/*
NAME
  audio-netsender - NetSender client for sending audio to NetReceiver

AUTHORS
  Alan Noble <alan@ausocean.org>
  Trek Hopton <trek@ausocean.org>

ACKNOWLEDGEMENTS
  A special thanks to Joel Jensen for his Go ALSA package.

LICENSE
  audio-netsender is Copyright (C) 2018 the Australian Ocean Lab (AusOcean).

  It is free software: you can redistribute it and/or modify them under
  the terms of the GNU General Public License as published by the
  Free Software Foundation, either version 3 of the License, or (at your
  option) any later version.

  It 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.

  You should have received a copy of the GNU General Public License
  along with https://bitbucket.org/ausocean/iot/src/master/gpl.txt.
  If not, see http://www.gnu.org/licenses.
*/

// Package audio-netsender is a NetSender client for sending audio to
// NetReceiver. Audio is captured by means of an ALSA recording
// device, specified by the NetReceiver "source" variable. It sent via
// HTTP to NetReceiver in raw audio form, i.e., as PCM data, where it
// is stored as BinaryData objects. Other NetReceiver variables are
// "rate", "period", "channels" and "bits", for specifiying the frame
// rate (Hz), audio period (seconds), number of channels and sample
// bit size respectively. For a description of NetReceiver see
// http://netreceiver.appspot.com/help.
package main

import (
	"errors"
	"flag"
	"io"
	"strconv"
	"sync"
	"time"

	yalsa "github.com/yobert/alsa"

	"bitbucket.org/ausocean/av/codec/pcm"
	"bitbucket.org/ausocean/iot/pi/netsender"
	"bitbucket.org/ausocean/iot/pi/sds"
	"bitbucket.org/ausocean/iot/pi/smartlogger"
	"bitbucket.org/ausocean/utils/logging"
	"bitbucket.org/ausocean/utils/pool"
)

const (
	progName         = "audio-netsender"
	logPath          = "/var/log/netsender"
	retryPeriod      = 5 * time.Second
	defaultFrameRate = 48000
	defaultPeriod    = 5 // seconds
	defaultChannels  = 2
	defaultBits      = 16
	rbDuration       = 300 // seconds
	rbTimeout        = 100 * time.Millisecond
	rbNextTimeout    = 100 * time.Millisecond
)

// audioClient holds everything we need to know about the client.
// NB: At 44100 Hz frame rate, 2 channels and 16-bit samples, a period of 5 seconds
// results in PCM data chunks of 882000 bytes! A longer period exceeds datastore's 1MB blob limit.
type audioClient struct {
	mu sync.Mutex // mu protects the audioClient.

	parameters

	// internals
	dev *yalsa.Device     // audio input device
	pb  pcm.Buffer        // Buffer to contain the direct audio from ALSA.
	buf *pool.Buffer      // Ring buffer to contain processed audio ready to be read.
	ns  *netsender.Sender // our NetSender
	vs  int               // our "var sum" to track var changes
}

type parameters struct {
	mode     string // operating mode, either "Normal" or "Paused"
	source   string // name of audio source, or empty for the default source
	rate     int    // frame rate in Hz, 44100Hz by default
	period   int    // audio period in seconds, 5s by default
	channels int    // number of audio channels, 1 for mono, 2 for stereo
	bits     int    // sample bit size, 16 by default
}

var log logging.Logger

func main() {
	var logLevel int
	flag.IntVar(&logLevel, "LogLevel", int(logging.Debug), "Specifies log level")
	flag.Parse()

	validLogLevel := true
	if logLevel < int(logging.Debug) || logLevel > int(logging.Fatal) {
		logLevel = int(logging.Info)
		validLogLevel = false
	}

	logSender := smartlogger.New(logPath)
	log = logging.New(int8(logLevel), &logSender.LogRoller, true)
	log.Info("log-netsender: Logger Initialized")
	if !validLogLevel {
		log.Error("invalid log level was defaulted to Info")
	}

	var ac audioClient
	var err error
	ac.ns, err = netsender.New(log, nil, sds.ReadSystem, nil)
	if err != nil {
		log.Fatal("netsender.Init failed", "error", err.Error())
	}

	// Get audio params and store the current var sum.
	vars, err := ac.ns.Vars()
	if err != nil {
		log.Warning("netsender.Vars failed; using defaults", "error", err.Error())
	}
	ac.params(vars)
	ac.vs = ac.ns.VarSum()

