Fix audio quality with per-user mixing buffer and prevent TUI layout break on log overflow
This commit is contained in:
Jose Luis Montañes Ojados
@@ -11,7 +11,7 @@ import (
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"github.com/moutend/go-wca/pkg/wca"
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)
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// Player handles WASAPI audio playback
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// Player handles WASAPI audio playback with mixing support
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type Player struct {
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client *wca.IAudioClient
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renderClient *wca.IAudioRenderClient
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@@ -23,24 +23,21 @@ type Player struct {
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running bool
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stopChan chan struct{}
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// Audio buffer - accumulates incoming audio
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audioBuffer []int16
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// User buffers for mixing
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// map[SenderID] -> AudioQueue
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userBuffers map[uint16][]int16
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bufferMu sync.Mutex
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// Frame queue (960 samples = 20ms at 48kHz)
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frameQueue chan []int16
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}
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const frameSamples = 960 // 20ms at 48kHz
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const (
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frameSamples = 960 // 20ms at 48kHz
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)
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// NewPlayer creates a new WASAPI audio player
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func NewPlayer() (*Player, error) {
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// Initialize COM using go-ole
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if err := ole.CoInitializeEx(0, ole.COINIT_APARTMENTTHREADED); err != nil {
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// Ignore if already initialized
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}
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// Initialize COM
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ole.CoInitializeEx(0, ole.COINIT_APARTMENTTHREADED)
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// Get default audio endpoint
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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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@@ -59,13 +56,11 @@ func NewPlayer() (*Player, error) {
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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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@@ -76,8 +71,7 @@ func NewPlayer() (*Player, error) {
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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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duration := wca.REFERENCE_TIME(100 * 10000) // 100ms buffer
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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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@@ -90,14 +84,12 @@ func NewPlayer() (*Player, error) {
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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 render client
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var renderClient *wca.IAudioRenderClient
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if err := audioClient.GetService(wca.IID_IAudioRenderClient, &renderClient); err != nil {
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audioClient.Release()
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@@ -112,8 +104,7 @@ func NewPlayer() (*Player, error) {
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volume: 1.0,
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muted: false,
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stopChan: make(chan struct{}),
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audioBuffer: make([]int16, 0, frameSamples*50), // ~1 second buffer
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frameQueue: make(chan []int16, 100), // ~2 seconds of frames
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userBuffers: make(map[uint16][]int16),
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}, nil
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}
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@@ -125,15 +116,14 @@ func (p *Player) Start() error {
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return nil
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}
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p.running = true
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p.stopChan = make(chan struct{})
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p.mu.Unlock()
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if err := p.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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// Playback loop writes frames from queue to WASAPI
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go p.playbackLoop()
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return nil
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}
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@@ -163,43 +153,25 @@ func (p *Player) Close() {
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ole.CoUninitialize()
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}
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// PlayPCM queues PCM audio for playback
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// Accumulates samples and queues complete 960-sample frames
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func (p *Player) PlayPCM(samples []int16) {
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// PlayPCM adds audio samples to a specific user's buffer
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func (p *Player) PlayPCM(senderID uint16, samples []int16) {
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if p.muted {
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return
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}
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// Apply volume
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adjusted := samples
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if p.volume != 1.0 {
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adjusted = make([]int16, len(samples))
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for i, s := range samples {
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adjusted[i] = int16(float32(s) * p.volume)
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}
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}
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p.bufferMu.Lock()
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p.audioBuffer = append(p.audioBuffer, adjusted...)
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defer p.bufferMu.Unlock()
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// Queue complete 960-sample frames
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for len(p.audioBuffer) >= frameSamples {
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frame := make([]int16, frameSamples)
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copy(frame, p.audioBuffer[:frameSamples])
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p.audioBuffer = p.audioBuffer[frameSamples:]
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// Append to user's specific buffer
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// This ensures sequential playback for the same user
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p.userBuffers[senderID] = append(p.userBuffers[senderID], samples...)
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select {
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case p.frameQueue <- frame:
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default:
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// Queue full, drop oldest frame
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select {
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case <-p.frameQueue:
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default:
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}
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p.frameQueue <- frame
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}
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// Limit buffer size per user to avoid memory leaks if stalled
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if len(p.userBuffers[senderID]) > 48000*2 { // 2 seconds max
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// Drop oldest
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drop := len(p.userBuffers[senderID]) - 48000
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p.userBuffers[senderID] = p.userBuffers[senderID][drop:]
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}
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p.bufferMu.Unlock()
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}
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// SetVolume sets playback volume (0.0 to 1.0)
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@@ -237,7 +209,6 @@ func (p *Player) IsMuted() bool {
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}
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func (p *Player) playbackLoop() {
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// Use 20ms ticker matching TeamSpeak frame rate
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ticker := time.NewTicker(20 * time.Millisecond)
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defer ticker.Stop()
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@@ -252,7 +223,6 @@ func (p *Player) playbackLoop() {
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}
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func (p *Player) writeFrame() {
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// Get current padding (samples already in buffer)
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var padding uint32
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if err := p.client.GetCurrentPadding(&padding); err != nil {
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return
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@@ -260,27 +230,65 @@ func (p *Player) writeFrame() {
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available := p.bufferSize - padding
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if available < frameSamples {
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return // Not enough space for a full frame
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return
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}
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// Try to get a frame from the queue
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select {
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case frame := <-p.frameQueue:
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var buffer *byte
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if err := p.renderClient.GetBuffer(frameSamples, &buffer); err != nil {
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return
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p.bufferMu.Lock()
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// Mix audio from all active user buffers
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mixed := make([]int32, frameSamples)
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activeUsers := 0
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for id, buf := range p.userBuffers {
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if len(buf) > 0 {
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activeUsers++
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// Take up to frameSamples from this user
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toTake := frameSamples
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if len(buf) < frameSamples {
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toTake = len(buf)
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}
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for i := 0; i < toTake; i++ {
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mixed[i] += int32(buf[i])
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}
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// Advance buffer
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if len(buf) <= frameSamples {
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delete(p.userBuffers, id)
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} else {
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p.userBuffers[id] = buf[frameSamples:]
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}
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}
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// Write frame to WASAPI buffer
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bufSlice := unsafe.Slice(buffer, frameSamples*2)
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for i := 0; i < frameSamples; i++ {
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binary.LittleEndian.PutUint16(bufSlice[i*2:], uint16(frame[i]))
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}
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p.renderClient.ReleaseBuffer(frameSamples, 0)
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default:
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// No audio available - optionally write silence
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// (skip for now to avoid crackling)
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}
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p.bufferMu.Unlock()
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// Get WASAPI buffer
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var buffer *byte
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if err := p.renderClient.GetBuffer(uint32(frameSamples), &buffer); err != nil {
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return
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}
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p.mu.Lock()
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vol := p.volume
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p.mu.Unlock()
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// Write mixed samples with clipping protection and volume application
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bufSlice := unsafe.Slice(buffer, int(frameSamples)*2)
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for i := 0; i < int(frameSamples); i++ {
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val := mixed[i]
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// Apply volume
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val = int32(float32(val) * vol)
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// Hard clipping
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if val > 32767 {
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val = 32767
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} else if val < -32768 {
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val = -32768
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}
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binary.LittleEndian.PutUint16(bufSlice[i*2:], uint16(val))
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}
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p.renderClient.ReleaseBuffer(uint32(frameSamples), 0)
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}
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