Fix audio clock drift and adaptive jitter buffering
This commit is contained in:
+2
-1
@@ -20,7 +20,8 @@ The supported source-of-truth layout is:
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## Release gates
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- Run real multi-person calls on Windows, macOS, and physical iOS hardware.
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- Run real multi-person calls on Windows, macOS, and physical iOS hardware, including adaptive
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20/40/60 ms buffering, duration-aware DRED/FEC, and mismatched input/output endpoints.
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- Complete NVDA and VoiceOver navigation/announcement passes.
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- Exercise iOS background/lock, interruption, Bluetooth, route-change, ReplayKit, and iOS 27
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ScreenCaptureKit paths on devices.
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@@ -139,7 +139,7 @@ internal sealed class MacAudioBackend : IAudioDeviceBackend
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private sealed class Playback : IAudioPlayback
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{
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private readonly PcmRing pcm = new(32_768);
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private readonly AdaptivePcmBuffer pcm = new(2);
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private readonly short[] renderScratch = new short[8_192];
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private readonly AVAudioEngine engine = new();
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private readonly AVAudioFormat format;
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@@ -187,6 +187,7 @@ internal sealed class MacAudioBackend : IAudioDeviceBackend
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}
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public void Write(ReadOnlySpan<short> stereoPcm) => pcm.TryWrite(stereoPcm);
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public int BufferMilliseconds { get => pcm.BufferMilliseconds; set => pcm.BufferMilliseconds = value; }
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private unsafe int Render(IntPtr isSilence, IntPtr _, uint frameCount, IntPtr outputData)
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{
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@@ -199,8 +200,6 @@ internal sealed class MacAudioBackend : IAudioDeviceBackend
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int frames = checked((int)frameCount);
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int requested = checked(frames * 2);
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if (bufferCount != 2 || requested > renderScratch.Length) return Fail("Core Audio returned an invalid planar stereo playback buffer.");
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Span<short> discard = stackalloc short[1_920];
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while (pcm.Count > 11_520) pcm.Read(discard[..Math.Min(discard.Length, pcm.Count - 11_520)]);
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Span<short> source = renderScratch.AsSpan(0, requested);
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int read = pcm.Read(source);
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source[read..].Clear();
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@@ -19,6 +19,7 @@ internal sealed class MacSettings
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internal bool AuxiliaryEnabled { get; set; }
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internal string? AuxiliaryDeviceId { get; set; }
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internal float AuxiliaryGain { get; set; } = 1f;
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internal int AudioBufferMilliseconds { get; set; } = 40;
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internal bool EventSounds { get; set; } = true;
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internal bool SpokenEvents { get; set; }
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internal float EventVolume { get; set; } = 1f;
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@@ -28,9 +29,9 @@ internal sealed class MacSettings
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internal void Load()
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{
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NSString[] keys = [(NSString)"voice.inputMode", (NSString)"voice.vadThreshold", (NSString)"voice.inputGain", (NSString)"voice.outputGain",
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(NSString)"voice.pttKeyCode", (NSString)"voice.auxGain", (NSString)"feedback.sounds", (NSString)"feedback.volume"];
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(NSString)"voice.pttKeyCode", (NSString)"voice.auxGain", (NSString)"voice.audioBufferMs", (NSString)"feedback.sounds", (NSString)"feedback.volume"];
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NSObject[] defaults = [NSNumber.FromInt32((int)AudioInputMode.VoiceActivation), NSNumber.FromFloat(0.05f),
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NSNumber.FromFloat(1f), NSNumber.FromFloat(0.8f), NSNumber.FromInt32(0x60), NSNumber.FromFloat(1f),
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NSNumber.FromFloat(1f), NSNumber.FromFloat(0.8f), NSNumber.FromInt32(0x60), NSNumber.FromFloat(1f), NSNumber.FromInt32(40),
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NSNumber.FromBoolean(true), NSNumber.FromFloat(1f)];
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values.RegisterDefaults(new NSDictionary<NSString, NSObject>(keys, defaults));
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InputMode = Enum.IsDefined(typeof(AudioInputMode), (int)values.IntForKey("voice.inputMode"))
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@@ -46,6 +47,7 @@ internal sealed class MacSettings
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AuxiliaryEnabled = values.BoolForKey("voice.auxEnabled");
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AuxiliaryDeviceId = EmptyToNull(values.StringForKey("voice.auxDeviceUID"));
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AuxiliaryGain = Math.Clamp(values.FloatForKey("voice.auxGain"), 0f, 4f);
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int audioBuffer = checked((int)values.IntForKey("voice.audioBufferMs")); AudioBufferMilliseconds = audioBuffer is 20 or 40 or 60 ? audioBuffer : 40;
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EventSounds = values.BoolForKey("feedback.sounds");
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SpokenEvents = values.BoolForKey("feedback.speech");
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EventVolume = Math.Clamp(values.FloatForKey("feedback.volume"), 0f, 1f);
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@@ -59,7 +61,7 @@ internal sealed class MacSettings
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values.SetFloat(InputGain, "voice.inputGain"); values.SetFloat(OutputGain, "voice.outputGain");
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values.SetBool(InputNoiseReduction, "voice.inputNoiseReduction"); values.SetBool(StereoMicrophone, "voice.stereoMic");
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values.SetInt(PushToTalkKeyCode, "voice.pttKeyCode"); Set("voice.inputDevice", InputDeviceId); Set("voice.outputDevice", OutputDeviceId);
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values.SetBool(AuxiliaryEnabled, "voice.auxEnabled"); Set("voice.auxDeviceUID", AuxiliaryDeviceId); values.SetFloat(AuxiliaryGain, "voice.auxGain");
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values.SetBool(AuxiliaryEnabled, "voice.auxEnabled"); Set("voice.auxDeviceUID", AuxiliaryDeviceId); values.SetFloat(AuxiliaryGain, "voice.auxGain"); values.SetInt(AudioBufferMilliseconds, "voice.audioBufferMs");
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values.SetBool(EventSounds, "feedback.sounds"); values.SetBool(SpokenEvents, "feedback.speech"); values.SetFloat(EventVolume, "feedback.volume");
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values.SetBool(SelfTalkSounds, "feedback.selfTalk"); values.SetBool(PushToTalkSound, "feedback.ptt"); values.Synchronize();
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}
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@@ -210,6 +210,8 @@ internal sealed class MainWindowController : NSWindowController
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if (settings.InputMode != AudioInputMode.PushToTalk) { client.Audio.PushToTalk = false; pushToTalkEngaged = false; }
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client.Audio.InputGain = settings.InputGain; client.Audio.OutputGain = settings.OutputGain;
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client.Audio.InputNoiseReduction = settings.InputNoiseReduction;
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client.Audio.DeviceBufferMilliseconds = settings.AudioBufferMilliseconds;
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if (playback is not null) playback.BufferMilliseconds = settings.AudioBufferMilliseconds;
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if (auxiliaryStreamId != 0) client.Audio.SetLocalGain(auxiliaryStreamId, settings.AuxiliaryGain);
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}
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@@ -367,6 +369,7 @@ internal sealed class MainWindowController : NSWindowController
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activeOutputDevice = settings.OutputDeviceId ?? SelectedDevice(outputDevice);
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activeStereo = settings.StereoMicrophone;
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playback = audioBackend.OpenPlayback(activeOutputDevice);
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playback.BufferMilliseconds = settings.AudioBufferMilliseconds;
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client.Audio.MixedPcm += PlayMixedPcm;
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StreamInfo stream = await client.StartStreamAsync(StreamKind.StreamMic, "Microphone", settings.StereoMicrophone ? 2 : 1);
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microphoneStreamId = stream.StreamId;
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@@ -26,15 +26,18 @@ internal sealed class SettingsWindowController : NSWindowController
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private readonly NSButton pttSound = NSButton.CreateCheckbox("Push-to-talk cue", () => { });
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private readonly NSButton pttKey = new(new CGRect(220, 12, 200, 32)) { Title = "Change PTT key…" };
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private readonly NSSlider eventVolume = new(new CGRect(160, 505, 260, 24)) { MinValue = 0, MaxValue = 100 };
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private readonly NSPopUpButton audioBuffer = new(new CGRect(160, 545, 260, 28), false);
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internal SettingsWindowController(MainWindowController owner, MacSettings settings, MacAudioBackend audio) : base(
