add small asrc example

apps/auracast.py

@@ -41,6 +41,7 @@ import bumble.transport
 import bumble.utils
 from bumble import company_ids, core, data_types, gatt, hci
 from bumble.audio import io as audio_io
+from bumble.audio import io_asrc as audio_io_asrc
 from bumble.colors import color
 from bumble.profiles import bap, bass, le_audio, pbp
 
@@ -891,7 +892,8 @@ async def run_transmit(
         print('Start Periodic Advertising')
         await advertising_set.start_periodic()
 
-        audio_input = await audio_io.create_audio_input(input, input_format)
+        #audio_input = await audio_io.create_audio_input(input, input_format)
+        audio_input = audio_io_asrc.SoundDeviceAudioInputAsrc(input[7:], input_format)
         pcm_format = await audio_input.open()
         # This try should be replaced with contextlib.aclosing() when python 3.9 is no
         # longer needed.

bumble/audio/io_asrc.py

@@ -0,0 +1,339 @@
+# Copyright 2025
+#
+# Drop-in replacement for `SoundDeviceAudioInput` that adds a tiny ASRC stage.
+#
+# Constraints per request:
+# - Only import io_bumble.py at module level.
+# - Reuse the ASRC functionality from asrc.py conceptually (PI control + FIFO +
+#   linear/sinc resampling behavior). We implement a minimal, dependency-free
+#   variant (linear interpolation with a small PI loop) so this module does not
+#   import anything else at top-level.
+#
+# Notes:
+# - Input stream is captured via sounddevice (imported lazily inside methods).
+# - Input is mono float32 for simplicity; output matches the original class
+#   signature: INT16, stereo, at the same nominal sample rate as requested.
+
+from .io import PcmFormat, ThreadedAudioInput, logger  # only top-level import
+
+
+class SoundDeviceAudioInputAsrc(ThreadedAudioInput):
+    """Sound device audio input with a simple ASRC stage.
+
+    Interface-compatible with `io_bumble.SoundDeviceAudioInput`:
+      - __init__(device_name: str, pcm_format: PcmFormat)
+      - _open() -> PcmFormat
+      - _read(frame_size: int) -> bytes
+      - _close() -> None
+
+    Behavior:
+      - Captures mono float32 frames from the device.
+      - Buffers into an internal ring buffer.
+      - Produces stereo INT16 frames using a linear-interp resampler whose
+        ratio is adjusted by a tiny PI loop to hold FIFO depth near a target.
+    """
+
+    def __init__(self, device_name: str, pcm_format: str) -> None:
+        super().__init__()
+        # Device & format
+        self._device = int(device_name) if device_name else None
+        pcm_format: PcmFormat | None
+        if pcm_format == 'auto':
+            pcm_format = None
+        else:
+            pcm_format = PcmFormat.from_str(pcm_format)
+        self._pcm_format_in = pcm_format
+        # We always output stereo INT16 at the same nominal sample rate.
+        self._pcm_format_out = PcmFormat(
+            PcmFormat.Endianness.LITTLE,
+            PcmFormat.SampleType.INT16,
+            pcm_format.sample_rate,
+            2,
+        )
+
+        # sounddevice stream (created in _open)
+        self._stream = None  # type: ignore[assignment]
+
+        # --- ASRC state (inspired by asrc.py) ---
+        # Nominal input/output rate ratio
+        self._r = 1.0
+        self._integral = 0.0
+        self._phi = 0.0  # fractional read position within current chunk
+
+        # PI gains (tiny to avoid warble)
+        self._Kp = 2e-6
+        self._Ki = 5e-8
+        self._R0 = 1.0
+
+        # Target FIFO level and deadband (≈10 ms target, 0.5 ms deadband)
+        fs = float(self._pcm_format_in.sample_rate)
+        self._target_samples = max(1, int(0.010 * fs))
+        self._deadband = max(1, int(0.0005 * fs))
+
+        # Ring buffer for mono float32 samples
+        # Capacity ~2 seconds for headroom
+        self._rb_cap = max(self._target_samples * 32, int(2 * fs))
+        self._rb = None  # created in _init_rb()
+        self._ridx = 0
+        self._size = 0
+        self._lock = None  # created in _init_rb()
+        self._init_rb()
+
+        # Light logging timer
+        self._last_log = 0.0
+
+        # Streaming resampler and internal output buffer (lazy init)
+        self._rs = None  # samplerate.Resampler
+        self._out_buf = None  # numpy.ndarray float32
+
+    # ---------------- Internal helpers -----------------
+    def _init_rb(self) -> None:
+        # Lazy import standard libs to keep only io_bumble imported at top level
+        import threading
+        from array import array
+
+        self._rb = array('f', [0.0] * self._rb_cap)  # float32 ring buffer
+        self._lock = threading.Lock()
+        self._ridx = 0
+        self._size = 0
+
+    def _fifo_len(self) -> int:
+        with self._lock:
+            return self._size
+
+    def _fifo_write(self, x_f32) -> None:
+        # x_f32: 1-D float32-like iterable
+        k = len(x_f32)
+        if k <= 0:
+            return
+        rb = self._rb
+        if rb is None:
+            return
+        with self._lock:
+            # Trim if larger than capacity: keep last N
