Merge pull request #26 from rsp4jack/patch1

Add chorus, reverb dry and master gain for sonivoxrender

arm-wt-22k/lib_src/eas_chorus.c

@@ -85,7 +85,6 @@ static const EAS_I16 EAS_chorusShape[CHORUS_SHAPE_SIZE] = {
 static EAS_RESULT ChorusInit (EAS_DATA_HANDLE pEASData, EAS_VOID_PTR *pInstData)
 {
     S_CHORUS_OBJECT *pChorusData;
-    S_CHORUS_PRESET *pPreset;
     EAS_I32 index;
 
     /* check Configuration Module for data allocation */
@@ -109,29 +108,7 @@ static EAS_RESULT ChorusInit (EAS_DATA_HANDLE pEASData, EAS_VOID_PTR *pInstData)
 
     /* set some default values */
     pChorusData->bypass =       EAS_CHORUS_BYPASS_DEFAULT;
-    pChorusData->preset =       EAS_CHORUS_PRESET_DEFAULT;
-    pChorusData->m_nLevel =     EAS_CHORUS_LEVEL_DEFAULT;
-    pChorusData->m_nRate =      EAS_CHORUS_RATE_DEFAULT;
-    pChorusData->m_nDepth =     EAS_CHORUS_DEPTH_DEFAULT;
-
-    //chorus rate and depth need some massaging from preset value (which is sample rate independent)
-
-    //convert rate from steps of .05 Hz to value which can be used as phase increment,
-    //with current CHORUS_SHAPE_SIZE and rate limits, this fits into 16 bits
-    //want to compute ((shapeSize * 65536) * (storedRate/20))/sampleRate;
-    //computing it as below allows rate steps to be evenly spaced
-    //uses 32 bit divide, but only once when new value is selected
-    pChorusData->m_nRate = (EAS_I16)
-        ((((EAS_I32)CHORUS_SHAPE_SIZE<<16)/(20*(EAS_I32)_OUTPUT_SAMPLE_RATE)) * pChorusData->m_nRate);
-
-    //convert depth from steps of .05 ms, to samples, with 16 bit whole part, discard fraction
-    //want to compute ((depth * sampleRate)/20000)
-    //use the following approximation since 105/32 is roughly 65536/20000
-    /*lint -e{704} use shift for performance */
-    pChorusData->m_nDepth = (EAS_I16)
-        (((((EAS_I32)pChorusData->m_nDepth * _OUTPUT_SAMPLE_RATE)>>5) * 105) >> 16);
-
-    pChorusData->m_nLevel = pChorusData->m_nLevel;
+    pChorusData->m_nNextChorus = EAS_CHORUS_PRESET_DEFAULT;
 
     //zero delay memory for chorus
     for (index = CHORUS_L_SIZE - 1; index >= 0; index--)
@@ -157,18 +134,7 @@ static EAS_RESULT ChorusInit (EAS_DATA_HANDLE pEASData, EAS_VOID_PTR *pInstData)
     pChorusData->chorusTapPosition = (EAS_I16)((CHORUS_DELAY_MS * _OUTPUT_SAMPLE_RATE)/1000);
 
     //now copy from the new preset into Chorus
-    pPreset = &pChorusData->m_sPreset.m_sPreset[pChorusData->m_nNextChorus];
-
-    pChorusData->m_nLevel = pPreset->m_nLevel;
-    pChorusData->m_nRate =  pPreset->m_nRate;
-    pChorusData->m_nDepth = pPreset->m_nDepth;
-
-    pChorusData->m_nRate = (EAS_I16)
-        ((((EAS_I32)CHORUS_SHAPE_SIZE<<16)/(20*(EAS_I32)_OUTPUT_SAMPLE_RATE)) * pChorusData->m_nRate);
-
-    /*lint -e{704} use shift for performance */
-    pChorusData->m_nDepth = (EAS_I16)
-        (((((EAS_I32)pChorusData->m_nDepth * _OUTPUT_SAMPLE_RATE)>>5) * 105) >> 16);
+    ChorusUpdate(pChorusData);
 
