revid: started modifying audio-netsender to be a general audio input

revid/audio.go

@@ -1,46 +1,7 @@
-/*
+package revid
-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.
-*/
-
-// 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"
@@ -51,16 +12,12 @@ import (
 	"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/logger"
 	"bitbucket.org/ausocean/utils/ring"
 )
 
 const (
-	progName         = "audio-netsender"
+	defaultRate     = 48000
-	logPath          = "/var/log/netsender"
-	retryPeriod      = 5 * time.Second
-	defaultFrameRate = 48000
 	defaultPeriod   = 5 // seconds
 	defaultChannels = 2
 	defaultBits     = 16
@@ -69,19 +26,18 @@ const (
 	rbNextTimeout   = 100 * time.Millisecond
 )
 
-// audioClient holds everything we need to know about the client.
+// audioInput holds everything we need to know about the audio input stream.
 // 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 {
+type audioInput struct {
-	mu sync.Mutex // mu protects the audioClient.
+	mu sync.Mutex // mu protects the audioInput.
 
 	parameters
 
 	// internals
 	dev *alsa.Device // audio input device
 	ab  alsa.Buffer  // ALSA's buffer
-	rb  *ring.Buffer      // our buffer
+	rb  *ring.Buffer // our buffer //TODO: change this to output stream, doesn't have to be ring buffer
-	ns  *netsender.Sender // our NetSender
 	vs  int          // our "var sum" to track var changes
 }
 
@@ -94,32 +50,8 @@ type parameters struct {
 	bits     int    // sample bit size, 16 by default
 }
 
-var log *logger.Logger
+func NewAudioInput() {
-
+	var ac audioInput
-func main() {
-	var logLevel int
-	flag.IntVar(&logLevel, "LogLevel", int(logger.Debug), "Specifies log level")
-	flag.Parse()
-
-	validLogLevel := true
-	if logLevel < int(logger.Debug) || logLevel > int(logger.Fatal) {
-		logLevel = int(logger.Info)
-		validLogLevel = false
-	}
-
-	logSender := smartlogger.New(logPath)
-	log = logger.New(int8(logLevel), &logSender.LogRoller)
-	log.Log(logger.Info, "log-netsender: Logger Initialized")
-	if !validLogLevel {
-		log.Log(logger.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.Log(logger.Fatal, "netsender.Init failed", "error", err.Error())
-	}
 
 	// Get audio params and store the current var sum.
 	vars, err := ac.ns.Vars()
@@ -144,11 +76,13 @@ func main() {
 	go ac.input()
 
 	ac.output()
+
+	return stream
 }
 
 // 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.
+// See audioInput for a description of the params and their limits.
-func (ac *audioClient) params(vars map[string]string) bool {
+func (ac *audioInput) 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
@@ -166,7 +100,7 @@ func (ac *audioClient) params(vars map[string]string) bool {
 	}
 	val, err := strconv.Atoi(vars["rate"])
 	if err != nil {
-		val = defaultFrameRate
+		val = defaultRate
 	}
 	if p.rate != val {
 		p.rate = val
@@ -209,7 +143,7 @@ func (ac *audioClient) params(vars map[string]string) bool {
 
 // 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 {
+func (ac *audioInput) open() error {
 	if ac.dev != nil {
 		log.Log(logger.Debug, "Closing", "source", ac.source)
 		ac.dev.Close()
@@ -279,11 +213,11 @@ func (ac *audioClient) open() error {
 	// If no easily divisible rate is found, then use the default rate.
 	if !foundRate {
 		log.Log(logger.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)
+		_, err = ac.dev.NegotiateRate(defaultRate)
 		if err != nil {
 			return err
 		}
-		log.Log(logger.Debug, "Sample rate set", "rate", defaultFrameRate)
+		log.Log(logger.Debug, "Sample rate set", "rate", defaultRate)
 	}
 
 	var fmt alsa.FormatType
@@ -318,7 +252,7 @@ func (ac *audioClient) open() 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 ALSA buffer, i.e.,  _while_ it is recording to avoid any gaps.
-func (ac *audioClient) input() {
+func (ac *audioInput) input() {
 	for {
 		ac.mu.Lock()
 		mode := ac.mode
@@ -369,7 +303,7 @@ func (ac *audioClient) input() {
 // 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() {
+func (ac *audioInput) output() {
 	// Calculate the size of the output data based on wanted channels and rate.
 	outLen := (((len(ac.ab.Data) / ac.ab.Format.Channels) * ac.channels) / ac.ab.Format.Rate) * ac.rate
 	buf := make([]byte, outLen)
@@ -510,7 +444,7 @@ func read(rb *ring.Buffer, buf []byte) (int, error) {
 
 // formatBuffer returns an ALSA buffer that has the recording data from the ac's original ALSA buffer but stored
 // in the desired format specified by the ac's parameters.
-func (ac *audioClient) formatBuffer() alsa.Buffer {
+func (ac *audioInput) formatBuffer() alsa.Buffer {
 	var err error
 	ac.mu.Lock()
 	wantChannels := ac.channels