refactor(audio): remove commented debug info and clean up tracing usage
This commit is contained in:
parent
df1df2a294
commit
5d42526121
47
src/audio.rs
47
src/audio.rs
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@ -9,7 +9,7 @@ use symphonia::core::formats::{FormatOptions, FormatReader};
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use symphonia::core::io::MediaSourceStream;
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use symphonia::core::meta::MetadataOptions;
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use symphonia::core::probe::Hint;
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use tracing::{//info!, debug, error, warn};
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use tracing::{error, warn};
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pub struct AsioPlayer {
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device: Device,
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@ -38,7 +38,7 @@ impl AsioPlayer {
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let mut devices = host.output_devices()?;
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let device = devices.next()
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.ok_or_else(|| anyhow::anyhow!("No ASIO output device found"))?;
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//info!!("Using ASIO device: {}", device.name()?);
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//info!("Using ASIO device: {}", device.name()?);
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Ok(Self {
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device,
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@ -75,17 +75,17 @@ impl AsioPlayer {
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}
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self.forced_sample_rate = Some(sample_rate);
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//info!!("Forced sample rate set to {} Hz", sample_rate);
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//info!("Forced sample rate set to {} Hz", sample_rate);
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Ok(())
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}
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pub fn clear_sample_rate(&mut self) {
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self.forced_sample_rate = None;
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////info!!("Sample rate forcing cleared - will use automatic selection");
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//info!("Sample rate forcing cleared - will use automatic selection");
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}
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pub async fn play_url(&mut self, url: &str) -> Result<()> {
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////info!!("Starting playback of: {}", url);
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//info!("Starting playback of: {}", url);
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// Stop any existing playback
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self.stop()?;
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@ -104,7 +104,7 @@ impl AsioPlayer {
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stream.play()?;
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self.start_time = Some(Instant::now());
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self.paused_position = None;
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//info!!("Playback started");
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//info!("Playback started");
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}
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Ok(())
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@ -127,7 +127,7 @@ impl AsioPlayer {
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buffer.paused = true;
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}
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//info!!("Playback paused");
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//info!("Playback paused");
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Ok(())
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} else {
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Err(anyhow::anyhow!("No active stream to pause"))
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@ -150,7 +150,7 @@ impl AsioPlayer {
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}
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stream.play()?;
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//info!!("Playback resumed");
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//info!("Playback resumed");
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Ok(())
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} else {
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Err(anyhow::anyhow!("No active stream to resume"))
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@ -176,9 +176,10 @@ impl AsioPlayer {
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self.start_time = None;
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self.paused_position = None;
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////info!!("Playback stopped and buffer cleared");
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//info!("Playback stopped and buffer cleared");
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Ok(())
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} else {
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// Not an error if already stopped
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Ok(())
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}
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}
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@ -206,7 +207,7 @@ impl AsioPlayer {
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}
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}
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////info!!("Seeked to position: {:.1}%", position * 100.0);
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//info!("Seeked to position: {:.1}%", position * 100.0);
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Ok(())
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} else {
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Err(anyhow::anyhow!("No active stream to seek"))
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@ -220,7 +221,7 @@ impl AsioPlayer {
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if let Some(buffer) = &self.stream_buffer {
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let mut buffer = buffer.lock().unwrap();
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buffer.volume = volume;
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////info!!("Volume set to: {:.0}%", volume * 100.0);
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//info!("Volume set to: {:.0}%", volume * 100.0);
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}
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Ok(())
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@ -241,12 +242,12 @@ impl AsioPlayer {
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}
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async fn download_and_decode(&self, url: &str) -> Result<(Vec<f32>, u32, u16, u64)> {
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debug!("Downloading audio from: {}", url);
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//debug!("Downloading audio from: {}", url);
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let response = reqwest::get(url).await?;
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let bytes = response.bytes().await?;
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debug!("Downloaded {} bytes, decoding...", bytes.len());
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//debug!("Downloaded {} bytes, decoding...", bytes.len());
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// Create media source
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let cursor = std::io::Cursor::new(bytes);
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@ -276,7 +277,7 @@ impl AsioPlayer {
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let mut decoder = symphonia::default::get_codecs()
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.make(&track.codec_params, &decoder_opts)?;
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////info!!!("Decoding: {} channels, {} Hz", channels, sample_rate);
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//info!("Decoding: {} channels, {} Hz", channels, sample_rate);
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let mut samples = Vec::new();
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@ -348,12 +349,12 @@ impl AsioPlayer {
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}
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}
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_ => {
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debug!("Unsupported audio format");
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//debug!("Unsupported audio format");
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}
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}
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}
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Err(e) => {
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debug!("Decode error: {}", e);
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//debug!("Decode error: {}", e);
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break;
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}
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}
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@ -366,7 +367,7 @@ impl AsioPlayer {
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let num_frames = samples.len() / channels as usize;
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let duration_ms = (num_frames as f64 / sample_rate as f64 * 1000.0) as u64;
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//info!!("Decoded {} samples, duration: {}ms", samples.len(), duration_ms);
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//info!("Decoded {} samples, duration: {}ms", samples.len(), duration_ms);
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Ok((samples, sample_rate, channels, duration_ms))
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}
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@ -386,7 +387,7 @@ impl AsioPlayer {
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config.channels() == source_channels {
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selected_config = Some(config.with_sample_rate(SampleRate(forced_rate)));
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target_sample_rate = forced_rate;
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//info!!("Using forced sample rate: {} Hz", forced_rate);
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//info!("Using forced sample rate: {} Hz", forced_rate);
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break;
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}
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}
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@ -446,8 +447,8 @@ impl AsioPlayer {
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self.device.default_output_config()?
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};
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//info!!("Selected config: {} channels, {} Hz, format: {:?}",
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// config.channels(), config.sample_rate().0, config.sample_format());
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//info!("Selected config: {} channels, {} Hz, format: {:?}",
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//config.channels(), config.sample_rate().0, config.sample_format());
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// Resample if necessary
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let final_samples = if target_sample_rate != source_sample_rate {
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@ -521,7 +522,7 @@ impl AsioPlayer {
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fn build_f32_stream(&mut self, config: StreamConfig, audio_buffer: Arc<Mutex<StreamBuffer>>) -> Result<()> {
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let stream = self.device.build_output_stream(
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&config,
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move |data: &mut [f32], _: &cpal::OutputCallback//info!| {
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move |data: &mut [f32], _: &cpal::OutputCallbackInfo| {
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let mut buffer = audio_buffer.lock().unwrap();
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// If paused, just output silence
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@ -566,7 +567,7 @@ impl AsioPlayer {
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fn build_i32_stream(&mut self, config: StreamConfig, audio_buffer: Arc<Mutex<StreamBuffer>>) -> Result<()> {
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let stream = self.device.build_output_stream(
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&config,
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move |data: &mut [i32], _: &cpal::OutputCallback//info!| {
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move |data: &mut [i32], _: &cpal::OutputCallbackInfo| {
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let mut buffer = audio_buffer.lock().unwrap();
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// If paused, just output silence
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@ -601,7 +602,7 @@ impl AsioPlayer {
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fn build_i16_stream(&mut self, config: StreamConfig, audio_buffer: Arc<Mutex<StreamBuffer>>) -> Result<()> {
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let stream = self.device.build_output_stream(
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&config,
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move |data: &mut [i16], _: &cpal::OutputCallback//info!| {
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move |data: &mut [i16], _: &cpal::OutputCallbackInfo| {
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let mut buffer = audio_buffer.lock().unwrap();
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// If paused, just output silence
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