Sunday 17 February 2019

Mod player in Rust - part 4. Finally some music

In this post we will finally have some music. Below a quick preview of the player in action. 

In this post I will spend some time discussing how the Amiga sound hardware worked. The original mod file format and how it is played back is intimately linked with the Amiga hardware.
Tracking the play position

The speed at which the mod is played is based on counting vertical blank interrupts ( the interrupt triggered at each hardware screen refresh ) On PAL systems the VBI frequency is 50hz and on the NTSC systems it is 60hz. The song's playing speed is defined as the number of vertical blanks between two lines in a pattern. ( This means that songs developed on a PAL system will play slightly too fast on an NTSC system. )
Mod files also have an option to specify the playback speed in beats per minute (BPM) but this was not used often because many of the mod songs were intended to be used with demos. Demos often ran all of their code on the vertical blank interrupt but for many values of BPM the player could not have run on the same interrupt, making it much harder to manage the processor time. ( There also seems to be a lot variability between players and mods over how exactly the BPM should be handled ) For now I am going to concentrate on mods that used VBI based speed.
When playing a mod with a VBI based speed on an Amiga, keeping track of the speed was just a matter of checking how many times the interrupt had been called since playing the last line. This is not an option for my Rust player as it does not run on an interrupt. Instead I can use the device sample rate to track theoretical vblanks using the following formula
samplesPerVblank = \frac {deviceSampleRate}{vblankFrequency}
If the playback device has a sample rate of 48000 and I am simulating the playback on a PAL device, the samplesPerVblank is 48000/50 = 960. That means that after every 960 samples it is time to increment the vertical blank counter by one, and if it matches the playback speed it is time to play a new line.
Once we have played all the lines from the current pattern it is time to move to the next pattern.
The PlayerState keeps track of all the above information.
struct PlayerState{
    song_pattern_position: u32,             // where in the pattern table are we currently
    current_line: u32,                      // current position in the pattern
    song_speed: u32,                        // in vblanks
    current_vblank : u32,                   // how many vblanks since last play line
    current_vblank_sample : u32,            // how many device samples have we played for the current 'vblank'
    samples_per_vblank: u32,                // how many device samples per 'vblank'

Playing samples

The mod files use frequency shifting to player different notes of the same sound. If the original sound effect was sampled at 10kh, playing it back at double the original speed increases its frequency to 20kh and plays it one octave higher.
On the original Amiga hardware the DMA (Direct Memory Access) is responsible for transferring audio data from memory and sending it to the DAC ( Digital to Analog Converter ). The DMA speed is controlled by setting the number of system clock ticks between samples. On a PAL system each clock tick is 281.937 nanoseconds, thus there are 3,546,895 clock ticks per second ( on a NTSC system there a tick is 279.365 nanoseconds and there are 3,579,545 clock ticks per second).
For hardware design reasons the smallest interval between samples is 123 system clock ticks on a PAL system so this works out as 34.678 micro seconds per sample or about 28.8 khz. (On NTSC systems the minimum clock counts between samples is 124)
The note data in mod files specify the number of system clocks ticks each sample byte is played for before fetching a new one. This is the period in the note data. When the note states a period of 428 it means that means that each sample should be held for 120 microseconds (428 clocks * 0.281397 microsecs per clock = 120 microsecs) yielding a sample rate of about 8kHz. To play the same sample one octave higher a period of 214 would be used.
We can emulate this by behaviour by resampling the sounds at a speed that matches intended playback speed. We just need to map device sample duration into Amiga clock ticks; If the playback device has a sample rate of 48,000 we know that each device sample corresponds to 73.89 clock ticks (3,546,895/48,000 = 73.89). So each time we calculate a new sample we advance the playback position by;
\frac {clockTicksPerSecond}{period*deviceSampleRate} = \frac{clockTicksPerDeviceSample}{period}
To keep track of clockTicksPerDeviceSampleI add an additional variable to the PlayerState;
    clock_ticks_per_device_sample : f32,    // how many amiga hardware clock ticks per device sample
Some of the mod effects are created by varying the channel's period value while the note is playing. This is why the channel stores the period value rather than the value for clockTicksPerDeviceSample/period
In addition to the period. Each channel need to track what it's sample number, position within the sample and playback volume;
struct ChannelInfo {
    sample_num: u8,         // which sample is playing 
    sample_pos: f32,         
    period : f32,           
    volume: f32, 

