Support for stereo mic.

This commit is contained in:
Storm Dragon
2026-02-09 22:33:17 -05:00
parent cfbefd3f7d
commit e3b6eac2a0
2 changed files with 536 additions and 389 deletions

View File

@@ -1,35 +1,35 @@
package gumble
import (
"time"
"time"
)
const (
// AudioSampleRate is the audio sample rate (in hertz) for incoming and
// outgoing audio.
AudioSampleRate = 48000
// AudioSampleRate is the audio sample rate (in hertz) for incoming and
// outgoing audio.
AudioSampleRate = 48000
// AudioDefaultInterval is the default interval that audio packets are sent
// at.
AudioDefaultInterval = 10 * time.Millisecond
// AudioDefaultInterval is the default interval that audio packets are sent
// at.
AudioDefaultInterval = 10 * time.Millisecond
// AudioMonoChannels is the number of channels used for voice transmission
AudioMonoChannels = 1
// AudioMonoChannels is the number of channels used for voice transmission
AudioMonoChannels = 1
// AudioChannels is the number of channels used for playback
AudioChannels = 2
// AudioChannels is the number of channels used for playback
AudioChannels = 2
// AudioDefaultFrameSize is the number of audio frames that should be sent in
// a 10ms window (mono samples)
AudioDefaultFrameSize = AudioSampleRate / 100
// AudioDefaultFrameSize is the number of audio frames that should be sent in
// a 10ms window (mono samples)
AudioDefaultFrameSize = AudioSampleRate / 100
// AudioMaximumFrameSize is the maximum audio frame size from another user
// that will be processed (accounting for stereo)
AudioMaximumFrameSize = (AudioSampleRate / 1000 * 60) * AudioChannels
// AudioMaximumFrameSize is the maximum audio frame size from another user
// that will be processed (accounting for stereo)
AudioMaximumFrameSize = (AudioSampleRate / 1000 * 60) * AudioChannels
// AudioDefaultDataBytes is the default number of bytes that an audio frame
// can use.
AudioDefaultDataBytes = 40
// AudioDefaultDataBytes is the default number of bytes that an audio frame
// can use.
AudioDefaultDataBytes = 40
)
// AudioListener is the interface that must be implemented by types wishing to
@@ -39,52 +39,55 @@ const (
// implementer's responsibility to continuously process AudioStreamEvent.C
// until it is closed.
type AudioListener interface {
OnAudioStream(e *AudioStreamEvent)
OnAudioStream(e *AudioStreamEvent)
}
// AudioStreamEvent is event that is passed to AudioListener.OnAudioStream.
type AudioStreamEvent struct {
Client *Client
User *User
C <-chan *AudioPacket
Client *Client
User *User
C <-chan *AudioPacket
}
// AudioBuffer is a slice of PCM audio samples.
type AudioBuffer []int16
func (a AudioBuffer) writeAudio(client *Client, seq int64, final bool) error {
// Choose encoder based on whether stereo is enabled
encoder := client.AudioEncoder
if client.IsStereoEncoderEnabled() && client.AudioEncoderStereo != nil {
encoder = client.AudioEncoderStereo
}
if encoder == nil {
return nil
}
dataBytes := client.Config.AudioDataBytes
raw, err := encoder.Encode(a, len(a), dataBytes)
if final {
defer encoder.Reset()
}
if err != nil {
return err
}
// Choose encoder based on whether buffer size indicates stereo or mono
encoder := client.AudioEncoder
frameSize := client.Config.AudioFrameSize()
if len(a) == frameSize*AudioChannels && client.AudioEncoderStereo != nil {
encoder = client.AudioEncoderStereo
} else if client.IsStereoEncoderEnabled() && client.AudioEncoderStereo != nil {
encoder = client.AudioEncoderStereo
}
if encoder == nil {
return nil
}
dataBytes := client.Config.AudioDataBytes
raw, err := encoder.Encode(a, len(a), dataBytes)
if final {
defer encoder.Reset()
}
if err != nil {
return err
}
var targetID byte
if target := client.VoiceTarget; target != nil {
targetID = byte(target.ID)
}
return client.Conn.WriteAudio(byte(4), targetID, seq, final, raw, nil, nil, nil)
var targetID byte
if target := client.VoiceTarget; target != nil {
targetID = byte(target.ID)
}
return client.Conn.WriteAudio(byte(4), targetID, seq, final, raw, nil, nil, nil)
}
// AudioPacket contains incoming audio samples and information.
