples per read
	size_t spr = 65536 / bytes_per_sample;
	std::vector<char> buf;
	for (int64_t i = start_sample; i < end_sample; i += spr) {
		spr = std::min<size_t>(spr, end_sample - i);
		buf.resize(spr * bytes_per_sample);
		provider.GetAudio(&buf[0], i, spr);
		out.write(buf);
	}
}
}