     *pInstData = pChorusData;
 
@@ -491,6 +457,7 @@ static EAS_RESULT ChorusSetParam (EAS_VOID_PTR pInstData, EAS_I32 param, EAS_I32
                 value!=EAS_PARAM_CHORUS_PRESET3 && value!=EAS_PARAM_CHORUS_PRESET4)
                 return EAS_ERROR_INVALID_PARAMETER;
             p->m_nNextChorus = (EAS_I8)value;
+            ChorusUpdate(p); // update parameters immediately
             break;
         case EAS_PARAM_CHORUS_RATE:
             if(value<EAS_CHORUS_RATE_MIN || value>EAS_CHORUS_RATE_MAX)
@@ -590,9 +557,19 @@ static EAS_RESULT ChorusUpdate(S_CHORUS_OBJECT *pChorusData)
     pChorusData->m_nRate =  pPreset->m_nRate;
     pChorusData->m_nDepth = pPreset->m_nDepth;
 
+    //chorus rate and depth need some massaging from preset value (which is sample rate independent)
+
+    //convert rate from steps of .05 Hz to value which can be used as phase increment,
+    //with current CHORUS_SHAPE_SIZE and rate limits, this fits into 16 bits
+    //want to compute ((shapeSize * 65536) * (storedRate/20))/sampleRate;
+    //computing it as below allows rate steps to be evenly spaced
+    //uses 32 bit divide, but only once when new value is selected
     pChorusData->m_nRate = (EAS_I16)
         ((((EAS_I32)CHORUS_SHAPE_SIZE<<16)/(20*(EAS_I32)_OUTPUT_SAMPLE_RATE)) * pChorusData->m_nRate);
 
+    //convert depth from steps of .05 ms, to samples, with 16 bit whole part, discard fraction
+    //want to compute ((depth * sampleRate)/20000)
+    //use the following approximation since 105/32 is roughly 65536/20000
     /*lint -e{704} use shift for performance */
     pChorusData->m_nDepth = (EAS_I16)
         (((((EAS_I32)pChorusData->m_nDepth * _OUTPUT_SAMPLE_RATE)>>5) * 105) >> 16);

arm-wt-22k/lib_src/eas_chorusdata.h

@@ -38,9 +38,6 @@
 
 #define EAS_CHORUS_BYPASS_DEFAULT   1
 #define EAS_CHORUS_PRESET_DEFAULT   0
-#define EAS_CHORUS_RATE_DEFAULT     30
-#define EAS_CHORUS_DEPTH_DEFAULT    39
-#define EAS_CHORUS_LEVEL_DEFAULT    32767
 
 #define EAS_CHORUS_LEVEL_MIN        0
 #define EAS_CHORUS_LEVEL_MAX        32767
@@ -91,7 +88,6 @@ typedef struct
     EAS_PCM chorusDelayR[CHORUS_R_SIZE];
 
     EAS_BOOL    bypass;
-    EAS_I8      preset;
 
     EAS_I16     m_nCurrentChorus;           // preset number for current Chorus
     EAS_I16     m_nNextChorus;              // preset number for next Chorus

arm-wt-22k/lib_src/eas_reverb.c

@@ -168,13 +168,7 @@ static EAS_RESULT ReverbInit(EAS_DATA_HANDLE pEASData, EAS_VOID_PTR *pInstData)
     // for debugging purposes, allow bypass
     pReverbData->m_bBypass = EAS_TRUE;  //EAS_FALSE;
 
-    pReverbData->m_nNextRoom = 1;
-
-    pReverbData->m_nCurrentRoom = pReverbData->m_nNextRoom + 1; // force update on first iteration
-
-    pReverbData->m_nWet = REVERB_DEFAULT_WET;
-
-    pReverbData->m_nDry = REVERB_DEFAULT_DRY;
+    pReverbData->m_nNextRoom = REVERB_DEFAULT_ROOM_NUMBER;
 