type AudioPacket struct {
Client *Client
Sender *User
Target *VoiceTarget
Client *Client
Sender *User
Target *VoiceTarget
AudioBuffer
AudioBuffer
HasPosition bool
X, Y, Z float32
HasPosition bool
X, Y, Z float32
}

View File

@@ -1,442 +1,586 @@
package gumbleopenal
import (
"encoding/binary"
"errors"
"os/exec"
"time"
"encoding/binary"
"errors"
"os/exec"
"time"
"git.stormux.org/storm/barnard/audio"
"git.stormux.org/storm/barnard/gumble/gumble"
"git.stormux.org/storm/barnard/gumble/go-openal/openal"
"git.stormux.org/storm/barnard/audio"
"git.stormux.org/storm/barnard/gumble/go-openal/openal"
"git.stormux.org/storm/barnard/gumble/gumble"
"git.stormux.org/storm/barnard/noise"
)
// NoiseProcessor interface for noise suppression
type NoiseProcessor interface {
ProcessSamples(samples []int16)
IsEnabled() bool
ProcessSamples(samples []int16)
IsEnabled() bool
}
// EffectsProcessor interface for voice effects
type EffectsProcessor interface {
ProcessSamples(samples []int16)
IsEnabled() bool
ProcessSamples(samples []int16)
IsEnabled() bool
}
// FilePlayer interface for file playback
type FilePlayer interface {
GetAudioFrame() []int16
IsPlaying() bool
GetAudioFrame() []int16
IsPlaying() bool
}
const (
maxBufferSize = 11520 // Max frame size (2880) * bytes per stereo sample (4)
maxBufferSize = 11520 // Max frame size (2880) * bytes per stereo sample (4)
)
var (
ErrState = errors.New("gumbleopenal: invalid state")
ErrMic = errors.New("gumbleopenal: microphone disconnected or misconfigured")
ErrInputDevice = errors.New("gumbleopenal: invalid input device or parameters")
ErrOutputDevice = errors.New("gumbleopenal: invalid output device or parameters")
ErrState = errors.New("gumbleopenal: invalid state")
ErrMic = errors.New("gumbleopenal: microphone disconnected or misconfigured")
ErrInputDevice = errors.New("gumbleopenal: invalid input device or parameters")
ErrOutputDevice = errors.New("gumbleopenal: invalid output device or parameters")
)
func beep() {
cmd := exec.Command("beep")
cmdout, err := cmd.Output()
if err != nil {
panic(err)
}
if cmdout != nil {
}
cmd := exec.Command("beep")
cmdout, err := cmd.Output()
if err != nil {
panic(err)
}
if cmdout != nil {
}
}
type Stream struct {
client *gumble.Client
link gumble.Detacher
client *gumble.Client
link gumble.Detacher
deviceSource *openal.CaptureDevice
sourceFrameSize int
micVolume float32
sourceStop chan bool
deviceSource *openal.CaptureDevice
sourceFormat openal.Format
sourceChannels int
sourceFrameSize int
micVolume float32
sourceStop chan bool
deviceSink *openal.Device
contextSink *openal.Context
deviceSink *openal.Device
contextSink *openal.Context
noiseProcessor NoiseProcessor
micAGC *audio.AGC
effectsProcessor EffectsProcessor
filePlayer FilePlayer
noiseProcessor NoiseProcessor
noiseProcessorRight NoiseProcessor
micAGC *audio.AGC
micAGCRight *audio.AGC
effectsProcessor EffectsProcessor
effectsProcessorRight EffectsProcessor
filePlayer FilePlayer
}
func New(client *gumble.Client, inputDevice *string, outputDevice *string, test bool) (*Stream, error) {
