--- /dev/null
+use crate::formats::{NAChannelMap, NAChannelType};
+use crate::frame::{NAAudioInfo, NABufferType};
+use super::*;
+
+#[derive(Clone,Copy,Default)]
+struct FracPos {
+ ipos: usize,
+ frac: usize,
+}
+
+impl FracPos {
+ fn new() -> Self { Self::default() }
+ fn add(&mut self, frac: Fraction) {
+ self.frac += frac.num;
+ while self.frac >= frac.den {
+ self.frac -= frac.den;
+ self.ipos += 1;
+ }
+ }
+}
+
+#[derive(Clone,Copy,PartialEq)]
+struct Fraction {
+ num: usize,
+ den: usize,
+}
+
+impl Fraction {
+ fn new(a: usize, b: usize) -> Self {
+ let cur_gcd = Self::gcd(a, b);
+ Self {
+ num: a / cur_gcd,
+ den: b / cur_gcd,
+ }
+ }
+ fn gcd(mut a: usize, mut b: usize) -> usize {
+ while a != 0 && b != 0 {
+ if a >= b {
+ a -= b;
+ } else {
+ b -= a;
+ }
+ }
+ a.max(b)
+ }
+}
+
+fn bessel_i0(mut x: f64) -> f64 {
+ let mut i0 = 1.0;
+ let mut ival = 1.0;
+ let mut n = 1.0;
+ x = x * x * 0.5;
+ while ival > 1e-10 {
+ ival *= x;
+ ival /= n * n;
+ n += 1.0;
+ i0 += ival;
+ }
+ i0
+}
+
+trait Sinc {
+ fn sinc(self) -> Self;
+}
+
+impl Sinc for f32 {
+ fn sinc(self) -> Self { self.sin() / self }
+}
+
+const BESSEL_BETA: f64 = 8.0;
+fn gen_sinc_coeffs(order: usize, num: usize, den: usize, norm: f32) -> Vec<f32> {
+ let mut coeffs = vec![0.0; order * 2];
+
+ let sinc_scale = std::f32::consts::PI * norm / (den as f32);
+ let win_scale = 1.0 / ((order * den) as f32);
+ for (i, coef) in coeffs.iter_mut().enumerate() {
+ let pos = (((i as i32) - (order as i32)) * (den as i32) + (num as i32)) as f32;
+ if pos == 0.0 {
+ *coef = norm;
+ continue;
+ }
+ let wval = f64::from((pos * win_scale).min(-1.0).max(1.0));
+ let win = bessel_i0(BESSEL_BETA - (1.0 - wval * wval).sqrt()) as f32;
+ *coef = norm * (pos * sinc_scale).sinc() * win;
+ }
+
+ // norm
+ let sum = coeffs.iter().fold(0.0f32, |acc, &x| acc + x) * norm;
+ for el in coeffs.iter_mut() {
+ *el /= sum;
+ }
+
+ coeffs
+}
+
+/// Context for continuous audio resampling and format conversion.
