First draft

src/Tools/psxfileconv/src/audio/my_xa/mod.rs

@@ -9,8 +9,8 @@ use tool_helper::{Error, Input};
 pub struct Arguments {
     #[clap(value_enum, value_parser, default_value_t=Frequency::High)]
     frequency:    Frequency,
-    #[clap(value_enum, value_parser, default_value_t=Channels::Stereo)]
+    #[clap(value_enum, value_parser, default_value_t=Orality::Stereo)]
-    channels:     Channels,
+    orality:      Orality,
     #[clap(value_enum, value_parser, default_value_t=SampleDepth::Normal)]
     sample_depth: SampleDepth,
 }
@@ -21,8 +21,8 @@ pub enum Frequency {
     Low,
 }
 
-#[derive(Copy, Clone, ValueEnum)]
+#[derive(Copy, Clone, PartialEq, ValueEnum)]
-pub enum Channels {
+pub enum Orality {
     Stereo,
     Mono,
 }
@@ -35,5 +35,5 @@ pub enum SampleDepth {
 
 pub fn convert(args: Arguments, input: Input, output: &mut dyn Write) -> Result<(), Error> {
     let prepared_xa_audio = raw_audio::load_as_i16_audio(input, xa_audio::audio_conversion_settings(&args))?;
-    xa_audio::convert(prepared_xa_audio, output, &args)
+    xa_audio::encode(prepared_xa_audio, output, &args)
 }

src/Tools/psxfileconv/src/audio/my_xa/xa_audio/mod.rs

@@ -1,45 +1,52 @@
+mod xapcm;
+
+use crate::audio::my_xa::Orality;
+
 use super::Arguments;
 use super::raw_audio::{I16Samples, Settings};
 use std::io::Write;
 use symphonia::core::audio::Layout;
 use tool_helper::Error;
 
-const HIGH_FREQUENCY:u32 = 37_800;
+const BLOCKS_PER_SECTOR:usize = 18;
-const  LOW_FREQUENCY:u32 = 18_900;
+const HIGH_FREQUENCY:u32      = 37_800;
+const  LOW_FREQUENCY:u32      = 18_900;
 
 pub fn audio_conversion_settings(arguments: &Arguments) -> Settings {
     let frequency = match arguments.frequency {
         super::Frequency::High => HIGH_FREQUENCY,
         super::Frequency::Low  => LOW_FREQUENCY,
     };
-    let layout    = match arguments.channels {
+    let layout    = match arguments.orality {
-        super::Channels::Stereo => Layout::Stereo,
+        super::Orality::Stereo => Layout::Stereo,
-        super::Channels::Mono   => Layout::Mono,
+        super::Orality::Mono   => Layout::Mono,
     };
 
     Settings::new(frequency, layout)
 }
 
-pub fn convert(input: I16Samples, _output: &mut dyn Write, arguments: &Arguments) -> Result<(), Error> {
+pub fn encode(input: I16Samples, output: &mut dyn Write, arguments: &Arguments) -> Result<(), Error> {
     let (samples_per_block, sample_count) = init_values(&input, arguments);
 
-    for (iteration, sampled_id) in (0..sample_count).step_by(samples_per_block).enumerate() {
+    for (sectors, sampled_id) in (0..sample_count).step_by(samples_per_block).enumerate() {
+        let slice = &input[sampled_id..];
+
         print!("\rIteration: {}; Sample: {}", iteration, sampled_id);
     }
     println!();
     Err(Error::not_implemented("XA conversion"))
 }
 
-fn init_values(input: &I16Samples, arguments: &Arguments) -> (usize, u32) {
+fn init_values(input: &I16Samples, arguments: &Arguments) -> (usize, usize) {
     let samples_per_block = match arguments.sample_depth {
         super::SampleDepth::Normal => 224,
         super::SampleDepth::High   => 112,
     };
 
-    let sample_count = match arguments.channels {
+    let sample_count = match arguments.orality {
-        super::Channels::Stereo => input.len() / 2,
+        super::Orality::Stereo => input.len()/2,
-        super::Channels::Mono   => input.len(),
+        super::Orality::Mono   => input.len(),
     };
 