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new NSWindow(new CGRect(0, 0, 460, 620), NSWindowStyle.Titled | NSWindowStyle.Closable,
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new NSWindow(new CGRect(0, 0, 460, 660), NSWindowStyle.Titled | NSWindowStyle.Closable,
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NSBackingStore.Buffered, false))
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{
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this.owner = owner; this.settings = settings; this.audio = audio;
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Window!.Title = "VoiceCat Settings"; Window.Center();
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NSView view = Window.ContentView!;
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mode.AddItems(["Voice activation", "Push to talk", "Always on"]);
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audioBuffer.AddItems(["Low latency (20 ms)", "Balanced (40 ms)", "Stable (60 ms)"]);
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Add(view, "Audio buffering", audioBuffer, 555);
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Add(view, "Sound volume", eventVolume, 515);
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Add(view, "Input mode", mode, 465); Add(view, "VAD sensitivity", vad, 425); Add(view, "Microphone volume", inputGain, 385);
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noiseReduction.Frame = new CGRect(160, 340, 260, 24); stereo.Frame = new CGRect(160, 315, 260, 24);
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@@ -46,7 +49,7 @@ internal sealed class SettingsWindowController : NSWindowController
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selfTalk.Frame = new CGRect(265, 45, 180, 24); pttSound.Frame = new CGRect(25, 18, 180, 24);
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view.AddSubview(sounds); view.AddSubview(speech); view.AddSubview(selfTalk); view.AddSubview(pttSound); view.AddSubview(pttKey);
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foreach (NSControl control in new NSControl[] { mode, vad, inputGain, noiseReduction, stereo, input, output,
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outputGain, auxiliary, auxiliaryDevice, auxiliaryGain, sounds, speech, selfTalk, pttSound, eventVolume }) control.Activated += Changed;
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outputGain, auxiliary, auxiliaryDevice, auxiliaryGain, sounds, speech, selfTalk, pttSound, eventVolume, audioBuffer }) control.Activated += Changed;
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pttKey.Activated += CapturePushToTalkKey;
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Populate(input, true, settings.InputDeviceId); Populate(output, false, settings.OutputDeviceId); Populate(auxiliaryDevice, true, settings.AuxiliaryDeviceId);
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LoadValues(); SetAccessibility();
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@@ -76,6 +79,7 @@ internal sealed class SettingsWindowController : NSWindowController
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eventVolume.DoubleValue = settings.EventVolume * 100;
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selfTalk.State = State(settings.SelfTalkSounds); pttSound.State = State(settings.PushToTalkSound);
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pttKey.Title = $"PTT key code: {settings.PushToTalkKeyCode}";
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audioBuffer.SelectItem(settings.AudioBufferMilliseconds switch { 20 => 0, 60 => 2, _ => 1 });
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UpdateEnabled();
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}
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@@ -102,6 +106,7 @@ internal sealed class SettingsWindowController : NSWindowController
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settings.AuxiliaryGain = (float)auxiliaryGain.DoubleValue / 100; settings.EventSounds = On(sounds);
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settings.EventVolume = (float)eventVolume.DoubleValue / 100; settings.SpokenEvents = On(speech);
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settings.SelfTalkSounds = On(selfTalk); settings.PushToTalkSound = On(pttSound);
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settings.AudioBufferMilliseconds = audioBuffer.IndexOfSelectedItem switch { 0 => 20, 2 => 60, _ => 40 };
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settings.Save(); UpdateEnabled(); await owner.ApplySettingsAsync();
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}
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@@ -113,6 +118,7 @@ internal sealed class SettingsWindowController : NSWindowController
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((INSAccessibility)output).AccessibilityLabel = "Output device"; ((INSAccessibility)outputGain).AccessibilityLabel = "Output volume";
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((INSAccessibility)auxiliaryDevice).AccessibilityLabel = "Auxiliary input device"; ((INSAccessibility)auxiliaryGain).AccessibilityLabel = "Auxiliary input volume";
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((INSAccessibility)eventVolume).AccessibilityLabel = "Event sound volume";
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((INSAccessibility)audioBuffer).AccessibilityLabel = "Audio buffering";
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}
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private static bool On(NSButton button) => button.State == NSCellStateValue.On;
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private static NSCellStateValue State(bool value) => value ? NSCellStateValue.On : NSCellStateValue.Off;
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@@ -266,6 +266,8 @@ internal sealed class AppModel
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owner.Audio.InputMode = settings.InputMode; owner.Audio.VadThreshold = settings.VadThreshold;
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owner.Audio.InputGain = settings.InputGain; owner.Audio.OutputGain = settings.OutputGain;
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owner.Audio.InputNoiseReduction = settings.InputNoiseReduction;
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owner.Audio.DeviceBufferMilliseconds = settings.AudioBufferMilliseconds;
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IosAudioEngine.Shared.BufferMilliseconds = settings.AudioBufferMilliseconds;
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}
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private void PollAudio()
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@@ -8,7 +8,7 @@ namespace VoiceCat.iOS;
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internal sealed class IosAudioEngine
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{
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internal static IosAudioEngine Shared { get; } = new();
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private readonly PcmRing playbackRing = new(131_072);
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private readonly AdaptivePcmBuffer playbackRing = new(2, capacityFrames: 65_536);
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private readonly short[] renderScratch = new short[16_384];
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private AVAudioEngine? engine;
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private AVAudioSourceNode? source;
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@@ -28,6 +28,7 @@ internal sealed class IosAudioEngine
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private bool inputProvided;
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private bool tapInstalled;
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internal bool IsConnected { get; private set; }
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internal int BufferMilliseconds { get => playbackRing.BufferMilliseconds; set => playbackRing.BufferMilliseconds = value; }
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private IosAudioEngine() { microphoneWorker = PumpMicrophoneAsync(microphoneStop.Token); }
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@@ -17,13 +17,14 @@ internal sealed class IosSettings
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internal float EventVolume { get; set; } = 1f;
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internal bool SelfTalkSounds { get; set; }
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internal bool PushToTalkSound { get; set; }
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internal int AudioBufferMilliseconds { get; set; } = 40;
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internal void Load()
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{
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NSString[] keys = [(NSString)"voice.inputMode", (NSString)"voice.vadThreshold", (NSString)"voice.inputGain",
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(NSString)"voice.outputGain", (NSString)"feedback.sounds", (NSString)"feedback.volume"];
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(NSString)"voice.outputGain", (NSString)"voice.audioBufferMs", (NSString)"feedback.sounds", (NSString)"feedback.volume"];
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NSObject[] defaults = [NSNumber.FromInt32((int)AudioInputMode.VoiceActivation), NSNumber.FromFloat(0.025f),
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NSNumber.FromFloat(1f), NSNumber.FromFloat(1f), NSNumber.FromBoolean(true), NSNumber.FromFloat(1f)];
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NSNumber.FromFloat(1f), NSNumber.FromFloat(1f), NSNumber.FromInt32(40), NSNumber.FromBoolean(true), NSNumber.FromFloat(1f)];
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values.RegisterDefaults(new NSDictionary<NSString, NSObject>(keys, defaults));
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int inputMode = checked((int)values.IntForKey("voice.inputMode"));
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InputMode = Enum.IsDefined(typeof(AudioInputMode), inputMode) ? (AudioInputMode)inputMode : AudioInputMode.VoiceActivation;
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@@ -31,6 +32,7 @@ internal sealed class IosSettings
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InputGain = Math.Clamp(values.FloatForKey("voice.inputGain"), 0f, 4f);
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OutputGain = Math.Clamp(values.FloatForKey("voice.outputGain"), 0f, 1f);
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InputNoiseReduction = values.BoolForKey("voice.inputNoiseReduction");
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int audioBuffer = checked((int)values.IntForKey("voice.audioBufferMs")); AudioBufferMilliseconds = audioBuffer is 20 or 40 or 60 ? audioBuffer : 40;
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EventSounds = values.BoolForKey("feedback.sounds");
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SpokenEvents = values.BoolForKey("feedback.speech");
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EventVolume = Math.Clamp(values.FloatForKey("feedback.volume"), 0f, 1f);