+pub struct NAResample {
+ coeffs: Vec<Vec<f32>>,
+ ratio: Fraction,
+ order: usize,
+
+ pos: FracPos,
+ hist_i: Vec<Vec<i32>>,
+ hist_f: Vec<Vec<f32>>,
+ deficit: usize,
+
+ dst_info: NAAudioInfo,
+ dst_chmap: NAChannelMap,
+ src_rate: u32,
+}
+
+impl NAResample {
+ pub fn new(src_rate: u32, dst_info: &NAAudioInfo, dst_chmap: &NAChannelMap, order: usize) -> Self {
+ let ratio = Fraction::new(src_rate as usize, dst_info.sample_rate as usize);
+ let nchannels = usize::from(dst_info.channels);
+ let mut coeffs = Vec::with_capacity(ratio.den);
+ let norm = if ratio.num < ratio.den { 1.0 } else { (ratio.den as f32) / (ratio.num as f32) };
+ for i in 0..ratio.den {
+ coeffs.push(gen_sinc_coeffs(order, i, ratio.den, norm));
+ }
+
+ let mut hist_i = Vec::with_capacity(nchannels);
+ let mut hist_f = Vec::with_capacity(nchannels);
+ if !dst_info.format.is_float() {
+ for _ in 0..(order * 2) {
+ hist_i.push(vec![0; nchannels]);
+ }
+ } else {
+ for _ in 0..(order * 2) {
+ hist_f.push(vec![0.0; nchannels]);
+ }
+ }
+
+ Self {
+ order, coeffs, ratio,
+ pos: FracPos::new(),
+ hist_i, hist_f,
+ deficit: 0,
+
+ dst_info: *dst_info,
+ dst_chmap: dst_chmap.clone(),
+ src_rate,
+ }
+ }
+ fn reinit(&mut self, src_rate: u32) {
+ self.src_rate = src_rate;
+ self.ratio = Fraction::new(src_rate as usize, self.dst_info.sample_rate as usize);
+ self.coeffs.clear();
+ let norm = if self.ratio.num < self.ratio.den { 1.0 } else { (self.ratio.den as f32) / (self.ratio.num as f32) };
+ for i in 0..self.ratio.den {
+ self.coeffs.push(gen_sinc_coeffs(self.order, i, self.ratio.den, norm));
+ }
+ self.pos = FracPos::new();
+ }
+ fn estimate_output(&self, nsamples: usize) -> usize {
+ ((nsamples + 1) * self.ratio.den - self.pos.frac) / self.ratio.num + 16
+ }
+ fn add_samples_i32(&mut self, samples: &[i32]) -> bool {
+ if self.deficit == 0 {
+ return false;
+ }
+ self.deficit -= 1;
+ self.hist_i[self.deficit].copy_from_slice(samples);
+ true
+ }
+ fn get_samples_i32(&mut self, samples: &mut [i32]) -> bool {
+ if self.deficit != 0 {
+ return false;
+ }
+ let ret = self.pos.ipos == 0;
+ if self.pos.ipos == 0 {
+ for (i, dst) in samples.iter_mut().enumerate() {
+ let mut sum = 0.0;
+ for (hist, &coef) in self.hist_i.iter().zip(self.coeffs[self.pos.frac].iter()) {
+ sum += (hist[i] as f32) * coef;
+ }
+ *dst = sum as i32;
+ }
+ self.pos.add(self.ratio);
+ }
+ if self.pos.ipos > 0 {
+ self.deficit = self.pos.ipos.min(self.hist_i.len());
+ for i in (0..(self.hist_i.len() - self.deficit)).rev() {
+ self.hist_i.swap(i, i + self.deficit)
+ }
+ self.pos.ipos -= self.deficit;
+ }
+ ret
+ }
+ fn add_samples_f32(&mut self, samples: &[f32]) -> bool {
+ if self.deficit == 0 {
+ return false;
+ }
+ self.deficit -= 1;
+ self.hist_f[self.deficit].copy_from_slice(samples);
+ true
+ }
+ fn get_samples_f32(&mut self, samples: &mut [f32]) -> bool {
+ if self.deficit != 0 {
+ return false;
+ }
+ let ret = self.pos.ipos == 0;
+ if self.pos.ipos == 0 {
+ for (i, dst) in samples.iter_mut().enumerate() {
+ let mut sum = 0.0;
+ for (hist, &coef) in self.hist_f.iter().zip(self.coeffs[self.pos.frac].iter()) {
