feat(audio): optimize equalizer with stereo support and gain caching
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This commit is contained in:
Jose Luis Montañes Ojados
2026-01-17 20:49:16 +01:00
parent 711eb148df
commit be929ce55a
5 changed files with 203 additions and 138 deletions

View File

@@ -69,11 +69,11 @@ func NewPlayer() (*Player, error) {
waveFormat := &wca.WAVEFORMATEX{
WFormatTag: wca.WAVE_FORMAT_PCM,
NChannels: 1,
NChannels: 2, // STEREO
NSamplesPerSec: 48000,
WBitsPerSample: 16,
NBlockAlign: 2,
NAvgBytesPerSec: 96000,
NBlockAlign: 4, // 16bit * 2 channels / 8 = 4 bytes
NAvgBytesPerSec: 192000, // 48000 * 4
CbSize: 0,
}
@@ -167,21 +167,33 @@ func (p *Player) PlayPCM(senderID uint16, samples []int16) {
return
}
// ---------------------------------------------------------
// PHASE 1: Read Configuration (Safe Copy)
// ---------------------------------------------------------
p.bufferMu.Lock()
defer p.bufferMu.Unlock()
// Check per-user mute
if settings, ok := p.userSettings[senderID]; ok && settings.Muted {
settings, hasSettings := p.userSettings[senderID]
if hasSettings && settings.Muted {
p.bufferMu.Unlock()
return
}
// Apply EQ Filters if gains are non-zero
p.ensureEQ(senderID)
// Get EQ Instance (Create if needed)
if _, ok := p.userEQs[senderID]; !ok {
p.userEQs[senderID] = NewEQChain(48000)
}
userEQ := p.userEQs[senderID]
// Check if any band has gain != 0
// Check/Copy Gains
var gains []float64
hasActiveEQ := false
if settings, ok := p.userSettings[senderID]; ok && len(settings.Gains) == 5 {
for _, g := range settings.Gains {
if hasSettings && len(settings.Gains) == 5 {
// Copy gains to avoid race if UI changes them while we process
gains = make([]float64, 5)
copy(gains, settings.Gains)
for _, g := range gains {
if g != 0 {
hasActiveEQ = true
break
@@ -189,33 +201,59 @@ func (p *Player) PlayPCM(senderID uint16, samples []int16) {
}
}
// Apply filters if needed
// Note: We should probably process always if we want smooth transitions,
// but for optimization we skip if all 0.
// However, skipping might cause clicks if we jump from filtered to non-filtered state abruptly.
// For "Pro" audio, always process. For TUI app, let's process if active.
if hasActiveEQ {
if eq, ok := p.userEQs[senderID]; ok {
// Update gains from settings
// (Ideally we only do this on change, but doing it here ensures sync)
gains := p.userSettings[senderID].Gains
for i, g := range gains {
eq.SetGain(i, g)
}
p.bufferMu.Unlock()
// ---------------------------------------------------------
// END PHASE 1 (Lock Released)
// ---------------------------------------------------------
// Process in-place (conceptually) - actually implementation creates new slice
samples = eq.Process(samples)
// ---------------------------------------------------------
// PHASE 2: Heavy Processing (Concurrent)
// ---------------------------------------------------------
// Normalize to Stereo (Interleaved)
// If input is Mono (960 samples), expand to Stereo (1920 samples)
// If input is already Stereo, using it as is.
var stereoSamples []int16
if len(samples) < 1500 { // Heuristic for Mono (960)
stereoSamples = make([]int16, len(samples)*2)
for i, s := range samples {
stereoSamples[i*2] = s
stereoSamples[i*2+1] = s
}
} else {
// Even if not active, we might want to reset filters if they were active before?
// Or just leave them alone.
// Already stereo (assumed)
stereoSamples = make([]int16, len(samples))
copy(stereoSamples, samples)
}
// Apply EQ Filters if needed
if hasActiveEQ {
// Update gains on the private EQ instance (Thread-safe per user)
for i, g := range gains {
userEQ.SetGain(i, g)
}
// Process Stereo
stereoSamples = userEQ.Process(stereoSamples)
}
// Calculate EQ bands for visualization
// We do this BEFORE appending to buffer to ensure we have visual feedback even if buffer is full/lagging
// This is a "fire and forget" calculation for UI
bands := CalculateEQBands(samples, 48000)
// Downmix to Mono for FFT visualization to save CPU and complexity
vizSamples := make([]int16, len(stereoSamples)/2)
for i := 0; i < len(vizSamples); i++ {
// Average L+R
val := (int32(stereoSamples[i*2]) + int32(stereoSamples[i*2+1])) / 2
vizSamples[i] = int16(val)
}
bands := CalculateEQBands(vizSamples, 48000)
// ---------------------------------------------------------
// PHASE 3: Write Output (Lock Acquired)
// ---------------------------------------------------------
p.bufferMu.Lock()
defer p.bufferMu.Unlock()
// Re-check existence (could have disconnected?)
