feat: Implement TeamSpeak 3 client with connection management, event handling, and audio support.
This commit is contained in:
Jose Luis Montañes Ojados
261
pkg/audio/capture.go
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261
pkg/audio/capture.go
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@@ -0,0 +1,261 @@
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package audio
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import (
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"encoding/binary"
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"fmt"
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"sync"
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"time"
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"unsafe"
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"github.com/go-ole/go-ole"
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"github.com/moutend/go-wca/pkg/wca"
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)
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const captureFrameSamples = 960 // 20ms at 48kHz
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// Capturer handles WASAPI audio capture from microphone
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type Capturer struct {
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client *wca.IAudioClient
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captureClient *wca.IAudioCaptureClient
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waveFormat *wca.WAVEFORMATEX
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bufferSize uint32
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running bool
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mu sync.Mutex
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stopChan chan struct{}
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// Callback for captured audio (called with 960-sample frames)
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onAudio func(samples []int16)
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// Sample accumulation buffer
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sampleBuffer []int16
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bufferMu sync.Mutex
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// Current audio level (0-100)
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currentLevel int
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levelMu sync.RWMutex
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}
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// NewCapturer creates a new WASAPI audio capturer
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func NewCapturer() (*Capturer, error) {
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// Initialize COM (may already be initialized)
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ole.CoInitializeEx(0, ole.COINIT_APARTMENTTHREADED)
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// Get default capture endpoint (microphone)
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var deviceEnumerator *wca.IMMDeviceEnumerator
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if err := wca.CoCreateInstance(
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wca.CLSID_MMDeviceEnumerator,
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0,
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wca.CLSCTX_ALL,
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wca.IID_IMMDeviceEnumerator,
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&deviceEnumerator,
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); err != nil {
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return nil, fmt.Errorf("failed to create device enumerator: %w", err)
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}
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defer deviceEnumerator.Release()
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var device *wca.IMMDevice
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if err := deviceEnumerator.GetDefaultAudioEndpoint(wca.ECapture, wca.EConsole, &device); err != nil {
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return nil, fmt.Errorf("failed to get default capture device: %w", err)
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}
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defer device.Release()
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// Activate audio client
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var audioClient *wca.IAudioClient
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if err := device.Activate(wca.IID_IAudioClient, wca.CLSCTX_ALL, nil, &audioClient); err != nil {
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return nil, fmt.Errorf("failed to activate audio client: %w", err)
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}
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// Set up format for 48kHz mono 16-bit (TeamSpeak format)
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waveFormat := &wca.WAVEFORMATEX{
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WFormatTag: wca.WAVE_FORMAT_PCM,
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NChannels: 1,
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NSamplesPerSec: 48000,
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WBitsPerSample: 16,
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NBlockAlign: 2,
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NAvgBytesPerSec: 96000,
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CbSize: 0,
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}
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// Initialize in shared mode - 100ms buffer
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duration := wca.REFERENCE_TIME(100 * 10000) // 100ms in 100-nanosecond units
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if err := audioClient.Initialize(
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wca.AUDCLNT_SHAREMODE_SHARED,
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wca.AUDCLNT_STREAMFLAGS_AUTOCONVERTPCM|wca.AUDCLNT_STREAMFLAGS_SRC_DEFAULT_QUALITY,
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duration,
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0,
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waveFormat,
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nil,
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); err != nil {
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audioClient.Release()
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return nil, fmt.Errorf("failed to initialize audio client: %w", err)
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}
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// Get buffer size
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var bufferSize uint32
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if err := audioClient.GetBufferSize(&bufferSize); err != nil {
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audioClient.Release()
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return nil, fmt.Errorf("failed to get buffer size: %w", err)
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}
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// Get capture client
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var captureClient *wca.IAudioCaptureClient
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if err := audioClient.GetService(wca.IID_IAudioCaptureClient, &captureClient); err != nil {
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audioClient.Release()
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return nil, fmt.Errorf("failed to get capture client: %w", err)
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}
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return &Capturer{
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client: audioClient,
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captureClient: captureClient,
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waveFormat: waveFormat,
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bufferSize: bufferSize,
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stopChan: make(chan struct{}),
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sampleBuffer: make([]int16, 0, captureFrameSamples*50), // ~1 second buffer
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}, nil
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}
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// SetCallback sets the callback for captured audio (receives 960-sample frames)
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func (c *Capturer) SetCallback(fn func(samples []int16)) {
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c.mu.Lock()
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c.onAudio = fn