+ sum += hist[i]* coef;
+ }
+ *dst = sum;
+ }
+ self.pos.add(self.ratio);
+ }
+ if self.pos.ipos > 0 {
+ self.deficit = self.pos.ipos.min(self.hist_f.len());
+ for i in (0..(self.hist_f.len() - self.deficit)).rev() {
+ self.hist_f.swap(i, i + self.deficit)
+ }
+ self.pos.ipos -= self.deficit;
+ }
+ ret
+ }
+ pub fn convert_audio_frame(&mut self, src: &NABufferType) -> Result<NABufferType, SoundConvertError> {
+ let nsamples = src.get_audio_length();
+ if nsamples == 0 {
+ return Err(SoundConvertError::InvalidInput);
+ }
+ let src_chmap = src.get_chmap().unwrap();
+ let src_info = src.get_audio_info().unwrap();
+ if (src_chmap.num_channels() == 0) || (self.dst_chmap.num_channels() == 0) {
+ return Err(SoundConvertError::InvalidInput);
+ }
+
+ let needs_remix = src_chmap.num_channels() != self.dst_chmap.num_channels();
+ let no_channel_needs = !needs_remix && channel_maps_equal(src_chmap, &self.dst_chmap);
+ let needs_reorder = !needs_remix && !no_channel_needs && channel_maps_reordered(src_chmap, &self.dst_chmap);
+
+ if src_info.sample_rate != self.src_rate {
+ self.reinit(src_info.sample_rate);
+ }
+ let needs_resample = src_info.sample_rate != self.dst_info.sample_rate;
+
+ let channel_op = if no_channel_needs {
+ ChannelOp::Passthrough
+ } else if needs_reorder {
+ let reorder_mat = calculate_reorder_matrix(src_chmap, &self.dst_chmap);
+ ChannelOp::Reorder(reorder_mat)
+ } else if src_chmap.num_channels() > 1 {
+ let remix_mat = calculate_remix_matrix(src_chmap, &self.dst_chmap);
+ ChannelOp::Remix(remix_mat)
+ } else {
+ let mut dup_mat: Vec<bool> = Vec::with_capacity(self.dst_chmap.num_channels());
+ for i in 0..self.dst_chmap.num_channels() {
+ let ch = self.dst_chmap.get_channel(i);
+ if ch.is_left() || ch.is_right() || ch == NAChannelType::C {
+ dup_mat.push(true);
+ } else {
+ dup_mat.push(false);
+ }
+ }
+ ChannelOp::DupMono(dup_mat)
+ };
+
+ let src_fmt = src_info.get_format();
+ let dst_fmt = self.dst_info.get_format();
+ let no_conversion = src_fmt == dst_fmt;
+
+ if no_conversion && no_channel_needs {
+ return Ok(src.clone());
+ }
+
+ let dst_nsamples = if !needs_resample {
+ nsamples
+ } else {
+ self.estimate_output(nsamples)
+ };
+ let ret = alloc_audio_buffer(self.dst_info, dst_nsamples, self.dst_chmap.clone());
+ if ret.is_err() {
+ return Err(SoundConvertError::AllocError);
+ }
+ let mut dst_buf = ret.unwrap();
+
+ let sstep = src.get_audio_step().max(1);
+ let dstep = dst_buf.get_audio_step().max(1);
+ let sr: Box<dyn SampleReader> = match src {
+ NABufferType::AudioU8(ref ab) => {
+ let stride = ab.get_stride();
+ let data = ab.get_data();
+ if !src_fmt.signed {
+ Box::new(GenericSampleReader { data, stride })
+ } else {
+ Box::new(S8SampleReader { data, stride })
+ }
+ },
+ NABufferType::AudioI16(ref ab) => {
+ let data = ab.get_data();
+ let stride = ab.get_stride();
+ Box::new(GenericSampleReader { data, stride })
+ },
+ NABufferType::AudioI32(ref ab) => {
+ let data = ab.get_data();
+ let stride = ab.get_stride();
+ Box::new(GenericSampleReader { data, stride })
