Merge b68e57e106 into b8804a0d2e

.github/workflows/build.yml

@@ -39,6 +39,11 @@ jobs:
           wget -q https://developer.arm.com/-/media/Files/downloads/gnu-a/10.3-2021.07/binrel/gcc-arm-10.3-2021.07-x86_64-aarch64-none-elf.tar.xz
           tar xf gcc-arm-10.3-2021.07-x86_64-aarch64-none-elf.tar.xz
 
+      # Patch Synth_Dexed PluginFX
+      - name: Patch Synth_Dexed
+        run: |
+          ( cd Synth_Dexed ; patch -p1 < ../src/patches/0001-PluginFX-change-gain-at-zero-crossing.patch )
+
       - name: Build for Raspberry Pi 5 (64-bit)
         run: |
           set -ex
@@ -124,6 +129,11 @@ jobs:
           wget -q https://developer.arm.com/-/media/Files/downloads/gnu-a/10.3-2021.07/binrel/gcc-arm-10.3-2021.07-x86_64-arm-none-eabi.tar.xz
           tar xf gcc-arm-10.3-2021.07-x86_64-arm-none-eabi.tar.xz
 
+      # Patch Synth_Dexed PluginFX
+      - name: Patch Synth_Dexed
+        run: |
+          ( cd Synth_Dexed ; patch -p1 < ../src/patches/0001-PluginFX-change-gain-at-zero-crossing.patch )
+
       - name: Build for Raspberry Pi 2 (32-bit)
         run: |
           set -ex

src/effect_mixer.hpp

@@ -22,7 +22,10 @@ public:
 	{
 		buffer_length=len;
 		for (uint8_t i=0; i<NN; i++)
+		{
 			multiplier[i] = UNITY_GAIN;
+			multiplier_w[i] = UNITY_GAIN;
+		}
 
 		sumbufL=new float32_t[buffer_length];
 		arm_fill_f32(0.0f, sumbufL, len);
@@ -52,7 +55,7 @@ public:
 			gain = MAX_GAIN;
 		else if (gain < MIN_GAIN)
 			gain = MIN_GAIN;
-		multiplier[channel] = powf(gain, 4); // see: https://www.dr-lex.be/info-stuff/volumecontrols.html#ideal2
+		multiplier_w[channel] = powf(gain, 4); // see: https://www.dr-lex.be/info-stuff/volumecontrols.html#ideal2
 	}
 
 	void gain(float32_t gain)
@@ -63,7 +66,7 @@ public:
 				gain = MAX_GAIN;
 			else if (gain < MIN_GAIN)
 				gain = MIN_GAIN;
-			multiplier[i] = powf(gain, 4); // see: https://www.dr-lex.be/info-stuff/volumecontrols.html#ideal2
+			multiplier_w[i] = powf(gain, 4); // see: https://www.dr-lex.be/info-stuff/volumecontrols.html#ideal2
 		} 
 	}
 
@@ -79,6 +82,7 @@ public:
 
 protected:
 	float32_t multiplier[NN];
+	float32_t multiplier_w[NN];
 	float32_t* sumbufL;
 	uint16_t buffer_length;
 };
@@ -92,6 +96,8 @@ public:
 		{
 			panorama[i][0] = UNITY_PANORAMA;
 			panorama[i][1] = UNITY_PANORAMA;
+			panorama_w[i][0] = UNITY_PANORAMA;
+			panorama_w[i][1] = UNITY_PANORAMA;
 		}
 
 		sumbufR=new float32_t[buffer_length];
@@ -103,6 +109,26 @@ public:
 		delete [] sumbufR;
 	}
 
+	uint32_t find_zc_f32(const float32_t *pSrc, uint32_t blockSize)
+	{
+		const float32_t *pIn = pSrc;
+		uint32_t blkCnt = blockSize;
+
+		pIn++;
+		blkCnt--;
+
+		while (blkCnt > 0U)
+		{
+			if (*pIn >= 0 && *(pIn - 1) < 0)
+				return blockSize - blkCnt;
+
+			pIn++;
+			blkCnt--;
+		}
+
+		return blockSize;
+	}
+
         void pan(uint8_t channel, float32_t pan)
 	{
 		if (channel >= NN) return;
@@ -113,43 +139,38 @@ public:
 			pan = MIN_PANORAMA;
 
 		// From: https://stackoverflow.com/questions/67062207/how-to-pan-audio-sample-data-naturally
-		panorama[channel][0]=arm_sin_f32(mapfloat(pan, MIN_PANORAMA, MAX_PANORAMA, 0.0, M_PI/2.0));
+		panorama_w[channel][0]=arm_cos_f32(mapfloat(pan, MIN_PANORAMA, MAX_PANORAMA, 0.0, M_PI/2.0));
-		panorama[channel][1]=arm_cos_f32(mapfloat(pan, MIN_PANORAMA, MAX_PANORAMA, 0.0, M_PI/2.0));
+		panorama_w[channel][1]=arm_sin_f32(mapfloat(pan, MIN_PANORAMA, MAX_PANORAMA, 0.0, M_PI/2.0));
 	}
 
 	void doAddMix(uint8_t channel, float32_t* in)
 	{
 		float32_t tmp[buffer_length];
-
+		uint32_t zc;
 		assert(in);
 
