Keep samples in 64 bit

CHANGELOG.md

@@ -11,11 +11,13 @@ and this project adheres to [Semantic Versioning](https://semver.org/spec/v2.0.0
 - [playback] Add support for dithering with `--dither` for lower requantization error (breaking)
 - [playback] Add `--volume-range` option to set dB range and control `log` and `cubic` volume control curves
 - [playback] `alsamixer`: support for querying dB range from Alsa softvol
+- [playback] Add `--format F64` (supported by Alsa and GStreamer only)
 
 ### Changed
 - [audio, playback] Moved `VorbisDecoder`, `VorbisError`, `AudioPacket`, `PassthroughDecoder`, `PassthroughError`, `AudioError`, `AudioDecoder` and the `convert` module from `librespot-audio` to `librespot-playback`. The underlying crates `vorbis`, `librespot-tremor`, `lewton` and `ogg` should be used directly. (breaking)
 - [connect, playback] Moved volume controls from `librespot-connect` to `librespot-playback` crate
 - [connect] Synchronize player volume with mixer volume on playback
+- [playback] Store and pass samples in 64-bit floating point
 - [playback] Make cubic volume control available to all mixers with `--volume-ctrl cubic`
 - [playback] Normalize volumes to `[0.0..1.0]` instead of `[0..65535]` for greater precision and performance (breaking)
 - [playback] `alsamixer`: complete rewrite (breaking)

playback/src/audio_backend/alsa.rs

@@ -41,6 +41,7 @@ fn list_outputs() {
 fn open_device(dev_name: &str, format: AudioFormat) -> Result<(PCM, Frames), Box<Error>> {
     let pcm = PCM::new(dev_name, Direction::Playback, false)?;
     let alsa_format = match format {
+        AudioFormat::F64 => Format::float64(),
         AudioFormat::F32 => Format::float(),
         AudioFormat::S32 => Format::s32(),
         AudioFormat::S24 => Format::s24(),

playback/src/audio_backend/jackaudio.rs

@@ -70,9 +70,10 @@ impl Open for JackSink {
 }
 
 impl Sink for JackSink {
-    fn write(&mut self, packet: &AudioPacket, _: &mut Converter) -> io::Result<()> {
-        for s in packet.samples().iter() {
-            let res = self.send.send(*s);
+    fn write(&mut self, packet: &AudioPacket, converter: &mut Converter) -> io::Result<()> {
+        let samples_f32: &[f32] = &converter.f64_to_f32(packet.samples());
+        for sample in samples_f32.iter() {
+            let res = self.send.send(*sample);
             if res.is_err() {
                 error!("cannot write to channel");
             }

playback/src/audio_backend/mod.rs

@@ -35,21 +35,25 @@ macro_rules! sink_as_bytes {
             use zerocopy::AsBytes;
             match packet {
                 AudioPacket::Samples(samples) => match self.format {
-                    AudioFormat::F32 => self.write_bytes(samples.as_bytes()),
+                    AudioFormat::F64 => self.write_bytes(samples.as_bytes()),
+                    AudioFormat::F32 => {
+                        let samples_f32: &[f32] = &converter.f64_to_f32(samples);
+                        self.write_bytes(samples_f32.as_bytes())
+                    }
                     AudioFormat::S32 => {
-                        let samples_s32: &[i32] = &converter.f32_to_s32(samples);
+                        let samples_s32: &[i32] = &converter.f64_to_s32(samples);
                         self.write_bytes(samples_s32.as_bytes())
                     }
                     AudioFormat::S24 => {
-                        let samples_s24: &[i32] = &converter.f32_to_s24(samples);
+                        let samples_s24: &[i32] = &converter.f64_to_s24(samples);
                         self.write_bytes(samples_s24.as_bytes())
                     }
                     AudioFormat::S24_3 => {
-                        let samples_s24_3: &[i24] = &converter.f32_to_s24_3(samples);
+                        let samples_s24_3: &[i24] = &converter.f64_to_s24_3(samples);
                         self.write_bytes(samples_s24_3.as_bytes())
                     }
                     AudioFormat::S16 => {
-                        let samples_s16: &[i16] = &converter.f32_to_s16(samples);
+                        let samples_s16: &[i16] = &converter.f64_to_s16(samples);
                         self.write_bytes(samples_s16.as_bytes())
                     }
                 },

