revid: added concurrency support to start and stop

revid/audio-input.go

@@ -41,9 +41,9 @@ import (
 
 const (
 	logPath       = "/var/log/netsender"
-	rbDuration    = 300 // seconds
 	rbTimeout     = 100 * time.Millisecond
 	rbNextTimeout = 100 * time.Millisecond
+	rbLen         = 200
 )
 
 const (
@@ -77,7 +77,7 @@ type AudioConfig struct {
 	SampleRate int
 	Channels   int
 	BitDepth   int
-	RecPeriod  int
+	RecPeriod  float64
 	Codec      uint8
 }
 
@@ -109,7 +109,6 @@ func NewAudioDevice(cfg *AudioConfig) *audioDevice {
 	// Setup ring buffer to capture audio in periods of a.RecPeriod seconds, and buffer rbDuration seconds in total.
 	a.ab = a.dev.NewBufferDuration(time.Second * time.Duration(a.RecPeriod))
 	a.chunkSize = (((len(a.ab.Data) / a.dev.BufferFormat().Channels) * a.Channels) / a.dev.BufferFormat().Rate) * a.SampleRate
-	rbLen := rbDuration / a.RecPeriod
 	a.rb = ring.NewBuffer(rbLen, a.chunkSize, rbTimeout)
 
 	a.mode = paused
@@ -120,13 +119,11 @@ func NewAudioDevice(cfg *AudioConfig) *audioDevice {
 // Start will start recording audio and writing to the output.
 func (a *audioDevice) Start() {
 	a.mu.Lock()
-	mode := a.mode
+	switch a.mode {
-	a.mu.Unlock()
-	switch mode {
 	case paused:
 		// Start Recording
 		go a.input()
-		mode = running
+		a.mode = running
 	case stopped:
 		// Open the audio device and start recording.
 		err := a.open()
@@ -134,25 +131,23 @@ func (a *audioDevice) Start() {
 			log.Log(logger.Fatal, "alsa.open failed", "error", err.Error())
 		}
 		go a.input()
-		mode = running
+		a.mode = running
 	case running:
 		return
 	}
-	a.mu.Lock()
-	a.mode = mode
 	a.mu.Unlock()
 }
 
 // Stop will stop recording audio and close the device
 func (a *audioDevice) Stop() {
 	a.mu.Lock()
-	a.mode = stopped
-	a.mu.Unlock()
 	if a.dev != nil {
 		log.Log(logger.Debug, "Closing", "source", a.source)
 		a.dev.Close()
 		a.dev = nil
 	}
+	a.mode = stopped
+	a.mu.Unlock()
 
 }
 
@@ -273,17 +268,16 @@ func (a *audioDevice) open() error {
 func (a *audioDevice) input() {
 	for {
 		a.mu.Lock()
-		mode := a.mode
+		switch a.mode {
-		a.mu.Unlock()
-		switch mode {
 		case paused:
+			a.mu.Unlock()
 			time.Sleep(time.Duration(a.RecPeriod) * time.Second)
 			continue
 		case stopped:
-			break
+			a.mu.Unlock()
+			return
 		}
 		log.Log(logger.Debug, "Recording audio for period", "seconds", a.RecPeriod)
-		a.mu.Lock()
 		err := a.dev.Read(a.ab.Data)
 		a.mu.Unlock()
 		if err != nil {
@@ -318,6 +312,16 @@ func (a *audioDevice) input() {
 // 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 (a *audioDevice) Read(p []byte) (n int, err error) {
+	a.mu.Lock()
+	if a.rb == nil {
+		fmt.Println("READ: RB IS NIL")
+	}
+	switch a.mode {
+	case paused:
+		return 0, nil
+	case stopped:
+		return 0, nil
+	}
 	chunk, err := a.rb.Next(rbNextTimeout)
 	switch err {
 	case nil:
@@ -336,8 +340,8 @@ func (a *audioDevice) Read(p []byte) (n int, err error) {
 		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)
+	a.mu.Unlock()
 	return n, nil
 }
 

revid/audio-input_test.go

@@ -1,25 +1,20 @@
 package revid
 
 import (
+	"bytes"
+	"errors"
 	"testing"
+	"time"
 