-		// left
+		if ((panorama[channel][0] != panorama_w[channel][0] || multiplier[channel] != multiplier_w[channel]) &&
-		arm_scale_f32(in, panorama[channel][0], tmp, buffer_length);
+			(zc = find_zc_f32(in, buffer_length)) != buffer_length)
-		if(multiplier[channel]!=UNITY_GAIN)
+		{
-			arm_scale_f32(tmp,multiplier[channel],tmp,buffer_length);
+			arm_scale_f32(in, panorama[channel][0] * multiplier[channel], tmp, zc);
-		arm_add_f32(sumbufL, tmp, sumbufL, buffer_length);
+			arm_scale_f32(in + zc, panorama_w[channel][0] * multiplier_w[channel], tmp + zc, buffer_length - zc);
-		// right
+			arm_add_f32(sumbufL, tmp, sumbufL, buffer_length);
-		arm_scale_f32(in, panorama[channel][1], tmp, buffer_length);
+			panorama[channel][0] = panorama_w[channel][0];
-		if(multiplier[channel]!=UNITY_GAIN)
-			arm_scale_f32(tmp,multiplier[channel],tmp,buffer_length);
-		arm_add_f32(sumbufR, tmp, sumbufR, buffer_length);
-	}
-
-	void doAddMix(uint8_t channel, float32_t* inL, float32_t* inR)
-	{
-		float32_t tmp[buffer_length];
 
-		assert(inL);
+			arm_scale_f32(in, panorama[channel][1] * multiplier[channel], tmp, zc);
-		assert(inR);
+			arm_scale_f32(in + zc, panorama_w[channel][1] * multiplier_w[channel], tmp + zc, buffer_length - zc);
+			arm_add_f32(sumbufR, tmp, sumbufR, buffer_length);
+			panorama[channel][1] = panorama_w[channel][1];
 
-		// left
+			multiplier[channel] = multiplier_w[channel];
-		if(multiplier[channel]!=UNITY_GAIN)
+		}
-			arm_scale_f32(inL,multiplier[channel],tmp,buffer_length);
+		else
-		arm_add_f32(sumbufL, tmp, sumbufL, buffer_length);
+		{
-		// right
+			arm_scale_f32(in, panorama[channel][0] * multiplier[channel], tmp, buffer_length);
-		if(multiplier[channel]!=UNITY_GAIN)
+			arm_add_f32(sumbufL, tmp, sumbufL, buffer_length);
-			arm_scale_f32(inR,multiplier[channel],tmp,buffer_length);
+			arm_scale_f32(in, panorama[channel][1] * multiplier[channel], tmp, buffer_length);
-		arm_add_f32(sumbufR, tmp, sumbufR, buffer_length);
+			arm_add_f32(sumbufR, tmp, sumbufR, buffer_length);
+		}	
 	}
 
 	void getMix(float32_t* bufferL, float32_t* bufferR)
@@ -168,11 +189,27 @@ public:
 			arm_fill_f32(0.0f, sumbufR, buffer_length);
 	}
 
+	void getBuffers(float32_t (*buffers[2]))
+	{
+		buffers[0] = sumbufL;
+		buffers[1] = sumbufR;
+	}
+
+	void zeroFill()
+	{
+		if(sumbufL)
+			arm_fill_f32(0.0f, sumbufL, buffer_length);
+		if(sumbufR)
+			arm_fill_f32(0.0f, sumbufR, buffer_length);
+	}
+
 protected:
 	using AudioMixer<NN>::sumbufL;
 	using AudioMixer<NN>::multiplier;
+	using AudioMixer<NN>::multiplier_w;
 	using AudioMixer<NN>::buffer_length;
 	float32_t panorama[NN][2];
+	float32_t panorama_w[NN][2];
 	float32_t* sumbufR;
 };
 

src/minidexed.cpp

@@ -213,10 +213,6 @@ CMiniDexed::CMiniDexed (CConfig *pConfig, CInterruptSystem *pInterrupt,
 
 		m_pSoundDevice = new CHDMISoundBaseDevice (pInterrupt, pConfig->GetSampleRate (),
 							   pConfig->GetChunkSize ());
-
-		// The channels are swapped by default in the HDMI sound driver.
-		// TODO: Remove this line, when this has been fixed in the driver.
-		m_bChannelsSwapped = !m_bChannelsSwapped;
 #endif
 	}
 	else
@@ -1406,34 +1402,35 @@ void CMiniDexed::ProcessSound (void)
 
 			if(nMasterVolume > 0.0)
 			{
+				// get the mix buffer of all TGs
+				float32_t (*SampleBuffer[2]);
+				tg_mixer->getBuffers(SampleBuffer);
+
+				tg_mixer->zeroFill();
+
 				for (uint8_t i = 0; i < m_nToneGenerators; i++)
 				{
 					tg_mixer->doAddMix(i,m_OutputLevel[i]);
-					reverb_send_mixer->doAddMix(i,m_OutputLevel[i]);
 				}
 				// END TG mixing
 