-    (samples_per_block as usize, sample_count as u32)
+    (samples_per_block as usize, sample_count)
 }

src/Tools/psxfileconv/src/audio/my_xa/xa_audio/xapcm.rs

@@ -0,0 +1,244 @@
+use crate::audio::my_xa::{Arguments, Orality, SampleDepth};
+use tool_helper::Error;
+
+pub struct Encoder {
+    left:  ChannelState,
+    right: ChannelState
+}
+
+impl Encoder {
+    const XA_ADPCM_FILTER_COUNT: i32 = 4;
+    const FILTER_K1: [i16; 5] = [0, 60, 115,  98, 122];
+    const FILTER_K2: [i16; 5] = [0,  0, -52, -55, -60];
+
+    pub fn encode_xa(&mut self, samples: &[i16], sample_limit: i32, data: &mut [u8], args: Arguments) -> Result<(), Error> {
+        const SHIFT_RANGE_4BPS: i32 = 12;
+        const SHIFT_RANGE_8BPS: i32 = 8;
+
+        let channels = [&mut self.left, &mut self.right];
+        match args.sample_depth {
+            SampleDepth::Normal => {
+                let offset = if args.orality == Orality::Stereo {&STEREO_4BIT} else {&MONO_4BIT};
+                let (first_offset, second_offset) = offset;
+
+                for (offset_idx, offset_set) in [first_offset, second_offset].iter().enumerate() {
+                    for (idx, offset) in offset_set.iter().enumerate() {
+                        let byte = Self::encode(channels[idx%2], &samples[offset.sample..], sample_limit + offset.sample_limit, offset.pitch, &mut data[offset.data..], offset.data_shift, offset.data_pitch, Self::XA_ADPCM_FILTER_COUNT, SHIFT_RANGE_4BPS)?;
+                        data[idx + (offset_idx*8)]     = byte;
+                        data[idx + 4 + (offset_idx*8)] = byte;
+                    }                    
+                }
+            },
+            SampleDepth::High   => {
+                let offset_set = if args.orality == Orality::Stereo {&STEREO_8BIT} else {&MONO_8BIT};
+                for (idx, offset) in offset_set.iter().enumerate() {
+                    let byte = Self::encode(channels[idx%2], &samples[offset.sample..], sample_limit + offset.sample_limit, offset.pitch, &mut data[offset.data..], offset.data_shift, offset.data_pitch, Self::XA_ADPCM_FILTER_COUNT, SHIFT_RANGE_8BPS)?;    
+                    data[idx]     = byte;
+                    data[idx + 4] = byte;
+                }
+            }
+        }
+
+        Ok(())
+    }
+
+    fn encode(channel_state: &mut ChannelState, samples: &[i16], sample_limit: i32, pitch: i32, data: &mut [u8], data_shift: i32, data_pitch: i32, filter_count: i32, shift_range: i32) -> Result<u8, Error> {
+        let mut best_mse          = 1u64 << 50u64;
+        let mut best_filer        = 0;
+        let mut best_sample_shift = 0;
+
+        for filter in 0..filter_count {
+            let true_min_shift = Self::find_min_shift(channel_state, samples, sample_limit, pitch, filter as usize, shift_range)?;
+
+            // Testing has shown that the optimal shift can be off the true minimum shift
+            // by 1 in *either* direction.
+            // This is NOT the case when dither is used.
+            let min_shift = if true_min_shift - 1 < 0           {0}           else {true_min_shift - 1};
+            let max_shift = if true_min_shift + 1 > shift_range {shift_range} else {true_min_shift + 1};
+
+            for sample_shift in min_shift..max_shift {
+                let mut proposed = channel_state.clone();
+                Self::attempt_encode(&mut proposed, samples, sample_limit, pitch, data, data_shift, data_pitch, filter, sample_shift, shift_range)?;
+                if best_mse > proposed.mse {
+                    best_mse          = proposed.mse;
+                    best_filer        = filter;
+                    best_sample_shift = sample_shift;
+                }
+            }
+        }
+
+        Self::attempt_encode(channel_state, samples, sample_limit, pitch, data, data_shift, data_pitch, best_filer, best_sample_shift, shift_range)
+    }
+