	// Open the requested audio device.
	err = ac.open()
	if err != nil {
		log.Fatal("yalsa.open failed", "error", err.Error())
	}

	// Capture audio in periods of ac.period seconds, and buffer rbDuration seconds in total.
	ab := ac.dev.NewBufferDuration(time.Second * time.Duration(ac.period))
	sf, err := pcm.SFFromString(ab.Format.SampleFormat.String())
	if err != nil {
		log.Error(err.Error())
	}
	cf := pcm.BufferFormat{
		SFormat:  sf,
		Channels: uint(ab.Format.Channels),
		Rate:     uint(ab.Format.Rate),
	}
	ac.pb = pcm.Buffer{
		Format: cf,
		Data:   ab.Data,
	}

	cs := pcm.DataSize(
		uint(ac.parameters.rate),
		uint(ac.parameters.channels),
		uint(ac.parameters.bits),
		float64(ac.parameters.period),
	)
	rbLen := rbDuration / ac.period
	ac.buf = pool.NewBuffer(int(rbLen), cs, rbTimeout)

	go ac.input()

	ac.output()
}

// params extracts audio params from corresponding NetReceiver vars and returns true if anything has changed.
// See audioClient for a description of the params and their limits.
func (ac *audioClient) params(vars map[string]string) bool {
	// We are the only writers to this field
	// so we don't need to lock here.
	p := ac.parameters
	changed := false

	mode := vars["mode"]
	if p.mode != mode {
		p.mode = mode
		changed = true
	}
	source := vars["source"]
	if p.source != source {
		p.source = source
		changed = true
	}
	val, err := strconv.Atoi(vars["rate"])
	if err != nil {
		val = defaultFrameRate
	}
	if p.rate != val {
		p.rate = val
		changed = true
	}
	val, err = strconv.Atoi(vars["period"])
	if err != nil || val < 1 || 5 < val {
		val = defaultPeriod
	}
	if p.period != val {
		p.period = val
		changed = true
	}
	val, err = strconv.Atoi(vars["channels"])
	if err != nil || (val != 1 && val != 2) {
		val = defaultChannels
	}
	if p.channels != val {
		p.channels = val
		changed = true
	}
	val, err = strconv.Atoi(vars["bits"])
	if err != nil || (val != 16 && val != 32) {
		val = defaultBits
	}
	if p.bits != val {
		p.bits = val
		changed = true
	}

	if changed {
		ac.mu.Lock()
		ac.parameters = p
		ac.mu.Unlock()
		log.Debug("params changed")
	}
	log.Debug("parameters", "mode", p.mode, "source", p.source, "rate", p.rate, "period", p.period, "channels", p.channels, "bits", p.bits)
	return changed
}

// open or re-open the recording device with the given name and prepare it to record.
// If name is empty, the first recording device is used.
func (ac *audioClient) open() error {
	if ac.dev != nil {
		log.Debug("closing", "source", ac.source)
		ac.dev.Close()
		ac.dev = nil
	}
	log.Debug("opening", "source", ac.source)

	cards, err := yalsa.OpenCards()
	if err != nil {
		return err
	}
	defer yalsa.CloseCards(cards)

	for _, card := range cards {
		devices, err := card.Devices()
		if err != nil {
			return err
		}
		for _, dev := range devices {
			if dev.Type != yalsa.PCM || !dev.Record {
				continue
			}
			if dev.Title == ac.source || ac.source == "" {
				ac.dev = dev
				break
			}
		}
	}

	if ac.dev == nil {
		return errors.New("No audio source found")
	}
	log.Debug("found audio source", "source", ac.dev.Title)

	// ToDo: time out if Open takes too long.
	err = ac.dev.Open()
	if err != nil {
		return err
	}
	log.Debug("opened audio source")

	_, err = ac.dev.NegotiateChannels(defaultChannels)
	if err != nil {
		return err
	}

	// Try to negotiate a rate to record in that is divisible by the wanted rate
	// so that it can be easily downsampled to the wanted rate.
	// Note: if a card thinks it can record at a rate but can't actually, this can cause a failure. Eg.
	// the audioinjector is supposed to record at 8000Hz and 16000Hz but it can't due to a firmware issue,
	// to fix this 8000 and 16000 must be removed from this slice.
	rates := [8]int{8000, 16000, 32000, 44100, 48000, 88200, 96000, 192000}
	foundRate := false
	for _, r := range rates {
		if r < ac.rate {
			continue
		}
		if r%ac.rate == 0 {
			_, err = ac.dev.NegotiateRate(r)
			if err == nil {
				foundRate = true
				log.Debug("sample rate set", "rate", r)
				break
			}
		}
	}

