Fix/audio stream continuity (#16095)

* fix(audio): add streaming resampler * fix(audio): preserve stream resampling state * fix(audio): keep playback callback nonblocking * fix(audio): decouple capture conversion from dasp * fix(audio): support stateful samplerate backend * refactor(audio): isolate stream callback state * refactor(audio): group capture output options * fix(audio): clear stale playback state after startup failure Reset non-Linux playback state when stream startup fails to prevent new-format audio from using the previous stream or resampler. Add regression tests for failed format changes and successful playback. Signed-off-by: fufesou <linlong1266@gmail.com> * fix(audio): honor capture resampler selection and reuse buffers Use the selected resampling backend for fixed-frame capture. Convert samples directly into the input queue and reuse the PCM frame buffer. Add tests for anti-aliasing, thread transfer, and partial-frame draining. Signed-off-by: fufesou <linlong1266@gmail.com> * refact: reduce diffs Signed-off-by: fufesou <linlong1266@gmail.com> * test(audio): check resampler output count and passband energy Signed-off-by: fufesou <linlong1266@gmail.com> * fix(audio): reset incompatible Linux playback state on startup failure Preserve compatible output streams when replacement startup fails. Clear state when no compatible stream exists and cover both paths in tests. Signed-off-by: fufesou <linlong1266@gmail.com> * perf(audio): reuse PCM buffers in the capture pipeline - Reuse capture framing, resampling, and channel conversion buffers - Deliver borrowed packets and write Sinc output into reusable storage - Add allocation and output-equivalence regression tests Signed-off-by: fufesou <linlong1266@gmail.com> * fix(audio): smooth buffer discard discontinuities Signal receiver PCM discards and fade from the current playback output when the callback reaches the new timeline. Signed-off-by: fufesou <linlong1266@gmail.com> * fix(audio): add missing Cargo.toml Signed-off-by: fufesou <linlong1266@gmail.com> * perf(audio): move capture encoding off the CPAL callback Move Opus encoding and service delivery to a dedicated worker. Use a preallocated bounded PCM queue with explicit loss reporting. Add tests for callback allocations and queue saturation. Signed-off-by: fufesou <linlong1266@gmail.com> * fix(audio): smooth capture gaps and report losses during backlog Signed-off-by: fufesou <linlong1266@gmail.com> * feat(audio): report capture queue high-water mark Track peak queued PCM packets and log the approximate queued audio duration alongside capture loss statistics. Signed-off-by: fufesou <linlong1266@gmail.com> * refact(audio): reduce diffs Signed-off-by: fufesou <linlong1266@gmail.com> * fix(audio): avoid blocking capture on encoder queue contention Use preallocated queues with try_lock in the capture callback. Count and drop the current packet on contention, preserving drop-oldest behavior on overflow. Add regressions for paused workers, buffer reuse, and sequence wrap. Signed-off-by: fufesou <linlong1266@gmail.com> * fix: add the missing files Signed-off-by: fufesou <linlong1266@gmail.com> * fix(audio): isolate zero-gate state per encoder Signed-off-by: fufesou <linlong1266@gmail.com> * refact: reduce diffs Signed-off-by: fufesou <linlong1266@gmail.com> * refact(audio): simple refactor Signed-off-by: fufesou <linlong1266@gmail.com> * fix(audio): avoid waiting on playback callback locks Use one PCM try_lock attempt and preserve queued samples during contention. Replace readiness locking with per-stream atomic status and report callback errors from the receiving thread. Cover callback progress, retained audio, recovery, and poisoned-buffer handling. * fix(audio): restart capture after processing errors Stop further processing until the service recreates the stream. Document the guard as defensive recovery for an unconfirmed failure. Group capture and resampler submodules under their parent directories. Signed-off-by: fufesou <linlong1266@gmail.com> * audio: report capture queue contention drops separately - Add contention_dropped to loss reports while preserving total drop counts - Document packet rejection on contention even when buffers are available - Extend existing contention and saturation test assertions Signed-off-by: fufesou <linlong1266@gmail.com> * refact unit tests Signed-off-by: fufesou <linlong1266@gmail.com> --------- Signed-off-by: fufesou <linlong1266@gmail.com>