+    fn attempt_encode(out_channel_state: &mut ChannelState, samples: &[i16], sample_limit: i32, pitch: i32, data: &mut [u8], data_shift: i32, data_pitch: i32, filter: i32, sample_shift: i32, shift_range: i32) -> Result<u8, Error> {
+        let sample_mask  = (0xFFFF >> shift_range) as u8;
+        let nondata_mask = (!(sample_mask << data_shift)) as u8;
+
+        let min_shift = sample_shift;
+        let k1          = Self::FILTER_K1[filter as usize] as i32;
+        let k2          = Self::FILTER_K2[filter as usize] as i32;
+
+        let hdr = ((min_shift & 0x0F) | ((filter as i32) << 4)) as u8;
+
+        out_channel_state.mse = 0;
+
+        for i in 0..28 {
+            // TODO: Code duplication with `find_min_shift`?
+            let     sample         = if i >= sample_limit {0} else {samples[(i*pitch) as usize] as i32 + out_channel_state.qerr};
+            let     previous_value = (k1*out_channel_state.prev1 + k2*out_channel_state.prev2 + (1 << 5)) >> 6;
+            
+            let mut sample_enc = sample - previous_value;
+            sample_enc <<= min_shift;
+            sample_enc +=  1 << (shift_range - 1);
+            sample_enc >>= shift_range;
+
+            // TODO: And this?
+            if sample_enc < (-0x8000 >> shift_range) {sample_enc = -0x8000 >> shift_range}
+            if sample_enc > ( 0x7FFF >> shift_range) {sample_enc =  0x7FFF >> shift_range}
+            sample_enc &= sample_mask as i32;
+
+            let mut sample_dec = ((sample_enc & sample_mask as i32) << shift_range) & 0xFFFF;
+            sample_dec >>= min_shift;
+            sample_dec += previous_value;
+            if sample_dec >  0x7FFF {sample_dec =  0x7FFF}
+            if sample_dec < -0x8000 {sample_dec = -0x8000}
+
+            let sample_error = sample_dec - sample;
+            if sample_error >=  (1 << 30) || sample_error <= -(1 << 30) {
+                return Err(Error::from_text(format!("Sample error exceeds 30bit: {}", sample_error)));
+            }
+
+            data[(i*data_pitch) as usize] = ((data[(i*pitch) as usize] & nondata_mask) as i32 | (sample_enc << data_shift)) as u8;
+
+            out_channel_state.mse += sample_error as u64*sample_error as u64;
+            out_channel_state.prev2 = out_channel_state.prev1;
+            out_channel_state.prev1 = sample_dec;
+        }
+
+        Ok(hdr)
+    }
+
+    fn find_min_shift(channel_state: &ChannelState, samples: &[i16], sample_limit: i32, pitch: i32, filter: usize, shift_range: i32) -> Result<i32, Error> {
+        /* 
+            Assumption made:
+            There is value in shifting right one step further to allow the nibbles to clip.
+            However, given a possible shift value, there is no value in shifting one step less.
+
+            Having said that, this is not a completely accurate model of the encoder,
+            so maybe we will need to shift one step less.
+        */
+
+        let mut prev1       = channel_state.prev1;
+        let mut prev2       = channel_state.prev2;
+        let k1          = Self::FILTER_K1[filter] as i32;
+        let k2          = Self::FILTER_K2[filter] as i32;
+        let mut right_shift = 0;
+        let mut s_min       = 0;
+        let mut s_max       = 0;
+
+        for i in 0..28 {
+            let raw_sample  = if i >= sample_limit {0} else {samples[(i*pitch) as usize]} as i32;
+            let prev_values = (k1*prev1 + k2*prev2 + (1 << 5)) >> 6;
+            let sample      = raw_sample - prev_values;
+
+            if sample < s_min {
+                s_min = sample;
+            }
+
+            if sample > s_max {
+                s_max = sample;
+            }
+
+            prev2 = prev1;
+            prev1 = raw_sample;
+        }
+
+        while right_shift < shift_range && (s_max >> right_shift) > (0x7FFF >> shift_range) {
+            right_shift += 1;
+        }
+
+        while right_shift < shift_range && (s_min >> right_shift) < (-0x8000 >> shift_range) {
+            right_shift += 1;
+        }
