DDC support arbitrary sample rate by analytical spectrum analyser and reverse calculation of peaking equation

DDC support arbitrary sample rate by analytical spectrum analyser and reverse calculation of peaking equation

Author: james34602

Audio_Engine/eclipse_libjamesdsp_free_bp/jni/Application.mk

@@ -1,3 +1,3 @@
 APP_OPTIM := release
 APP_PLATFORM := android-21
-APP_ABI := armeabi-v7a x86
+APP_ABI := armeabi-v7a arm64-v8a x86

Audio_Engine/eclipse_libjamesdsp_free_bp/jni/EffectDSPMain.cpp

@@ -19,7 +19,7 @@ const double interpFreq[NUM_BANDS] = { 25.0, 40.0, 63.0, 100.0, 160.0, 250.0, 40
 
 EffectDSPMain::EffectDSPMain()
 	: DSPbufferLength(1024), inOutRWPosition(0), equalizerEnabled(0), ramp(1.0), pregain(12.0), threshold(-60.0), knee(30.0), ratio(12.0), attack(0.001), release(0.24), isBenchData(0), mPreset(0), reverbEnabled(0), threadResult(0)
-	, mMatrixMCoeff(1.0), mMatrixSCoeff(1.0), bassBoostLp(0), FIREq(0), convolver(0), fullStereoConvolver(0), sosCount(0), df441(0), df48(0)
+	, mMatrixMCoeff(1.0), mMatrixSCoeff(1.0), bassBoostLp(0), FIREq(0), convolver(0), fullStereoConvolver(0), sosCount(0), resampledSOSCount(0), usedSOSCount(0), df441(0), df48(0), dfResampled(0)
 	, tempImpulseIncoming(0), tempImpulsedouble(0), finalImpulse(0), convolverReady(-1), bassLpReady(-1), analogModelEnable(0), tubedrive(2.0), eqFilterType(0), arbEq(0), xaxis(0), yaxis(0), eqFIRReady(0)
 {
 	double c0[12] = { 2.138018534150542e-5, 4.0608501987194246e-5, 7.950414700590711e-5, 1.4049065318523225e-4, 2.988065284903209e-4, 0.0013061668170781858, 0.0036204239724680425, 0.008959629624060151, 0.027083658741258742, 0.08156916666666666, 0.1978822177777778, 0.4410733777777778 };
@@ -90,6 +90,14 @@ EffectDSPMain::~EffectDSPMain()
 		free(df441);
 		free(df48);
 		sosCount = 0;
+		if (resampledSOSCount)
+		{
+			for (int i = 0; i < resampledSOSCount; i++)
+				free(dfResampled[i]);
+			free(dfResampled);
+			dfResampled = 0;
+			resampledSOSCount = 0;
+		}
 	}
 #ifdef DEBUG
 	LOGI("Buffer freed");
@@ -587,18 +595,22 @@ int32_t EffectDSPMain::command(uint32_t cmdCode, uint32_t cmdSize, void* pCmdDat
 				int16_t oldVal = viperddcEnabled;
 				if (mSamplingRate == 44100.0 && df441)
 				{
-					viperddcEnabled = ((int16_t *)cep)[8];
 					sosPointer = df441;
+					usedSOSCount = sosCount;
+					viperddcEnabled = ((int16_t *)cep)[8];
 				}
 				else if (mSamplingRate == 48000.0 && df48)
 				{
-					viperddcEnabled = ((int16_t *)cep)[8];
 					sosPointer = df48;
+					usedSOSCount = sosCount;
+					viperddcEnabled = ((int16_t *)cep)[8];
 				}
 				else
 				{
-					viperddcEnabled = 0;
-					sosPointer = 0;
+					resampledSOSCount = PeakingFilterResampler(df48, 48000.0, &dfResampled, mSamplingRate, sosCount);
+					usedSOSCount = resampledSOSCount;
+					sosPointer = dfResampled;
+					viperddcEnabled = ((int16_t *)cep)[8];
 				}
 #ifdef DEBUG
 				LOGI("viperddcEnabled: %d", viperddcEnabled);
@@ -622,6 +634,14 @@ int32_t EffectDSPMain::command(uint32_t cmdCode, uint32_t cmdSize, void* pCmdDat