frmsz := 480
if !test {
frmsz = client.Config.AudioFrameSize()
}
frmsz := 480
if !test {
frmsz = client.Config.AudioFrameSize()
}
// Always use mono for input device
idev := openal.CaptureOpenDevice(*inputDevice, gumble.AudioSampleRate, openal.FormatMono16, uint32(frmsz))
if idev == nil {
return nil, ErrInputDevice
}
inputFormat := openal.FormatStereo16
sourceChannels := 2
idev := openal.CaptureOpenDevice(*inputDevice, gumble.AudioSampleRate, inputFormat, uint32(frmsz))
if idev == nil {
inputFormat = openal.FormatMono16
sourceChannels = 1
idev = openal.CaptureOpenDevice(*inputDevice, gumble.AudioSampleRate, inputFormat, uint32(frmsz))
}
if idev == nil {
return nil, ErrInputDevice
}
odev := openal.OpenDevice(*outputDevice)
if odev == nil {
idev.CaptureCloseDevice()
return nil, ErrOutputDevice
}
odev := openal.OpenDevice(*outputDevice)
if odev == nil {
idev.CaptureCloseDevice()
return nil, ErrOutputDevice
}
if test {
idev.CaptureCloseDevice()
odev.CloseDevice()
return nil, nil
}
if test {
idev.CaptureCloseDevice()
odev.CloseDevice()
return nil, nil
}
s := &Stream{
client: client,
sourceFrameSize: frmsz,
micVolume: 1.0,
micAGC: audio.NewAGC(), // Always enable AGC for outgoing mic
}
s := &Stream{
client: client,
sourceFormat: inputFormat,
sourceChannels: sourceChannels,
sourceFrameSize: frmsz,
micVolume: 1.0,
micAGC: audio.NewAGC(), // Always enable AGC for outgoing mic
}
if sourceChannels == 2 {
s.micAGCRight = audio.NewAGC()
}
s.deviceSource = idev
if s.deviceSource == nil {
return nil, ErrInputDevice
}
s.deviceSource = idev
if s.deviceSource == nil {
return nil, ErrInputDevice
}
s.deviceSink = odev
if s.deviceSink == nil {
return nil, ErrOutputDevice
}
s.contextSink = s.deviceSink.CreateContext()
if s.contextSink == nil {
s.Destroy()
return nil, ErrOutputDevice
}
s.contextSink.Activate()
s.deviceSink = odev
if s.deviceSink == nil {
return nil, ErrOutputDevice
}
s.contextSink = s.deviceSink.CreateContext()
if s.contextSink == nil {
s.Destroy()
return nil, ErrOutputDevice
}
s.contextSink.Activate()
return s, nil
return s, nil
}
func (s *Stream) AttachStream(client *gumble.Client) {
s.link = client.Config.AttachAudio(s)
s.link = client.Config.AttachAudio(s)
}
func (s *Stream) SetNoiseProcessor(np NoiseProcessor) {
s.noiseProcessor = np
s.noiseProcessor = np
s.noiseProcessorRight = cloneNoiseProcessor(np)
}
func (s *Stream) SetEffectsProcessor(ep EffectsProcessor) {
s.effectsProcessor = ep
s.effectsProcessor = ep
s.effectsProcessorRight = cloneEffectsProcessor(ep)
}
func (s *Stream) GetEffectsProcessor() EffectsProcessor {
return s.effectsProcessor
return s.effectsProcessor
}
func (s *Stream) SetFilePlayer(fp FilePlayer) {
s.filePlayer = fp
s.filePlayer = fp
}
func (s *Stream) GetFilePlayer() FilePlayer {
return s.filePlayer
return s.filePlayer
}
func (s *Stream) Destroy() {
if s.link != nil {
s.link.Detach()
}
if s.deviceSource != nil {
s.StopSource()
s.deviceSource.CaptureCloseDevice()
s.deviceSource = nil
}
if s.deviceSink != nil {
s.contextSink.Destroy()
s.deviceSink.CloseDevice()