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@@ -42,6 +44,7 @@ internal sealed class IosSettings
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{
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values.SetInt((int)InputMode, "voice.inputMode"); values.SetFloat(VadThreshold, "voice.vadThreshold");
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values.SetFloat(InputGain, "voice.inputGain"); values.SetFloat(OutputGain, "voice.outputGain");
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values.SetInt(AudioBufferMilliseconds, "voice.audioBufferMs");
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values.SetBool(InputNoiseReduction, "voice.inputNoiseReduction"); values.SetBool(EventSounds, "feedback.sounds");
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values.SetBool(SpokenEvents, "feedback.speech"); values.SetFloat(EventVolume, "feedback.volume");
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values.SetBool(SelfTalkSounds, "feedback.selfTalk"); values.SetBool(PushToTalkSound, "feedback.ptt"); values.Synchronize();
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@@ -14,23 +14,23 @@ internal sealed class SettingsController : UITableViewController
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public override void ViewDidLoad() { base.ViewDidLoad(); TableView.RegisterClassForCellReuse(typeof(UITableViewCell), "setting"); }
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private void Reload() => TableView.ReloadData();
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public override nint NumberOfSections(UITableView tableView) => 5;
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public override nint RowsInSection(UITableView tableView, nint section) => section switch { 0 => 4, 1 => 4, 2 => 5, 3 => model.Client?.Permissions is { } p && (p.IsAdmin || p.CanAdminAccounts) ? 1 : 0, _ => 2 };
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public override nint RowsInSection(UITableView tableView, nint section) => section switch { 0 => 5, 1 => 4, 2 => 5, 3 => model.Client?.Permissions is { } p && (p.IsAdmin || p.CanAdminAccounts) ? 1 : 0, _ => 2 };
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public override string? TitleForHeader(UITableView tableView, nint section) => section switch { 0 => "Audio", 1 => "Voice", 2 => "Notifications", 3 => "Administration", _ => "Server" };
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public override UITableViewCell GetCell(UITableView tableView, NSIndexPath path)
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{
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UITableViewCell cell = tableView.DequeueReusableCell("setting", path); cell.AccessoryView = null; cell.Accessory = UITableViewCellAccessory.None;
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string title = path.Section switch
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{
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0 => path.Row switch { 0 => "Audio preset", 1 => "Speaker output", 2 => "Advanced audio", _ => model.ScreenSharing ? "Stop sharing screen audio" : "Share screen audio" },
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0 => path.Row switch { 0 => "Audio preset", 1 => "Speaker output", 2 => "Advanced audio", 3 => "Audio buffering", _ => model.ScreenSharing ? "Stop sharing screen audio" : "Share screen audio" },
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1 => path.Row switch { 0 => "Input mode", 1 => $"VAD threshold: {model.Settings.VadThreshold:0.000}", 2 => $"Microphone volume: {model.Settings.InputGain:P0}", _ => "Microphone noise reduction" },
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2 => path.Row switch { 0 => "Event sounds", 1 => $"Sound volume: {model.Settings.EventVolume:P0}", 2 => "Speak events", 3 => "Own voice activity sounds", _ => "Push-to-talk cue" },
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3 => "Manage accounts", _ => path.Row == 0 ? "Disconnect" : "VoiceCat 0.0.1"
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};
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var content = cell.DefaultContentConfiguration; content.Text = title; content.SecondaryText = path.Section == 0 && path.Row == 0 ? IosAudioRouter.Shared.Preset.ToString() : path.Section == 1 && path.Row == 0 ? model.Settings.InputMode.ToString() : path.Section == 0 && path.Row == 3 && model.ScreenSharing ? "Sharing" : null; cell.ContentConfiguration = content; cell.AccessibilityLabel = title;
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var content = cell.DefaultContentConfiguration; content.Text = title; content.SecondaryText = path.Section == 0 && path.Row == 0 ? IosAudioRouter.Shared.Preset.ToString() : path.Section == 1 && path.Row == 0 ? model.Settings.InputMode.ToString() : path.Section == 0 && path.Row == 3 ? $"{model.Settings.AudioBufferMilliseconds} ms" : path.Section == 0 && path.Row == 4 && model.ScreenSharing ? "Sharing" : null; cell.ContentConfiguration = content; cell.AccessibilityLabel = title + (content.SecondaryText is null ? "" : ", " + content.SecondaryText);
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if (path.Section == 0 && path.Row == 1) cell.AccessoryView = Toggle(IosAudioRouter.Shared.ForceSpeaker, "Speaker output", (_, _) => IosAudioRouter.Shared.SetForceSpeaker(((UISwitch)cell.AccessoryView!).On));
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else if (path.Section == 1 && path.Row == 3) cell.AccessoryView = Toggle(model.Settings.InputNoiseReduction, title, (_, _) => { model.Settings.InputNoiseReduction = ((UISwitch)cell.AccessoryView!).On; model.ApplyVoiceSettings(); });
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else if (path.Section == 2 && path.Row is 0 or 2 or 3 or 4) { bool value = path.Row switch { 0 => model.Settings.EventSounds, 2 => model.Settings.SpokenEvents, 3 => model.Settings.SelfTalkSounds, _ => model.Settings.PushToTalkSound }; int row = path.Row; cell.AccessoryView = Toggle(value, title, (_, _) => { bool on = ((UISwitch)cell.AccessoryView!).On; if (row == 0) model.Settings.EventSounds = on; else if (row == 2) model.Settings.SpokenEvents = on; else if (row == 3) model.Settings.SelfTalkSounds = on; else model.Settings.PushToTalkSound = on; model.Save(); }); }
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else if (path.Section == 0 && path.Row == 2 || path.Section == 3) cell.Accessory = UITableViewCellAccessory.DisclosureIndicator;
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else if (path.Section == 0 && path.Row is 2 or 3 || path.Section == 3) cell.Accessory = UITableViewCellAccessory.DisclosureIndicator;
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return cell;
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}
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public override async void RowSelected(UITableView tableView, NSIndexPath path)
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@@ -39,7 +39,8 @@ internal sealed class SettingsController : UITableViewController
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{
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if (path.Section == 0 && path.Row == 0) Choice("Audio preset", Enum.GetValues<IosAudioPreset>().Select(value => value.ToString()).ToArray(), index => IosAudioRouter.Shared.SelectPreset(Enum.GetValues<IosAudioPreset>()[index]));
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else if (path.Section == 0 && path.Row == 2) NavigationController?.PushViewController(new AdvancedAudioController(), true);
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else if (path.Section == 0 && path.Row == 3) ShowScreenSharing();
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else if (path.Section == 0 && path.Row == 3) Choice("Audio buffering", ["Low latency (20 ms)", "Balanced (40 ms)", "Stable (60 ms)"], index => { model.Settings.AudioBufferMilliseconds = index switch { 0 => 20, 2 => 60, _ => 40 }; model.ApplyVoiceSettings(); });
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else if (path.Section == 0 && path.Row == 4) ShowScreenSharing();
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else if (path.Section == 1 && path.Row == 0) Choice("Input mode", ["Voice activation", "Push to talk", "Always on"], index => { model.Settings.InputMode = (AudioInputMode)index; model.ApplyVoiceSettings(); });
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else if (path.Section == 1 && path.Row == 1) Slider("Voice activation threshold", 0.001f, 0.1f, model.Settings.VadThreshold, value => { model.Settings.VadThreshold = value; model.ApplyVoiceSettings(); });
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else if (path.Section == 1 && path.Row == 2) Slider("Microphone volume", 0, 4, model.Settings.InputGain, value => { model.Settings.InputGain = value; model.ApplyVoiceSettings(); });
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@@ -34,7 +34,7 @@ public sealed class WasapiAudioBackend : IAudioDeviceBackend
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private sealed class Playback : IAudioPlayback
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{
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private readonly string? deviceId;
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private readonly PcmRing pcm = new(32768);
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private readonly AdaptivePcmBuffer pcm = new(2);
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private readonly ManualResetEventSlim initialized = new(false);
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private readonly Thread thread;
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private volatile bool running = true;
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@@ -47,6 +47,7 @@ public sealed class WasapiAudioBackend : IAudioDeviceBackend
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thread.Start();
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if (!initialized.Wait(5000) || !ready) { Dispose(); throw new InvalidOperationException("WASAPI playback could not start."); }
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}
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public int BufferMilliseconds { get => pcm.BufferMilliseconds; set => pcm.BufferMilliseconds = value; }
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public void Write(ReadOnlySpan<short> stereoPcm) => pcm.TryWrite(stereoPcm);
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private unsafe void Work()
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{
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@@ -68,7 +69,6 @@ public sealed class WasapiAudioBackend : IAudioDeviceBackend
|
||||
render = (IAudioRenderClient)output;
|
||||
if (client.Start() < 0) return;
|
||||
ready = true; initialized.Set();
|
||||
Span<short> discard = stackalloc short[1920];
|
||||
while (running)
|
||||
{
|
||||
bufferReady.WaitOne(20); // Scheduling wait is outside the buffer-fill cycle.
|
||||
@@ -77,8 +77,6 @@ public sealed class WasapiAudioBackend : IAudioDeviceBackend
|
||||
if (frames == 0) continue;
|
||||
if (render.GetBuffer(frames, out nint buffer) < 0) break;
|
||||
var destination = new Span<short>((void*)buffer, checked((int)frames * 2));
|
||||
// Keep queued playback bounded to ~120 ms, then fill underflow with silence.