2026-09-19 10:51:01 +03:00 · 2026-09-10 16:00:58 +08:00
parent 978e2e28b9
commit c4221469d8
20 changed files with 2957 additions and 221 deletions
--- a/src/audio_resampler/allocation_tests.rs
+++ b/src/audio_resampler/allocation_tests.rs
@@ -0,0 +1,142 @@
+use super::{AudioResamplerConfig, FixedFrameAudioResampler};
+use std::alloc::{GlobalAlloc, Layout, System};
+use std::cell::Cell;
+
+struct CountingAllocator;
+
+thread_local! {
+    static ALLOCATIONS: Cell<Option<usize>> = const { Cell::new(None) };
+}
+
+fn record_allocation() {
+    let _ = ALLOCATIONS.try_with(|count| {
+        if let Some(value) = count.get() {
+            count.set(Some(value + 1));
+        }
+    });
+}
+
+unsafe impl GlobalAlloc for CountingAllocator {
+    unsafe fn alloc(&self, layout: Layout) -> *mut u8 {
+        record_allocation();
+        unsafe { System.alloc(layout) }
+    }
+
+    unsafe fn alloc_zeroed(&self, layout: Layout) -> *mut u8 {
+        record_allocation();
+        unsafe { System.alloc_zeroed(layout) }
+    }
+
+    unsafe fn realloc(&self, ptr: *mut u8, layout: Layout, size: usize) -> *mut u8 {
+        record_allocation();
+        unsafe { System.realloc(ptr, layout, size) }
+    }
+
+    unsafe fn dealloc(&self, ptr: *mut u8, layout: Layout) {
+        unsafe { System.dealloc(ptr, layout) }
+    }
+}
+
+#[global_allocator]
+static ALLOCATOR: CountingAllocator = CountingAllocator;
+
+pub(crate) fn assert_no_allocations(process: impl FnOnce()) {
+    struct ResetCounter;
+    impl Drop for ResetCounter {
+        fn drop(&mut self) {
+            ALLOCATIONS.with(|count| count.set(None));
+        }
+    }
+
+    ALLOCATIONS.with(|count| assert!(count.replace(Some(0)).is_none()));
+    let reset = ResetCounter;
+    process();
+    let allocations = ALLOCATIONS.with(|count| count.get().unwrap());
+    drop(reset);
+    assert_eq!(
+        allocations, 0,
+        "PCM processing allocated on the capture thread"
+    );
+}
+
+#[test]
+fn capture_resampling_reuses_buffers() {
+    const PACKETS_PER_SECOND: usize = 100;
+    const PACKET_COUNT: usize = 100;
+    const MAX_STARTUP_DELAY_PACKETS: usize = 1;
+    const SIGNAL_LEVEL: f32 = 0.25;
+    const RATE_PAIRS: [(u32, u32); 6] = [
+        (32_000, 24_000),
+        (44_100, 24_000),
+        (44_100, 48_000),
+        (48_000, 24_000),
+        (96_000, 48_000),
+        (192_000, 48_000),
+    ];
+
+    for (input_rate, output_rate) in RATE_PAIRS {
+        for channels in [1, 2, 4, 6, 8] {
+            let config = AudioResamplerConfig {
+                input_rate,
+                output_rate,
+                channels,
+            };
+            let input =
+                vec![SIGNAL_LEVEL; input_rate as usize / PACKETS_PER_SECOND * channels as usize];
+            let frames = output_rate as usize / PACKETS_PER_SECOND;
+            let mut resampler = FixedFrameAudioResampler::new(config, frames).unwrap();
+            let mut packets = 0;
+            let mut energy = 0.0;
+            assert_no_allocations(|| {
+                for _ in 0..PACKET_COUNT {
+                    resampler
+                        .process_with(&input, |packet| {
+                            assert_eq!(packet.len(), frames * channels as usize);
+                            energy += packet.iter().map(|sample| sample * sample).sum::<f32>();
+                            packets += 1;
+                        })
+                        .unwrap();
+                }
+            });
+            assert!((PACKET_COUNT - MAX_STARTUP_DELAY_PACKETS..=PACKET_COUNT).contains(&packets));
+            assert!(energy > SIGNAL_LEVEL);
+        }
+    }
+}
+
+#[cfg(all(feature = "use_samplerate", not(feature = "use_dasp")))]