	// If no easily divisible rate is found, then use the default rate.
	if !foundRate {
		log.Warning("no available device sample-rates are divisible by the requested rate. Default rate will be used. Resampling may fail.", "rateRequested", ac.rate)
		_, err = ac.dev.NegotiateRate(defaultFrameRate)
		if err != nil {
			return err
		}
		log.Debug("sample rate set", "rate", defaultFrameRate)
	}

	var fmt yalsa.FormatType
	switch ac.bits {
	case 16:
		fmt = yalsa.S16_LE
	case 32:
		fmt = yalsa.S32_LE
	default:
		return errors.New("unsupported sample bits")
	}
	_, err = ac.dev.NegotiateFormat(fmt)
	if err != nil {
		return err
	}

	// Either 8192 or 16384 bytes is a reasonable ALSA buffer size.
	_, err = ac.dev.NegotiateBufferSize(8192, 16384)
	if err != nil {
		return err
	}

	if err = ac.dev.Prepare(); err != nil {
		return err
	}
	log.Debug("successfully negotiated ALSA params")
	return nil
}

// input continously records audio and writes it to the ringbuffer.
// Re-opens the device and tries again if ASLA returns an error.
// Spends a lot of time sleeping in Paused mode.
// ToDo: Currently, reading audio and writing to the ringbuffer are synchronous.
// Need a way to asynchronously read from the buf, i.e.,  _while_ it is recording to avoid any gaps.
func (ac *audioClient) input() {
	for {
		ac.mu.Lock()
		mode := ac.mode
		ac.mu.Unlock()
		if mode == "Paused" {
			time.Sleep(time.Duration(ac.period) * time.Second)
			continue
		}
		log.Debug("recording audio for period", "seconds", ac.period)
		ac.mu.Lock()
		err := ac.dev.Read(ac.pb.Data)
		ac.mu.Unlock()
		if err != nil {
			log.Debug("device.Read failed", "error", err.Error())
			ac.mu.Lock()
			err = ac.open() // re-open
			if err != nil {
				log.Fatal("yalsa.open failed", "error", err.Error())
			}
			ac.mu.Unlock()
			continue
		}

		toWrite := ac.formatBuffer()

		log.Debug("audio format conversion has been performed where needed")

		var n int
		n, err = ac.buf.Write(toWrite.Data)
		switch err {
		case nil:
			log.Debug("wrote audio to ringbuffer", "length", n)
		case pool.ErrDropped:
			log.Warning("dropped audio")
		default:
			log.Error("unexpected ringbuffer error", "error", err.Error())
			return
		}
	}
}

// output continously reads audio from the ringbuffer and sends it to NetReceiver via poll requests.
// When "B0" is configured as one of the NetReceiver inputs, audio data is posted as "B0".
// When "B0" is not an input, the poll request happens without any audio data
// (although other inputs may still be present via URL parameters).
// When paused, polling continues but without sending audio (B0) data.
// Sending is throttled so as to complete one pass of this loop approximately every audio period,
// since cycling more frequently is pointless.
// Finally while audio data is sent every audio period, other data is reported only every monitor period.
// This function also handles NetReceiver configuration requests and updating of NetReceiver vars.
func (ac *audioClient) output() {
	// Calculate the size of the output data based on wanted channels and rate.
	outLen := (((len(ac.pb.Data) / int(ac.pb.Format.Channels)) * ac.channels) / int(ac.pb.Format.Rate)) * ac.rate
	buf := make([]byte, outLen)

	mime := "audio/x-wav;codec=pcm;rate=" + strconv.Itoa(ac.rate) + ";channels=" + strconv.Itoa(ac.channels) + ";bits=" + strconv.Itoa(ac.bits)
	ip := ac.ns.Param("ip")
	mp, err := strconv.Atoi(ac.ns.Param("mp"))
	if err != nil {
		log.Fatal("mp not an integer")
	}

	report := true         // Report non-audio data.
	reported := time.Now() // When we last did so.