|
||||
while (pcm.Count > 11520) pcm.Read(discard);
|
||||
int count = pcm.Read(destination); destination[count..].Clear();
|
||||
if (render.ReleaseBuffer(frames, 0) < 0) break;
|
||||
}
|
||||
|
||||
@@ -34,6 +34,7 @@ public sealed class AudioSettingsForm : Form
|
||||
private readonly bool _origAuxEnabled;
|
||||
private readonly string? _origAuxDeviceId;
|
||||
private readonly int _origAuxGain;
|
||||
private readonly int _origAudioBufferMilliseconds;
|
||||
|
||||
private readonly ComboBox _cboDevice;
|
||||
private readonly Button _btnRefresh;
|
||||
@@ -54,6 +55,7 @@ public sealed class AudioSettingsForm : Form
|
||||
private readonly Button _btnAuxRefresh;
|
||||
private readonly Label _lblAuxGain;
|
||||
private readonly TrackBar _trkAuxGain;
|
||||
private readonly ComboBox _cboBuffer;
|
||||
|
||||
private Keys _pttKey;
|
||||
|
||||
@@ -80,6 +82,7 @@ public sealed class AudioSettingsForm : Form
|
||||
_origAuxEnabled = settings.AuxEnabled;
|
||||
_origAuxDeviceId = settings.AuxDeviceId;
|
||||
_origAuxGain = settings.AuxGain;
|
||||
_origAudioBufferMilliseconds = settings.AudioBufferMilliseconds;
|
||||
|
||||
Text = "Audio settings";
|
||||
FormBorderStyle = FormBorderStyle.FixedDialog;
|
||||
@@ -87,7 +90,7 @@ public sealed class AudioSettingsForm : Form
|
||||
MinimizeBox = false;
|
||||
StartPosition = FormStartPosition.CenterParent;
|
||||
AutoScaleMode = AutoScaleMode.Font;
|
||||
ClientSize = new Size(420, 639);
|
||||
ClientSize = new Size(420, 690);
|
||||
|
||||
// ── Device row ────────────────────────────────────────────────────────
|
||||
var lblDevice = new Label
|
||||
@@ -319,19 +322,34 @@ public sealed class AudioSettingsForm : Form
|
||||
"Volume of the aux input stream. 100 is unity gain; range 0–400 percent.";
|
||||
_trkAuxGain.Scroll += TrkAuxGain_Scroll;
|
||||
|
||||
var lblBuffer = new Label { Text = "Audio &buffering:", Location = new Point(12, 590), AutoSize = true };
|
||||
_cboBuffer = new ComboBox
|
||||
{
|
||||
Location = new Point(130, 586), Width = 190, DropDownStyle = ComboBoxStyle.DropDownList,
|
||||
AccessibleName = "Audio buffering",
|
||||
AccessibleDescription = "Local capture and playback buffering. Lower values reduce latency; higher values tolerate more scheduling jitter."
|
||||
};
|
||||
_cboBuffer.Items.AddRange(["Low latency (20 ms)", "Balanced (40 ms)", "Stable (60 ms)"]);
|
||||
_cboBuffer.SelectedIndex = settings.AudioBufferMilliseconds switch { 20 => 0, 60 => 2, _ => 1 };
|
||||
_cboBuffer.SelectedIndexChanged += (_, _) =>
|
||||
{
|
||||
_settings.AudioBufferMilliseconds = _cboBuffer.SelectedIndex switch { 0 => 20, 2 => 60, _ => 40 };
|
||||
_client.SetAudioBufferMilliseconds(_settings.AudioBufferMilliseconds);
|
||||
};
|
||||
|
||||
// ── OK / Cancel ───────────────────────────────────────────────────────
|
||||
var btnOk = new Button
|
||||
{
|
||||
Text = "&OK",
|
||||
DialogResult = DialogResult.OK,
|
||||
Location = new Point(228, 600),
|
||||
Location = new Point(228, 642),
|
||||
Size = new Size(80, 27),
|
||||
};
|
||||
var btnCancel = new Button
|
||||
{
|
||||
Text = "&Cancel",
|
||||
DialogResult = DialogResult.Cancel,
|
||||
Location = new Point(316, 600),
|
||||
Location = new Point(316, 642),
|
||||
Size = new Size(80, 27),
|
||||
};
|
||||
|
||||
@@ -350,6 +368,7 @@ public sealed class AudioSettingsForm : Form
|
||||
_radioPtt, _lblPttKey, _btnChangePtt, _chkSystemWidePtt, _radioAlwaysOn,
|
||||
lblGain, _trkGain, _chkNoiseReduction, _chkStereoMic,
|
||||
_chkAux, _lblAuxDevice, _cboAuxDevice, _btnAuxRefresh, _lblAuxGain, _trkAuxGain,
|
||||
lblBuffer, _cboBuffer,
|
||||
btnOk, btnCancel,
|
||||
]);
|
||||
|
||||
@@ -534,6 +553,8 @@ public sealed class AudioSettingsForm : Form
|
||||
_settings.StereoMic = _origStereoMic;
|
||||
_settings.PttKey = (int)_origPttKey;
|
||||
_settings.SystemWidePtt = _origSystemWidePtt;
|
||||
_settings.AudioBufferMilliseconds = _origAudioBufferMilliseconds;
|
||||
_client.SetAudioBufferMilliseconds(_origAudioBufferMilliseconds);
|
||||
|
||||
if (_micStreamId != 0)
|
||||
{
|
||||
|
||||
@@ -109,8 +109,11 @@ public partial class MainForm : Form
|
||||
BootstrapFromServer();
|
||||
}
|
||||
|
||||
private void ApplyPersistedVoiceSettings() =>
|
||||
private void ApplyPersistedVoiceSettings()
|
||||
{
|
||||
_pttKey = (Keys)_voiceSettings.PttKey;
|
||||
_client.SetAudioBufferMilliseconds(_voiceSettings.AudioBufferMilliseconds);
|
||||
}
|
||||
|
||||
// ── Startup ──────────────────────────────────────────────────────────────
|
||||
|
||||
|
||||
@@ -59,6 +59,9 @@ public sealed class VoiceSettings
|
||||
/// to the captured PCM before feeding (the core's input gain is mic-only and global).</summary>
|
||||
public int AuxGain { get; set; } = 100;
|
||||
|
||||
/// <summary>Local capture/playback safety buffer in milliseconds: 20, 40, or 60.</summary>
|
||||
public int AudioBufferMilliseconds { get; set; } = 40;
|
||||
|
||||
private static readonly JsonSerializerOptions JsonOptions = new() { WriteIndented = true };
|
||||
|
||||
private static string AppDataDir => Path.Combine(
|
||||
@@ -72,7 +75,9 @@ public sealed class VoiceSettings
|
||||
{
|
||||
if (!File.Exists(FilePath)) return new VoiceSettings();
|
||||
string json = File.ReadAllText(FilePath);
|
||||
return JsonSerializer.Deserialize<VoiceSettings>(json) ?? new VoiceSettings();
|
||||
VoiceSettings result = JsonSerializer.Deserialize<VoiceSettings>(json) ?? new VoiceSettings();
|
||||
if (result.AudioBufferMilliseconds is not (20 or 40 or 60)) result.AudioBufferMilliseconds = 40;
|
||||
return result;
|
||||
}
|
||||
catch
|
||||
{
|
||||
|
||||
@@ -55,6 +55,16 @@ public sealed partial class VoiceCatClient
|
||||
public VcResult SetOutputVolume(float gain) { if (!float.IsFinite(gain) || gain is < 0 or > 4) return VcResult.InvalidArg; core.Audio.OutputGain = gain; return VcResult.Ok; }
|
||||
public VcResult SetInputGain(float gain) { if (!float.IsFinite(gain) || gain is < 0 or > 4) return VcResult.InvalidArg; core.Audio.InputGain = gain; return VcResult.Ok; }
|
||||
public VcResult SetInputNoiseReduction(bool enabled) { core.Audio.InputNoiseReduction = enabled; return VcResult.Ok; }
|
||||
public VcResult SetAudioBufferMilliseconds(int milliseconds)
|
||||
{
|
||||
try
|
||||
{
|
||||
core.Audio.DeviceBufferMilliseconds = milliseconds;
|
||||
if (Volatile.Read(ref playback) is { } output) output.BufferMilliseconds = milliseconds;
|
||||
return VcResult.Ok;
|
||||
}
|
||||
catch (ArgumentOutOfRangeException) { return VcResult.InvalidArg; }
|
||||
}
|
||||
public VcResult SetRemoteStream(uint userId, uint streamId, float gain, bool muted, bool nr)
|
||||
{ try { core.Audio.SetRemotePlayback(userId, streamId, gain, muted, nr); return VcResult.Ok; } catch { return VcResult.InvalidArg; } }
|
||||
public (VcResult Result, RemoteStreamState? State) GetRemoteStream(uint userId, uint streamId)
|
||||
|
||||
@@ -159,7 +159,11 @@ public sealed partial class VoiceCatClient : IDisposable
|
||||
{
|
||||
try
|
||||
{
|
||||
if (backend is not null && playback is null) playback = backend.OpenPlayback();
|
||||
if (backend is not null && playback is null)
|
||||
{
|
||||
playback = backend.OpenPlayback();
|
||||
playback.BufferMilliseconds = core.Audio.DeviceBufferMilliseconds;
|
||||
}
|
||||
var result = core.SubscribeVoiceAsync().GetAwaiter().GetResult();
|
||||
if (!result.Ok) { Interlocked.Exchange(ref playback, null)?.Dispose(); }
|
||||
return result.Ok ? VcResult.Ok : VcResult.Audio;
|
||||
|
||||
+4
-2
@@ -177,8 +177,10 @@ Noise reduction does not gate speech.