 						free(df48);
 						df48 = 0;
 						sosCount = 0;
+						if (resampledSOSCount)
+						{
+							for (int i = 0; i < resampledSOSCount; i++)
+								free(dfResampled[i]);
+							free(dfResampled);
+							dfResampled = 0;
+							resampledSOSCount = 0;
+						}
 						sosPointer = 0;
 					}
 				}
@@ -716,6 +736,14 @@ int32_t EffectDSPMain::command(uint32_t cmdCode, uint32_t cmdSize, void* pCmdDat
 						free(df48);
 						df48 = 0;
 						sosCount = 0;
+						if (resampledSOSCount)
+						{
+							for (int i = 0; i < resampledSOSCount; i++)
+								free(dfResampled[i]);
+							free(dfResampled);
+							dfResampled = 0;
+							resampledSOSCount = 0;
+						}
 						sosPointer = 0;
 					}
 				}
@@ -884,7 +912,7 @@ int32_t EffectDSPMain::command(uint32_t cmdCode, uint32_t cmdSize, void* pCmdDat
 }
 void EffectDSPMain::refreshTubeAmp()
 {
-	if (!InitTube(&tubeP[0], 0, mSamplingRate, tubedrive, 6.5, 5.6, 4.5, 5, 8192, 0))
+	if (!InitTube(&tubeP[0], 0, mSamplingRate, tubedrive, 8192, 0))
 		analogModelEnable = 0;
 	tubeP[1] = tubeP[0];
 	rightparams2.tube = tubeP;
@@ -1219,7 +1247,7 @@ int32_t EffectDSPMain::process(audio_buffer_t *in, audio_buffer_t *out)
 					for (i = 0; i < DSPbufferLength; i++)
 					{
 						double sampleOutL = inputBuffer[0][i], sampleOutR = inputBuffer[1][i];
-						for (int j = 0; j < sosCount; j++)
+						for (int j = 0; j < usedSOSCount; j++)
 							SOS_DF2_StereoProcess(sosPointer[j], sampleOutL, sampleOutR, &sampleOutL, &sampleOutR);
 						outputBuffer[0][i] = sampleOutL;
 						outputBuffer[1][i] = sampleOutR;
@@ -1353,7 +1381,7 @@ int32_t EffectDSPMain::process(audio_buffer_t *in, audio_buffer_t *out)
 					for (i = 0; i < DSPbufferLength; i++)
 					{
 						double sampleOutL = inputBuffer[0][i], sampleOutR = inputBuffer[1][i];
-						for (int j = 0; j < sosCount; j++)
+						for (int j = 0; j < usedSOSCount; j++)
 							SOS_DF2_StereoProcess(sosPointer[j], sampleOutL, sampleOutR, &sampleOutL, &sampleOutR);
 						outputBuffer[0][i] = sampleOutL;
 						outputBuffer[1][i] = sampleOutR;
@@ -1487,7 +1515,7 @@ int32_t EffectDSPMain::process(audio_buffer_t *in, audio_buffer_t *out)
 					for (i = 0; i < DSPbufferLength; i++)
 					{
 						double sampleOutL = inputBuffer[0][i], sampleOutR = inputBuffer[1][i];
-						for (int j = 0; j < sosCount; j++)
+						for (int j = 0; j < usedSOSCount; j++)
 							SOS_DF2_StereoProcess(sosPointer[j], sampleOutL, sampleOutR, &sampleOutL, &sampleOutR);
 						outputBuffer[0][i] = sampleOutL;
 						outputBuffer[1][i] = sampleOutR;

Audio_Engine/eclipse_libjamesdsp_free_bp/jni/EffectDSPMain.h

@@ -24,8 +24,8 @@ class EffectDSPMain : public Effect
 protected:
 	int stringLength;
 	char *stringEq;
-	DirectForm2 **df441, **df48, **sosPointer;
-	int sosCount;
+	DirectForm2 **df441, **df48, **dfResampled, **sosPointer;
+	int sosCount, resampledSOSCount, usedSOSCount;
 	typedef struct threadParamsConv {
 		AutoConvolverMono **conv;
 		double **in, **out;