Putting it together

I am going to quickly run through the main blocks of the playing code. This is all pretty straight forward Rust code.
First, I replace the instrument playing logic from previous version of the code with a call to next_sample which returns a tuple that has the left and right samples;
        cpal::StreamData::Output { buffer: cpal::UnknownTypeOutputBuffer::F32(mut buffer) } => {
            for sample in buffer.chunks_mut(format.channels as usize) {
                let ( left, right ) = next_sample(&song, &mut player_state);
                sample[0] = left;
                sample[1] = right;
The above uses destructuring to assign the members of the returned tuple into left and right. Sadly, Rust doesn't seem to allow for destructuring directly into the destination array.
The code for next_sample tracks how many samples it has played for the current vblank and whether enough vblanks have passed to play a new line. The code keeps track of vblanks and lines separately because many of the effects require updates that occur on the vblank.
fn next_sample(song: &Song, player_state: &mut PlayerState) -> (f32, f32) {
    let mut left = 0.0;
    let mut right = 0.0;

    // Have we reached a new vblank
    if player_state.current_vblank_sample >= player_state.samples_per_vblank {
        player_state.current_vblank_sample = 0;

        // Is it time to play a new note line
        if player_state.current_vblank >= player_state.song_speed {
            player_state.current_vblank = 0;
            play_line( song, player_state );
        player_state.current_vblank += 1;
    player_state.current_vblank_sample += 1;
    // .. handle sample play back
The next interesting bit is in play_line which updates the current line and pattern position and plays a note for each channel;
fn play_line(song: &Song, player_state: &mut PlayerState ) {
    // use the pattern table to work out which pattern we are in    
    let pattern_idx = song.pattern_table[player_state.song_pattern_position as usize];
    let pattern = &song.patterns[ pattern_idx as usize];

    pattern.print_line(player_state.current_line as usize);
    let line = &pattern.lines[ player_state.current_line as usize ];
    for channel_number in 0..line.len(){
        play_note(&line[ channel_number as usize ], player_state, channel_number, song);
    player_state.current_line += 1;
    if player_state.current_line >= 64 {
        // end of the pattern
        player_state.song_pattern_position += 1;
        player_state.current_line = 0;
The play_note function sets up the new sample ( if one is set ) and handles effects. For now, this code will only handle a minimal set of effects to keep the code size and complexity down. The function needs access to the PlayerState structure instead of just the channel data because some of the effects, like setting the speed, are global.
fn play_note(note: &Note, player_state: &mut PlayerState, channel_num: usize, song: &Song) {
    if note.sample_number > 0 {
        // sample number 0, means that the sample keeps playing. The sample indices starts at one, so subtract 1 to get to 0 based index
        player_state.channels[channel_num].volume = song.samples[(note.sample_number - 1) as usize].volume as f32 / 64.0;    // Get volume from sample
        player_state.channels[channel_num].sample_num = note.sample_number;
        player_state.channels[channel_num].sample_pos = 0.0;
        if note.period != 0 {
            player_state.channels[channel_num].period = note.period as f32;

    match note.effect {
        Effect::SetSpeed{ speed } => {
            player_state.song_speed = speed as u32;
        Effect::SetVolume{ volume } => {
            player_state.channels[channel_num].volume = volume as f32;
        Effect::None => {}
        _ => {
            println!("Unhandled effect" );
Finally, the second half of the next_sample performs the sample retrieval and mixing into device channels
for channel_number in 0..player_state.channels.len() {
    let channel_info: &mut ChannelInfo = &mut player_state.channels[channel_number];
    if channel_info.sample_num != 0 {
        let current_sample: &Sample = &song.samples[(channel_info.sample_num - 1) as usize];
        let mut channel_value: f32 = current_sample.samples[(channel_info.sample_pos as u32) as usize] as f32;             
        // max channel vol (64), sample range [ -128,127] scaled to [-1,1] 
        channel_value *= channel_info.volume / (128.0*64);

        // update position and check if we have reached the end of the sample
        channel_info.sample_pos +=  player_state.clock_ticks_per_device_sample / channel_info.period;
        if channel_info.sample_pos >= current_sample.size as f32 {
            channel_info.sample_num = 0;
        let channel_selector = ( channel_number as u8 ) & 0x0003; 
        if channel_selector == 0 || channel_number as u32 == 0 || channel_number == 3 {
            left += channel_value;
        } else {
            right += channel_value;
I need to control the range of the sound output from the mixer so it has a range that works with the playback device. As cpal uses the range [-1,1] for devices supporting floating point channels I have decided to use the same range for mixer output.
Mod files only support stereo. The mod channels 0 and 3 should be played on the left channel and 1 and 2 on the right. If there are more channels, the same allocation scheme is used ( 4 and 7 are let, 5 and 6 are right)