|
||||
`TimeProvider` timestamps; tests inject a clock. Threshold changes are atomic; all
|
||||
processing state otherwise has one owner. Codec/DSP processing methods allocate no
|
||||
managed memory after initialization, verified across 1,000 combined cycles. They run
|
||||
on a managed worker, never the native real-time device callback. Native device rings,
|
||||
jitter, mixer, and audio scheduling remain later work.
|
||||
on a managed worker, never the native real-time device callback. Managed capture/playback
|
||||
rings compensate independent clock drift around a configurable 20/40/60 ms target. Receive
|
||||
jitter is duration-aware, reserves one codec frame for DRED/FEC look-ahead, and decodes each
|
||||
channel packet at its actual duration before the fixed 20 ms mixer stage.
|
||||
|
||||
## Initial managed server
|
||||
|
||||
|
||||
+19
-7
@@ -117,6 +117,9 @@ Guidance baked into defaults / docs:
|
||||
(accumulating two 960-frames for a 40 ms channel, splitting each into two 480-frames for a
|
||||
10 ms channel, etc.). This keeps the hardware/`vc_stream_feed_pcm` contract a single 48 kHz
|
||||
clock regardless of the channel's window — see `vc_client::on_capture_frame`.
|
||||
The device/mixer quantum is not the Opus packet duration: managed send streams reframe it
|
||||
into the channel's 5/10/20/40/60 ms packets, and receive streams decode at that duration
|
||||
before slicing decoded PCM back into 20 ms mixer blocks.
|
||||
- **Mode/bitrate:** speech channels → `MONO`, `VOIP`, 24–32 kbps, DTX on, FEC on.
|
||||
Music/screen-audio channels → `STEREO`, `AUDIO`, 96–128 kbps, DTX off, FEC optional.
|
||||
- **`application`:** `VOIP` for talk, `AUDIO` for music/screen-share, `LOWDELAY` for
|
||||
@@ -152,26 +155,30 @@ straight through to PLC.
|
||||
## 5. Jitter buffer
|
||||
|
||||
Each receiver keeps an **adaptive jitter buffer per ssrc** with **bounded-depth playout**
|
||||
(`core/src/audio/audio_engine.cpp` — `JitterBuffer` + `AudioEngine::on_playback`).
|
||||
(`dotnet/src/VoiceCat.Audio/ReceiveStream.cs`).
|
||||
|
||||
- Frames are inserted by `timestamp`; playback reads in order at the device callback rate.
|
||||
- **The playout clock is always bounded against the stream's *leading edge* (newest buffered
|
||||
frame), never re-synced to the oldest.** The clock free-runs at the playback hardware rate,
|
||||
frame), never re-synced to the oldest.** The managed playout clock is isolated from hardware
|
||||
drift by the adaptive PCM ring,
|
||||
while the sender omits VAD/PTT/DTX silence from its timestamps, so the two diverge across gaps
|
||||
and late joins. Two corrections keep latency bounded:
|
||||
- **(Re)seed to the leading edge** on first frame, on a talkspurt `marker`, or when the clock
|
||||
has run past the newest frame (starved after silence). No artificial prebuffer — latency
|
||||
starts as low as possible; buffered frames still play oldest-first.
|
||||
has run past the newest frame (starved after silence). Playout retains the adaptive target;
|
||||
when DRED/FEC is enabled that includes one codec frame of recovery look-ahead.
|
||||
- **Frame-skip catch-up:** when the backlog grows past `target + hysteresis` (clock drift,
|
||||
bursty arrival, reordering), fast-forward the clock to leave `target` buffered and drop the
|
||||
now-stale frames. This is the downward force that prevents latency from ratcheting upward.
|
||||
- `target` is the adaptive jitter estimate (EWMA of inter-arrival gap vs. the per-frame gap),
|
||||
floored; silence gaps and reordered stragglers are rejected as outliers so they don't inflate
|
||||
it. The late-drop window tracks `target` (floored/capped at 500 ms).
|
||||
- `target` is an EWMA of arrival-gap variation measured in 48 kHz sample time. DRED or FEC
|
||||
reserves one complete channel Opus frame of look-ahead, variation can raise the target to
|
||||
120 ms, and packet history is bounded to 500 ms. Limits are durations rather than packet
|
||||
counts, so 5 ms and 60 ms channels receive the same policy.
|
||||
- Late frames past the playout point are dropped; gaps are filled by DRED (if the next frame
|
||||
arrived) or PLC.
|
||||
- The `marker` flag (start of talkspurt) — set by the sender on the first frame after a
|
||||
transmission gap — lets the buffer reseed cleanly after silence/DTX without accumulating drift.
|
||||
- One late device callback does not end a talkspurt. Capture resets only after 200 ms of
|
||||
continuous starvation, avoiding a marker/rebuffer cascade from an isolated scheduling miss.
|
||||
- Diagnostics per stream: `packets_lost`, `duplicates`, `underruns`, `target_depth_ms`.
|
||||
|
||||
```
|
||||
@@ -180,6 +187,11 @@ Each receiver keeps an **adaptive jitter buffer per ssrc** with **bounded-depth
|
||||
jitter estimate feeds depth
|
||||
```
|
||||
|
||||
Capture and playback use an allocation-free adaptive PCM ring between the managed 20 ms clock
|
||||
and the hardware clock. Linear-interpolation correction, limited to ±0.5%, holds the ring near
|
||||
its target instead of periodically dropping a block or rendering silence as device clocks drift.
|
||||
The local **Audio buffering** preset is 20, 40 (default), or 60 ms and is persisted per client.
|
||||
|
||||
## 6. UDP keepalive & NAT
|
||||
|
||||
- A `KEEPALIVE` (type 2) frame flows both directions on the media channel every ~5 s to
|
||||
|
||||
@@ -0,0 +1,97 @@
|
||||
namespace VoiceCat.Audio;
|
||||
|
||||
// SPSC PCM handoff with a small occupancy-controlled sample-rate correction. Producers and
|
||||
// consumers remain non-waiting; the correction keeps independent hardware and managed clocks
|
||||
// from periodically reaching the hard underflow/overflow edges of a conventional ring.
|
||||
public sealed class AdaptivePcmBuffer
|
||||
{
|
||||
private const int SampleRate = 48_000;
|
||||
private const double MaximumCorrection = 0.005;
|
||||
private readonly short[] samples;
|
||||
private readonly int channels, frameMask;
|
||||
private int readFrame, writtenFrame, producer, targetFrames;
|
||||
private double phase;
|
||||
private bool primed;
|
||||
|
||||
public AdaptivePcmBuffer(int channels, int bufferMilliseconds = 40, int capacityFrames = 16_384)
|
||||
{
|
||||
if (channels is not (1 or 2)) throw new ArgumentOutOfRangeException(nameof(channels));
|
||||
if (capacityFrames < 2 || (capacityFrames & (capacityFrames - 1)) != 0) throw new ArgumentOutOfRangeException(nameof(capacityFrames));
|
||||
this.channels = channels; samples = new short[checked(capacityFrames * channels)]; frameMask = capacityFrames - 1;
|
||||
BufferMilliseconds = bufferMilliseconds;
|
||||
}
|
||||
|
||||
public int Channels => channels;
|
||||
public int CountFrames => unchecked(Volatile.Read(ref writtenFrame) - Volatile.Read(ref readFrame));
|
||||
public int BufferMilliseconds
|
||||
{
|
||||
get => Volatile.Read(ref targetFrames) * 1000 / SampleRate;
|
||||
set
|
||||
{
|
||||
if (value is not (20 or 40 or 60)) throw new ArgumentOutOfRangeException(nameof(value));
|
||||
Volatile.Write(ref targetFrames, value * SampleRate / 1000);
|
||||
}
|
||||
}
|
||||
|
||||
public bool TryWrite(ReadOnlySpan<short> source)
|
||||
{
|
||||
if (source.Length == 0 || source.Length % channels != 0 || Interlocked.CompareExchange(ref producer, 1, 0) != 0) return false;
|
||||
try
|
||||
{
|
||||
int frames = source.Length / channels, index = writtenFrame;
|
||||
if (frames > frameMask + 1 - unchecked(index - Volatile.Read(ref readFrame))) return false;
|
||||
for (int frame = 0; frame < frames; frame++)
|
||||
{
|
||||
int target = ((index + frame) & frameMask) * channels;
|
||||
for (int channel = 0; channel < channels; channel++) samples[target + channel] = source[frame * channels + channel];
|
||||
}
|
||||
Volatile.Write(ref writtenFrame, unchecked(index + frames)); return true;
|
||||
}
|
||||
finally { Volatile.Write(ref producer, 0); }
|
||||
}
|
||||
|
||||
// Returns interleaved samples written. A zero return means the caller should treat the
|
||||
// already-cleared destination as silence. Once primed, short scheduling stalls re-prime
|
||||
// instead of repeatedly clicking at the ring edge.