     // set base index into circular buffer
     pReverbData->m_nBaseIndex = 0;
@@ -207,43 +201,7 @@ static EAS_RESULT ReverbInit(EAS_DATA_HANDLE pEASData, EAS_VOID_PTR *pInstData)
         pReverbData->m_nDelayLine[i] = 0;
     }
 
-    ////////////////////////////////
-    ///code from the EAS DEMO Reverb
-    //now copy from the new preset into the reverb
-    pPreset = &pReverbData->m_sPreset.m_sPreset[pReverbData->m_nNextRoom];
-
-    pReverbData->m_nLpfFbk = pPreset->m_nLpfFbk;
-    pReverbData->m_nLpfFwd = pPreset->m_nLpfFwd;
-
-    pReverbData->m_nEarly = pPreset->m_nEarly;
-    pReverbData->m_nWet = pPreset->m_nWet;
-    pReverbData->m_nDry = pPreset->m_nDry;
-
-    pReverbData->m_nMaxExcursion = pPreset->m_nMaxExcursion;
-    //stored as time based, convert to sample based
-    temp = pPreset->m_nXfadeInterval;
-    /*lint -e{702} shift for performance */
-    temp = (temp * _OUTPUT_SAMPLE_RATE) >> 16;
-    pReverbData->m_nXfadeInterval = (EAS_U16) temp;
-    //gsReverbObject.m_nXfadeInterval = pPreset->m_nXfadeInterval;
-
-    pReverbData->m_sAp0.m_nApGain = pPreset->m_nAp0_ApGain;
-    //stored as time based, convert to absolute sample value
-    temp = pPreset->m_nAp0_ApOut;
-    /*lint -e{702} shift for performance */
-    temp = (temp * _OUTPUT_SAMPLE_RATE) >> 16;
-    pReverbData->m_sAp0.m_zApOut = (EAS_U16) (pReverbData->m_sAp0.m_zApIn + temp);
-    //gsReverbObject.m_sAp0.m_zApOut = pPreset->m_nAp0_ApOut;
-
-    pReverbData->m_sAp1.m_nApGain = pPreset->m_nAp1_ApGain;
-    //stored as time based, convert to absolute sample value
-    temp = pPreset->m_nAp1_ApOut;
-    /*lint -e{702} shift for performance */
-    temp = (temp * _OUTPUT_SAMPLE_RATE) >> 16;
-    pReverbData->m_sAp1.m_zApOut = (EAS_U16) (pReverbData->m_sAp1.m_zApIn + temp);
-    //gsReverbObject.m_sAp1.m_zApOut = pPreset->m_nAp1_ApOut;
-    ///code from the EAS DEMO Reverb
-    ////////////////////////////////
+    ReverbUpdateRoom(pReverbData);
 
     *pInstData = pReverbData;
 
@@ -877,6 +835,7 @@ static EAS_RESULT ReverbSetParam (EAS_VOID_PTR pInstData, EAS_I32 param, EAS_I32
                 value!=EAS_PARAM_REVERB_CHAMBER && value!=EAS_PARAM_REVERB_ROOM)
                 return EAS_ERROR_INVALID_PARAMETER;
             p->m_nNextRoom = (EAS_I16)value;
+            ReverbUpdateRoom(p); // change parameters immediately
             break;
         case EAS_PARAM_REVERB_WET:
             if(value>EAS_REVERB_WET_MAX || value<EAS_REVERB_WET_MIN)

arm-wt-22k/lib_src/eas_reverbdata.h

@@ -163,9 +163,6 @@ EAS_I32 nPanG1 = -1.0 for cos
 #define DEFAULT_AP1_LENGTH              (int)(((double) (16.5/1000.0))  * ((double) _OUTPUT_SAMPLE_RATE))
 #define DEFAULT_AP1_GAIN                -19400
 