The code we have developed so far will play simple mod files that have minimal effects. The mod file I have included ( axelf.mod ) is an example of such a mod file. Most mods make heavy use of effects and cannot be played with this code.
In my next post I will look into supporting more effects so I can support a larger set of mod files. The code is also growing to be quite large so it is time to look at a Rust modules as a better of organizing code.
All the code for the current article is at

Monday 4 February 2019

Mod player in Rust - part 3. Audio and threads

I finished my last post with having a fully parsed mod file in memory and ready to be played. Before I start working on the code that handles the mod playing I need to get some important building blocks in place.
  • Audio. Rust does not come with audio capabilities so I need to add crate that provides audio streaming capability
  • Threads. In order to stream the music while letting other processing take place the player needs to run on its own thread. The main thread should still be able to control the player thread which means I need to be able to create threads and pass data between them.
By the end of this post I want to get the code into a state where the main thread lets the user pick instruments from the mod file and play them back on the audio thread.


Rust uses crates to add new functionality to programs. To access the functionality provided by a crate we need to add it as a dependency to the .toml file and add extern crate  to the .rs file that uses functionality from the crate.
So far I haven't really looked at .toml files. It is just something automatically created by cargo when I create a new project. To add a dependency I need to create a new section called [dependencies]which has a list of dependency names and the version numbers.
Dependencies often have their own dependencies. Rust will automatically include any sub-dependencies when it builds the project.
In my case I decided to use the cpal crate for audio streaming. To add it to my project I added the following lines to the .toml file
cpal = "0.8.2"
Next time I build the project, cargo will download and compile the cpal crate and any crates that it depends on. Rust will also keep track of which version(s) it has downloaded so it will only download the dependencies again if the version is explicitly changed.

cpal and closures

I decided to use cpal because it provides the right level of access to the audio hardware, has some good documentation and seems to be widely used and is still supported.
The cpal documentation gives a pretty good overview of how to setup streaming audio. Using their example as a starting point I created the following code to stream silence.
fn setup_stream( ) {
    let device = cpal::default_output_device().expect("Failed to get default output device");
    let format  = device.default_output_format().expect("Failed to get default output format");
    let event_loop = cpal::EventLoop::new();
    let stream_id = event_loop.build_output_stream(&device, &format).unwrap();
    event_loop.play_stream(stream_id.clone()); |_, data| {
        match data {
            cpal::StreamData::Output { buffer: cpal::UnknownTypeOutputBuffer::F32(mut buffer) } => {
                for sample in buffer.chunks_mut(format.channels as usize) {
                    sample[0] = 0.0;
                    sample[1] = 0.0;
The first half creates a device, picks a format and creates and starts an output stream. The interesting part is the call to, which transfer control of the current thread to the cpal streaming code that will run indefinitely. The closure passed in as an argument is called whenever it needs a new piece of audio to play back.
Closures in Rust are like anonymous functions that can take ownership of their immediate environment.
Closures are defined using the syntax |arg| code where arg is zero or more arguments and code is the code that is executed when the closure is called. Optionally, the closure definition can be prefixed by a move that transfers ownership of any used variables to the closure.
In our case the closure takes two arguments _ and _data. We define a two argument closure because that is what cpal expects to call. We don't care about the first argument, a stream id so the code uses _ which is a special variable name prefix in Rust to indicate that the variable is not used ( and thus shouldn't generate and unused variable warning).
The second argument is the data buffer that should receive the sound samples. The match syntax is used the extract an output buffer of type f32. For now, I will only deal with f32 sound buffers to keep the code simple. The inner loop copies the sound into the buffer. (The code assumes we are dealing with a stereo buffer. If that is not the case the code will panic)
The above code works but will block forever because the call to never returns.

