YM7128B

This library aims to emulate the Yamaha YM7128B Surround Processor.

The original goal is to contribute to the emulation of the AdLib Gold sound card, which used such integrated circuit to add surround effects to the audio output, as heard in the beautiful soundtrack of the Dune videogame, made specifically for that rare sound card.

As I do not know how the actual circuits are made, I cannot emulate the chip perfectly. Instead, my goal is to simulate the overall sound, based on the short information available, and with some assumptions based on other Yamaha chips of the same age, as far as I know.

If anybody can analyze a decapped chip, that would of course be the best information, as already done for many vintage sound chips.

I am willing to get some real chips, as long as they are available for purchase, at least for black-box comparisons with dedicated test vectors.

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Engines

The library provides chip emulation with the following engines:

• Fixed: it should emulate an actual chip, using fixed point arithmetics. It features all the actual chip constraints, like sample rates, the presence of an output oversampler, and saturated arithmetics. Algorithms are rather slow to compute, and resampling is typically heavy.

• Float: like Fixed, using floating point arithmetics instead. It increases the sound quality, while keeping the architecture similar to that of an actual chip. Algorithms are fast to compute, but resampling still has a significant impact on performance.

• Ideal: it represents an ideal implementation of the algorithms. Arithmetics are in floating point, there are no saturations, and no sample rate conversions, not even at the output stage. Algorithms are fast, despite the loss in accuracy with respect to an actual chip.

• Short: somewhat an hybrid of Fixed and Ideal, in that it uses full-range 16-bit fixed point arithmetics with saturations, but without sample rate conversions. Algorithms are fast, despite the loss in accuracy with respect to an actual chip.

In the following table, an overview of the features of each engine:

Feature	Fixed	Float	Ideal	Short
Input filter	suggested: 6th order 15 kHz	suggested: 6th order 15 kHz	not required	not required
Input signal	Q1.13	double, normalized	double	Q1.15
Input rate	23550 Hz	23550 Hz	suggested: above 40 kHz	suggested: above 40 kHz
Saturated arithmetics	yes	yes, normalized	no	yes
Signal operand	Q1.15	double	double	Q1.15
Gain operand	Q1.11	double	double	Q1.15
Feedback operand	Q1.5	double	double	Q1.5
Oversampler operand	Q1.11	double	no oversampling	no oversampling
Output rate	47100 Hz	47100 Hz	same as input	same as input
Output signal	Q1.13	double, normalized	double	Q1.15
Output filter	suggested: 3rd order 15 kHz	suggested: 3rd order 15 kHz	not required	not required
Status memory	allocated by the user	allocated by the user	allocated by the user	allocated by the user
Delay memory	part of the status	part of the status	dynamic heap allocation	dynamic heap allocation
Performance	very slow	slow	fast	fast
Accuracy	best?	good	poor	poor

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Usage

To use this library, just include YM7128B_emu.c and YM7128B_emu.h into your project.

All the engines implement the same conceptual flow:

1. Status memory allocation.
2. Call Ctor() method to invalidate internal data.
3. Call Reset() method to clear emulated registers.
4. Call Setup() to allocate internal delay memory (only for Ideal engine).
5. Call Start() to start the algorithms.
6. Processing loop:
   1. Filter input samples.
   2. Resample input samples.
   3. For each sample:
      1. Call Process() method, with appropriate data types.
   4. Resample output samples.
   5. Filter output samples.
7. Call Stop() method to stop the algorithms.
8. Call Dtor() method to deallocate and invalidate internal data.
9. Status memory deallocation.

Register access timings are not emulated.

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Sample rate conversion

The biggest concern is about the weird sampling rates of the original chip: 23.6 kHz for input, and 47.1 kHz for output.

Since modern common audio sample rates are based either on audio CD (44.1 kHz), or legacy professional audio (48 kHz), the sampling rate of most late '80s Yamaha products need sample rate conversion both at input and output to be emulated by software.

