Pushed v1.23 code

- PAT2SMP now has higher quality sound. Especially noticable in LO mode.
- PAT2SMP LO mode now uses 20864Hz (E-3 finetune 0) instead of 22168Hz
  (F-3 finetune +1) for slightly more room, and still has better quality
  than before because of the improvement above!
- The optional 2x downsample during sample load is now of better quality.
  Or it should be, at least...
- Fixed some bugs in the "sampling" screen (SAMPLER screen -> Sample)
- The sampling feature now takes up less CPU time while downsampling. Also
  fixed a small error in the sinc resampling interpolation.
- Don't block GUI input and don't show red mouse pointer for a while every time
  you cancel a requester. No idea why PT did this in the first place, what a
  workflow limiter!
- Bugfix: Don't reset playback time counter on "play pattern"
- protracker.ini: "A500LOWPASSFILTER" has been replaced with "FILTERMODEL"
  (A1200 or A500). Removed NORMALIZESAMPLING and SAMPLELOWPASS (deprecated).
- Some other minor changes not worth of a mention
- Code cleanup

.gitignore

@@ -22,3 +22,5 @@ vs2019_project/pt2-clone/Debug/pt2-clone.vcxproj.FileListAbsolute.txt
 *.db-wal
 *.db-shm
 pt_pal_editor/vs2019_project/pt_pal_editor/Debug/pt_pal_editor.vcxproj.FileListAbsolute.txt
+vs2019_project/pt2-clone/Release/pt2-clone-win64.exe.recipe
+*.recipe

release/help.txt

@@ -11,6 +11,9 @@ https://support.microsoft.com/en-us/help/2977003/the-latest-supported-visual-c-d
  * Is there a way to run the program in fullscreen mode?
  - Press F11. On some keyboards/configurations you may need to press fn+F11.
 
+ * How can I toggle the audio type between Amiga 500 and Amiga 1200?
+ - Press F12. You can also change "FILTERMODEL" in protracker.ini.
+
  * Alt+F5/ALT+F5 (copy/paste) doesn't work!
  - Windows: you need to make sure those keybinding are disabled in
    'GeForce Experience' if you have that software installed.
@@ -170,10 +173,10 @@ https://support.microsoft.com/en-us/help/2977003/the-latest-supported-visual-c-d
 
  - Quality mode: -
  HI: 28604Hz - Play the new sample at note A-3 (finetune +4)
- LO: 22168Hz - Play the new sample at note F-3 (finetune +1)
+ LO: 20864Hz - Play the new sample at note E-3 (finetune  0)
 
  HI has the highest possible quality, but can only fit 2.29 seconds of audio.
- LO is worse in quality, but can fit slightly more at 2.95 seconds.
+ LO is worse in quality, but can fit slightly more at 3.14 seconds.
 
  You should use this feature in sequences to fit as much rows as you need.
  F.ex. start from row 0 to row 23, then make another sample from row 24 to 47,

release/macos/protracker.ini

@@ -220,38 +220,15 @@ FREQUENCY=48000
 ;
 SAMPLINGFREQ=44100
 
-; Normalize sampled audio before converting to 8-bit
-;        Syntax: TRUE or FALSE
-; Default value: TRUE
-;       Comment: This one is for the audio sampling feature in the SAMPLER
-;         screen. If it's set to TRUE, it will normalize the gain before it
-;         converts the sample to 8-bit in the end. This will preserve as much
-;         amplitude information as possible to lower quantization noise.
-;
-NORMALIZESAMPLING=TRUE
-
-; Audio buffer size
-;        Syntax: Number, in samples
-; Default value: 1024
-;       Comment: Ranges from 128 to 8192. Should be a number that is 2^n
-;          (128, 256, 512, 1024, 2048, 4096, 8192, ...). The number you input
-;          isn't necessarily the actual value the audio API decides to use.
-;          Lower means less audio latency but possible audio issues, higher
-;          means more audio latency but less chance for issues.
+; Filter model (Amiga model)
+;        Syntax: A500 or A1200
+; Default value: A1200
+;       Comment: Selects what kind of Amiga to simulate (lp/hp filters).
+;         A1200 has sharper sound but more aliasing, while A500 has more
+;         filtered (muddy) sound but less aliasing. The filter model can
+;         also be toggled by pressing F12 in the program.
 ;
-BUFFERSIZE=1024
-
-; Amiga 500 low-pass filter (not the "LED" filter)
-;        Syntax: TRUE or FALSE
-; Default value: FALSE
-;       Comment: Enabling this will simulate the ~4421Hz 6dB/oct RC low-pass
-;         filter present in almost all Amigas. This will make the sound a bit
-;         muddier. On Amiga 1200, the cut-off is ~34kHz (sharper sound). This
-;         can also be toggled in the tracker by pressing F12. This must not be
-;         confused with the "LED" filter which can be toggled with the pattern
-;         command E0x.
-;
-A500LOWPASSFILTER=FALSE
+FILTERMODEL=A1200
 