playback/src/audio_backend/portaudio.rs

@@ -151,14 +151,15 @@ impl<'a> Sink for PortAudioSink<'a> {
         let samples = packet.samples();
         let result = match self {
             Self::F32(stream, _parameters) => {
-                write_sink!(ref mut stream, samples)
+                let samples_f32: &[f32] = &converter.f64_to_f32(samples);
+                write_sink!(ref mut stream, samples_f32)
             }
             Self::S32(stream, _parameters) => {
-                let samples_s32: &[i32] = &converter.f32_to_s32(samples);
+                let samples_s32: &[i32] = &converter.f64_to_s32(samples);
                 write_sink!(ref mut stream, samples_s32)
             }
             Self::S16(stream, _parameters) => {
-                let samples_s16: &[i16] = &converter.f32_to_s16(samples);
+                let samples_s16: &[i16] = &converter.f64_to_s16(samples);
                 write_sink!(ref mut stream, samples_s16)
             }
         };

playback/src/audio_backend/pulseaudio.rs

@@ -28,6 +28,9 @@ impl Open for PulseAudioSink {
             AudioFormat::S24 => pulse::sample::Format::S24_32le,
             AudioFormat::S24_3 => pulse::sample::Format::S24le,
             AudioFormat::S16 => pulse::sample::Format::S16le,
+            _ => {
+                unimplemented!("PulseAudio currently does not support {:?} output", format)
+            }
         };
 
         let ss = pulse::sample::Spec {

playback/src/audio_backend/rodio.rs

@@ -178,12 +178,16 @@ impl Sink for RodioSink {
         let samples = packet.samples();
         match self.format {
             AudioFormat::F32 => {
-                let source =
-                    rodio::buffer::SamplesBuffer::new(NUM_CHANNELS as u16, SAMPLE_RATE, samples);
+                let samples_f32: &[f32] = &converter.f64_to_f32(samples);
+                let source = rodio::buffer::SamplesBuffer::new(
+                    NUM_CHANNELS as u16,
+                    SAMPLE_RATE,
+                    samples_f32,
+                );
                 self.rodio_sink.append(source);
             }
             AudioFormat::S16 => {
-                let samples_s16: &[i16] = &converter.f32_to_s16(samples);
+                let samples_s16: &[i16] = &converter.f64_to_s16(samples);
                 let source = rodio::buffer::SamplesBuffer::new(
                     NUM_CHANNELS as u16,
                     SAMPLE_RATE,

playback/src/audio_backend/sdl.rs

@@ -94,16 +94,17 @@ impl Sink for SdlSink {
         let samples = packet.samples();
         match self {
             Self::F32(queue) => {
+                let samples_f32: &[f32] = &converter.f64_to_f32(samples);
                 drain_sink!(queue, AudioFormat::F32.size());
-                queue.queue(samples)
+                queue.queue(samples_f32)
             }
             Self::S32(queue) => {
-                let samples_s32: &[i32] = &converter.f32_to_s32(samples);
+                let samples_s32: &[i32] = &converter.f64_to_s32(samples);
                 drain_sink!(queue, AudioFormat::S32.size());
                 queue.queue(samples_s32)
             }
             Self::S16(queue) => {
-                let samples_s16: &[i16] = &converter.f32_to_s16(samples);
+                let samples_s16: &[i16] = &converter.f64_to_s16(samples);
                 drain_sink!(queue, AudioFormat::S16.size());
                 queue.queue(samples_s16)
             }