s.contextSink = nil
s.deviceSink = nil
}
if s.link != nil {
s.link.Detach()
}
if s.deviceSource != nil {
s.StopSource()
s.deviceSource.CaptureCloseDevice()
s.deviceSource = nil
}
if s.deviceSink != nil {
s.contextSink.Destroy()
s.deviceSink.CloseDevice()
s.contextSink = nil
s.deviceSink = nil
}
}
func (s *Stream) StartSource(inputDevice *string) error {
if s.sourceStop != nil {
return ErrState
}
if s.deviceSource == nil {
return ErrMic
}
s.deviceSource.CaptureStart()
s.sourceStop = make(chan bool)
go s.sourceRoutine(inputDevice)
return nil
if s.sourceStop != nil {
return ErrState
}
if s.deviceSource == nil {
return ErrMic
}
s.deviceSource.CaptureStart()
s.sourceStop = make(chan bool)
go s.sourceRoutine(inputDevice)
return nil
}
func (s *Stream) StopSource() error {
if s.deviceSource == nil {
return ErrMic
}
s.deviceSource.CaptureStop()
if s.sourceStop == nil {
return ErrState
}
close(s.sourceStop)
s.sourceStop = nil
return nil
if s.deviceSource == nil {
return ErrMic
}
s.deviceSource.CaptureStop()
if s.sourceStop == nil {
return ErrState
}
close(s.sourceStop)
s.sourceStop = nil
return nil
}
func (s *Stream) GetMicVolume() float32 {
return s.micVolume
return s.micVolume
}
func (s *Stream) SetMicVolume(change float32, relative bool) {
var val float32
if relative {
val = s.GetMicVolume() + change
} else {
val = change
}
if val >= 1 {
val = 1.0
}
if val <= 0 {
val = 0
}
s.micVolume = val
var val float32
if relative {
val = s.GetMicVolume() + change
} else {
val = change
}
if val >= 1 {
val = 1.0
}
if val <= 0 {
val = 0
}
s.micVolume = val
}
func (s *Stream) OnAudioStream(e *gumble.AudioStreamEvent) {
go func(e *gumble.AudioStreamEvent) {
var source = openal.NewSource()
e.User.AudioSource = &source
// Set initial gain based on volume and mute state
if e.User.LocallyMuted {
e.User.AudioSource.SetGain(0)
} else {
e.User.AudioSource.SetGain(e.User.Volume)
}
go func(e *gumble.AudioStreamEvent) {
var source = openal.NewSource()
e.User.AudioSource = &source
bufferCount := e.Client.Config.Buffers
if bufferCount < 64 {
bufferCount = 64
}
emptyBufs := openal.NewBuffers(bufferCount)
reclaim := func() {
if n := source.BuffersProcessed(); n > 0 {
reclaimedBufs := make(openal.Buffers, n)
source.UnqueueBuffers(reclaimedBufs)
emptyBufs = append(emptyBufs, reclaimedBufs...)
}
}
// Set initial gain based on volume and mute state
if e.User.LocallyMuted {
e.User.AudioSource.SetGain(0)
} else {
e.User.AudioSource.SetGain(e.User.Volume)
}
var raw [maxBufferSize]byte
for packet := range e.C {
// Skip processing if user is locally muted
if e.User.LocallyMuted {
continue
}
bufferCount := e.Client.Config.Buffers
if bufferCount < 64 {
bufferCount = 64
}
emptyBufs := openal.NewBuffers(bufferCount)
var boost uint16 = uint16(1)
samples := len(packet.AudioBuffer)
if samples > cap(raw)/2 {
continue
}
boost = e.User.Boost
reclaim := func() {
if n := source.BuffersProcessed(); n > 0 {
reclaimedBufs := make(openal.Buffers, n)
source.UnqueueBuffers(reclaimedBufs)
emptyBufs = append(emptyBufs, reclaimedBufs...)