+#[test]
+fn sinc_output_matches_the_existing_backend() {
+    use super::AudioResampler;
+
+    const INPUT_FRAMES: usize = 2_048;
+    const CHUNK_FRAMES: usize = 73;
+    const SIGNAL_STEP: f32 = 0.07;
+    for (input_rate, output_rate) in [(44_100, 24_000), (44_100, 48_000), (96_000, 48_000)] {
+        for channels in [1, 2, 4, 6, 8] {
+            let config = AudioResamplerConfig {
+                input_rate,
+                output_rate,
+                channels,
+            };
+            let input: Vec<_> = (0..INPUT_FRAMES * channels as usize)
+                .map(|sample| (sample as f32 * SIGNAL_STEP).sin())
+                .collect();
+            let mut actual = AudioResampler::new(config).unwrap();
+            let expected = samplerate::Samplerate::new(
+                samplerate::ConverterType::SincBestQuality,
+                input_rate,
+                output_rate,
+                channels as usize,
+            )
+            .unwrap();
+            for chunk in input.chunks(CHUNK_FRAMES * channels as usize) {
+                assert_eq!(
+                    actual.process(chunk).unwrap(),
+                    expected.process(chunk).unwrap()
+                );
+                assert_eq!(actual.process(&[]).unwrap(), expected.process(&[]).unwrap());
+            }
+        }
+    }
+}
--- a/src/audio_resampler/samplerate_tests.rs
+++ b/src/audio_resampler/samplerate_tests.rs
@@ -0,0 +1,157 @@
+use super::{AudioResampler, AudioResamplerConfig, AudioResamplerError, FixedFrameAudioResampler};
+
+const INPUT_RATE: u32 = 44_100;
+const OUTPUT_RATE: u32 = 48_000;
+const CHANNELS: u16 = 2;
+const INPUT_PACKET_FRAMES: usize = INPUT_RATE as usize / PACKETS_PER_SECOND;
+const OUTPUT_PACKET_FRAMES: usize = OUTPUT_RATE as usize / PACKETS_PER_SECOND;
+const PACKET_COUNT: usize = 20;
+const PACKETS_PER_SECOND: usize = 100;
+const MIN_CONTINUITY_PACKETS: usize = 2;
+const TONE_FREQUENCY_HZ: f32 = 997.0;
+const TONE_AMPLITUDE: f32 = 0.5;
+const MAX_BOUNDARY_RESIDUAL: f32 = 0.02;
+const INCOMPLETE_SAMPLE_COUNT: usize = 1;
+const DOWNSAMPLE_RATE: u32 = 24_000;
+const REJECTED_TONE_HZ: f64 = 18_000.0;
+const MAX_ALIAS_RMS: f64 = 0.01;
+const MIN_PASSBAND_RMS: f64 = 0.3;
+
+fn stereo_tone(frames: usize) -> Vec<f32> {
+    (0..frames)
+        .flat_map(|frame| {
+            let phase =
+                std::f32::consts::TAU * TONE_FREQUENCY_HZ * frame as f32 / INPUT_RATE as f32;
+            let sample = TONE_AMPLITUDE * phase.sin();
+            [sample, sample]
+        })
+        .collect()
+}
+
+fn maximum_boundary_residual(packets: &[Vec<f32>]) -> f32 {
+    packets.windows(2).fold(0.0, |maximum, pair| {
+        let previous = &pair[0];
+        let current = &pair[1];
+        let last = previous.len() - CHANNELS as usize;
+        let penultimate = last - CHANNELS as usize;
+        (0..CHANNELS as usize).fold(maximum, |maximum, channel| {
+            let predicted = previous[last + channel]
+                + (previous[last + channel] - previous[penultimate + channel]);
+            maximum.max((current[channel] - predicted).abs())
+        })
+    })
+}
+
+fn stereo_config() -> AudioResamplerConfig {
+    AudioResamplerConfig {
+        input_rate: INPUT_RATE,
+        output_rate: OUTPUT_RATE,
+        channels: CHANNELS,
+    }
+}
+
+#[test]
+fn moving_capture_resampler_preserves_pending_audio() {
+    let input = stereo_tone(INPUT_PACKET_FRAMES * PACKET_COUNT);
+    let packet_samples = INPUT_PACKET_FRAMES * CHANNELS as usize;
+    let mut expected_resampler =
+        FixedFrameAudioResampler::new(stereo_config(), OUTPUT_PACKET_FRAMES).unwrap();
+    let expected: Vec<_> = input
+        .chunks(packet_samples)
+        .flat_map(|packet| expected_resampler.process(packet).unwrap())
+        .collect();
+    let mut moved_resampler =
+        FixedFrameAudioResampler::new(stereo_config(), OUTPUT_PACKET_FRAMES).unwrap();
+    let mut output = moved_resampler.process(&input[..packet_samples]).unwrap();
+    let remaining = std::thread::spawn(move || {
+        input[packet_samples..]