	for {
		var rc int
		start := time.Now()
		audio := false
		var pins []netsender.Pin

		if ac.mode == "Paused" {

			// Only send X data when paused (if any).
			if report {
				pins = netsender.MakePins(ip, "X")
			}
		} else {
			n, err := read(ac.buf, buf)
			if err != nil {
				return
			}
			if n == 0 {
				goto sleep
			}
			if n != len(buf) {
				log.Error("unexpected length from read", "length", n)
				return
			}
			if report {
				pins = netsender.MakePins(ip, "")
			} else {
				pins = netsender.MakePins(ip, "B")
			}
			for i, pin := range pins {
				if pin.Name == "B0" {
					audio = true
					pins[i].Value = n
					pins[i].Data = buf
					pins[i].MimeType = mime
				}
			}
		}

		if !(report || audio) {
			goto sleep // nothing to do
		}

		// Populate X pins, if any.
		for i, pin := range pins {
			if pin.Name[0] == 'X' {
				err := sds.ReadSystem(&pins[i])
				if err != nil {
					log.Warning("sds.ReadSystem failed", "error", err.Error())
					// Pin.Value defaults to -1 upon error, so OK to continue.
				}
			}
		}
		_, rc, err = ac.ns.Send(netsender.RequestPoll, pins)
		if err != nil {
			log.Debug("netsender.Send failed", "error", err.Error())
			goto sleep
		}
		if report {
			reported = start
			report = false
		}
		if rc == netsender.ResponseUpdate {
			_, err = ac.ns.Config()
			if err != nil {
				log.Warning("netsender.Config failed", "error", err.Error())
				goto sleep
			}
			ip = ac.ns.Param("ip")
			mp, err = strconv.Atoi(ac.ns.Param("mp"))
			if err != nil {
				log.Fatal("mp not an integer")
			}
		}

		if ac.vs != ac.ns.VarSum() {
			vars, err := ac.ns.Vars()
			if err != nil {
				log.Error("netsender.Vars failed", "error", err.Error())
				goto sleep
			}
			ac.params(vars) // ToDo: re-open device if audio params have changed.
			ac.vs = ac.ns.VarSum()
		}

	sleep:
		pause := ac.period*1000 - int(time.Since(start).Seconds()*1000)
		if pause > 0 {
			time.Sleep(time.Duration(pause) * time.Millisecond)
		}
		if time.Since(reported).Seconds() >= float64(mp) {
			report = true
		}

	}
}

// read reads a full PCM chunk from the ringbuffer, returning the number of bytes read upon success.
// Any errors returned are unexpected and should be considered fatal.
func read(rb *pool.Buffer, buf []byte) (int, error) {
	chunk, err := rb.Next(rbNextTimeout)
	switch err {
	case nil:
		// Do nothing.
	case pool.ErrTimeout:
		return 0, nil
	case io.EOF:
		log.Error("unexpected EOF from pool.Next")
		return 0, io.ErrUnexpectedEOF
	default:
		log.Error("unexpected error from pool.Next", "error", err.Error())
		return 0, err
	}

	n, err := io.ReadFull(rb, buf[:chunk.Len()])
	if err != nil {
		log.Error("unexpected error from pool.Read", "error", err.Error())
		return n, err
	}

	log.Debug("read audio from ringbuffer", "length", n)
	return n, nil
}

// formatBuffer returns a Buffer that has the recording data from the ac's original Buffer but stored
// in the desired format specified by the ac's parameters.
func (ac *audioClient) formatBuffer() pcm.Buffer {
	var err error
	ac.mu.Lock()
	wantChannels := ac.channels
	wantRate := ac.rate
	ac.mu.Unlock()

	// If nothing needs to be changed, return the original.
	if int(ac.pb.Format.Channels) == wantChannels && int(ac.pb.Format.Rate) == wantRate {
		return ac.pb
	}

	formatted := pcm.Buffer{Format: ac.pb.Format}
	bufCopied := false
	if int(ac.pb.Format.Channels) != wantChannels {

		// Convert channels.
		if ac.pb.Format.Channels == 2 && wantChannels == 1 {
			if formatted, err = pcm.StereoToMono(ac.pb); err != nil {
				log.Warning("channel conversion failed, audio has remained stereo", "error", err.Error())
			} else {
				formatted.Format.Channels = 1
			}
			bufCopied = true
		}
	}

	if int(ac.pb.Format.Rate) != wantRate {

		// Convert rate.
		if bufCopied {
			formatted, err = pcm.Resample(formatted, uint(wantRate))
		} else {
			formatted, err = pcm.Resample(ac.pb, uint(wantRate))
		}
		if err != nil {
			log.Warning("rate conversion failed, audio has remained original rate", "error", err.Error())
		} else {
			formatted.Format.Rate = uint(wantRate)
		}
	}
	return formatted
}