av/cmd/audio-netsender/main.go

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/*
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/logger"
"bitbucket.org/ausocean/utils/ring"
)
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
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dev *yalsa.Device // audio input device
pb pcm.Buffer // Buffer to contain the direct audio from ALSA.
rb *ring.Buffer // Ring buffer to contain processed audio ready to be read.
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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 *logger.Logger
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)
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log = logger.New(int8(logLevel), &logSender.LogRoller, true)
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()
if err != nil {
log.Log(logger.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.Log(logger.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.Log(logger.Error, err.Error())
}
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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),
0,
)
rbLen := rbDuration / ac.period
ac.rb = ring.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.Log(logger.Debug, "params changed")
}
log.Log(logger.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.Log(logger.Debug, "closing", "source", ac.source)
ac.dev.Close()
ac.dev = nil
}
log.Log(logger.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.Log(logger.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.Log(logger.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.Log(logger.Debug, "sample rate set", "rate", r)
break
}
}
}
// 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)
if err != nil {
return err
}
log.Log(logger.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.Log(logger.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.
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// 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.Log(logger.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.Log(logger.Debug, "device.Read failed", "error", err.Error())
ac.mu.Lock()
err = ac.open() // re-open
if err != nil {
log.Log(logger.Fatal, "yalsa.open failed", "error", err.Error())
}
ac.mu.Unlock()
continue
}
toWrite := ac.formatBuffer()
log.Log(logger.Debug, "audio format conversion has been performed where needed")
var n int
n, err = ac.rb.Write(toWrite.Data)
switch err {
case nil:
log.Log(logger.Debug, "wrote audio to ringbuffer", "length", n)
case ring.ErrDropped:
log.Log(logger.Warning, "dropped audio")
default:
log.Log(logger.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.Log(logger.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.rb, buf)
if err != nil {
return
}
if n == 0 {
goto sleep
}
if n != len(buf) {
log.Log(logger.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.Log(logger.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.Log(logger.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.Log(logger.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.Log(logger.Fatal, "mp not an integer")
}
}
if ac.vs != ac.ns.VarSum() {
vars, err := ac.ns.Vars()
if err != nil {
log.Log(logger.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 *ring.Buffer, buf []byte) (int, error) {
chunk, err := rb.Next(rbNextTimeout)
switch err {
case nil:
// Do nothing.
case ring.ErrTimeout:
return 0, nil
case io.EOF:
log.Log(logger.Error, "unexpected EOF from ring.Next")
return 0, io.ErrUnexpectedEOF
default:
log.Log(logger.Error, "unexpected error from ring.Next", "error", err.Error())
return 0, err
}
n, err := io.ReadFull(rb, buf[:chunk.Len()])
if err != nil {
log.Log(logger.Error, "unexpected error from ring.Read", "error", err.Error())
return n, err
}
log.Log(logger.Debug, "read audio from ringbuffer", "length", n)
return n, nil
}
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// 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.
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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.Log(logger.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.Log(logger.Warning, "rate conversion failed, audio has remained original rate", "error", err.Error())
} else {
formatted.Format.Rate = uint(wantRate)
}
}
return formatted
}