Resampling and interleaving supported

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

@@ -4,6 +4,7 @@ mod xa_audio;
 use clap::{Args, ValueEnum};
 use std::io::Write;
 use tool_helper::{Error, Input};
+use xa_audio::{LOW_FREQUENCY, HIGH_FREQUENCY};
 
 #[derive(Args, Clone)]
 pub struct Arguments {
@@ -26,7 +27,14 @@ 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, args.frequency)?;
+    let prepared_xa_audio = raw_audio::load_as_i16_audio(input, frequency_to_value(args.frequency))?;
 
     xa_audio::encode(prepared_xa_audio, output, &args)
+}
+
+fn frequency_to_value(requested_freq: Frequency) -> u32 {
+    match requested_freq {
+        Frequency::High => HIGH_FREQUENCY,
+        Frequency::Low  => LOW_FREQUENCY,
+    }
 }

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

@@ -1,15 +1,14 @@
 mod error;
 
-use super::Frequency as RequestedFrequencyType;
 use rubato::{FftFixedInOut, Resampler};
 use symphonia::core::{
-    audio::{Layout, SampleBuffer},
+    audio::{AudioBuffer, Layout, SampleBuffer, Signal, SignalSpec},
     codecs::{Decoder, DecoderOptions, CODEC_TYPE_NULL},
     errors::Error as SymError,
     formats::{FormatOptions, FormatReader},
     io::MediaSourceStream,
     meta::MetadataOptions,
-    probe::Hint, sample
+    probe::Hint
 };
 use tool_helper::{Error, Input};
 
@@ -19,30 +18,21 @@ pub enum Orality {
     Mono,
 }
 
-impl Orality {
+pub struct CDAudioSamples {
-    fn as_channels(&self) -> usize {
-        match self {
-            Orality::Mono   => 1,
-            Orality::Stereo => 2,
-        }
-    }
-}
-
-pub struct PreparedAudioSamples {
     samples:   Vec::<i16>,
     orality:   Orality,
     frequency: u32,
 }
 
-impl PreparedAudioSamples {
+impl CDAudioSamples {
-    pub fn new(samples: Vec<i16>, channels: usize, frequency: u32) -> Result<PreparedAudioSamples, Error> {
+    pub fn new(samples: Vec<i16>, channels: usize, frequency: u32) -> Result<CDAudioSamples, Error> {
         let orality = match channels {
             0 => return Err(Error::from_str("Input file has no audio channels")),
             1 => Orality::Mono,
             2 => Orality::Stereo,
             _ => return Err(Error::from_str("Only Mono and Stereo input are supported")),
         };
-        Ok(PreparedAudioSamples{samples, orality, frequency})
+        Ok(CDAudioSamples{samples, orality, frequency})
     }
 
     pub fn samples(&self) -> &Vec::<i16> {
@@ -61,9 +51,8 @@ struct InternalAudioSamples {
 
 impl InternalAudioSamples {
     pub fn new(planar_samples: Vec<Vec<f32>>, frequency: u32) -> Result<InternalAudioSamples, Error> {
-        if planar_samples.len() < 1 {
+        if planar_samples.len() < 1 || planar_samples.len() > 2{
-            // TODO: Make better text
+            Err(Error::from_str("Audio samples need to be either mono or stereo"))
-            Err(Error::from_str("Audio samples can not be empty"))
         }
 
         else {
@@ -71,6 +60,20 @@ impl InternalAudioSamples {
         }
     }
 
+    pub fn into_audio_buffer(self) -> AudioBuffer<f32> {
+        let     duration         = self.sample_len() as u64;
+        let mut new_audio_buffer = AudioBuffer::new(duration, SignalSpec::new_with_layout(self.frequency, if self.planar_samples.len() == 1 {Layout::Mono} else {Layout::Stereo}));
+
+        new_audio_buffer.render_silence(None);
+        for (channel_idx, channel) in self.planar_samples.into_iter().enumerate() {
+            let dst_channel = new_audio_buffer.chan_mut(channel_idx);
+            for (sample_idx, sample) in channel.into_iter().enumerate() {
+                dst_channel[sample_idx] = sample;
+            }
+        }
+        new_audio_buffer
+    }
+
     pub fn sample_len(&self) -> usize {
         self.planar_samples[0].len()
     }
@@ -89,18 +92,20 @@ impl InternalAudioSamples {
     }
 }
 