 ; Stereo separation
 ;        Syntax: 0 to 100 (percent)
@@ -261,18 +238,15 @@ A500LOWPASSFILTER=FALSE
 ;
 STEREOSEPARATION=20
 
-
-; Low-pass samples before getting 2x downsampled during loading (if requested)
-;        Syntax: TRUE or FALSE
-; Default value: TRUE
-;       Comment: Set to false if you want slightly sharper sound when loading
-;         samples that are 2x downsampled (if requested).
-;         Keep in mind that you might get more aliasing in the sound if you
-;         disable this, and certain sounds with a lot of treble might sound
-;         weird (f.ex. hi-hats and cymbals). If the sample you load have a
-;         frequency below 22kHz, it will never be downsampled (and thus this
-;         setting changes nothing). I recommend keeping this set to TRUE.
+; Audio buffer size
+;        Syntax: Number, in unit of samples (not bytes)
+; Default value: 1024
+;       Comment: Ranges from 128 to 8192. Should be a number that is 2^n
+;          (128, 256, 512, 1024, 2048, 4096, 8192, ...). The number you input
+;          isn't necessarily the actual value the audio API decides to use.
+;          Lower means less audio latency but possible audio issues, higher
+;          means more audio latency but less chance for issues.
 ;
-SAMPLELOWPASS=TRUE
+BUFFERSIZE=1024
 
 ; End of config file

release/other/protracker.ini

@@ -220,38 +220,15 @@ FREQUENCY=48000
 ;
 SAMPLINGFREQ=44100
 
-; Normalize sampled audio before converting to 8-bit
-;        Syntax: TRUE or FALSE
-; Default value: TRUE
-;       Comment: This one is for the audio sampling feature in the SAMPLER
-;         screen. If it's set to TRUE, it will normalize the gain before it
-;         converts the sample to 8-bit in the end. This will preserve as much
-;         amplitude information as possible to lower quantization noise.
-;
-NORMALIZESAMPLING=TRUE
-
-; Audio buffer size
-;        Syntax: Number, in samples
-; Default value: 1024
-;       Comment: Ranges from 128 to 8192. Should be a number that is 2^n
-;          (128, 256, 512, 1024, 2048, 4096, 8192, ...). The number you input
-;          isn't necessarily the actual value the audio API decides to use.
-;          Lower means less audio latency but possible audio issues, higher
-;          means more audio latency but less chance for issues.
+; Filter model (Amiga model)
+;        Syntax: A500 or A1200
+; Default value: A1200
+;       Comment: Selects what kind of Amiga to simulate (lp/hp filters).
+;         A1200 has sharper sound but more aliasing, while A500 has more
+;         filtered (muddy) sound but less aliasing. The filter model can
+;         also be toggled by pressing F12 in the program.
 ;
-BUFFERSIZE=1024
-
-; Amiga 500 low-pass filter (not the "LED" filter)
-;        Syntax: TRUE or FALSE
-; Default value: FALSE
-;       Comment: Enabling this will simulate the ~4421Hz 6dB/oct RC low-pass
-;         filter present in almost all Amigas. This will make the sound a bit
-;         muddier. On Amiga 1200, the cut-off is ~34kHz (sharper sound). This
-;         can also be toggled in the tracker by pressing F12. This must not be
-;         confused with the "LED" filter which can be toggled with the pattern
-;         command E0x.
-;
-A500LOWPASSFILTER=FALSE
+FILTERMODEL=A1200
 