+	"bitbucket.org/ausocean/av/codec/lex"
 	"github.com/yobert/alsa"
 )
 
-func TestAudioDevice(t *testing.T) {
+// Check that a device exists with the given config parameters.
-	// We want to open a device with a standard configuration.
+func checkDevice(ac *AudioConfig) error {
-	ac := &AudioConfig{
-		SampleRate: 8000,
-		Channels:   1,
-		RecPeriod:  1,
-		BitDepth:   16,
-		Codec:      ADPCM,
-	}
-
-	// Check that a device exists with the desired parameters.
 	cards, err := alsa.OpenCards()
 	if err != nil {
-		t.Skip("skipping, no audio card found")
+		return errors.New("no audio cards found")
 	}
 	defer alsa.CloseCards(cards)
 	var testDev *alsa.Device
@@ -37,11 +32,15 @@ func TestAudioDevice(t *testing.T) {
 		}
 	}
 	if testDev == nil {
-		t.Skip("skipping, no suitable audio device found")
+		return errors.New("no suitable device found")
+	}
+	err = testDev.Open()
+	if err != nil {
+		return err
 	}
 	_, err = testDev.NegotiateChannels(2)
 	if err != nil {
-		t.Skip("skipping, no suitable audio device found")
+		return err
 	}
 	foundRate := false
 	for i := 0; i < len(rates) && !foundRate; i++ {
@@ -58,25 +57,57 @@ func TestAudioDevice(t *testing.T) {
 	if !foundRate {
 		_, err = testDev.NegotiateRate(defaultSampleRate)
 		if err != nil {
-			t.Skip("skipping, no suitable audio device found")
+			return err
 		}
 	}
-	_, err = testDev.NegotiateFormat(alsa.S16_LE)
+	var aFmt alsa.FormatType
+	switch ac.BitDepth {
+	case 16:
+		aFmt = alsa.S16_LE
+	case 32:
+		aFmt = alsa.S32_LE
+	default:
+		return errors.New("unsupported bitdepth")
+	}
+	_, err = testDev.NegotiateFormat(aFmt)
 	if err != nil {
-		t.Skip("skipping, no suitable audio device found")
+		return err
 	}
 	_, err = testDev.NegotiateBufferSize(8192, 16384)
 	if err != nil {
-		t.Skip("skipping, no suitable audio device found")
+		return err
 	}
 	if err = testDev.Prepare(); err != nil {
-		t.Skip("skipping, no suitable audio device found")
+		return err
 	}
+	if testDev != nil {
 		testDev.Close()
-
+	}
-	ai := NewAudioDevice(ac)
+	return nil
-
+}
-	ai.Start()
+
-
+func TestAudio(t *testing.T) {
-	ai.Stop()
+	// We want to open a device with a standard configuration.
+	ac := &AudioConfig{
+		SampleRate: 8000,
+		Channels:   1,
+		RecPeriod:  0.01,
+		BitDepth:   16,
+		Codec:      ADPCM,
+	}
+
+	// Skip if there are no suitable devices to test with.
+	err := checkDevice(ac)
+	if err != nil {
+		t.Error(err)
+	}
+
+	// Create a new audioDevice, start, read/lex, and then stop it.
+	ai := NewAudioDevice(ac)
+	dst := bytes.NewBuffer(make([]byte, 0, ai.ChunkSize()*4))
+	ai.Start()
+	go lex.ADPCM(dst, ai, time.Duration(ac.RecPeriod), ai.ChunkSize())
+	time.Sleep(time.Millisecond * 10)
+	ai.Stop()
+
 }

revid/config.go

@@ -75,7 +75,7 @@ type Config struct {
 
 	// Audio
 	SampleRate int     // Samples a second (Hz).
-	RecPeriod  int // How many seconds to record at a time.
+	RecPeriod  float64 // How many seconds to record at a time.
 	Channels   int     // Number of audio channels, 1 for mono, 2 for stereo.
 	BitDepth   int     // Sample bit depth.
 }
@@ -168,7 +168,7 @@ const (
 	defaultSampleRate      = 48000
 	defaultBitDepth        = 16
 	defaultChannels        = 1
-	defaultRecPeriod       = 1
+	defaultRecPeriod       = 1.0
 )
 
 // Validate checks for any errors in the config fields and defaults settings