+            .chunks(packet_samples)
+            .flat_map(|packet| moved_resampler.process(packet).unwrap())
+            .collect::<Vec<_>>()
+    })
+    .join()
+    .unwrap();
+    output.extend(remaining);
+
+    assert!(output.len() >= MIN_CONTINUITY_PACKETS);
+    assert!(output
+        .iter()
+        .all(|packet| packet.len() == OUTPUT_PACKET_FRAMES * CHANNELS as usize));
+    assert!(maximum_boundary_residual(&output) <= MAX_BOUNDARY_RESIDUAL);
+    assert_eq!(output, expected);
+}
+
+fn downsampled_rms(input: &[f32]) -> f64 {
+    let config = AudioResamplerConfig {
+        output_rate: DOWNSAMPLE_RATE,
+        ..stereo_config()
+    };
+    let output_frames = DOWNSAMPLE_RATE as usize / PACKETS_PER_SECOND;
+    let mut resampler = FixedFrameAudioResampler::new(config, output_frames).unwrap();
+    let output: Vec<f32> = input
+        .chunks(INPUT_PACKET_FRAMES * CHANNELS as usize)
+        .flat_map(|packet| resampler.process(packet).unwrap().into_iter().flatten())
+        .collect();
+
+    assert!(output.len() >= output_frames * CHANNELS as usize * MIN_CONTINUITY_PACKETS);
+    let mean_square = output
+        .iter()
+        .map(|sample| f64::from(*sample).powi(2))
+        .sum::<f64>()
+        / output.len() as f64;
+    mean_square.sqrt()
+}
+
+#[test]
+fn capture_downsampling_filters_out_of_band_audio() {
+    let input: Vec<_> = (0..INPUT_PACKET_FRAMES * PACKET_COUNT)
+        .flat_map(|frame| {
+            let phase =
+                std::f64::consts::TAU * REJECTED_TONE_HZ * frame as f64 / f64::from(INPUT_RATE);
+            let sample = (f64::from(TONE_AMPLITUDE) * phase.sin()) as f32;
+            [sample, sample]
+        })
+        .collect();
+    let rms = downsampled_rms(&input);
+    assert!(
+        rms < MAX_ALIAS_RMS,
+        "out-of-band output RMS {rms} exceeded {MAX_ALIAS_RMS}"
+    );
+
+    let input = stereo_tone(INPUT_PACKET_FRAMES * PACKET_COUNT);
+    let rms = downsampled_rms(&input);
+    assert!(
+        rms > MIN_PASSBAND_RMS,
+        "in-band output RMS {rms} fell below {MIN_PASSBAND_RMS}"
+    );
+}
+
+#[test]
+fn samplerate_backend_preserves_streaming_continuity() {
+    let input = stereo_tone(INPUT_PACKET_FRAMES * PACKET_COUNT);
+    let mut resampler = AudioResampler::new(stereo_config()).unwrap();
+    let packets: Vec<_> = input
+        .chunks(INPUT_PACKET_FRAMES * CHANNELS as usize)
+        .map(|packet| resampler.process(packet).unwrap())
+        .filter(|packet| !packet.is_empty())
+        .collect();
+
+    assert!(packets.len() >= MIN_CONTINUITY_PACKETS);
+    assert!(packets
+        .iter()
+        .all(|packet| packet.len() % CHANNELS as usize == 0));
+    assert!(maximum_boundary_residual(&packets) <= MAX_BOUNDARY_RESIDUAL);
+}
+
+#[test]
+fn samplerate_backend_reports_incomplete_frame_context() {
+    let mut resampler = AudioResampler::new(stereo_config()).unwrap();
+
+    assert_eq!(
+        resampler.process(&[0.0]).unwrap_err(),
+        AudioResamplerError::IncompleteFrame {
+            samples: INCOMPLETE_SAMPLE_COUNT,
+            channels: CHANNELS as usize,
+        }
+    );
+}
--- a/src/audio_resampler/sinc.rs
+++ b/src/audio_resampler/sinc.rs
@@ -0,0 +1,112 @@
+use super::{AudioResamplerConfig, AudioResamplerError};
+use libsamplerate_sys as sys;
+use std::ptr::NonNull;
+
+const OUTPUT_MARGIN_FRAMES: usize = 1;
+
+pub(super) struct SincAudioResampler {
+    state: NonNull<sys::SRC_STATE>,
+    config: AudioResamplerConfig,
+}
+
+impl SincAudioResampler {
+    pub(super) fn new(config: AudioResamplerConfig) -> Result<Self, AudioResamplerError> {
+        super::validate_config(config)?;
+        let ratio = f64::from(config.output_rate) / f64::from(config.input_rate);
+        if unsafe { sys::src_is_valid_ratio(ratio) } == 0 {
+            return Err(backend_error(
+                config,
+                samplerate::ErrorCode::BadSrcRatio as _,
+            ));
+        }
+        let mut error = 0;
+        // SAFETY: src_new allocates independent state; this owner releases it in Drop.