// Update user settings with new bands
if _, ok := p.userSettings[senderID]; !ok {
p.userSettings[senderID] = &UserSettings{Volume: 1.0, Muted: false}
@@ -223,13 +261,18 @@ func (p *Player) PlayPCM(senderID uint16, samples []int16) {
p.userSettings[senderID].EQBands = bands
// Append to user's specific buffer
// This ensures sequential playback for the same user
p.userBuffers[senderID] = append(p.userBuffers[senderID], samples...)
p.userBuffers[senderID] = append(p.userBuffers[senderID], stereoSamples...)
// Limit buffer size per user to avoid memory leaks if stalled
if len(p.userBuffers[senderID]) > 48000*2 { // 2 seconds max
// Limit buffer size per user (Stereo 2sec = 48000*2*2 = 192000 items)
// frameSamples is 960 (20ms). 2sec = 100 frames * 960 * 2 = 192000
const maxBufferSize = 48000 * 2 * 2 // 2 seconds stereo
if len(p.userBuffers[senderID]) > maxBufferSize {
// Drop oldest
drop := len(p.userBuffers[senderID]) - 48000
drop := len(p.userBuffers[senderID]) - maxBufferSize
// Ensure we drop aligned to stereo frame (even number)
if drop%2 != 0 {
drop++
}
p.userBuffers[senderID] = p.userBuffers[senderID][drop:]
}
}
@@ -389,7 +432,8 @@ func (p *Player) writeFrame() {
p.bufferMu.Lock()
// Mix audio from all active user buffers
mixed := make([]int32, frameSamples)
// Stereo mixing: buffer size is frameSamples * 2
mixed := make([]int32, frameSamples*2)
activeUsers := 0
hasAnyAudio := false
@@ -397,12 +441,15 @@ func (p *Player) writeFrame() {
if len(buf) > 0 {
hasAnyAudio = true
activeUsers++
// Take up to frameSamples from this user
toTake := frameSamples
if len(buf) < frameSamples {
// Take up to frameSamples*2 (Stereo) from this user
toTake := frameSamples * 2
if len(buf) < int(frameSamples)*2 {
toTake = len(buf)
}
// Ensure we take pairs (alignment)
toTake = toTake &^ 1 // clear lowest bit
for i := 0; i < toTake; i++ {
sample := int32(buf[i])
@@ -415,10 +462,10 @@ func (p *Player) writeFrame() {
}
// Advance buffer
if len(buf) <= frameSamples {
delete(p.userBuffers, id)
if len(buf) <= toTake {
delete(p.userBuffers, id) // Finished this buffer
} else {
p.userBuffers[id] = buf[frameSamples:]
p.userBuffers[id] = buf[toTake:]
}
}
}
@@ -441,8 +488,19 @@ func (p *Player) writeFrame() {
p.mu.Unlock()
// Write mixed samples with clipping protection and volume application
bufSlice := unsafe.Slice(buffer, int(frameSamples)*2)
for i := 0; i < int(frameSamples); i++ {
// Output buffer is for Stereo (frameSamples * 2 channels)
bufSlice := unsafe.Slice(buffer, int(frameSamples)*2*2) // *2 channels *2 bytes? No, unsafe.Slice takes count of Type.
// If buffer is *byte, we need bytes. frameSamples * 2 channels * 2 bytes/sample.
// Wait, GetBuffer returns BYTE pointer.
// Let's use uint16 slice.
// The logic below was: binary.LittleEndian.PutUint16(bufSlice[i*2:], ...)
// frameSamples was 960. loop 0..960.
// Now we have Stereo mixed buffer. Length = frameSamples * 2.
// We need to write frameSamples * 2 samples.
// Correct loop for Stereo:
for i := 0; i < int(frameSamples)*2; i++ { // Iterate over all samples (L, R, L, R...)
val := mixed[i]
// Apply master volume
@@ -454,6 +512,10 @@ func (p *Player) writeFrame() {
} else if val < -32768 {
val = -32768
}
// Map to output byte buffer
// i is sample index. Each sample is 2 bytes.
// Offset = i * 2.
binary.LittleEndian.PutUint16(bufSlice[i*2:], uint16(val))
}