 ; Stereo separation
 ;        Syntax: 0 to 100 (percent)
@@ -261,18 +238,15 @@ A500LOWPASSFILTER=FALSE
 ;
 STEREOSEPARATION=20
 
-
-; Low-pass samples before getting 2x downsampled during loading (if requested)
-;        Syntax: TRUE or FALSE
-; Default value: TRUE
-;       Comment: Set to false if you want slightly sharper sound when loading
-;         samples that are 2x downsampled (if requested).
-;         Keep in mind that you might get more aliasing in the sound if you
-;         disable this, and certain sounds with a lot of treble might sound
-;         weird (f.ex. hi-hats and cymbals). If the sample you load have a
-;         frequency below 22kHz, it will never be downsampled (and thus this
-;         setting changes nothing). I recommend keeping this set to TRUE.
+; Audio buffer size
+;        Syntax: Number, in unit of samples (not bytes)
+; Default value: 1024
+;       Comment: Ranges from 128 to 8192. Should be a number that is 2^n
+;          (128, 256, 512, 1024, 2048, 4096, 8192, ...). The number you input
+;          isn't necessarily the actual value the audio API decides to use.
+;          Lower means less audio latency but possible audio issues, higher
+;          means more audio latency but less chance for issues.
 ;
-SAMPLELOWPASS=TRUE
+BUFFERSIZE=1024
 
 ; End of config file

release/win32/protracker.ini

@@ -220,38 +220,15 @@ FREQUENCY=48000
 ;
 SAMPLINGFREQ=44100
 
-; Normalize sampled audio before converting to 8-bit
-;        Syntax: TRUE or FALSE
-; Default value: TRUE
-;       Comment: This one is for the audio sampling feature in the SAMPLER
-;         screen. If it's set to TRUE, it will normalize the gain before it
-;         converts the sample to 8-bit in the end. This will preserve as much
-;         amplitude information as possible to lower quantization noise.
-;
-NORMALIZESAMPLING=TRUE
-
-; Audio buffer size
-;        Syntax: Number, in samples
-; Default value: 1024
-;       Comment: Ranges from 128 to 8192. Should be a number that is 2^n
-;          (128, 256, 512, 1024, 2048, 4096, 8192, ...). The number you input
-;          isn't necessarily the actual value the audio API decides to use.
-;          Lower means less audio latency but possible audio issues, higher
-;          means more audio latency but less chance for issues.
+; Filter model (Amiga model)
+;        Syntax: A500 or A1200
+; Default value: A1200
+;       Comment: Selects what kind of Amiga to simulate (lp/hp filters).
+;         A1200 has sharper sound but more aliasing, while A500 has more
+;         filtered (muddy) sound but less aliasing. The filter model can
+;         also be toggled by pressing F12 in the program.
 ;
-BUFFERSIZE=1024
-
-; Amiga 500 low-pass filter (not the "LED" filter)
-;        Syntax: TRUE or FALSE
-; Default value: FALSE
-;       Comment: Enabling this will simulate the ~4421Hz 6dB/oct RC low-pass
-;         filter present in almost all Amigas. This will make the sound a bit
-;         muddier. On Amiga 1200, the cut-off is ~34kHz (sharper sound). This
-;         can also be toggled in the tracker by pressing F12. This must not be
-;         confused with the "LED" filter which can be toggled with the pattern
-;         command E0x.
-;
-A500LOWPASSFILTER=FALSE
+FILTERMODEL=A1200
 
 ; Stereo separation
 ;        Syntax: 0 to 100 (percent)
@@ -261,18 +238,15 @@ A500LOWPASSFILTER=FALSE
 ;
 STEREOSEPARATION=20
 
