#include <pocketsphinx.h>
#include <iostream>
#include <boost/filesystem.hpp>
#include "phone_extraction.h"
#include "audio_input/SampleRateConverter.h"
#include "audio_input/ChannelDownmixer.h"
#include "platform_tools.h"
using std::runtime_error;
using std::unique_ptr;
using std::shared_ptr;
using std::string;
using std::map;
using boost::filesystem::path;

unique_ptr<AudioStream> to16kHzMono(unique_ptr<AudioStream> stream) {
	// Downmix, if required
	if (stream->getChannelCount() != 1) {
		stream.reset(new ChannelDownmixer(std::move(stream)));
	}

	// Downsample, if required
	if (stream->getFrameRate() < 16000) {
		throw runtime_error("Sample rate must not be below 16kHz.");
	}
	if (stream->getFrameRate() != 16000) {
		stream.reset(new SampleRateConverter(std::move(stream), 16000));
	}

	return stream;
}

// Converts a float in the range -1..1 to a signed 16-bit int
int16_t floatSampleToInt16(float sample) {
	sample = std::max(sample, -1.0f);
	sample = std::min(sample, 1.0f);
	return static_cast<int16_t>(((sample + 1) / 2) * (INT16_MAX - INT16_MIN) + INT16_MIN);
}

map<centiseconds, Phone> detectPhones(unique_ptr<AudioStream> audioStream) {
	// Convert audio stream to the exact format PocketSphinx requires
	audioStream = to16kHzMono(std::move(audioStream));

	// Create PocketSphinx configuration
	path binDirectory(getBinDirectory());
	path resDirectory(binDirectory.parent_path() / "res");
	shared_ptr<cmd_ln_t> config(
		cmd_ln_init(
			nullptr, ps_args(), true,
			// Set acoustic model
			"-hmm", (resDirectory / "sphinx/acoustic_model").string().c_str(),
			// Set phonetic language model
			"-allphone", (resDirectory / "sphinx/en-us-phone.lm.bin").string().c_str(),
			"-allphone_ci", "yes",
			// The following settings are taken from http://cmusphinx.sourceforge.net/wiki/phonemerecognition
			// Set beam width applied to every frame in Viterbi search
			"-beam", "1e-20",
			// Set beam width applied to phone transitions
			"-pbeam", "1e-20",
			// Set language model probability weight
			"-lw", "2.0",
			nullptr),
		[](cmd_ln_t* config) { cmd_ln_free_r(config); });
	if (!config) throw runtime_error("Error creating configuration.");

	// Create phone recognizer
	shared_ptr<ps_decoder_t> recognizer(
		ps_init(config.get()),
		[](ps_decoder_t* recognizer) { ps_free(recognizer); });
	if (!recognizer) throw runtime_error("Error creating speech recognizer.");

	// Start recognition
	int error = ps_start_utt(recognizer.get());
	if (error) throw runtime_error("Error starting utterance processing.");

	// Process entire sound file
	std::vector<int16_t> buffer;
	const int capacity = 1600; // 0.1 second capacity
	buffer.reserve(capacity);
	int sampleCount = 0;
	do {
		// Read to buffer
		buffer.clear();
		while (buffer.size() < capacity) {
			float sample;
			if (!audioStream->getNextSample(sample)) break;
			buffer.push_back(floatSampleToInt16(sample));
		}

		// Analyze buffer
		int searchedFrameCount = ps_process_raw(recognizer.get(), buffer.data(), buffer.size(), false, false);
		if (searchedFrameCount < 0) throw runtime_error("Error decoding raw audio data.");

		sampleCount += buffer.size();
	} while (buffer.size());
	error = ps_end_utt(recognizer.get());
	if (error) throw runtime_error("Error ending utterance processing.");

	// Collect results into map
	map<centiseconds, Phone> result;
	ps_seg_t *segmentationIter;
	int32 score;
	int endFrame;
	for (segmentationIter = ps_seg_iter(recognizer.get(), &score); segmentationIter; segmentationIter = ps_seg_next(segmentationIter)) {
		// Get phone
		char const *phone = ps_seg_word(segmentationIter);

		// Get timing
		int startFrame;
		ps_seg_frames(segmentationIter, &startFrame, &endFrame);

		result[centiseconds(startFrame)] = stringToPhone(phone);
	}
	// Add dummy entry past the last phone
	result[centiseconds(endFrame + 1)] = Phone::None;
	return result;
}