|
||||
public int Read(Span<short> destination)
|
||||
{
|
||||
if (destination.Length % channels != 0) throw new ArgumentException("PCM must contain complete interleaved frames.", nameof(destination));
|
||||
int requestedFrames = destination.Length / channels;
|
||||
if (requestedFrames == 0) return 0;
|
||||
int available = CountFrames, target = Volatile.Read(ref targetFrames);
|
||||
if (!primed)
|
||||
{
|
||||
if (available < target) { destination.Clear(); return 0; }
|
||||
primed = true; phase = 0;
|
||||
}
|
||||
if (available <= 0) { primed = false; phase = 0; destination.Clear(); return 0; }
|
||||
|
||||
double correction = Math.Clamp((available - target) / (SampleRate * 2.0), -MaximumCorrection, MaximumCorrection);
|
||||
double step = 1.0 + correction;
|
||||
int produced = 0, read = readFrame;
|
||||
for (int frame = 0; frame < requestedFrames; frame++)
|
||||
{
|
||||
int baseOffset = (read & frameMask) * channels;
|
||||
int nextOffset = ((read + 1) & frameMask) * channels;
|
||||
int remaining = unchecked(Volatile.Read(ref writtenFrame) - read);
|
||||
if (remaining <= 0) break;
|
||||
double fraction = phase;
|
||||
for (int channel = 0; channel < channels; channel++)
|
||||
{
|
||||
int first = samples[baseOffset + channel];
|
||||
int second = remaining > 1 ? samples[nextOffset + channel] : first;
|
||||
destination[produced++] = (short)Math.Clamp((int)Math.Round(first + (second - first) * fraction), short.MinValue, short.MaxValue);
|
||||
}
|
||||
phase += step;
|
||||
int advance = (int)phase;
|
||||
if (advance > remaining) advance = remaining;
|
||||
read = unchecked(read + advance); phase -= advance;
|
||||
}
|
||||
Volatile.Write(ref readFrame, read);
|
||||
if (produced < destination.Length)
|
||||
{
|
||||
destination[produced..].Clear(); primed = false; phase = 0;
|
||||
}
|
||||
return produced;
|
||||
}
|
||||
}
|
||||
@@ -3,7 +3,11 @@ namespace VoiceCat.Audio;
|
||||
public sealed record AudioDeviceInfo(string Id, string Name, bool IsDefault);
|
||||
public delegate void CapturePcmHandler(ReadOnlySpan<short> pcm, int channels);
|
||||
public interface IAudioCapture : IDisposable { }
|
||||
public interface IAudioPlayback : IDisposable { void Write(ReadOnlySpan<short> stereoPcm); }
|
||||
public interface IAudioPlayback : IDisposable
|
||||
{
|
||||
int BufferMilliseconds { get; set; }
|
||||
void Write(ReadOnlySpan<short> stereoPcm);
|
||||
}
|
||||
public interface IAudioDeviceBackend
|
||||
{
|
||||
IReadOnlyList<AudioDeviceInfo> Enumerate(bool input);
|
||||
|
||||
@@ -23,6 +23,17 @@ public sealed class AudioEngine : IDisposable
|
||||
public volatile float InputGain = 1, OutputGain = 1, VadThreshold = 0.02f;
|
||||
public volatile bool InputNoiseReduction, MicMuted, Deafened, PushToTalk;
|
||||
public volatile AudioInputMode InputMode = AudioInputMode.VoiceActivation;
|
||||
private int deviceBufferMilliseconds = 40;
|
||||
public int DeviceBufferMilliseconds
|
||||
{
|
||||
get => Volatile.Read(ref deviceBufferMilliseconds);
|
||||
set
|
||||
{
|
||||
if (value is not (20 or 40 or 60)) throw new ArgumentOutOfRangeException(nameof(value));
|
||||
Volatile.Write(ref deviceBufferMilliseconds, value);
|
||||
foreach (LocalStream stream in Volatile.Read(ref routes).Local) stream.BufferMilliseconds = value;
|
||||
}
|
||||
}
|
||||
public Exception? Failure { get; private set; }
|
||||
|
||||
public AudioEngine(EncodedVoiceSender sender, bool startWorker = true)
|
||||
@@ -42,7 +53,7 @@ public sealed class AudioEngine : IDisposable
|
||||
lock (gate)
|
||||
{
|
||||
ObjectDisposedException.ThrowIf(disposed != 0, this);
|
||||
var stream = new LocalStream(info, captureChannels);
|
||||
var stream = new LocalStream(info, captureChannels, DeviceBufferMilliseconds);
|
||||
Routes previous = routes;
|
||||
var locals = previous.Local.Where(s => s.Info.StreamId != info.StreamId).Append(stream).ToArray();
|
||||
Publish(locals, previous.Remote);
|
||||
@@ -58,7 +69,7 @@ public sealed class AudioEngine : IDisposable
|
||||
lock (gate)
|
||||
{
|
||||
var current = routes.Local.FirstOrDefault(s => s.Info.StreamId == streamId) ?? throw new ArgumentException("Stream not found.");
|
||||
if (current.CaptureChannels != channels) Publish(routes.Local.Select(s => s == current ? new LocalStream(s.Info, channels) : s).ToArray(), routes.Remote);
|
||||
if (current.CaptureChannels != channels) Publish(routes.Local.Select(s => s == current ? new LocalStream(s.Info, channels, DeviceBufferMilliseconds) : s).ToArray(), routes.Remote);
|
||||
}
|
||||
}
|
||||
|
||||
|
||||
@@ -15,7 +15,7 @@ internal sealed class LocalStream : IDisposable
|
||||
{
|
||||
internal readonly StreamInfo Info;
|
||||
internal readonly int CaptureChannels;
|
||||
internal readonly PcmRing Input = new(16384);
|
||||
internal readonly AdaptivePcmBuffer Input;
|
||||
internal volatile float Level;
|
||||
internal volatile bool Talking;
|
||||
internal volatile float Gain = 1;
|
||||
@@ -27,6 +27,7 @@ internal sealed class LocalStream : IDisposable
|
||||
private readonly short[] converted = new short[16384];
|
||||
private int feeding;
|
||||
private int buffered;
|
||||
private int starvedSamples;
|
||||
private uint timestamp;
|
||||
private bool wasTransmitting, marker;
|
||||
|
||||
@@ -45,9 +46,12 @@ internal sealed class LocalStream : IDisposable
|
||||
finally { Volatile.Write(ref feeding, 0); }
|
||||
}
|
||||
|
||||
internal LocalStream(StreamInfo stream, int captureChannels)
|
||||
internal int BufferMilliseconds { get => Input.BufferMilliseconds; set => Input.BufferMilliseconds = value; }
|
||||
|
||||
internal LocalStream(StreamInfo stream, int captureChannels, int bufferMilliseconds = 40)
|
||||
{
|
||||
Info = stream.Clone(); CaptureChannels = captureChannels;
|
||||
Input = new(captureChannels, bufferMilliseconds);
|
||||
encoder = new(new()
|
||||
{
|
||||
Channels = stream.Audio.Mode == ChannelMode.ModeStereo ? 2 : 1,
|
||||
@@ -64,10 +68,14 @@ internal sealed class LocalStream : IDisposable
|
||||
internal void Process(AudioEngine engine, EncodedVoiceSender sender)
|
||||
{
|
||||
var input = capture.AsSpan(0, 960 * CaptureChannels);
|
||||
if (Input.Count < input.Length) { Level = 0; Talking = false; buffered = 0; wasTransmitting = false; return; }
|
||||
while (Input.Count > input.Length * 6) Input.Read(input);
|
||||
if (Input.Read(input) != input.Length)
|
||||
{
|
||||
Level = 0; Talking = false; starvedSamples += 960;
|
||||
if (starvedSamples >= 9600) { buffered = 0; wasTransmitting = false; }
|
||||
return;
|
||||
}
|
||||
starvedSamples = 0;
|
||||
if (buffered == 0) timestamp = engine.SampleClock;
|
||||
Input.Read(input);
|
||||
bool mic = Info.Kind == StreamKind.StreamMic;
|
||||