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c.mu.Unlock()
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}
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// Start begins audio capture
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func (c *Capturer) Start() error {
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c.mu.Lock()
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if c.running {
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c.mu.Unlock()
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return nil
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}
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c.running = true
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c.stopChan = make(chan struct{}) // Recreate channel for each start
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c.mu.Unlock()
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if err := c.client.Start(); err != nil {
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return fmt.Errorf("failed to start audio client: %w", err)
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}
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go c.captureLoop()
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return nil
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}
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// Stop stops audio capture
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func (c *Capturer) Stop() {
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c.mu.Lock()
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if !c.running {
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c.mu.Unlock()
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return
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}
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c.running = false
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c.mu.Unlock()
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close(c.stopChan)
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c.client.Stop()
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}
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// Close releases all resources
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func (c *Capturer) Close() {
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c.Stop()
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if c.captureClient != nil {
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c.captureClient.Release()
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}
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if c.client != nil {
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c.client.Release()
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}
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}
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// GetLevel returns the current audio input level (0-100)
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func (c *Capturer) GetLevel() int {
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c.levelMu.RLock()
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defer c.levelMu.RUnlock()
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return c.currentLevel
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}
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// IsRunning returns whether capture is active
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func (c *Capturer) IsRunning() bool {
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c.mu.Lock()
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defer c.mu.Unlock()
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return c.running
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}
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func (c *Capturer) captureLoop() {
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ticker := time.NewTicker(10 * time.Millisecond) // Check more often than 20ms
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defer ticker.Stop()
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for {
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select {
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case <-c.stopChan:
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return
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case <-ticker.C:
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c.readFromBuffer()
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}
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}
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}
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func (c *Capturer) readFromBuffer() {
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// Read all available packets
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for {
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var packetLength uint32
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if err := c.captureClient.GetNextPacketSize(&packetLength); err != nil {
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return
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}
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if packetLength == 0 {
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break
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}
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var buffer *byte
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var numFrames uint32
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var flags uint32
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if err := c.captureClient.GetBuffer(&buffer, &numFrames, &flags, nil, nil); err != nil {
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return
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}
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if numFrames == 0 {
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c.captureClient.ReleaseBuffer(numFrames)
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continue
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}
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samples := make([]int16, numFrames)
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bufSlice := unsafe.Slice(buffer, numFrames*2)
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for i := uint32(0); i < numFrames; i++ {
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samples[i] = int16(binary.LittleEndian.Uint16(bufSlice[i*2:]))
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}
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c.captureClient.ReleaseBuffer(numFrames)
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// Skip silent buffers
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if flags&wca.AUDCLNT_BUFFERFLAGS_SILENT != 0 {
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continue
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}
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// Add to sample buffer
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c.bufferMu.Lock()
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c.sampleBuffer = append(c.sampleBuffer, samples...)
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// Calculate level from latest samples
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level := CalculateRMSLevel(samples)
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c.levelMu.Lock()
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c.currentLevel = level
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c.levelMu.Unlock()
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// Send complete 960-sample frames
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for len(c.sampleBuffer) >= captureFrameSamples {
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frame := make([]int16, captureFrameSamples)
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copy(frame, c.sampleBuffer[:captureFrameSamples])
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c.sampleBuffer = c.sampleBuffer[captureFrameSamples:]
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// Call callback with the frame
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c.mu.Lock()
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callback := c.onAudio
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c.mu.Unlock()
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if callback != nil {
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callback(frame)
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}
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}
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c.bufferMu.Unlock()
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}
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}
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