-pub fn load_as_i16_audio(input: Input, freq_type: RequestedFrequencyType) -> Result<PreparedAudioSamples, Error> {
+pub fn load_as_i16_audio(input: Input, target_frequency: u32) -> Result<CDAudioSamples, Error> {
     let raw_audio = load_raw_audio(input)?;
-    let raw_audio = resample(raw_audio, 37_800)?;
+    let raw_audio = resample(raw_audio, target_frequency)?;
 
-    test_write_wav(down_sample_interleave(raw_audio)?)?;
+    down_sample_interleave(raw_audio)
-    Err(Error::not_implemented("Resampling not implemented"))
 }
 
-fn test_write_wav(audio_samples: PreparedAudioSamples) -> Result<(), Error> {
+fn _test_write_wav(audio_samples: CDAudioSamples) -> Result<(), Error> {
     let spec = hound::WavSpec {
-        channels: 2,
+        channels: match audio_samples.orality {
-        sample_rate: 37_800,
+            Orality::Mono   => 1,
+            Orality::Stereo => 2,
+        },
+        sample_rate: audio_samples.frequency,
         bits_per_sample: 16,
         sample_format: hound::SampleFormat::Int,
     };
@@ -251,8 +256,14 @@ fn resample(input: InternalAudioSamples, target_frequency: u32) -> Result<Intern
     InternalAudioSamples::new(planar_output, target_frequency)
 }
 
-fn down_sample_interleave(input: InternalAudioSamples) -> Result<PreparedAudioSamples, Error> {
+fn down_sample_interleave(input: InternalAudioSamples) -> Result<CDAudioSamples, Error> {
-    Err(Error::not_implemented("down_sample_interleave"))
+    let     channels      = input.channels();
+    let     frequency     = input.frequency;
+    let     audio_buffer  = input.into_audio_buffer();
+    let mut sample_buffer = SampleBuffer::<i16>::new(audio_buffer.capacity() as u64, audio_buffer.spec().clone());
+
+    sample_buffer.copy_interleaved_typed::<f32>(&audio_buffer);
+    CDAudioSamples::new(sample_buffer.samples().to_vec(), channels, frequency)
 }
 
 fn load_to_ram(mut input: Input) -> Result<std::io::Cursor<Vec<u8>>, Error> {

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

@@ -1,24 +1,18 @@
 mod xapcm;
 
 use super::Arguments;
-use super::raw_audio::PreparedAudioSamples;
+use super::raw_audio::CDAudioSamples;
 use std::io::Write;
 use tool_helper::Error;
 
-const HIGH_FREQUENCY:u32 = 37_800;
+pub const HIGH_FREQUENCY:u32 = 37_800;
-const  LOW_FREQUENCY:u32 = 18_900;
+pub const  LOW_FREQUENCY:u32 = 18_900;
 
-pub fn encode(input: PreparedAudioSamples, output: &mut dyn Write, arguments: &Arguments) -> Result<(), Error> {
+pub fn encode(input: CDAudioSamples, output: &mut dyn Write, arguments: &Arguments) -> Result<(), Error> {
-    let mut encoder      = xapcm::Encoder::new(&input.samples(), arguments.clone(), input.orality());
+    let mut encoder = xapcm::Encoder::new(&input.samples(), arguments.clone(), input.orality());
-    let mut sector_count = 0;
 
     while let Some(xa_sector) = encoder.encode_next_xa_sector()? {
         output.write(&xa_sector)?;
-
-        sector_count += 1;
-        print!("\rSector: {} written", sector_count);
     }
-    println!();
-    //Err(Error::not_implemented("XA conversion"))
     Ok(())
 }

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

@@ -13,7 +13,7 @@ pub struct Encoder<'a> {
 
 impl<'a> Encoder<'a> {
     const BLOCKS_PER_SECTOR:usize    = 18;
-    const XA_ADPCM_FILTER_COUNT: i32 = 4;
+    const XA_ADPCM_FILTER_COUNT: i32 = 3;
     const FILTER_K1: [i16; 5]        = [0, 60, 115,  98, 122];
     const FILTER_K2: [i16; 5]        = [0,  0, -52, -55, -60];