playback/src/config.rs

@@ -33,6 +33,7 @@ impl Default for Bitrate {
 
 #[derive(Clone, Copy, Debug, Hash, PartialOrd, Ord, PartialEq, Eq)]
 pub enum AudioFormat {
+    F64,
     F32,
     S32,
     S24,
@@ -44,6 +45,7 @@ impl TryFrom<&String> for AudioFormat {
     type Error = ();
     fn try_from(s: &String) -> Result<Self, Self::Error> {
         match s.to_uppercase().as_str() {
+            "F64" => Ok(Self::F64),
             "F32" => Ok(Self::F32),
             "S32" => Ok(Self::S32),
             "S24" => Ok(Self::S24),
@@ -65,6 +67,8 @@ impl AudioFormat {
     #[allow(dead_code)]
     pub fn size(&self) -> usize {
         match self {
+            Self::F64 => mem::size_of::<f64>(),
+            Self::F32 => mem::size_of::<f32>(),
             Self::S24_3 => mem::size_of::<i24>(),
             Self::S16 => mem::size_of::<i16>(),
             _ => mem::size_of::<i32>(), // S32 and S24 are both stored in i32
@@ -127,11 +131,11 @@ pub struct PlayerConfig {
     pub normalisation: bool,
     pub normalisation_type: NormalisationType,
     pub normalisation_method: NormalisationMethod,
-    pub normalisation_pregain: f32,
-    pub normalisation_threshold: f32,
-    pub normalisation_attack: f32,
-    pub normalisation_release: f32,
-    pub normalisation_knee: f32,
+    pub normalisation_pregain: f64,
+    pub normalisation_threshold: f64,
+    pub normalisation_attack: f64,
+    pub normalisation_release: f64,
+    pub normalisation_knee: f64,
 
     // pass function pointers so they can be lazily instantiated *after* spawning a thread
     // (thereby circumventing Send bounds that they might not satisfy)
@@ -160,10 +164,10 @@ impl Default for PlayerConfig {
 // fields are intended for volume control range in dB
 #[derive(Clone, Copy, Debug)]
 pub enum VolumeCtrl {
-    Cubic(f32),
+    Cubic(f64),
     Fixed,
     Linear,
-    Log(f32),
+    Log(f64),
 }
 
 impl FromStr for VolumeCtrl {
@@ -183,9 +187,9 @@ impl VolumeCtrl {
     pub const MAX_VOLUME: u16 = std::u16::MAX;
 
     // Taken from: https://www.dr-lex.be/info-stuff/volumecontrols.html
-    pub const DEFAULT_DB_RANGE: f32 = 60.0;
+    pub const DEFAULT_DB_RANGE: f64 = 60.0;
 
-    pub fn from_str_with_range(s: &str, db_range: f32) -> Result<Self, <Self as FromStr>::Err> {
+    pub fn from_str_with_range(s: &str, db_range: f64) -> Result<Self, <Self as FromStr>::Err> {
         use self::VolumeCtrl::*;
         match s.to_lowercase().as_ref() {
             "cubic" => Ok(Cubic(db_range)),

playback/src/convert.rs

@@ -30,34 +30,37 @@ impl Converter {
         }
     }
 
-    // Denormalize and dither
-    pub fn scale(&mut self, sample: f32, factor: i64) -> f32 {
+    const SCALE_S32: f64 = 2147483648.;
+    const SCALE_S24: f64 = 8388608.;
+    const SCALE_S16: f64 = 32768.;
+
+    pub fn scale(&mut self, sample: f64, factor: f64) -> f64 {
         let dither = match self.ditherer {
             Some(ref mut d) => d.noise(),
             None => 0.0,
         };
 
         // From the many float to int conversion methods available, match what
         // the reference Vorbis implementation uses: sample * 32768 (for 16 bit)
-        let int_value = sample * factor as f32 + dither;
+        let int_value = sample * factor + dither;
 
         // Casting float to integer rounds towards zero by default, i.e. it
         // truncates, and that generates larger error than rounding to nearest.
         // Absolute lowest error is gained from rounding ties to even.
-        math::round::half_to_even(int_value.into(), 0) as f32
+        math::round::half_to_even(int_value, 0)
     }
 
     // Special case for samples packed in a word of greater bit depth (e.g.
     // S24): clamp between min and max to ensure that the most significant
     // byte is zero. Otherwise, dithering may cause an overflow. This is not
     // necessary for other formats, because casting to integer will saturate
     // to the bounds of the primitive.
-    pub fn clamping_scale(&mut self, sample: f32, factor: i64) -> f32 {
+    pub fn clamping_scale(&mut self, sample: f64, factor: f64) -> f64 {
         let int_value = self.scale(sample, factor);
 
