Make audio fetch parameters tunable

This change collects all those audio fetch parameters that were defined as
static constants into a dedicated struct, AudioFetchParams.
This struct can be read and set statically, allowing a user of the library to
modify those parameters without the need to recompile.

audio/src/fetch/mod.rs

@@ -6,7 +6,7 @@ use std::{
     io::{self, Read, Seek, SeekFrom},
     sync::{
         atomic::{AtomicBool, AtomicUsize, Ordering},
-        Arc,
+        Arc, OnceLock,
     },
     time::Duration,
 };
@@ -55,42 +55,75 @@ impl From<AudioFileError> for Error {
     }
 }
 
+#[derive(Clone)]
+pub struct AudioFetchParams {
     /// The minimum size of a block that is requested from the Spotify servers in one request.
     /// This is the block size that is typically requested while doing a `seek()` on a file.
     /// The Symphonia decoder requires this to be a power of 2 and > 32 kB.
     /// Note: smaller requests can happen if part of the block is downloaded already.
-pub const MINIMUM_DOWNLOAD_SIZE: usize = 64 * 1024;
+    pub minimum_download_size: usize,
 
     /// The minimum network throughput that we expect. Together with the minimum download size,
     /// this will determine the time we will wait for a response.
-pub const MINIMUM_THROUGHPUT: usize = 8 * 1024;
+    pub minimum_throughput: usize,
 
     /// The ping time that is used for calculations before a ping time was actually measured.
-pub const INITIAL_PING_TIME_ESTIMATE: Duration = Duration::from_millis(500);
+    pub initial_ping_time_estimate: Duration,
 
     /// If the measured ping time to the Spotify server is larger than this value, it is capped
     /// to avoid run-away block sizes and pre-fetching.
-pub const MAXIMUM_ASSUMED_PING_TIME: Duration = Duration::from_millis(1500);
+    pub maximum_assumed_ping_time: Duration,
 
     /// Before playback starts, this many seconds of data must be present.
     /// Note: the calculations are done using the nominal bitrate of the file. The actual amount
     /// of audio data may be larger or smaller.
-pub const READ_AHEAD_BEFORE_PLAYBACK: Duration = Duration::from_secs(1);
+    pub read_ahead_before_playback: Duration,
 
     /// While playing back, this many seconds of data ahead of the current read position are
     /// requested.
     /// Note: the calculations are done using the nominal bitrate of the file. The actual amount
     /// of audio data may be larger or smaller.
-pub const READ_AHEAD_DURING_PLAYBACK: Duration = Duration::from_secs(5);
+    pub read_ahead_during_playback: Duration,
 
     /// If the amount of data that is pending (requested but not received) is less than a certain amount,
     /// data is pre-fetched in addition to the read ahead settings above. The threshold for requesting more
     /// data is calculated as `<pending bytes> < PREFETCH_THRESHOLD_FACTOR * <ping time> * <nominal data rate>`
-pub const PREFETCH_THRESHOLD_FACTOR: f32 = 4.0;
+    pub prefetch_threshold_factor: f32,
 
     /// The time we will wait to obtain status updates on downloading.
-pub const DOWNLOAD_TIMEOUT: Duration =
-    Duration::from_secs((MINIMUM_DOWNLOAD_SIZE / MINIMUM_THROUGHPUT) as u64);
+    pub download_timeout: Duration,
+}
+
+impl Default for AudioFetchParams {
+    fn default() -> Self {
+        let minimum_download_size = 64 * 1024;
+        let minimum_throughput = 8 * 1024;
+        Self {
+            minimum_download_size,
+            minimum_throughput,
+            initial_ping_time_estimate: Duration::from_millis(500),
+            maximum_assumed_ping_time: Duration::from_millis(1500),
+            read_ahead_before_playback: Duration::from_secs(1),
+            read_ahead_during_playback: Duration::from_secs(5),
+            prefetch_threshold_factor: 4.0,
+            download_timeout: Duration::from_secs(
+                (minimum_download_size / minimum_throughput) as u64,
+            ),
+        }
+    }
+}
+
+static AUDIO_FETCH_PARAMS: OnceLock<AudioFetchParams> = OnceLock::new();
+
+impl AudioFetchParams {
+    pub fn set(params: AudioFetchParams) -> Result<(), AudioFetchParams> {
+        AUDIO_FETCH_PARAMS.set(params)
+    }
+
+    pub fn get() -> &'static AudioFetchParams {
+        AUDIO_FETCH_PARAMS.get_or_init(AudioFetchParams::default)
+    }
+}
 
 pub enum AudioFile {
     Cached(fs::File),
@@ -183,6 +216,7 @@ impl StreamLoaderController {
 