-#define REVERB_DEFAULT_WET              32767
-#define REVERB_DEFAULT_DRY              0
-
 #define EAS_REVERB_WET_MAX              32767
 #define EAS_REVERB_WET_MIN              0
 #define EAS_REVERB_DRY_MAX              32767

example/sonivoxrender.1

@@ -1,105 +1,105 @@
-.\" Automatically generated by Pandoc 2.14.0.3
+.\" Automatically generated by Pandoc 3.1.11.1
 .\"
-.TH "SONIVOXRENDER" "1" "October 6, 2024" "sonivox 3.6.14.0" "Sonivox MIDI File Renderer"
-.hy
+.TH "SONIVOXRENDER" "1" "October 25, 2024" "sonivox 3.6.15.0" "Sonivox MIDI File Renderer"
 .SH NAME
-.PP
 \f[B]sonivoxrender\f[R] \[em] Render standard MIDI files into raw PCM
 audio
 .SH SYNOPSIS
 .PP
-\f[B]sonivoxrender\f[R] [\f[B]-h\f[R]] [\f[B]-d\f[R] \f[I]file.dls\f[R]]
-[\f[B]-r\f[R] \f[I]0..4\f[R]] [\f[B]-w\f[R] \f[I]0..32765\f[R]]
-\f[I]midi_file\f[R]
+\f[B]sonivoxrender\f[R] [\f[B]\-h\f[R]] [\f[B]\-d\f[R]
+\f[I]file.dls\f[R]] [\f[B]\-r\f[R] \f[I]0..4\f[R]] [\f[B]\-w\f[R]
+\f[I]0..32767\f[R]] [\f[B]\-n\f[R] \f[I]0..32767\f[R]] [\f[B]\-c\f[R]
+\f[I]0..4\f[R]] [\f[B]\-l\f[R] \f[I]0..32767\f[R]] [\f[B]\-v\f[R]
+\f[I]0..100\f[R]] \f[I]midi_file\f[R]
 .SH DESCRIPTION
-.PP
 This program is a MIDI file renderer based on the sonivox synthesizer
 library.
 It reads .MID (Standard MIDI Files) file format, and writes an audio
 stream to the standard output as raw 16 bit stereo PCM samples.
 .SS Options
 .TP
--h
+\-h
 Prints brief usage information.
 .TP
--d \f[I]file.dls\f[R]
+\-d \f[I]file.dls\f[R]
 Optional DLS soundfont file name.
 If not provided, it uses an internal embedded soundfont.
 .TP
--r \f[I]reverb_preset\f[R]
+\-r \f[I]reverb_preset\f[R]
 Reverb preset between 0 and 4: 0=no, 1=large hall, 2=hall, 3=chamber,
 4=room.
 .TP
--w \f[I]reverb_wet\f[R]
-Reverb wet level between 0 and 32765.
+\-w \f[I]reverb_wet\f[R]
+Reverb wet level between 0 and 32767.
+.TP
+\-n \f[I]reverb_dry\f[R]
+Reverb dry level between 0 and 32767.
+.TP
+\-c \f[I]chorus_preset\f[R]
+Chorus preset between 0 and 4: 0=no, 1..4=presets.
+.TP
+\-l \f[I]chorus_level\f[R]
+Chorus level between 0 and 32767.
+.TP
+\-v \f[I]master_volume\f[R]
+Master volume between 0 and 100, default to 90.
 .SS Arguments
 .TP
 \f[I]midi_file\f[R]
 Input MID file name.
 .SH EXAMPLES
-.PP
 The following examples assume the default option USE_44KHZ=ON, which
 means an output sample rate = 44100 Hz.
 .PP
 Example 1: Render a MIDI file and save the rendered audio as a raw audio
 file:
 .IP
-.nf
-\f[C]
+.EX