+        let state = unsafe {
+            sys::src_new(
+                sys::SRC_SINC_BEST_QUALITY as _,
+                config.channels.into(),
+                &mut error,
+            )
+        };
+        let state = NonNull::new(state).ok_or_else(|| backend_error(config, error))?;
+        Ok(Self { state, config })
+    }
+
+    pub(super) fn process_into(
+        &mut self,
+        input: &[f32],
+        output: &mut Vec<f32>,
+    ) -> Result<(), AudioResamplerError> {
+        super::validate_input(input, self.config.channels as usize)?;
+        let mut consumed = 0;
+        loop {
+            let (used, generated) = self.process_block(&input[consumed..], output)?;
+            consumed += used;
+            if consumed == input.len() {
+                return Ok(());
+            }
+            if used == 0 && generated == 0 {
+                return Err(AudioResamplerError::Backend(
+                    "libsamplerate made no progress while input remained".to_owned(),
+                ));
+            }
+        }
+    }
+
+    fn process_block(
+        &mut self,
+        input: &[f32],
+        output: &mut Vec<f32>,
+    ) -> Result<(usize, usize), AudioResamplerError> {
+        let channels = self.config.channels as usize;
+        let input_frames = input.len() / channels;
+        let output_frames = input_frames
+            .checked_mul(self.config.output_rate as usize)
+            .map(|frames| frames / self.config.input_rate as usize)
+            .and_then(|frames| frames.checked_add(OUTPUT_MARGIN_FRAMES))
+            .ok_or(AudioResamplerError::CapacityOverflow)?;
+        let start = output.len();
+        let end = output_frames
+            .checked_mul(channels)
+            .and_then(|samples| start.checked_add(samples))
+            .ok_or(AudioResamplerError::CapacityOverflow)?;
+        let mut data = sys::SRC_DATA {
+            data_in: input.as_ptr(),
+            input_frames: input_frames
+                .try_into()
+                .map_err(|_| AudioResamplerError::CapacityOverflow)?,
+            output_frames: output_frames
+                .try_into()
+                .map_err(|_| AudioResamplerError::CapacityOverflow)?,
+            src_ratio: f64::from(self.config.output_rate) / f64::from(self.config.input_rate),
+            ..Default::default()
+        };
+        output.resize(end, 0.0);
+        data.data_out = output[start..].as_mut_ptr();
+        // SAFETY: state is exclusively owned; disjoint slices cover the declared frame counts.
+        let error = unsafe { sys::src_process(self.state.as_ptr(), &mut data) };
+        let generated = data.output_frames_gen as usize * channels;
+        output.truncate(start + generated);
+        if error != 0 {
+            return Err(backend_error(self.config, error));
+        }
+        Ok((data.input_frames_used as usize * channels, generated))
+    }
+}
+
+impl Drop for SincAudioResampler {
+    fn drop(&mut self) {
+        // SAFETY: this owner holds the only handle returned by src_new.