if (mic && engine.InputNoiseReduction)
|
||||
{
|
||||
|
||||
@@ -20,28 +20,36 @@ internal sealed class ReceiveStream : IDisposable
|
||||
private readonly byte[][] packets = Enumerable.Range(0, 64).Select(_ => new byte[1275]).ToArray();
|
||||
private readonly VoiceFrameHeader[] headers = new VoiceFrameHeader[64];
|
||||
private readonly int[] lengths = new int[64];
|
||||
private readonly long[] arrivals = new long[64];
|
||||
private int read, written;
|
||||
private readonly byte[][] jitter = Enumerable.Range(0, 6).Select(_ => new byte[1275]).ToArray();
|
||||
private readonly uint[] timestamps = new uint[6];
|
||||
private readonly int[] sizes = new int[6];
|
||||
private readonly byte[][] jitter;
|
||||
private readonly uint[] timestamps;
|
||||
private readonly int[] sizes;
|
||||
private int count, available, offset, missing, waiting;
|
||||
private uint expected;
|
||||
private bool started, hasTimestamp;
|
||||
private bool hasMarker;
|
||||
private uint lastMarker;
|
||||
private readonly int channels, frameSamples, maximumDepth;
|
||||
private long lastArrival;
|
||||
private uint lastArrivalTimestamp;
|
||||
private double jitterSamples;
|
||||
private readonly TimeProvider clock;
|
||||
internal int Depth => count;
|
||||
internal int TargetDepthSamples => TargetSamples();
|
||||
internal int ConcealedFrames { get; private set; }
|
||||
internal int DredFrames { get; private set; }
|
||||
internal int FecFrames { get; private set; }
|
||||
|
||||
internal ReceiveStream(uint userId, StreamInfo info)
|
||||
internal ReceiveStream(uint userId, StreamInfo info, TimeProvider? clock = null)
|
||||
{
|
||||
if (info.Audio.FrameMs is not (5 or 10 or 20 or 40 or 60) || !Enum.IsDefined(info.Audio.Mode)) throw new ArgumentException("Unsupported remote audio configuration.", nameof(info));
|
||||
UserId = userId; Info = info.Clone();
|
||||
UserId = userId; Info = info.Clone(); this.clock = clock ?? TimeProvider.System;
|
||||
channels = info.Audio.Mode == ChannelMode.ModeStereo ? 2 : 1;
|
||||
frameSamples = checked((int)info.Audio.FrameMs * 48);
|
||||
maximumDepth = Math.Clamp(120 / (int)info.Audio.FrameMs, 2, 6);
|
||||
maximumDepth = Math.Clamp(500 / (int)info.Audio.FrameMs + 2, 8, 104);
|
||||
jitter = Enumerable.Range(0, maximumDepth).Select(_ => new byte[1275]).ToArray();
|
||||
timestamps = new uint[maximumDepth]; sizes = new int[maximumDepth];
|
||||
decoder = new(48000, channels);
|
||||
try { left = new(); } catch { decoder.Dispose(); throw; }
|
||||
try { right = new(); } catch { left.Dispose(); decoder.Dispose(); throw; }
|
||||
@@ -53,7 +61,7 @@ internal sealed class ReceiveStream : IDisposable
|
||||
{
|
||||
int index = written;
|
||||
if (payload.Length is < 1 or > 1275 || unchecked(index - Volatile.Read(ref read)) >= 64) return false;
|
||||
int slot = index & 63; payload.CopyTo(packets[slot]); headers[slot] = header; lengths[slot] = payload.Length;
|
||||
int slot = index & 63; payload.CopyTo(packets[slot]); headers[slot] = header; lengths[slot] = payload.Length; arrivals[slot] = clock.GetTimestamp();
|
||||
Volatile.Write(ref written, unchecked(index + 1)); return true;
|
||||
}
|
||||
|
||||
@@ -67,8 +75,8 @@ internal sealed class ReceiveStream : IDisposable
|
||||
if ((headers[source].Flags & VoiceFrameFlags.Marker) != 0 && (!hasMarker || unchecked((int)(timestamp - lastMarker)) > 0))
|
||||
{
|
||||
hasMarker = true; lastMarker = timestamp;
|
||||
sizes.AsSpan().Clear(); count = available = offset = missing = waiting = 0;
|
||||
expected = timestamp; started = false; delta = 0;
|
||||
DropBefore(timestamp); available = offset = missing = waiting = 0;
|
||||
expected = timestamp; started = false; delta = 0; lastArrival = 0; jitterSamples = 0;
|
||||
}
|
||||
bool duplicate = false;
|
||||
for (int i = 0; i < sizes.Length; i++) if (sizes[i] != 0 && timestamps[i] == timestamp) duplicate = true;
|
||||
@@ -79,14 +87,61 @@ internal sealed class ReceiveStream : IDisposable
|
||||
int oldest = Oldest(); sizes[oldest] = 0; count--;
|
||||
}
|
||||
int target = Array.IndexOf(sizes, 0);
|
||||
timestamps[target] = timestamp; sizes[target] = lengths[source]; packets[source].AsSpan(0, lengths[source]).CopyTo(jitter[target]); count++;
|
||||
int oldestRemaining = Oldest();
|
||||
if (count >= maximumDepth && unchecked((int)(timestamps[oldestRemaining] - expected)) > 0) expected = timestamps[oldestRemaining];
|
||||
timestamps[target] = timestamp; sizes[target] = lengths[source];
|
||||
packets[source].AsSpan(0, lengths[source]).CopyTo(jitter[target]); count++;
|
||||
ObserveArrival(timestamp, arrivals[source]);
|
||||
}
|
||||
Volatile.Write(ref read, unchecked(read + 1));
|
||||
}
|
||||
}
|
||||
|
||||
private void ObserveArrival(uint timestamp, long arrival)
|
||||
{
|
||||
if (lastArrival != 0)
|
||||
{
|
||||
int timestampDelta = unchecked((int)(timestamp - lastArrivalTimestamp));
|
||||
if (timestampDelta <= 0) return;
|
||||
if (timestampDelta > 0 && timestampDelta <= frameSamples * 10)
|
||||
{
|
||||
double arrivalDelta = clock.GetElapsedTime(lastArrival, arrival).TotalSeconds * 48_000;
|
||||
double deviation = Math.Abs(arrivalDelta - timestampDelta);
|
||||
jitterSamples += (deviation - jitterSamples) / 16.0;
|
||||
}
|
||||
}
|
||||
lastArrival = arrival; lastArrivalTimestamp = timestamp;
|
||||
}
|
||||
|
||||
private int TargetSamples()
|
||||
{
|
||||
int recovery = Info.Audio.Dred || Info.Audio.Fec ? frameSamples : 0;
|
||||
int variation = checked((int)Math.Ceiling(4 * jitterSamples / frameSamples)) * frameSamples;
|
||||
return Math.Min(5760, recovery + variation);
|
||||
}
|
||||
|
||||
private int Newest()
|
||||
{
|
||||
int newest = -1;
|
||||
for (int i = 0; i < sizes.Length; i++) if (sizes[i] != 0 && (newest < 0 || unchecked((int)(timestamps[i] - timestamps[newest])) > 0)) newest = i;
|
||||
return newest;
|
||||
}
|
||||
|
||||
private void DropBefore(uint timestamp)
|
||||
{
|
||||
for (int i = 0; i < sizes.Length; i++)
|
||||
if (sizes[i] != 0 && unchecked((int)(timestamps[i] - timestamp)) < 0) { sizes[i] = 0; count--; }
|
||||
}
|
||||
|
||||
private void CatchUp()
|
||||
{
|
||||
if (!started || available != 0 || count == 0) return;
|
||||
int newest = Newest(); int target = TargetSamples();
|
||||
int lead = unchecked((int)(timestamps[newest] - expected));
|
||||
if (lead <= target + frameSamples) return;
|
||||
int keepBehind = target / frameSamples * frameSamples;
|
||||
expected = unchecked(timestamps[newest] - (uint)keepBehind);
|
||||
DropBefore(expected); missing = 0;
|
||||
}
|
||||
|
||||
private int Oldest()
|
||||
{
|
||||
int oldest = -1;
|
||||
@@ -108,11 +163,12 @@ internal sealed class ReceiveStream : IDisposable
|
||||
else
|
||||
{
|
||||