}
}
// Check if sample count suggests stereo data
isStereo := samples > gumble.AudioDefaultFrameSize && samples%2 == 0
format := openal.FormatMono16
if isStereo {
format = openal.FormatStereo16
samples = samples / 2
}
var raw [maxBufferSize]byte
rawPtr := 0
if isStereo {
// Process stereo samples as pairs
for i := 0; i < samples*2; i += 2 {
// Process left channel with saturation protection
sample := packet.AudioBuffer[i]
if boost > 1 {
boosted := int32(sample) * int32(boost)
if boosted > 32767 {
sample = 32767
} else if boosted < -32767 {
sample = -32767
} else {
sample = int16(boosted)
}
}
binary.LittleEndian.PutUint16(raw[rawPtr:], uint16(sample))
rawPtr += 2
for packet := range e.C {
// Skip processing if user is locally muted
if e.User.LocallyMuted {
continue
}
// Process right channel with saturation protection
sample = packet.AudioBuffer[i+1]
if boost > 1 {
boosted := int32(sample) * int32(boost)
if boosted > 32767 {
sample = 32767
} else if boosted < -32767 {
sample = -32767
} else {
sample = int16(boosted)
}
}
binary.LittleEndian.PutUint16(raw[rawPtr:], uint16(sample))
rawPtr += 2
}
} else {
// Process mono samples with saturation protection
for i := 0; i < samples; i++ {
sample := packet.AudioBuffer[i]
if boost > 1 {
boosted := int32(sample) * int32(boost)
if boosted > 32767 {
sample = 32767
} else if boosted < -32767 {
sample = -32767
} else {
sample = int16(boosted)
}
}
binary.LittleEndian.PutUint16(raw[rawPtr:], uint16(sample))
rawPtr += 2
}
}
var boost uint16 = uint16(1)
samples := len(packet.AudioBuffer)
if samples > cap(raw)/2 {
continue
}
reclaim()
if len(emptyBufs) == 0 {
continue
}
boost = e.User.Boost
last := len(emptyBufs) - 1
buffer := emptyBufs[last]
emptyBufs = emptyBufs[:last]
// Check if sample count suggests stereo data
isStereo := samples > gumble.AudioDefaultFrameSize && samples%2 == 0
format := openal.FormatMono16
if isStereo {
format = openal.FormatStereo16
samples = samples / 2
}
buffer.SetData(format, raw[:rawPtr], gumble.AudioSampleRate)
source.QueueBuffer(buffer)
rawPtr := 0
if isStereo {
// Process stereo samples as pairs
for i := 0; i < samples*2; i += 2 {
// Process left channel with saturation protection
sample := packet.AudioBuffer[i]
if boost > 1 {
boosted := int32(sample) * int32(boost)
if boosted > 32767 {
sample = 32767
} else if boosted < -32767 {
sample = -32767
} else {
sample = int16(boosted)
}
}
binary.LittleEndian.PutUint16(raw[rawPtr:], uint16(sample))
rawPtr += 2
if source.State() != openal.Playing {
source.Play()
}
}
reclaim()
emptyBufs.Delete()
source.Delete()
}(e)
// Process right channel with saturation protection
sample = packet.AudioBuffer[i+1]
if boost > 1 {
boosted := int32(sample) * int32(boost)
if boosted > 32767 {
sample = 32767
} else if boosted < -32767 {
sample = -32767
} else {
sample = int16(boosted)
}
}
binary.LittleEndian.PutUint16(raw[rawPtr:], uint16(sample))
rawPtr += 2
}
} else {
// Process mono samples with saturation protection
for i := 0; i < samples; i++ {
sample := packet.AudioBuffer[i]
if boost > 1 {
boosted := int32(sample) * int32(boost)
