Added channel downmix logic

Author: nefarius

DS4AudioStreamer/Sound/Downmixer.cs

@@ -0,0 +1,96 @@
+namespace DS4AudioStreamer.Sound;
+
+public static class Downmixer
+{
+    public static void DownmixToStereo(float[] input, float[] output, int frames, int channels)
+    {
+        if (channels < 2)
+        {
+            throw new ArgumentException("Need at least 2 channels to downmix");
+        }
+
+        int inIdx = 0;
+        int outIdx = 0;
+
+        for (int i = 0; i < frames; i++)
+        {
+            float left = input[inIdx]; // Front Left
+            float right = input[inIdx + 1]; // Front Right
+
+            if (channels > 2)
+            {
+                // Center
+                if (channels > 2)
+                {
+                    left += input[inIdx + 2] * 0.7f;
+                    right += input[inIdx + 2] * 0.7f;
+                }
+
+                // LFE
+                if (channels > 3)
+                {
+                    left += input[inIdx + 3] * 0.5f;
+                    right += input[inIdx + 3] * 0.5f;
+                }
+
+                // SL/SR
+                if (channels > 4)
+                {
+                    left += input[inIdx + 4] * 0.7f;
+                }
+
+                if (channels > 5)
+                {
+                    right += input[inIdx + 5] * 0.7f;
+                }
+
+                // SBL/SBR (7.1)
+                if (channels > 6)
+                {
+                    left += input[inIdx + 6] * 0.7f;
+                }
+
+                if (channels > 7)
+                {
+                    right += input[inIdx + 7] * 0.7f;
+                }
+            }
+
+            // Prevent clipping by scaling down
+            output[outIdx] = left * 0.5f;
+            output[outIdx + 1] = right * 0.5f;
+
+            inIdx += channels;
+            outIdx += 2;
+        }
+    }
+
+    public static unsafe void Downmix6To2(Span<float> input, Span<float> output, int frames)
+    {
+        int inIdx = 0;
+        int outIdx = 0;
+
+        for (int i = 0; i < frames; i++)
+        {
+            // Load 6 floats (L, R, C, LFE, SL, SR)
+            // We can’t load all 6 at once with Vector<float> (needs multiple of Vector<float>.Count),
+            // so SIMD wins mostly for bulk copy/scale. For mixing, plain scalar math is fine.
+
+            float l = input[inIdx];
+            float r = input[inIdx + 1];
+            float c = input[inIdx + 2];
+            float lfe = input[inIdx + 3];
+            float sl = input[inIdx + 4];
+            float sr = input[inIdx + 5];
+
+            float left = l + (c * 0.7f) + (lfe * 0.5f) + (sl * 0.7f);
+            float right = r + (c * 0.7f) + (lfe * 0.5f) + (sr * 0.7f);
+
+            output[outIdx] = left * 0.5f;
+            output[outIdx + 1] = right * 0.5f;
+
+            inIdx += 6;
+            outIdx += 2;
+        }
+    }
+}

DS4AudioStreamer/Sound/SbcAudioStream.cs

@@ -1,3 +1,4 @@
+using System.Buffers;
 using System.Diagnostics;
 using System.Runtime.InteropServices;
 
@@ -59,7 +60,7 @@ public SbcAudioStream()
 
         _captureDevice = new BufferedLoopbackCapture(device);
         _captureDevice.DataAvailable += OnAudioCaptured;
-        
+
         Console.WriteLine($"Wave format: {_captureDevice.WaveFormat}");
 
         // Resampler
@@ -155,7 +156,16 @@ private unsafe void OnAudioCaptured(object? sender, WaveInEventArgs e)
         short* finalShortFormat = stackalloc short[outSamples];
         int finalSampleCount = 0;
 
-        fixed (float* pcmBuffer = MemoryMarshal.Cast<byte, float>(e.Buffer))
+        Span<float> pcmData = MemoryMarshal.Cast<byte, float>(e.Buffer);
+        float[] downmixed = ArrayPool<float>.Shared.Rent(inSamples);
+        
+        // TODO: experimental
+        if (_captureDevice.WaveFormat.Channels == 6)
+        {
+            Downmixer.Downmix6To2(pcmData, downmixed, inFrameCount);
+        }
+
+        fixed (float* pcmBuffer = _captureDevice.WaveFormat.Channels == 6 ? downmixed : pcmData)
         {
             if (SbcSampleRate != _captureDevice.WaveFormat.SampleRate)
             {