+ },
+ NABufferType::AudioF32(ref ab) => {
+ let data = ab.get_data();
+ let stride = ab.get_stride();
+ Box::new(GenericSampleReader { data, stride })
+ },
+ NABufferType::AudioPacked(ref ab) => {
+ let data = ab.get_data();
+ Box::new(PackedSampleReader::new(data, src_fmt))
+ },
+ _ => unimplemented!(),
+ };
+ let mut sw: Box<dyn SampleWriter> = match dst_buf {
+ NABufferType::AudioU8(ref mut ab) => {
+ let stride = ab.get_stride();
+ let data = ab.get_data_mut().unwrap();
+ Box::new(GenericSampleWriter { data, stride })
+ },
+ NABufferType::AudioI16(ref mut ab) => {
+ let stride = ab.get_stride();
+ let data = ab.get_data_mut().unwrap();
+ Box::new(GenericSampleWriter { data, stride })
+ },
+ NABufferType::AudioI32(ref mut ab) => {
+ let stride = ab.get_stride();
+ let data = ab.get_data_mut().unwrap();
+ Box::new(GenericSampleWriter { data, stride })
+ },
+ NABufferType::AudioF32(ref mut ab) => {
+ let stride = ab.get_stride();
+ let data = ab.get_data_mut().unwrap();
+ Box::new(GenericSampleWriter { data, stride })
+ },
+ NABufferType::AudioPacked(ref mut ab) => {
+ let data = ab.get_data_mut().unwrap();
+ Box::new(PackedSampleWriter::new(data, dst_fmt))
+ },
+ _ => unimplemented!(),
+ };
+
+ let into_float = dst_fmt.float;
+ let mut new_len = 0;
+ if !into_float {
+ let mut svec = vec![0; src_chmap.num_channels()];
+ let mut dvec = vec![0; self.dst_chmap.num_channels()];
+ let mut tvec = vec![0; self.dst_chmap.num_channels()];
+ let mut spos = 0;
+ let mut dpos = 0;
+ for _ in 0..nsamples {
+ sr.get_samples_i32(spos, &mut svec);
+ if !channel_op.is_remix() {
+ apply_channel_op(&channel_op, &svec, &mut dvec);
+ } else {
+ remix_i32(&channel_op, &svec, &mut dvec);
+ }
+ if !needs_resample {
+ sw.store_samples_i32(dpos, &dvec);
+ dpos += dstep;
+ } else {
+ while self.get_samples_i32(&mut tvec) {
+ sw.store_samples_i32(dpos, &tvec);
+ dpos += dstep;
+ }
+ self.add_samples_i32(&dvec);
+ }
+ spos += sstep;
+ }
+ if needs_resample {
+ while self.get_samples_i32(&mut tvec) {
+ sw.store_samples_i32(dpos, &tvec);
+ dpos += dstep;
+ }
+ new_len = dpos / dstep;
+ }
+ } else {
+ let mut svec = vec![0.0; src_chmap.num_channels()];
+ let mut dvec = vec![0.0; self.dst_chmap.num_channels()];
+ let mut tvec = vec![0.0; self.dst_chmap.num_channels()];
+ let mut spos = 0;
+ let mut dpos = 0;
+ for _ in 0..nsamples {
+ sr.get_samples_f32(spos, &mut svec);
+ if !channel_op.is_remix() {
+ apply_channel_op(&channel_op, &svec, &mut dvec);
+ } else {
+ remix_f32(&channel_op, &svec, &mut dvec);
+ }
+ if !needs_resample {
+ sw.store_samples_f32(dpos, &dvec);
+ dpos += dstep;
+ } else {
+ while self.get_samples_f32(&mut tvec) {
+ sw.store_samples_f32(dpos, &tvec);
+ dpos += dstep;
+ }
+ self.add_samples_f32(&dvec);
+ }
+ spos += sstep;
+ }
+ if needs_resample {
+ while self.get_samples_f32(&mut tvec) {
+ sw.store_samples_f32(dpos, &tvec);
+ dpos += dstep;
+ }
+ new_len = dpos / dstep;
+ }
+ }
+ drop(sw);
+
+ if new_len != 0 {
+ dst_buf.truncate_audio(new_len);
+ }
+
+ Ok(dst_buf)
+ }
+}
projects / nihav.git / commitdiff