if boosted > 32767 {
sample = 32767
} else if boosted < -32767 {
sample = -32767
} else {
sample = int16(boosted)
}
}
binary.LittleEndian.PutUint16(raw[rawPtr:], uint16(sample))
rawPtr += 2
}
}
reclaim()
if len(emptyBufs) == 0 {
continue
}
last := len(emptyBufs) - 1
buffer := emptyBufs[last]
emptyBufs = emptyBufs[:last]
buffer.SetData(format, raw[:rawPtr], gumble.AudioSampleRate)
source.QueueBuffer(buffer)
if source.State() != openal.Playing {
source.Play()
}
}
reclaim()
emptyBufs.Delete()
source.Delete()
}(e)
}
func (s *Stream) sourceRoutine(inputDevice *string) {
interval := s.client.Config.AudioInterval
frameSize := s.client.Config.AudioFrameSize()
interval := s.client.Config.AudioInterval
frameSize := s.client.Config.AudioFrameSize()
if frameSize != s.sourceFrameSize {
s.deviceSource.CaptureCloseDevice()
s.sourceFrameSize = frameSize
// Always use mono for input
s.deviceSource = openal.CaptureOpenDevice(*inputDevice, gumble.AudioSampleRate, openal.FormatMono16, uint32(s.sourceFrameSize))
}
if frameSize != s.sourceFrameSize {
s.deviceSource.CaptureCloseDevice()
s.sourceFrameSize = frameSize
s.deviceSource = openal.CaptureOpenDevice(*inputDevice, gumble.AudioSampleRate, s.sourceFormat, uint32(s.sourceFrameSize))
if s.deviceSource == nil && s.sourceFormat == openal.FormatStereo16 {
s.sourceFormat = openal.FormatMono16
s.sourceChannels = 1
s.deviceSource = openal.CaptureOpenDevice(*inputDevice, gumble.AudioSampleRate, s.sourceFormat, uint32(s.sourceFrameSize))
}
}
if s.deviceSource == nil {
return
}
ticker := time.NewTicker(interval)
defer ticker.Stop()
ticker := time.NewTicker(interval)
defer ticker.Stop()
stop := s.sourceStop
stop := s.sourceStop
outgoing := s.client.AudioOutgoing()
defer close(outgoing)
outgoing := s.client.AudioOutgoing()
defer close(outgoing)
for {
select {
case <-stop:
return
case <-ticker.C:
// Initialize buffer with silence
int16Buffer := make([]int16, frameSize)
for {
select {
case <-stop:
return
case <-ticker.C:
sampleCount := frameSize * s.sourceChannels
int16Buffer := make([]int16, sampleCount)
// Capture microphone if available
hasMicInput := false
buff := s.deviceSource.CaptureSamples(uint32(frameSize))
if len(buff) == frameSize*2 {
hasMicInput = true
for i := range int16Buffer {
sample := int16(binary.LittleEndian.Uint16(buff[i*2:]))
if s.micVolume != 1.0 {
sample = int16(float32(sample) * s.micVolume)
}
int16Buffer[i] = sample
}
// Capture microphone if available
hasMicInput := false
buff := s.deviceSource.CaptureSamples(uint32(frameSize))
if len(buff) == sampleCount*2 {
hasMicInput = true
for i := 0; i < sampleCount; i++ {
sample := int16(binary.LittleEndian.Uint16(buff[i*2:]))
if s.micVolume != 1.0 {
sample = int16(float32(sample) * s.micVolume)
}
int16Buffer[i] = sample
}
// Apply noise suppression if available and enabled
if s.noiseProcessor != nil && s.noiseProcessor.IsEnabled() {
s.noiseProcessor.ProcessSamples(int16Buffer)
}
if s.sourceChannels == 1 {
s.processMonoSamples(int16Buffer)
} else {
s.processStereoSamples(int16Buffer, frameSize)
}
}