threads and message channels

Creating a thread in Rust is simple. The call thread::spawn( closure ) will create a new thread and start its execution at the given closure.
I can modify the earlier code to wrap the inside a thread closure to let it run in its own thread. Both closures are move closures so the thread will take ownership of any variables that are used in the event loop.
   thread::spawn( move || { |_, data| {
Now I have two threads; the main thread and the audio thread created specifically to run the audio event loop.
I want to able to control what the audio stream is playing from the main thread so they need some way of exchanging data. Rust's std::sync library has support for message passing between threads. The messages are passed along a channel with a transmitter and receiver.
A channel is created with
    let (tx,rx) = mpsc::channel();
where tx is the transmitter and rx is the receiver. Because I want the main thread to send messages to the audio thread I will return tx as a return value from the setup_stream function.
Before I can declare the return type of setup_stream I need to define the type of messages that will travel on the channel.
enum PlayerCommand{
    PlayInstrument{ index : u8 }
Now I can change the call signature of setup_stream to indicate that it returns a PlayerCommand channel sender;
fn setup_stream(  ) -> mpsc::Sender<PlayerCommand> {
The player thread needs to handle commands when it receives them. So for each iteration of audio streaming I want to check if there is a new message and process it. The channel Receiver has a non-blocking try_recv() method that will return a Result that is Ok if a new message was available. The first bit of code in the event_loop closure is now;
    let message = rx.try_recv();
    if message.is_ok() {
        match message.unwrap() {
            PlayerCommand::PlayInstrument{ index } => { 
                instrument_number = index;
                instrument_pos = 0;
The code above is pretty straightforward. It checks if a new message is available and then uses match to handle it. instrument_number and instrument_pos are used to track current playing instrument and the playing position of the instument. Both of the variables are defined in the scope setup_stream but because both surrounding closures are marked as move the thread owns them.
The main thread can now store the sender and use it to send messages
    let tx = setup_stream();
        // loop for picking instrument
        // ...
        tx.send(PlayerCommand::PlayInstrument{ index : instrument } );

Sharing data between threads

I can now send commands to the player thread but it also needs to have access to the mod data.
If I try to pass a &Song to the thread I get complaints from the compiler about it needing a static lifetime. The problem is that the compiler does not know how long the player thread runs for and it could potentially run longer than the original Song object exists for. This would leave the player thread with an invalid reference.
I could fix the issue by doing what the compiler suggests and give &Song argument a static lifetime mofifier but this is not what I really want. I potentially do want to change the song for another song at some later point in the program and free up the resoures taken by the first song. I need smart pointers.
Rust comes with a range of different smart pointers that are designed for different use cases. The one I need here is Arc<> or Atomic Reference Counted pointers. It is safe to use between threads because it is atomic and thus cant be left in an undefined state even if two thread try to access it at the exactly same time. Because it is reference counted it does not need to worry about setting life times, once the last reference to Ar<> goes away,it is automatically destroyed.
Because Arc<> is reference counted it uses the method clone to create a new instances of itself. Calling clone does not make a copy of the underlying data but merely increments the reference counter by one and hands back a new reference to the same underlying data.
A new instance of Arc<> is created using sync::Arc::new( data ) so the lines for creating the song and setting up audio thread become.
    let song = sync::Arc::new( read_mod_file("AXELF.MOD") );
    let tx = setup_stream(song.clone());
I also need to make the matching change to setup_stream so it accepts a Arc
fn setup_stream( song : sync::Arc<Song> ) -> mpsc::Sender<PlayerCommand> {

Playing an instrument

I can now rewrite the streaming code to play out the selected instrument from the mod file.
    // handle messages like before...
    match data {
        cpal::StreamData::Output { buffer: cpal::UnknownTypeOutputBuffer::F32(mut buffer) } => {
            for sample in buffer.chunks_mut(format.channels as usize) {
                let mut sound_sample = 0.0;
                if instrument_number > 0 {
                    let the_sample : &Sample = &current_song.samples[ instrument_number as usize];
                    sound_sample = the_sample.samples[ instrument_pos as usize ] as f32/ 256.0;
                    instrument_pos += 1;
                    if instrument_pos >= the_sample.size {
                        instrument_number = 0;
                sample[0] = sound_sample;
                sample[1] = sound_sample;
        _ => (),
This is all pretty standard Rust code. Initially the sound_sample is set to zero but if the instrument_number is non-zero we play play keep playing its sample data until we reach the end and switch back to instrument zero to stop playing.
The code now allows us to playback individual sounds samples from the mod file. This gives us the first audible confirmation that the code is doing the right thing.
In the next post I want to play back the note data so I can start hearing the actual mod music.
All the code for the current article is at

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