+        unsafe { sys::src_delete(self.state.as_ptr()) };
+    }
+}
+
+fn backend_error(config: AudioResamplerConfig, code: i32) -> AudioResamplerError {
+    AudioResamplerError::Backend(format!(
+        "input_rate={}, output_rate={}, channels={}: {:?}",
+        config.input_rate,
+        config.output_rate,
+        config.channels,
+        samplerate::Error::from_int(code)
+    ))
+}
--- a/src/audio_resampler/tests.rs
+++ b/src/audio_resampler/tests.rs
@@ -0,0 +1,178 @@
+use super::{AudioResampler, AudioResamplerConfig, FixedFrameAudioResampler};
+
+const INPUT_RATE: u32 = 24_000;
+const OUTPUT_RATE: u32 = 48_000;
+const CHANNELS: u16 = 2;
+const CHUNK_FRAMES: usize = 240;
+const CHUNK_COUNT: usize = 4;
+const TONE_FREQUENCY_HZ: f32 = 997.0;
+const TONE_AMPLITUDE: f32 = 0.5;
+const MAX_BOUNDARY_RESIDUAL: f32 = 0.02;
+const LOOK_AHEAD_OUTPUT_FRAMES: usize = 1;
+const UNEVEN_CHUNK_FRAMES: usize = 73;
+const MONO_CHANNELS: u16 = 1;
+const UNIT_RATE: u32 = 1;
+const DOUBLE_RATE: u32 = 2;
+const FIRST_DOWNSAMPLE_PACKET: [f32; 3] = [0.0, 1.0, 2.0];
+const SECOND_DOWNSAMPLE_PACKET: [f32; 4] = [3.0, 4.0, 5.0, 6.0];
+const EXPECTED_DOWNSAMPLED_OUTPUT: [f32; 4] = [0.0, 2.0, 4.0, 6.0];
+const PACKETS_PER_SECOND: usize = 100;
+const OUTPUT_PACKET_FRAMES: usize = OUTPUT_RATE as usize / PACKETS_PER_SECOND;
+const RATE_44_1_KHZ: u32 = 44_100;
+const FLOAT_TOLERANCE: f32 = 0.000_001;
+const MIN_CONTINUITY_PACKETS: usize = 2;
+
+fn stereo_tone_at_rate(frames: usize, sample_rate: u32) -> Vec<f32> {
+    (0..frames)
+        .flat_map(|frame| {
+            let phase =
+                std::f32::consts::TAU * TONE_FREQUENCY_HZ * frame as f32 / sample_rate as f32;
+            let sample = TONE_AMPLITUDE * phase.sin();
+            [sample, sample]
+        })
+        .collect()
+}
+
+fn stereo_tone(frames: usize) -> Vec<f32> {
+    stereo_tone_at_rate(frames, INPUT_RATE)
+}
+
+fn maximum_tone_prediction_residual(sample_rate: u32) -> f32 {
+    let half_step = std::f32::consts::PI * TONE_FREQUENCY_HZ / sample_rate as f32;
+    4.0 * TONE_AMPLITUDE * half_step.sin().powi(2)
+}
+
+fn maximum_boundary_residual(chunks: &[Vec<f32>]) -> f32 {
+    chunks.windows(2).fold(0.0, |maximum, pair| {
+        let previous = &pair[0];
+        let current = &pair[1];
+        let last = previous.len() - CHANNELS as usize;
+        let penultimate = last - CHANNELS as usize;
+        (0..CHANNELS as usize).fold(maximum, |maximum, channel| {
+            let predicted = previous[last + channel]
+                + (previous[last + channel] - previous[penultimate + channel]);
+            maximum.max((current[channel] - predicted).abs())
+        })
+    })
+}
+
+fn stereo_config() -> AudioResamplerConfig {
+    AudioResamplerConfig {
+        input_rate: INPUT_RATE,
+        output_rate: OUTPUT_RATE,
+        channels: CHANNELS,
+    }
+}
+
+#[test]
+fn preserves_decoded_packet_continuity_and_output_ratio() {
+    let input = stereo_tone(CHUNK_FRAMES * CHUNK_COUNT);
+    let mut whole_resampler = AudioResampler::new(stereo_config()).unwrap();
+    let whole_output = whole_resampler.process(&input).unwrap();
+    let expected_frames = CHUNK_FRAMES * CHUNK_COUNT * OUTPUT_RATE as usize / INPUT_RATE as usize
+        - LOOK_AHEAD_OUTPUT_FRAMES;
+
+    for chunk_frames in [CHUNK_FRAMES, UNEVEN_CHUNK_FRAMES] {
+        let mut resampler = AudioResampler::new(stereo_config()).unwrap();
+        let output: Vec<_> = input
+            .chunks(chunk_frames * CHANNELS as usize)