// Apply AGC to outgoing microphone audio (always enabled)
if s.micAGC != nil {
s.micAGC.ProcessSamples(int16Buffer)
}
// Mix with or use file audio if playing
hasFileAudio := false
var outputBuffer []int16
// Apply voice effects if available and enabled
if s.effectsProcessor != nil && s.effectsProcessor.IsEnabled() {
s.effectsProcessor.ProcessSamples(int16Buffer)
}
}
if s.filePlayer != nil && s.filePlayer.IsPlaying() {
fileAudio := s.filePlayer.GetAudioFrame()
if fileAudio != nil && len(fileAudio) > 0 {
hasFileAudio = true
// File audio is stereo - send as stereo when file is playing
// Create stereo buffer (frameSize * 2 channels)
outputBuffer = make([]int16, frameSize*2)
// Mix with or use file audio if playing
hasFileAudio := false
var outputBuffer []int16
if hasMicInput {
if s.sourceChannels == 2 {
// Mix stereo mic with stereo file
for i := 0; i < frameSize; i++ {
idx := i * 2
if idx+1 < len(fileAudio) {
left := int32(int16Buffer[idx]) + int32(fileAudio[idx])
if left > 32767 {
left = 32767
} else if left < -32768 {
left = -32768
}
outputBuffer[idx] = int16(left)
if s.filePlayer != nil && s.filePlayer.IsPlaying() {
fileAudio := s.filePlayer.GetAudioFrame()
if fileAudio != nil && len(fileAudio) > 0 {
hasFileAudio = true
// File audio is stereo - send as stereo when file is playing
// Create stereo buffer (frameSize * 2 channels)
outputBuffer = make([]int16, frameSize*2)
right := int32(int16Buffer[idx+1]) + int32(fileAudio[idx+1])
if right > 32767 {
right = 32767
} else if right < -32768 {
right = -32768
}
outputBuffer[idx+1] = int16(right)
}
}
} else {
// Mix mono mic with stereo file
for i := 0; i < frameSize; i++ {
idx := i * 2
if idx+1 < len(fileAudio) {
left := int32(int16Buffer[i]) + int32(fileAudio[idx])
if left > 32767 {
left = 32767
} else if left < -32768 {
left = -32768
}
outputBuffer[idx] = int16(left)
if hasMicInput {
// Mix mono mic with stereo file
for i := 0; i < frameSize; i++ {
if i*2+1 < len(fileAudio) {
// Left channel: mic + file left
left := int32(int16Buffer[i]) + int32(fileAudio[i*2])
if left > 32767 {
left = 32767
} else if left < -32768 {
left = -32768
}
outputBuffer[i*2] = int16(left)
right := int32(int16Buffer[i]) + int32(fileAudio[idx+1])
if right > 32767 {
right = 32767
} else if right < -32768 {
right = -32768
}
outputBuffer[idx+1] = int16(right)
}
}
}
} else {
// Use file audio only (already stereo)
copy(outputBuffer, fileAudio[:frameSize*2])
}
}
}
// Right channel: mic + file right
right := int32(int16Buffer[i]) + int32(fileAudio[i*2+1])
if right > 32767 {
right = 32767
} else if right < -32768 {
right = -32768
}
outputBuffer[i*2+1] = int16(right)
}
}
} else {
// Use file audio only (already stereo)
copy(outputBuffer, fileAudio[:frameSize*2])
}
}
}
// Determine what to send
if hasFileAudio {
// Send stereo buffer when file is playing
outgoing <- gumble.AudioBuffer(outputBuffer)
} else if hasMicInput {
// Send mono mic when no file is playing
outgoing <- gumble.AudioBuffer(int16Buffer)
}
}
}
// Determine what to send
if hasFileAudio {
// Send stereo buffer when file is playing
