package jmri.jmrit.vsdecoder;

import com.jogamp.openal.AL;
import com.jogamp.openal.ALException;
import com.jogamp.openal.util.ALut;
import java.io.InputStream;
import java.nio.ByteBuffer;
import java.nio.ByteOrder;
import java.util.ArrayList;
import java.util.Arrays;
import java.util.List;
import jmri.jmrit.audio.AudioFactory;
import jmri.jmrit.audio.AudioBuffer;
import org.slf4j.Logger;
import org.slf4j.LoggerFactory;

/**
 * Utility class for doing "VSD-special" things with the JMRI Audio classes.
 *
 * <hr>
 * This file is part of JMRI.
 * <p>
 * JMRI is free software; you can redistribute it and/or modify it under the
 * terms of version 2 of the GNU General Public License as published by the Free
 * Software Foundation. See the "COPYING" file for a copy of this license.
 * <p>
 * JMRI is distributed in the hope that it will be useful, but WITHOUT ANY
 * WARRANTY; without even the implied warranty of MERCHANTABILITY or FITNESS FOR
 * A PARTICULAR PURPOSE. See the GNU General Public License for more details.
 *
 * @author Mark Underwood copyright (c) 2009, 2013
 */
public class AudioUtil {
    //------------------------
    // New Methods to allow creating a set of (sub)Buffers built off a single
    // input stream.  Should probably look closely at this and see if it could
    // be pushed up to AbstractAudioBuffer.  Or better yet, moved OVER to AbstractAudioFactory.

    /**
     * Split the WAV data from an InputStream into a series of ByteBuffers
     * of size (approximately) between max_time and min_time, split on zero
     * crossings.
     *
     * The method will try to create Buffers of max_time length (plus whatever's
     * needed to hit the next zero crossing), until there's not enough bytes
     * left in the InputStream. The last Buffer will be of at least min_time
     * length, else the remaining data (less than min_time length) will be
     * discarded. Buffers will be cut on zero-crossing boundaries.
     *
     * @param stream   : Input Stream, assumed to be WAV-format data
     * @param max_time : maximum (target) length of each split Buffer, in
     *                 milliseconds
     * @param min_time : minimum length of buffer to return, in milliseconds.
     *                 Any buffer that would be smaller than this will be
     *                 discarded.
     *
     * @return (List<ByteBuffer>) List of ByteBuffers containing the
     *         (split-up) data from stream
     */
    private static List<ByteBuffer> splitInputStream(InputStream stream, int max_time, int min_time) {
        List<ByteBuffer> rlist = new ArrayList<>();
        int[] format = new int[1];
        ByteBuffer[] data = new ByteBuffer[1];
        int[] size = new int[1];
        int[] freq = new int[1];
        int[] loop = new int[1];

        // Pull the WAV data into the "data" buffer.
        try {
            ALut.alutLoadWAVFile(stream, format, data, size, freq, loop);
        } catch (ALException e) {
            log.warn("Error loading JoalAudioBuffer", e);
            return null;
        }

        // OK, for now, we're only going to support 8-bit and 16-bit Mono data.
        // I'll have to figure out later how to extend this to multiple data formats.
        if ((format[0] != AL.AL_FORMAT_MONO8) && (format[0] != AL.AL_FORMAT_MONO16)) {
            log.warn("Invalid Format for splitting! Failing out.{}", parseFormat(format[0]));
            return null;
        }

        while (data[0].remaining() > 0) {
            log.debug("while loop. Source: {}", data[0]);
            ByteBuffer ab = getSubBuffer(data[0], max_time, min_time, format[0], freq[0]);
            // getSubBuffer returning null isn't necessarily an error.  It could mean there weren't enough bytes left, so we truncated.
            // In this case, we will have already gotten (via get()) the remaining bytes from data[0], so we should exit the while loop
            // normally.
            if (ab != null) {
                ab.rewind();
                rlist.add(ab);
            }
        }
        return rlist;
    }

    public static ByteBuffer getWavData(InputStream stream) {
        int[] format = new int[1];
        int[] size = new int[1];
        ByteBuffer[] data = new ByteBuffer[1];
        int[] freq = new int[1];
        int[] loop = new int[1];