+            .map(|chunk| resampler.process(chunk).unwrap())
+            .collect();
+        let residual = maximum_boundary_residual(&output);
+        assert!(
+            residual <= MAX_BOUNDARY_RESIDUAL,
+            "packet boundary residual {residual} exceeded {MAX_BOUNDARY_RESIDUAL}, chunk_frames={chunk_frames}"
+        );
+        let output_frames = output.iter().map(Vec::len).sum::<usize>() / CHANNELS as usize;
+        assert_eq!(output_frames, expected_frames);
+        assert_eq!(output.concat(), whole_output, "chunk_frames={chunk_frames}");
+    }
+}
+
+#[test]
+fn rejects_incomplete_interleaved_frames() {
+    let mut resampler = AudioResampler::new(stereo_config()).unwrap();
+
+    assert!(resampler.process(&[TONE_AMPLITUDE]).is_err());
+}
+
+#[test]
+fn interpolates_mono_samples() {
+    let config = AudioResamplerConfig {
+        input_rate: UNIT_RATE,
+        output_rate: DOUBLE_RATE,
+        channels: MONO_CHANNELS,
+    };
+    let mut resampler = AudioResampler::new(config).unwrap();
+
+    assert_eq!(
+        resampler.process(&[0.0, 1.0, 2.0]).unwrap(),
+        [0.0, 0.5, 1.0, 1.5, 2.0]
+    );
+}
+
+#[test]
+fn rejects_zero_rate_configuration() {
+    let config = AudioResamplerConfig {
+        input_rate: 0,
+        output_rate: OUTPUT_RATE,
+        channels: CHANNELS,
+    };
+
+    assert!(AudioResampler::new(config).is_err());
+}
+
+#[test]
+fn downsamples_across_packet_boundaries() {
+    let config = AudioResamplerConfig {
+        input_rate: DOUBLE_RATE,
+        output_rate: UNIT_RATE,
+        channels: MONO_CHANNELS,
+    };
+    let mut resampler = AudioResampler::new(config).unwrap();
+    let mut output = resampler.process(&FIRST_DOWNSAMPLE_PACKET).unwrap();
+    output.extend(resampler.process(&SECOND_DOWNSAMPLE_PACKET).unwrap());
+
+    assert_eq!(output, EXPECTED_DOWNSAMPLED_OUTPUT);
+}
+
+#[test]
+fn sender_resampler_emits_only_complete_continuous_frames() {
+    let input = stereo_tone(CHUNK_FRAMES * CHUNK_COUNT);
+    let mut resampler =
+        FixedFrameAudioResampler::new(stereo_config(), OUTPUT_PACKET_FRAMES).unwrap();
+    let output: Vec<_> = input
+        .chunks(CHUNK_FRAMES * CHANNELS as usize)
+        .flat_map(|chunk| resampler.process(chunk).unwrap())
+        .collect();
+
+    assert!(output.len() >= MIN_CONTINUITY_PACKETS);
+    assert!(output
+        .iter()
+        .all(|packet| packet.len() == OUTPUT_PACKET_FRAMES * CHANNELS as usize));
+    assert!(maximum_boundary_residual(&output) <= MAX_BOUNDARY_RESIDUAL);
+}
+
+#[test]
+fn sender_downsampling_preserves_packet_continuity() {
+    let input_packet_frames = RATE_44_1_KHZ as usize / PACKETS_PER_SECOND;
+    let input = stereo_tone_at_rate(input_packet_frames * CHUNK_COUNT, RATE_44_1_KHZ);
+    let config = AudioResamplerConfig {
+        input_rate: RATE_44_1_KHZ,
+        output_rate: INPUT_RATE,
+        channels: CHANNELS,
+    };
+    let mut resampler =
+        FixedFrameAudioResampler::new(config, INPUT_RATE as usize / PACKETS_PER_SECOND).unwrap();
+    let packets: Vec<_> = input
+        .chunks(input_packet_frames * CHANNELS as usize)
+        .flat_map(|packet| resampler.process(packet).unwrap())
+        .collect();
+    let residual = maximum_boundary_residual(&packets);
+
+    assert_eq!(packets.len(), CHUNK_COUNT);
+    assert!(
+        residual <= maximum_tone_prediction_residual(INPUT_RATE) + FLOAT_TOLERANCE,
+        "sender packet boundary residual {residual} exceeded the tone curvature"
+    );
+}