Quality realtime conversions of such unusual sample rates intrinsically require more CPU time than the chip emulation itself, so expect them to be the actual bootleneck of the emulation.

Also, sample rate conversions add delays to their outputs, which are not welcome to realtime processing.

Libraries

This library does not provide sample rate conversion itself, because proper conversion (without audible distortion) is not trivial at all. Instead, there are many libraries available, each with its quality rating, performance, and licensing. You can find a comprehensive comparison at Infinite Wave's website.

Personally, I have tried the following open source libraries with success:

• secret rabbit code 0.1.9 (download - github): A nice mature C library, easy to use, licensed as 2-clause BSD. Not the most lightweight for realtime, nor in performance, nor in code size.

• r8brain free 4.6 (github): Free version of a commercial C++ library, licensed as 2-clause BSD. Suitable for realtime, and with small footprint.

• zita-resampler 1.6.2 (download - github): A nice modern C++ library, licensed as GPL3. Suitable for realtime, and with small footprint.

Analog filtering

The datasheet suggests some input and output analog filtering, to reduce aliasing effects, more specifically a 6th order low-pass input filter, and a 3rd order low-pass output filter. Such filters should be considered into the system emulation, because an actual physical system needs them. I do not know how the AdLib Gold filtered them.

I do not have specifications about them, but they should be Butterworth filters (common analog audio filters), and I guess the cut-off frequency is around 15 kHz (common for FM radio, Sound Blaster, etc.).

Again, analog filters are not provided by the library, to give the user freedom about their implementation (e.g. Robert Bristow-Johnson's Audio EQ Cookbook - text).

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Implementation details

Here you can find some descriptions and discussions about implementation details.

Language

I chose C89 with a bit of C99. I was going to use C++20 for my own pleasure, but instead I find good old and mature C to be the best for such a tiny library. C is also easier to integrate with other languages, and the mighty features of a colossal language like C++ are more of a burden than for actual use in such case.

Cross-platform support

The code itself should be cross-platform and clean enough not to give compilation errors or ambiguity.

Currently the library is developed and tested under Windows with MSVC++ 14. Of course, I will at least provide support for gcc and clang under Linux. I do not have access to macOS, but I guess the Linux support should fit.

Code style

I chose the path of verbosity for variable declaration, to help debugging fixed point math and buffering. Indeed, when compiled for performance, all those variable declarations get optimized away easily.

I did not write the code for explicit vectoring, preferring a KISS approach at this stage of development. Actually I am not satisfied about how the MSVC++ compiler is generating machine code, and I guess that optimizing the code for vectoring should improve the performance by some margin, especially the parts for parallel 8-tap delay and output oversampling.

Sample format

The datasheet claims 14-bit floating point sampling for both input and output. There is no information about such esoteric format itself, but there are datasheets of Yamaha products of the same age that do. I know only the YM3014B for reference, which claims 16-bit (linear) dynamic range, against floating point samples with 3-bit exponent and 10-bit mantissa.

Anyway, the datasheet clearly states that the analog input is converted to 14-bit digital signal, so we can assume that samples are 14-bit wide. There is no mention of sign bits, as the A/D converter is monopolar (Vdd/2 center voltage bias), but I think the 14-bit samples are unbiased (signed) by the A/D converter itself, to agree with all the two-complement computations. The reverse operation is done by the D/A converter.

I am actually concerned about the bit size of each sample. Its sounds like the 14-bit multiplication results sound pretty awful for small signals, the way I am emulating the system right now. Tests with 16-bit sample emulation sound less worse, so I am wondering whether signals are actually processed as 14-bit or more.

Multiplication

Being a DSP, one of the most common operations is multiplication. Since this operation requires many logic gates, I guess there is only one multiplier in the chip, reused within the hundreds of clock ticks per sample.

The only information about multiplication comes from the description of feedback coefficients in the datasheet, which states that the 6 data bits of the registers are mapped onto the 6 most significant bits of the 12-bit two-complement operand, sign bit included. I think that the remaining 6 bits are not wasted only for such operation, but instead the very same multiplier is shared also for gain (decibel) multiplications.