        // Pull the WAV data into the "data" buffer.
        try {
            ALut.alutLoadWAVFile(stream, format, data, size, freq, loop);
        } catch (ALException e) {
            log.warn("Error loading JoalAudioBuffer from stream", e);
            return null;
        }
        log.debug("WAV data: {}, order: {}, size: {}", data[0], data[0].order(), size[0]);
        return data[0];
    }

    public static int[] getWavFormats(InputStream stream) {
        int[] format = new int[1];
        int[] size = new int[1];
        ByteBuffer[] data = new ByteBuffer[1];
        int[] freq = new int[1];
        int[] loop = new int[1];

        int[] formats = new int[3];

        // Pull the WAV data into the "data" buffer.
        try {
            ALut.alutLoadWAVFile(stream, format, data, size, freq, loop);
        } catch (ALException e) {
            log.warn("Error loading JoalAudioBuffer from stream", e);
            return formats;
        }
        // OK, for now, we're only going to support 8-bit and 16-bit Mono data.
        // I'll have to figure out later how to extend this to multiple data formats.
        if ((format[0] != AL.AL_FORMAT_MONO8) && (format[0] != AL.AL_FORMAT_MONO16)) {
            log.warn("Invalid Format! Failing out.{}", parseFormat(format[0]));
            return formats;
        }
        formats[0] = format[0];
        formats[1] = freq[0];
        formats[2] = frameSize(format[0]);
        return formats;
    }

    static public boolean isAudioRunning() {
        AudioFactory af = jmri.InstanceManager.getDefault(jmri.AudioManager.class).getActiveAudioFactory();
        if (af == null) {
            return false;
        } else {
            return ((jmri.jmrit.audio.AudioThread) af.getCommandThread()).alive();
        }
    }

    public static List<ByteBuffer> getByteBufferList(InputStream stream, int max_time, int min_time) {
        return splitInputStream(stream, max_time, min_time);
    }

    // This is here only because the AbstractAudioBuffer.getFrameSize() doesn't look
    // at the AL versions of the format strings.  And because it must be static.
    private static int frameSize(int format) {
        int frameSize;
        switch (format) {
            case AudioBuffer.FORMAT_16BIT_7DOT1:
                frameSize = 16;
                break;
            case AudioBuffer.FORMAT_8BIT_7DOT1:
                frameSize = 8;
                break;
            case AudioBuffer.FORMAT_16BIT_6DOT1:
                frameSize = 14;
                break;
            case AudioBuffer.FORMAT_8BIT_6DOT1:
                frameSize = 7;
                break;
            case AudioBuffer.FORMAT_16BIT_5DOT1:
                frameSize = 12;
                break;
            case AudioBuffer.FORMAT_8BIT_5DOT1:
                frameSize = 6;
                break;
            case AudioBuffer.FORMAT_16BIT_QUAD:
                frameSize = 8;
                break;
            case AudioBuffer.FORMAT_8BIT_QUAD:
            case AudioBuffer.FORMAT_16BIT_STEREO:
                frameSize = 4;
                break;
            case AL.AL_FORMAT_MONO16:
            case AL.AL_FORMAT_STEREO8:
                frameSize = 2;
                break;
            case AL.AL_FORMAT_MONO8:
            default:
                // Note this will be wrong for all the modes we don't support.
                frameSize = 1;
        }
        return frameSize;
    }

    private static String parseFormat(int fmt) {
        switch (fmt) {
            case AL.AL_FORMAT_MONO8:
                return "8-bit mono";
            case AL.AL_FORMAT_MONO16:
                return "16-bit mono";
            case AL.AL_FORMAT_STEREO8:
                return "8-bit stereo";
            case AL.AL_FORMAT_STEREO16:
                return "16-bit stereo";
            default:
                return "Something Multichannel: val=" + fmt;
        }
    }

    /**
     * Calculates the number of bytes offset that corresponds to the given time
     * interval, with the given data format and sample frequency.
     *
     * bytes = frame_size * time_ms * sample_frequency / 1000
     *
     * @param fmt     : audio data format
     * @param freq    : sample frequency in Hz
     * @param time_ms : time interval in milliseconds
     *
     * @return (int) number of bytes.
     */
    private static int calcTimeIndex(int fmt, int freq, int time_ms) {
        // freq == samples per second.  time_us = microseconds to calculate.
        // samples = time_us * freq / 1e3.
        // This will be approximate due to integer rounding.
        int rv = frameSize(fmt) * (time_ms * freq / 1000);
        log.debug("calcTimeIndex: freq = {} time_us = {} rv = {}", freq, time_ms, rv);
        return rv;
    }