         // In two's complement, there are more negative than positive values.
-        let min = -factor as f32;
-        let max = (factor - 1) as f32;
+        let min = -factor;
+        let max = factor - 1.0;
 
         if int_value < min {
             return min;
@@ -67,38 +70,42 @@ impl Converter {
         int_value
     }
 
-    pub fn f32_to_s32(&mut self, samples: &[f32]) -> Vec<i32> {
+    pub fn f64_to_f32(&mut self, samples: &[f64]) -> Vec<f32> {
+        samples.iter().map(|sample| *sample as f32).collect()
+    }
+
+    pub fn f64_to_s32(&mut self, samples: &[f64]) -> Vec<i32> {
         samples
             .iter()
-            .map(|sample| self.scale(*sample, 0x80000000) as i32)
+            .map(|sample| self.scale(*sample, Self::SCALE_S32) as i32)
             .collect()
     }
 
     // S24 is 24-bit PCM packed in an upper 32-bit word
-    pub fn f32_to_s24(&mut self, samples: &[f32]) -> Vec<i32> {
+    pub fn f64_to_s24(&mut self, samples: &[f64]) -> Vec<i32> {
         samples
             .iter()
-            .map(|sample| self.clamping_scale(*sample, 0x800000) as i32)
+            .map(|sample| self.clamping_scale(*sample, Self::SCALE_S24) as i32)
             .collect()
     }
 
     // S24_3 is 24-bit PCM in a 3-byte array
-    pub fn f32_to_s24_3(&mut self, samples: &[f32]) -> Vec<i24> {
+    pub fn f64_to_s24_3(&mut self, samples: &[f64]) -> Vec<i24> {
         samples
             .iter()
             .map(|sample| {
                 // Not as DRY as calling f32_to_s24 first, but this saves iterating
                 // over all samples twice.
-                let int_value = self.clamping_scale(*sample, 0x800000) as i32;
+                let int_value = self.clamping_scale(*sample, Self::SCALE_S24) as i32;
                 i24::from_s24(int_value)
             })
             .collect()
     }
 
-    pub fn f32_to_s16(&mut self, samples: &[f32]) -> Vec<i16> {
+    pub fn f64_to_s16(&mut self, samples: &[f64]) -> Vec<i16> {
         samples
             .iter()
-            .map(|sample| self.scale(*sample, 0x8000) as i16)
+            .map(|sample| self.scale(*sample, Self::SCALE_S16) as i16)
             .collect()
     }
 }

playback/src/decoder/lewton_decoder.rs

@@ -1,6 +1,7 @@
 use super::{AudioDecoder, AudioError, AudioPacket};
 
 use lewton::inside_ogg::OggStreamReader;
+use lewton::samples::InterleavedSamples;
 
 use std::error;
 use std::fmt;
@@ -35,11 +36,8 @@ where
         use lewton::OggReadError::NoCapturePatternFound;
         use lewton::VorbisError::{BadAudio, OggError};
         loop {
-            match self
-                .0
-                .read_dec_packet_generic::<lewton::samples::InterleavedSamples<f32>>()
-            {
-                Ok(Some(packet)) => return Ok(Some(AudioPacket::Samples(packet.samples))),
+            match self.0.read_dec_packet_generic::<InterleavedSamples<f32>>() {
+                Ok(Some(packet)) => return Ok(Some(AudioPacket::samples_from_f32(packet.samples))),
                 Ok(None) => return Ok(None),
 
                 Err(BadAudio(AudioIsHeader)) => (),

playback/src/decoder/mod.rs

@@ -7,12 +7,17 @@ mod passthrough_decoder;
 pub use passthrough_decoder::{PassthroughDecoder, PassthroughError};
 
 pub enum AudioPacket {
-    Samples(Vec<f32>),
+    Samples(Vec<f64>),
     OggData(Vec<u8>),
 }
 
 impl AudioPacket {
-    pub fn samples(&self) -> &[f32] {
+    pub fn samples_from_f32(f32_samples: Vec<f32>) -> Self {
+        let f64_samples = f32_samples.iter().map(|sample| *sample as f64).collect();
+        AudioPacket::Samples(f64_samples)
+    }
+
+    pub fn samples(&self) -> &[f64] {
         match self {
             AudioPacket::Samples(s) => s,
             AudioPacket::OggData(_) => panic!("can't return OggData on samples"),