So, we have a multiplier that has a signal operand 14-bit wide, and a gain operand 12-bit wide. The result should be a signal itself, 14-bit wide.

The multiplication is implemented as a classic 16-bit x 16-bit signed fixed-point multiplication (aka Q15), keeping only the 14 most significant bits.

Actually, since 14-bit operand sounds awful, unlike the few recordings found on the internet, I am using 16-bit operands right now.

Feedback coefficients

The datasheet is very clear about feedback coefficients: they are 6-bit values mapped directly from the register data to the 12-bit two-complement operand of the multiplier, padded with zeros at the least significant operand bits.

Feedback coefficients seem to be a special case of gains, being mapped directly as operands, while gains are remapped with a lookup table.

Gain coefficients

The datasheet mentions the gain coefficients in terms of decibels, tabulated into a table with 32 entries. Entries go from unity gain (0 dB) at the highest index, down to -60 dB at the second-lowest index, in steps of 2 dB. The lowest index (zero) is reserved for silence. This gives a good granularity to volume levels (2 dB is a common step size in incremental volume control).

I think that such entries are saved into a single table with non-negative 11-bit linear coefficients, from silence (all bits 0) to maximum volume (all bits 1).

I guess the negative coefficients are actually loaded as the one-complement (bit flip) instead of the two-complement, as often seen in Yamaha's synthesizers of the same age, to save silicon area despite a tiny gain error.

Digital delay line

The digital delay line allows up to 100 ms of delay at the nominal input rate. This leads to a buffer of at least 2355 samples. Such buffer should be a shifting FIFO with 32 pre-determined tap positions.

The delay line emulation is actually implemented as a random-access ring buffer, as shifting the whole buffer at each input sample is a waste of time.

Oversampler

The chip includes a 2x oversampling interpolator at the output stage, to help reduce the analog circuitry to reconstruct the output from the 47.1 kHz oversampled D/A data.

I guess the interpolator is a FIR filter that reuses the same DSP circuitry for gain and coefficient fixed-point multiplication, to save silicon area.

The datasheet shows only the magnitude versus freuquency response. So, based on such information, I tried to match the reponse with Iowa Hills FIR Filter Designer 7.0 by trial and error. The parameters I found do not look exacly as per the diagram of the datasheet, but the actual response should not differ too much:

I also think that the kernel is not minimum-phase, to save further silicon area, thanks to the mirrored coefficient values. I am no expert, but it looks like minimum-phase is also not welcome to audio, because of phase-incoherence among frequencies, despite having shorter delays. I left the possibility to choose the minimum-phase feature by configuring the YM7128B_USE_MINPHASE preprocessor symbol.

Floating-point

It is possible to configure the floating point data type used for processing, via the YM7128B_FLOAT preprocessor symbol. It defaults to double (double precision).

Please note that, contrary to common beliefs, the double data type is actually very fast on machines with hardware support for it.

I think that you should switch to float (single precision) only if:

• the hardware is limited to it, or
• if machine code vectorization gets faster, or
• conversion from/to buffer data to/from double precision is slower.

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YM7128B_pipe example

This repository provides a fully-featured example. It is a stream processor, in that it processes sample data coming from standard input, elaborates it, and generates outputs on the standard output.

Please refer to its own help page, by calling the canonical YM7128B_pipe --help, or reading it embedded in its source code.

Usage example with Lubuntu 20.04

1. Ensure the following packages are installed:

sudo apt install alsa-utils build-essential

2. Enter example folder and run make_gcc.sh:

cd example
bash make_gcc.sh

3. You should find the generated executable file as YM7128B_pipe.

4. Play some audio directly with aplay (the \ is for command line continuation), using the dune/warsong preset:

./YM7128B_pipe -r 23550 -f S16_LE --preset dune/warsong < sample_mono_23550Hz_S16LE.raw \
| aplay -c 2 -r 47100 -f S16_LE