 $ sonivoxrender ants.mid > ants.pcm
-\f[R]
-.fi
+.EE
 .PP
 Example 2: pipe the rendered audio thru the Linux ALSA \f[B]aplay\f[R]
 utility:
 .IP
-.nf
-\f[C]
-$ sonivoxrender ants.mid | aplay -c 2 -f S16_LE -r 44100
-\f[R]
-.fi
+.EX
+$ sonivoxrender ants.mid | aplay \-c 2 \-f S16_LE \-r 44100
+.EE
 .PP
 is equivalent to:
 .IP
-.nf
-\f[C]
-$ sonivoxrender ants.mid | aplay -f cd
-\f[R]
-.fi
+.EX
+$ sonivoxrender ants.mid | aplay \-f cd
+.EE
 .PP
 Example 3: pipe the rendered audio thru the \f[B]lame\f[R] utility
 creating a MP3 file:
 .IP
-.nf
-\f[C]
-$ sonivoxrender ants.mid | lame -r -s 44100 - ants.mp3
-\f[R]
-.fi
+.EX
+$ sonivoxrender ants.mid | lame \-r \-s 44100 \- ants.mp3
+.EE
 .PP
 Example 4: pipe the rendered audio thru the \f[B]sox\f[R] utility
 creating a WAV file:
 .IP
-.nf
-\f[C]
-$ sonivoxrender ants.mid | sox -t s16 -c 2 -r 44100 - ants.wav
-\f[R]
-.fi
+.EX
+$ sonivoxrender ants.mid | sox \-t s16 \-c 2 \-r 44100 \- ants.wav
+.EE
 .PP
 Example 5: pipe the rendered audio thru the PulseAudio\[cq]s
 \f[B]pacat\f[R] utility:
 .IP
-.nf
-\f[C]
+.EX
 $ sonivoxrender ants.mid | pacat
-\f[R]
-.fi
+.EE
 .SH BUGS
-.PP
-See Tickets at GitHub <https://github.com/pedrolcl/sonivox/issues/>
+See Tickets at GitHub \c
+.UR https://github.com/pedrolcl/sonivox/issues/
+.UE \c
 .SH LICENSE AND COPYRIGHT
-.PP
 Licensed under the Apache License, Version 2.0
 .PP
-Copyright (c) 2022-2024 Pedro L\['o]pez-Cabanillas and contributors
+Copyright (c) 2022\-2024 Pedro López\-Cabanillas and contributors
 .SH AUTHORS
-Pedro L\['o]pez-Cabanillas <plcl@users.sf.net>.
+Pedro López\-Cabanillas \c
+.MT plcl@users.sf.net
+.ME \c.

example/sonivoxrender.1.md

@@ -7,7 +7,7 @@
 
 # SYNOPSIS
 
-| **sonivoxrender** [**-h**] [**-d** _file.dls_] [**-r** _0..4_] [**-w** _0..32765_] _midi_file_
+| **sonivoxrender** [**-h**] [**-d** _file.dls_] [**-r** _0..4_] [**-w** _0..32767_] [**-n** _0..32767_] [**-c** _0..4_] [**-l** _0..32767_] [**-v** _0..100_]  _midi_file_
 
 # DESCRIPTION
 
@@ -30,7 +30,23 @@ It reads .MID (Standard MIDI Files) file format, and writes an audio stream to t
 
 -w  _reverb_wet_
 
-:   Reverb wet level between 0 and 32765.
+:   Reverb wet level between 0 and 32767.
+
+-n  _reverb_dry_
+
+:   Reverb dry level between 0 and 32767.
+
+-c  _chorus_preset_
+
+:   Chorus preset between 0 and 4: 0=no, 1..4=presets.
+
+-l _chorus_level_
+
+:   Chorus level between 0 and 32767.
+
+-v _master_volume_
+
+:   Master volume between 0 and 100, default to 90.
 
 ## Arguments