         if let Some(ref shared) = self.stream_shared {
             let mut download_status = shared.download_status.lock();
+            let download_timeout = AudioFetchParams::get().download_timeout;
 
             while range.length
                 > download_status
@@ -191,7 +225,7 @@ impl StreamLoaderController {
             {
                 if shared
                     .cond
-                    .wait_for(&mut download_status, DOWNLOAD_TIMEOUT)
+                    .wait_for(&mut download_status, download_timeout)
                     .timed_out()
                 {
                     return Err(AudioFileError::WaitTimeout.into());
@@ -297,7 +331,7 @@ impl AudioFileShared {
         if ping_time_ms > 0 {
             Duration::from_millis(ping_time_ms as u64)
         } else {
-            INITIAL_PING_TIME_ESTIMATE
+            AudioFetchParams::get().initial_ping_time_estimate
         }
     }
 
@@ -395,6 +429,8 @@ impl AudioFileStreaming {
             trace!("Streaming from {}", url);
         }
 
+        let minimum_download_size = AudioFetchParams::get().minimum_download_size;
+
         // When the audio file is really small, this `download_size` may turn out to be
         // larger than the audio file we're going to stream later on. This is OK; requesting
         // `Content-Range` > `Content-Length` will return the complete file with status code
@@ -402,7 +438,7 @@ impl AudioFileStreaming {
         let mut streamer =
             session
                 .spclient()
-                .stream_from_cdn(&cdn_url, 0, MINIMUM_DOWNLOAD_SIZE)?;
+                .stream_from_cdn(&cdn_url, 0, minimum_download_size)?;
 
         // Get the first chunk with the headers to get the file size.
         // The remainder of that chunk with possibly also a response body is then
@@ -490,9 +526,10 @@ impl Read for AudioFileStreaming {
             return Ok(0);
         }
 
+        let read_ahead_during_playback = AudioFetchParams::get().read_ahead_during_playback;
         let length_to_request = if self.shared.is_download_streaming() {
             let length_to_request = length
-                + (READ_AHEAD_DURING_PLAYBACK.as_secs_f32() * self.shared.bytes_per_second as f32)
+                + (read_ahead_during_playback.as_secs_f32() * self.shared.bytes_per_second as f32)
                     as usize;
 
             // Due to the read-ahead stuff, we potentially request more than the actual request demanded.
@@ -515,11 +552,12 @@ impl Read for AudioFileStreaming {
                 .map_err(|err| io::Error::new(io::ErrorKind::BrokenPipe, err))?;
         }
 
+        let download_timeout = AudioFetchParams::get().download_timeout;
         while !download_status.downloaded.contains(offset) {
             if self
                 .shared
                 .cond
-                .wait_for(&mut download_status, DOWNLOAD_TIMEOUT)
+                .wait_for(&mut download_status, download_timeout)
                 .timed_out()
             {
                 return Err(io::Error::new(

audio/src/fetch/receive.rs

@@ -16,9 +16,8 @@ use librespot_core::{http_client::HttpClient, session::Session, Error};
 use crate::range_set::{Range, RangeSet};
 
 use super::{
-    AudioFileError, AudioFileResult, AudioFileShared, StreamLoaderCommand, StreamingRequest,
-    MAXIMUM_ASSUMED_PING_TIME, MINIMUM_DOWNLOAD_SIZE, MINIMUM_THROUGHPUT,
-    PREFETCH_THRESHOLD_FACTOR,
+    AudioFetchParams, AudioFileError, AudioFileResult, AudioFileShared, StreamLoaderCommand,
+    StreamingRequest,
 };
 
 struct PartialFileData {
@@ -151,6 +150,8 @@ struct AudioFileFetch {
     file_data_tx: mpsc::UnboundedSender<ReceivedData>,
     complete_tx: Option<oneshot::Sender<NamedTempFile>>,
     network_response_times: Vec<Duration>,
+
+    params: AudioFetchParams,
 }
 
 // Might be replaced by enum from std once stable
@@ -166,8 +167,8 @@ impl AudioFileFetch {
     }
 
     fn download_range(&mut self, offset: usize, mut length: usize) -> AudioFileResult {
-        if length < MINIMUM_DOWNLOAD_SIZE {
-            length = MINIMUM_DOWNLOAD_SIZE;
+        if length < self.params.minimum_download_size {
+            length = self.params.minimum_download_size;
         }
 