    /**
     * Looks at the (last) three samples in buf (with sample size defined by
     *
     * @format) and determines whether they represent a positive-going zero
     * crossing event.
     *
     * Works only for AL.AL_FORMAT_MONO8 or AL.AL_FORMAT_MONO16. Returns false
     * otherwise.
     *
     * @param buf    : (minimum) 3-sample buffer of WAV data
     * @param len    : size of buf. Minimum 3 bytes for 8-bit, 6 bytes for
     *               16-bit mono data.
     * @param format : AL.format identifier.
     * @param order  : ByteOrder of data in buf
     *
     * @return true if a zero crossing is detected.
     */
    private static Boolean isZeroCross(byte[] buf, int len, int format, ByteOrder order) {
        switch (format) {
            case AL.AL_FORMAT_MONO8:
                if (len < 3) {
                    return false;
                } else {
                    return (((0xFF & buf[len - 3]) < 128) && ((0xFF & buf[len - 2]) < 128) && ((0xFF & buf[len - 1]) >= 128));
                }
            case AL.AL_FORMAT_MONO16:
                if (len < 6) {
                    return false;
                }
                short[] sbuf = new short[len / 2];
                // Below assumes little-endian
                ByteBuffer bb = ByteBuffer.wrap(buf);
                bb.order(order);
                //bb.reset();
                sbuf[0] = bb.getShort();
                sbuf[1] = bb.getShort();
                sbuf[2] = bb.getShort();
                return ((sbuf[0] < 0) && (sbuf[1] < 0) && (sbuf[2] >= 0));
            default:
                return false;
        }
    }

    /**
     * Extract a sub-buffer of (at least) specified size, extended to nearest
     * zero crossing, from the given ByteBuffer.
     *
     * Returns null if there are fewer than (min_time * sample rate) samples in
     * "source" Returns between min_time and (max_time + samples to next zero
     * crossing) samples if enough bytes are available.
     *
     * @param source   : ByteBuffer of source data.
     * @param max_time : time interval (ms) to slice the source buffer.
     * @param min_time : minimum size (ms) of the output buffer.
     * @param format   : audio format of source data
     * @param freq     : sample frequency of source data (in Hz)
     *
     * @return ByteBuffer of data copied from "source". Note: source position
     *         will be incremented by the number of bytes copied.
     */
    private static ByteBuffer getSubBuffer(ByteBuffer source, int max_time, int min_time, int format, int freq) {
        int time_size = calcTimeIndex(format, freq, max_time);
        int bufcount;
        int frameSize = frameSize(format);
        ByteBuffer retbuf;
        byte[] retbytes = new byte[source.remaining() + 1];

        log.debug("Creating sub buffer.  interval = {} freq = {} time_size = {} sample size= {}", max_time, freq, time_size, frameSize(format));
        log.debug("\tBefore: Source = {}", source);

        if (time_size < source.remaining()) {
            log.debug("Extracting slice.  Remaining = {}", source.remaining());
            // Enough bytes remaining to pull out a chunk.
            // First, copy over time_size bytes
            source.get(retbytes, 0, time_size);
            bufcount = time_size;
            // Now, find the zero crossing and add bytes up to it
            // Loop until we run out of samples or find a zero crossing
            while ((!isZeroCross(Arrays.copyOfRange(retbytes, bufcount - 6, bufcount), 6, format, source.order())) && (source.remaining() >= frameSize)) {
                source.get(retbytes, bufcount, frameSize);
                bufcount += frameSize;
            }
        } else {
            log.debug("Not enough bytes.  Copying remaining bytes = {}", source.remaining());
            // Not enough bytes remaning to pull out a chunk. Just copy/return the rest of the buffer.
            bufcount = source.remaining();
            source.get(retbytes, 0, bufcount);
        }
        // Now create the ByteBuffer for return... IF there's enough bytes to mess with.  If the size of the array
        // is smaller than the specified minimum time interval, return null.
        if (bufcount > calcTimeIndex(format, freq, min_time)) {
            retbuf = ByteBuffer.allocate(bufcount);
            retbuf.order(source.order()); // set new buffer's byte order to match source buffer.
            retbuf.put(retbytes, 0, bufcount);
            log.debug("\tAfter: source= {}bufcount={} retbuf= {}", source, bufcount, retbuf);
        } else {
            log.debug("Remaining bytes less than minimum time interval.  Discarding.");
            return null;
        }
        return retbuf;
    }

    private static final Logger log = LoggerFactory.getLogger(AudioUtil.class);

}