+
+        let min_shift = shift_range - right_shift;
+        if 0 <= min_shift && min_shift <= shift_range {
+            Ok(min_shift)
+        }
+
+        else {
+            Err(Error::from_text(format!("0 <= {} && {} <= {} was not satisfied with min_shift: {}", min_shift, min_shift, shift_range, min_shift)))
+        }
+    }
+}
+
+impl std::default::Default for Encoder {
+    fn default() -> Self {
+        Encoder{left: ChannelState::default(), right: ChannelState::default()}
+    }
+}
+
+#[derive(Clone)]
+struct ChannelState {
+    qerr:  i32, // quanitisation error
+    mse:   u64, // mean square error
+    prev1: i32, 
+    prev2: i32
+}
+
+impl std::default::Default for ChannelState {
+    fn default() -> Self { 
+        ChannelState{qerr: 0, mse: 0, prev1: 0, prev2: 0}
+    }
+}
+
+struct EncodingOffsets {
+    sample: usize,
+    sample_limit: i32,
+    pitch: i32,
+    data: usize,
+    data_shift: i32,
+    data_pitch: i32,
+}
+
+const STEREO_4BIT: ([EncodingOffsets; 4], [EncodingOffsets; 4]) = (
+    [
+        EncodingOffsets{sample: 0,      sample_limit:   0, pitch: 2, data: 0x10, data_shift: 0, data_pitch: 4},
+        EncodingOffsets{sample: 1,      sample_limit:   0, pitch: 2, data: 0x10, data_shift: 4, data_pitch: 4},
+        EncodingOffsets{sample: 56,     sample_limit: -28, pitch: 2, data: 0x11, data_shift: 0, data_pitch: 4},
+        EncodingOffsets{sample: 56 + 1, sample_limit: -28, pitch: 2, data: 0x11, data_shift: 4, data_pitch: 4},
+    ],
+    [
+        EncodingOffsets{sample: 56*2,     sample_limit: -28*2, pitch: 2, data: 0x12, data_shift: 0, data_pitch: 4},
+        EncodingOffsets{sample: 56*2 + 1, sample_limit: -28*2, pitch: 2, data: 0x12, data_shift: 4, data_pitch: 4},
+        EncodingOffsets{sample: 56*3,     sample_limit: -28*3, pitch: 2, data: 0x13, data_shift: 0, data_pitch: 4},
+        EncodingOffsets{sample: 56*3 + 1, sample_limit: -28*3, pitch: 2, data: 0x13, data_shift: 4, data_pitch: 4}
+    ]
+);
+const MONO_4BIT: ([EncodingOffsets; 4], [EncodingOffsets; 4]) = (
+    [
+        EncodingOffsets{sample: 0,    sample_limit:     0, pitch: 1, data: 0x10, data_shift: 0, data_pitch: 4},
+        EncodingOffsets{sample: 28,   sample_limit:   -28, pitch: 1, data: 0x10, data_shift: 4, data_pitch: 4},
+        EncodingOffsets{sample: 28*2, sample_limit: -28*2, pitch: 1, data: 0x11, data_shift: 0, data_pitch: 4},
+        EncodingOffsets{sample: 28*3, sample_limit: -28*3, pitch: 1, data: 0x11, data_shift: 4, data_pitch: 4},
+    ],
+    [
+        EncodingOffsets{sample: 28*4, sample_limit: -28*4, pitch: 1, data: 0x12, data_shift: 0, data_pitch: 4},
+        EncodingOffsets{sample: 28*5, sample_limit: -28*5, pitch: 1, data: 0x12, data_shift: 4, data_pitch: 4},
+        EncodingOffsets{sample: 28*6, sample_limit: -28*6, pitch: 1, data: 0x13, data_shift: 0, data_pitch: 4},
+        EncodingOffsets{sample: 28*7, sample_limit: -28*7, pitch: 1, data: 0x13, data_shift: 4, data_pitch: 4}
+    ]
+);
+
+const STEREO_8BIT: [EncodingOffsets;4] = [
+    EncodingOffsets{sample:  0,     sample_limit:   0, pitch: 2, data: 0x10, data_shift: 0, data_pitch: 4},
+    EncodingOffsets{sample:  1,     sample_limit:   0, pitch: 2, data: 0x11, data_shift: 0, data_pitch: 4},
+    EncodingOffsets{sample: 56,     sample_limit: -28, pitch: 2, data: 0x12, data_shift: 0, data_pitch: 4},
+    EncodingOffsets{sample: 56 + 1, sample_limit: -28, pitch: 2, data: 0x13, data_shift: 0, data_pitch: 4},
+];
+const MONO_8BIT:   [EncodingOffsets;4] = [
+    EncodingOffsets{sample:    0, sample_limit:     0, pitch: 1, data: 0x10, data_shift: 0, data_pitch: 4},
+    EncodingOffsets{sample:   28, sample_limit:   -28, pitch: 1, data: 0x11, data_shift: 0, data_pitch: 4},
+    EncodingOffsets{sample: 28*2, sample_limit: -28*2, pitch: 1, data: 0x12, data_shift: 0, data_pitch: 4},
+    EncodingOffsets{sample: 28*3, sample_limit: -28*3, pitch: 1, data: 0x13, data_shift: 0, data_pitch: 4},
+];