int next = Oldest(); missing++;
|
||||
if (next >= 0 && unchecked((int)(timestamps[next] - expected)) == frameSamples)
|
||||
int recoveryOffset = next < 0 ? 0 : unchecked((int)(timestamps[next] - expected));
|
||||
if (next >= 0 && recoveryOffset > 0 && recoveryOffset % frameSamples == 0)
|
||||
{
|
||||
var packet = jitter[next].AsSpan(0, sizes[next]);
|
||||
if (dred?.TryRecover(decoder, packet, pcm, frameSamples) == true) { decoded = true; DredFrames++; }
|
||||
else if (Info.Audio.Fec && decoder.TryDecode(packet, pcm, frameSamples, out int recovered, true) && recovered == frameSamples) { decoded = true; FecFrames++; }
|
||||
if (dred?.TryRecover(decoder, packet, pcm, frameSamples, recoveryOffset) == true) { decoded = true; DredFrames++; }
|
||||
else if (recoveryOffset == frameSamples && Info.Audio.Fec && decoder.TryDecode(packet, pcm, frameSamples, out int recovered, true) && recovered == frameSamples) { decoded = true; FecFrames++; }
|
||||
}
|
||||
}
|
||||
int maximumConcealment = Math.Max(1, 200 / (int)Info.Audio.FrameMs);
|
||||
@@ -127,11 +183,16 @@ internal sealed class ReceiveStream : IDisposable
|
||||
internal void Mix(Span<int> output, bool deafened, PcmStreamHandler? sink)
|
||||
{
|
||||
Drain();
|
||||
CatchUp();
|
||||
if (!started)
|
||||
{
|
||||
waiting++;
|
||||
if (!hasTimestamp || count < Math.Min(3, maximumDepth) && waiting < 3) return;
|
||||
expected = timestamps[Oldest()]; started = true;
|
||||
if (!hasTimestamp || count == 0) return;
|
||||
int oldest = Oldest(), newest = Newest();
|
||||
if (unchecked((int)(timestamps[newest] - timestamps[oldest])) < TargetSamples()) return;
|
||||
int keepBehind = TargetSamples() / frameSamples * frameSamples;
|
||||
expected = unchecked(timestamps[newest] - (uint)keepBehind);
|
||||
DropBefore(expected); started = true; waiting = 0;
|
||||
}
|
||||
int copied = 0;
|
||||
while (copied < 960)
|
||||
|
||||
@@ -7,6 +7,84 @@ namespace VoiceCat.Tests;
|
||||
|
||||
public class AudioEngineTests
|
||||
{
|
||||
[Theory]
|
||||
[InlineData(-1000)]
|
||||
[InlineData(1000)]
|
||||
public void AdaptivePcmBufferAbsorbsIndependentClockDrift(int partsPerMillion)
|
||||
{
|
||||
var buffer = new AdaptivePcmBuffer(1, 40); short[] input = new short[962], output = new short[960];
|
||||
input.AsSpan().Fill(1234); Assert.True(buffer.TryWrite(input.AsSpan(0, 960)));
|
||||
Assert.True(buffer.TryWrite(input.AsSpan(0, 960)));
|
||||
double produced = 0;
|
||||
for (int cycle = 0; cycle < 10_000; cycle++)
|
||||
{
|
||||
produced += 960 * (1 + partsPerMillion / 1_000_000.0);
|
||||
int frames = (int)produced; produced -= frames;
|
||||
Assert.True(buffer.TryWrite(input.AsSpan(0, frames)));
|
||||
Assert.Equal(output.Length, buffer.Read(output));
|
||||
}
|
||||
Assert.InRange(buffer.CountFrames, 480, 3840);
|
||||
long before = GC.GetAllocatedBytesForCurrentThread();
|
||||
for (int i = 0; i < 100; i++) { buffer.TryWrite(input.AsSpan(0, 960)); buffer.Read(output); }
|
||||
Assert.Equal(0, GC.GetAllocatedBytesForCurrentThread() - before);
|
||||
}
|
||||
|
||||
[Fact]
|
||||
public void OneCaptureMissDoesNotRestartTalkspurtButSustainedStarvationDoes()
|
||||
{
|
||||
var sent = new List<(uint Timestamp, VoiceFrameFlags Flags)>();
|
||||
using var engine = new AudioEngine((_, timestamp, _, flags) => { sent.Add((timestamp, flags)); return true; }, false)
|
||||
{ InputMode = AudioInputMode.AlwaysOn, DeviceBufferMilliseconds = 20 };
|
||||
engine.AddLocalStream(Stream()); short[] tone = Tone();
|
||||
engine.FeedPcm(1, tone, 1); engine.ProcessCycle();
|
||||
engine.ProcessCycle();
|
||||
engine.FeedPcm(1, tone, 1); engine.ProcessCycle();
|
||||
Assert.Equal(2, sent.Count); Assert.True((sent[0].Flags & VoiceFrameFlags.Marker) != 0); Assert.Equal(VoiceFrameFlags.None, sent[1].Flags & VoiceFrameFlags.Marker);
|
||||
for (int i = 0; i < 10; i++) engine.ProcessCycle();
|
||||
engine.FeedPcm(1, tone, 1); engine.ProcessCycle();
|
||||
Assert.True((sent[^1].Flags & VoiceFrameFlags.Marker) != 0);
|
||||
}
|
||||
|
||||
[Theory]
|
||||
[InlineData(5)] [InlineData(10)] [InlineData(20)] [InlineData(40)] [InlineData(60)]
|
||||
public void RecoveryLookaheadTracksChannelFrameDuration(int frameMilliseconds)
|
||||
{
|
||||
using var stream = new ReceiveStream(2, Stream(frameMilliseconds));
|
||||
Assert.Equal(frameMilliseconds * 48, stream.TargetDepthSamples);
|
||||
}
|
||||
|
||||
[Fact]
|
||||
public void JitterTargetAdaptsInSampleTimeAndRemainsCapped()
|
||||
{
|
||||
var clock = new ManualAudioClock(); StreamInfo info = Stream(); info.Audio.Fec = false;
|
||||
using var stream = new ReceiveStream(2, info, clock); using var encoder = new OpusEncoder(new() { Bitrate = 32000 });
|
||||
byte[] packet = new byte[1275]; int length = encoder.Encode(Tone(), packet);
|
||||
for (uint i = 0; i < 40; i++)
|
||||
{
|
||||
clock.Advance(i % 2 == 0 ? 5 : 35);
|
||||
stream.Enqueue(new(MediaFrameType.Voice, 0, 0, 42, i, i * 960), packet.AsSpan(0, length));
|
||||
}
|
||||
int[] output = new int[1920]; stream.Mix(output, false, null);
|
||||
Assert.InRange(stream.TargetDepthSamples, 960, 5760);
|
||||
}
|
||||
|
||||
[Fact]
|
||||
public void DredUsesTimestampOffsetForConsecutiveMissingShortFrames()
|
||||
{
|
||||
using var probe = new OpusEncoder();
|
||||
if (!probe.SupportsDeepRedundancy) return;
|
||||
StreamInfo info = Stream(10, dred: true); using var stream = new ReceiveStream(2, info);
|
||||
using var encoder = new OpusEncoder(new() { FrameDurationMilliseconds = 10, DeepRedundancy = true, ExpectedPacketLossPercent = 30, Bitrate = 64000 });
|
||||
byte[] packet = new byte[1275]; short[] tone = new short[480]; int[] output = new int[1920];
|
||||
for (uint i = 0; i < 50; i++)
|
||||
{
|
||||
CodecTests.FillTone(tone, 480, 1, 48000, (int)i); int length = encoder.Encode(tone, packet);
|
||||
if (i is not (25 or 26)) stream.Enqueue(new(MediaFrameType.Voice, 0, 0, 42, i, i * 480), packet.AsSpan(0, length));
|
||||
if (i % 2 == 1) { output.AsSpan().Clear(); stream.Mix(output, false, null); }
|
||||
}
|
||||
Assert.True(stream.DredFrames >= 2);
|
||||
}
|
||||
|
||||
[Theory]
|
||||
[InlineData(true)]
|
||||
[InlineData(false)]
|
||||
@@ -114,4 +192,12 @@ public class AudioEngineTests
|
||||
for (int i = 0; i < 15; i++) { send.FeedPcm(1, stereo, 2); send.ProcessCycle(); receive.ProcessCycle(); }
|
||||
Assert.True(energy > 100000);
|
||||
}
|
||||
|
||||
private sealed class ManualAudioClock : TimeProvider
|
||||
{
|
||||
private long milliseconds;
|
||||
public override long TimestampFrequency => 1000;
|
||||
public override long GetTimestamp() => milliseconds;
|
||||
internal void Advance(int value) => milliseconds += value;
|
||||
}
|
||||
}
|
||||
|
||||
Reference in New Issue
Block a user