         // If we are in streaming mode (so not seeking) then start downloading as large
@@ -258,13 +259,13 @@ impl AudioFileFetch {
     fn handle_file_data(&mut self, data: ReceivedData) -> Result<ControlFlow, Error> {
         match data {
             ReceivedData::Throughput(mut throughput) => {
-                if throughput < MINIMUM_THROUGHPUT {
+                if throughput < self.params.minimum_throughput {
                     warn!(
                         "Throughput {} kbps lower than minimum {}, setting to minimum",
                         throughput / 1000,
-                        MINIMUM_THROUGHPUT / 1000,
+                        self.params.minimum_throughput / 1000,
                     );
-                    throughput = MINIMUM_THROUGHPUT;
+                    throughput = self.params.minimum_throughput;
                 }
 
                 let old_throughput = self.shared.throughput();
@@ -287,13 +288,13 @@ impl AudioFileFetch {
                 self.shared.set_throughput(avg_throughput);
             }
             ReceivedData::ResponseTime(mut response_time) => {
-                if response_time > MAXIMUM_ASSUMED_PING_TIME {
+                if response_time > self.params.maximum_assumed_ping_time {
                     warn!(
                         "Time to first byte {} ms exceeds maximum {}, setting to maximum",
                         response_time.as_millis(),
-                        MAXIMUM_ASSUMED_PING_TIME.as_millis()
+                        self.params.maximum_assumed_ping_time.as_millis()
                     );
-                    response_time = MAXIMUM_ASSUMED_PING_TIME;
+                    response_time = self.params.maximum_assumed_ping_time;
                 }
 
                 let old_ping_time_ms = self.shared.ping_time().as_millis();
@@ -423,6 +424,8 @@ pub(super) async fn audio_file_fetch(
         initial_request,
     ));
 
+    let params = AudioFetchParams::get();
+
     let mut fetch = AudioFileFetch {
         session: session.clone(),
         shared,
@@ -431,6 +434,8 @@ pub(super) async fn audio_file_fetch(
         file_data_tx,
         complete_tx: Some(complete_tx),
         network_response_times: Vec::with_capacity(3),
+
+        params: params.clone(),
     };
 
     loop {
@@ -472,7 +477,7 @@ pub(super) async fn audio_file_fetch(
             let throughput = fetch.shared.throughput();
 
             let desired_pending_bytes = max(
-                (PREFETCH_THRESHOLD_FACTOR
+                (params.prefetch_threshold_factor
                     * ping_time_seconds
                     * fetch.shared.bytes_per_second as f32) as usize,
                 (ping_time_seconds * throughput as f32) as usize,

audio/src/lib.rs

@@ -7,5 +7,4 @@ mod fetch;
 mod range_set;
 
 pub use decrypt::AudioDecrypt;
-pub use fetch::{AudioFile, AudioFileError, StreamLoaderController};
-pub use fetch::{MINIMUM_DOWNLOAD_SIZE, READ_AHEAD_BEFORE_PLAYBACK, READ_AHEAD_DURING_PLAYBACK};
+pub use fetch::{AudioFetchParams, AudioFile, AudioFileError, StreamLoaderController};

playback/src/decoder/symphonia_decoder.rs

@@ -36,7 +36,7 @@ impl SymphoniaDecoder {
         R: MediaSource + 'static,
     {
         let mss_opts = MediaSourceStreamOptions {
-            buffer_len: librespot_audio::MINIMUM_DOWNLOAD_SIZE,
+            buffer_len: librespot_audio::AudioFetchParams::get().minimum_download_size,
         };
         let mss = MediaSourceStream::new(Box::new(input), mss_opts);
 

playback/src/player.rs

@@ -24,10 +24,7 @@ use symphonia::core::io::MediaSource;
 use tokio::sync::{mpsc, oneshot};
 
 use crate::{
-    audio::{
-        AudioDecrypt, AudioFile, StreamLoaderController, READ_AHEAD_BEFORE_PLAYBACK,
-        READ_AHEAD_DURING_PLAYBACK,
-    },
+    audio::{AudioDecrypt, AudioFetchParams, AudioFile, StreamLoaderController},
     audio_backend::Sink,
     config::{Bitrate, NormalisationMethod, NormalisationType, PlayerConfig},
     convert::Converter,
@@ -2223,13 +2220,14 @@ impl PlayerInternal {
             ..
         } = self.state
         {
+            let read_ahead_during_playback = AudioFetchParams::get().read_ahead_during_playback;
             // Request our read ahead range
             let request_data_length =
-                (READ_AHEAD_DURING_PLAYBACK.as_secs_f32() * bytes_per_second as f32) as usize;
+                (read_ahead_during_playback.as_secs_f32() * bytes_per_second as f32) as usize;
 
             // Request the part we want to wait for blocking. This effectively means we wait for the previous request to partially complete.
             let wait_for_data_length =
-                (READ_AHEAD_BEFORE_PLAYBACK.as_secs_f32() * bytes_per_second as f32) as usize;
+                (read_ahead_during_playback.as_secs_f32() * bytes_per_second as f32) as usize;
 
             stream_loader_controller
                 .fetch_next_and_wait(request_data_length, wait_for_data_length)