-
-; Low-pass samples before getting 2x downsampled during loading (if requested)
-;        Syntax: TRUE or FALSE
-; Default value: TRUE
-;       Comment: Set to false if you want slightly sharper sound when loading
-;         samples that are 2x downsampled (if requested).
-;         Keep in mind that you might get more aliasing in the sound if you
-;         disable this, and certain sounds with a lot of treble might sound
-;         weird (f.ex. hi-hats and cymbals). If the sample you load have a
-;         frequency below 22kHz, it will never be downsampled (and thus this
-;         setting changes nothing). I recommend keeping this set to TRUE.
+; Audio buffer size
+;        Syntax: Number, in unit of samples (not bytes)
+; Default value: 1024
+;       Comment: Ranges from 128 to 8192. Should be a number that is 2^n
+;          (128, 256, 512, 1024, 2048, 4096, 8192, ...). The number you input
+;          isn't necessarily the actual value the audio API decides to use.
+;          Lower means less audio latency but possible audio issues, higher
+;          means more audio latency but less chance for issues.
 ;
-SAMPLELOWPASS=TRUE
+BUFFERSIZE=1024
 
 ; End of config file

release/win64/protracker.ini

@@ -220,38 +220,15 @@ FREQUENCY=48000
 ;
 SAMPLINGFREQ=44100
 
-; Normalize sampled audio before converting to 8-bit
-;        Syntax: TRUE or FALSE
-; Default value: TRUE
-;       Comment: This one is for the audio sampling feature in the SAMPLER
-;         screen. If it's set to TRUE, it will normalize the gain before it
-;         converts the sample to 8-bit in the end. This will preserve as much
-;         amplitude information as possible to lower quantization noise.
-;
-NORMALIZESAMPLING=TRUE
-
-; Audio buffer size
-;        Syntax: Number, in samples
-; Default value: 1024
-;       Comment: Ranges from 128 to 8192. Should be a number that is 2^n
-;          (128, 256, 512, 1024, 2048, 4096, 8192, ...). The number you input
-;          isn't necessarily the actual value the audio API decides to use.
-;          Lower means less audio latency but possible audio issues, higher
-;          means more audio latency but less chance for issues.
+; Filter model (Amiga model)
+;        Syntax: A500 or A1200
+; Default value: A1200
+;       Comment: Selects what kind of Amiga to simulate (lp/hp filters).
+;         A1200 has sharper sound but more aliasing, while A500 has more
+;         filtered (muddy) sound but less aliasing. The filter model can
+;         also be toggled by pressing F12 in the program.
 ;
-BUFFERSIZE=1024
-
-; Amiga 500 low-pass filter (not the "LED" filter)
-;        Syntax: TRUE or FALSE
-; Default value: FALSE
-;       Comment: Enabling this will simulate the ~4421Hz 6dB/oct RC low-pass
-;         filter present in almost all Amigas. This will make the sound a bit
-;         muddier. On Amiga 1200, the cut-off is ~34kHz (sharper sound). This
-;         can also be toggled in the tracker by pressing F12. This must not be
-;         confused with the "LED" filter which can be toggled with the pattern
-;         command E0x.
-;
-A500LOWPASSFILTER=FALSE
+FILTERMODEL=A1200
 
 ; Stereo separation
 ;        Syntax: 0 to 100 (percent)
@@ -261,18 +238,15 @@ A500LOWPASSFILTER=FALSE
 ;
 STEREOSEPARATION=20
 
-
-; Low-pass samples before getting 2x downsampled during loading (if requested)
-;        Syntax: TRUE or FALSE
-; Default value: TRUE
-;       Comment: Set to false if you want slightly sharper sound when loading
-;         samples that are 2x downsampled (if requested).
-;         Keep in mind that you might get more aliasing in the sound if you
-;         disable this, and certain sounds with a lot of treble might sound
-;         weird (f.ex. hi-hats and cymbals). If the sample you load have a
-;         frequency below 22kHz, it will never be downsampled (and thus this
-;         setting changes nothing). I recommend keeping this set to TRUE.
+; Audio buffer size
+;        Syntax: Number, in unit of samples (not bytes)
+; Default value: 1024
+;       Comment: Ranges from 128 to 8192. Should be a number that is 2^n
+;          (128, 256, 512, 1024, 2048, 4096, 8192, ...). The number you input
+;          isn't necessarily the actual value the audio API decides to use.
+;          Lower means less audio latency but possible audio issues, higher
+;          means more audio latency but less chance for issues.
 ;
-SAMPLELOWPASS=TRUE
+BUFFERSIZE=1024
 
 ; End of config file