-				// BEGIN create SampleBuffer for holding audio data
-				float32_t SampleBuffer[2][nFrames];
-				// END create SampleBuffer for holding audio data
-
-				// get the mix of all TGs
-				tg_mixer->getMix(SampleBuffer[indexL], SampleBuffer[indexR]);
-
 				// BEGIN adding reverb
 				if (m_nParameter[ParameterReverbEnable])
 				{
 					float32_t ReverbBuffer[2][nFrames];
-					float32_t ReverbSendBuffer[2][nFrames];
 
-					arm_fill_f32(0.0f, ReverbBuffer[indexL], nFrames);
+					float32_t (*ReverbSendBuffer[2]);
-					arm_fill_f32(0.0f, ReverbBuffer[indexR], nFrames);
+					reverb_send_mixer->getBuffers(ReverbSendBuffer);
-					arm_fill_f32(0.0f, ReverbSendBuffer[indexR], nFrames);
+
-					arm_fill_f32(0.0f, ReverbSendBuffer[indexL], nFrames);
+					reverb_send_mixer->zeroFill();
+
+					for (uint8_t i = 0; i < m_nToneGenerators; i++)
+					{
+						reverb_send_mixer->doAddMix(i,m_OutputLevel[i]);
+					}
 
 					m_ReverbSpinLock.Acquire ();
 
-					reverb_send_mixer->getMix(ReverbSendBuffer[indexL], ReverbSendBuffer[indexR]);
 					reverb->doReverb(ReverbSendBuffer[indexL],ReverbSendBuffer[indexR],ReverbBuffer[indexL], ReverbBuffer[indexR],nFrames);
 
 					// scale down and add left reverb buffer by reverb level 

src/patches/0001-PluginFX-change-gain-at-zero-crossing.patch

@@ -0,0 +1,73 @@
+From 7daa07bd662d1894bc312b81c6b9e79260e7a66a Mon Sep 17 00:00:00 2001
+From: Gergo Koteles <soyer@irl.hu>
+Date: Fri, 11 Jul 2025 20:28:52 +0200
+Subject: [PATCH] PluginFX change gain at zero crossing
+
+---
+ src/PluginFx.cpp | 26 +++++++++++++++++++-------
+ src/PluginFx.h   |  2 ++
+ 2 files changed, 21 insertions(+), 7 deletions(-)
+
+diff --git a/src/PluginFx.cpp b/src/PluginFx.cpp
+index 7c3a1ab..2b7bb9b 100644
+--- a/src/PluginFx.cpp
++++ b/src/PluginFx.cpp
+@@ -56,6 +56,7 @@ PluginFx::PluginFx() {
+   Cutoff = 1.0;
+   Reso = 0.0;
+   Gain = 1.0;
++  aGain = 1.0;
+ }
+ 
+ void PluginFx::init(FRAC_NUM sr) {
+@@ -117,16 +118,27 @@ void PluginFx::process(float *work, uint16_t sampleSize) {
+ 
+   dc_od = work[sampleSize - 1];
+ 
+-  // Gain
+-  if (Gain == 0.0)
+-  {
++  if (Gain != aGain) {
+     for (uint16_t i = 0; i < sampleSize; i++ )
+-      work[i] = 0.0;
++    {
++      if (i != 0 && work[i] >= 0 && work[i-1] <= 0)
++        aGain = Gain;
++      work[i] *= aGain;
++    }
+   }
+-  else if ( Gain != 1.0)
++  else
+   {
+-    for (uint16_t i = 0; i < sampleSize; i++ )
+-      work[i] *= Gain;
++    // Gain
++    if (aGain == 0.0)
++    {
++      for (uint16_t i = 0; i < sampleSize; i++ )
++        work[i] = 0.0;
++    }
++    else if ( aGain != 1.0)
++    {
++      for (uint16_t i = 0; i < sampleSize; i++ )
++        work[i] *= aGain;
++    }
+   }
+ 
+   // don't apply the LPF if the cutoff is to maximum
+diff --git a/src/PluginFx.h b/src/PluginFx.h
+index 090e241..c27ccc5 100644
+--- a/src/PluginFx.h
++++ b/src/PluginFx.h
+@@ -67,6 +67,8 @@ class PluginFx {
+     float Reso;
+     float Gain;
+ 
++    float aGain;
++
+     void init(FRAC_NUM sampleRate);
+     void process(float *work, uint16_t sampleSize);
+     float getGain(void);
+-- 
+2.50.1
+