outgoing <- gumble.AudioBuffer(outputBuffer)
} else if hasMicInput {
// Send mic when no file is playing
outgoing <- gumble.AudioBuffer(int16Buffer)
}
}
}
}
func (s *Stream) processMonoSamples(samples []int16) {
s.processChannel(samples, s.noiseProcessor, s.micAGC, s.effectsProcessor)
}
func (s *Stream) processStereoSamples(samples []int16, frameSize int) {
if frameSize == 0 || len(samples) < frameSize*2 {
return
}
s.ensureStereoProcessors()
s.syncStereoProcessors()
left := make([]int16, frameSize)
right := make([]int16, frameSize)
for i := 0; i < frameSize; i++ {
idx := i * 2
left[i] = samples[idx]
right[i] = samples[idx+1]
}
s.processChannel(left, s.noiseProcessor, s.micAGC, s.effectsProcessor)
s.processChannel(right, s.noiseProcessorRight, s.micAGCRight, s.effectsProcessorRight)
for i := 0; i < frameSize; i++ {
idx := i * 2
samples[idx] = left[i]
samples[idx+1] = right[i]
}
}
func (s *Stream) processChannel(samples []int16, noiseProcessor NoiseProcessor, micAGC *audio.AGC, effectsProcessor EffectsProcessor) {
if noiseProcessor != nil && noiseProcessor.IsEnabled() {
noiseProcessor.ProcessSamples(samples)
}
if micAGC != nil {
micAGC.ProcessSamples(samples)
}
if effectsProcessor != nil && effectsProcessor.IsEnabled() {
effectsProcessor.ProcessSamples(samples)
}
}
func (s *Stream) ensureStereoProcessors() {
if s.micAGCRight == nil {
s.micAGCRight = audio.NewAGC()
}
if s.noiseProcessorRight == nil {
s.noiseProcessorRight = cloneNoiseProcessor(s.noiseProcessor)
}
if s.effectsProcessorRight == nil {
s.effectsProcessorRight = cloneEffectsProcessor(s.effectsProcessor)
}
}
func (s *Stream) syncStereoProcessors() {
leftSuppressor, leftOk := s.noiseProcessor.(*noise.Suppressor)
rightSuppressor, rightOk := s.noiseProcessorRight.(*noise.Suppressor)
if leftOk && rightOk {
if leftSuppressor.IsEnabled() != rightSuppressor.IsEnabled() {
rightSuppressor.SetEnabled(leftSuppressor.IsEnabled())
}
if leftSuppressor.GetThreshold() != rightSuppressor.GetThreshold() {
rightSuppressor.SetThreshold(leftSuppressor.GetThreshold())
}
}
leftEffects, leftOk := s.effectsProcessor.(*audio.EffectsProcessor)
rightEffects, rightOk := s.effectsProcessorRight.(*audio.EffectsProcessor)
if leftOk && rightOk {
if leftEffects.IsEnabled() != rightEffects.IsEnabled() {
rightEffects.SetEnabled(leftEffects.IsEnabled())
}
if leftEffects.GetCurrentEffect() != rightEffects.GetCurrentEffect() {
rightEffects.SetEffect(leftEffects.GetCurrentEffect())
}
}
}
func cloneNoiseProcessor(np NoiseProcessor) NoiseProcessor {
if np == nil {
return nil
}
if suppressor, ok := np.(*noise.Suppressor); ok {
clone := noise.NewSuppressor()
clone.SetEnabled(suppressor.IsEnabled())
clone.SetThreshold(suppressor.GetThreshold())
return clone
}
return nil
}
func cloneEffectsProcessor(ep EffectsProcessor) EffectsProcessor {
if ep == nil {
return nil
}
if processor, ok := ep.(*audio.EffectsProcessor); ok {
clone := audio.NewEffectsProcessor(gumble.AudioSampleRate)
clone.SetEnabled(processor.IsEnabled())
clone.SetEffect(processor.GetCurrentEffect())
return clone
}
return nil
}