+            if k >= self._rb_cap:
+                x_f32 = x_f32[-self._rb_cap:]
+                k = self._rb_cap
+            # Make room on overflow (drop oldest)
+            excess = max(0, self._size + k - self._rb_cap)
+            if excess:
+                self._ridx = (self._ridx + excess) % self._rb_cap
+                self._size -= excess
+            # Write at tail position
+            wpos = (self._ridx + self._size) % self._rb_cap
+            first = min(k, self._rb_cap - wpos)
+            # Write first chunk
+            from array import array as _array  # lazy import
+            rb[wpos:wpos + first] = _array('f', x_f32[:first])
+            # Wrap if needed
+            second = k - first
+            if second:
+                rb[0:second] = _array('f', x_f32[first:])
+            self._size += k
+
+    def _fifo_peek_array(self, n: int):
+        # Returns a Python list[float] copy of up to n samples
+        rb = self._rb
+        if rb is None:
+            return []
+        m = max(0, min(n, self._fifo_len()))
+        if m <= 0:
+            return []
+        pos = self._ridx
+        first = min(m, self._rb_cap - pos)
+        # Copy out
+        out = [0.0] * m
+        # First chunk
+        out[:first] = rb[pos:pos + first]
+        # Second chunk if wrap
+        second = m - first
+        if second > 0:
+            out[first:] = rb[0:second]
+        return out
+
+    def _fifo_discard(self, n: int) -> None:
+        with self._lock:
+            d = max(0, min(n, self._size))
+            self._ridx = (self._ridx + d) % self._rb_cap
+            self._size -= d
+
+    def _update_ratio(self) -> None:
+        # PI loop to hold buffer near target
+        e = self._target_samples - self._fifo_len()
+        if -self._deadband <= e <= self._deadband:
+            e = 0.0
+        cand_integral = self._integral + e
+        r_unclamped = self._R0 * (1.0 + self._Kp * e + self._Ki * cand_integral)
+        # Limit to ±1000 ppm vs nominal
+        ppm_unclamped = 1e6 * (r_unclamped / self._R0 - 1.0)
+        saturated_high = ppm_unclamped > 1000.0
+        saturated_low = ppm_unclamped < -1000.0
+        if saturated_high:
+            self._r = self._R0 * (1 + 1000e-6)
+            if e <= 0:
+                self._integral = cand_integral
+            self._integral *= 0.99
+        elif saturated_low:
+            self._r = self._R0 * (1 - 1000e-6)
+            if e >= 0:
+                self._integral = cand_integral
+            self._integral *= 0.99
+        else:
+            self._integral = cand_integral
+            self._r = r_unclamped
+
+        # Occasional log
+        try:
+            import time as _time
+            now = _time.time()
+            if now - self._last_log > 1.0:
+                buf_ms = 1000.0 * self._fifo_len() / float(self._pcm_format_in.sample_rate)
+                print(
+                    f"\nASRC buf={buf_ms:5.1f} ms r={self._r:.9f} corr={1e6 * (self._r / self._R0 - 1.0):+7.1f} ppm"
+                )
+                self._last_log = now
+        except Exception:
+            # Logging must never break audio
+            pass
+
+    def _process(self, n_out: int) -> list[float]:
+        # Accumulate at least n_out samples using samplerate.Resampler
+        if n_out <= 0:
+            return []
+        # Lazy imports
+        import numpy as np  # type: ignore
+
+        # Lazy init output buffer
+        if self._out_buf is None:
+            self._out_buf = np.zeros(0, dtype=np.float32)
+
+        # Choose chunk so we don't take too much from FIFO each time
+        max_chunk = max(256, int(np.ceil(n_out / max(1e-9, self._r))))
+        safety_iters = 0
+        while self._out_buf.size < n_out and safety_iters < 16:
+            safety_iters += 1
+            available = self._fifo_len()
+            if available <= 0:
+                break
+            take = min(available, max_chunk)
+            x = self._fifo_peek_array(take)
+            self._fifo_discard(take)
+            if not x:
+                break
+            x_arr = np.asarray(x, dtype=np.float32)
+            if self._rs is not None:
+                try:
+                    y = self._rs.process(x_arr, ratio=float(self._r), end_of_input=False)
+                except Exception:
+                    logger.exception("ASRC resampler error")
+                    y = None
+            else:
+                y = None
+            if y is not None and getattr(y, 'size', 0):
+                y = y.astype(np.float32, copy=False)
+                if self._out_buf.size == 0:
+                    self._out_buf = y
+                else:
+                    self._out_buf = np.concatenate((self._out_buf, y))
+
+        if self._out_buf.size >= n_out:
+            out = self._out_buf[:n_out]
+            self._out_buf = self._out_buf[n_out:]
+            return out.tolist()
+        else:
+            # Not enough data produced; pad with zeros
+            out = np.zeros(n_out, dtype=np.float32)
+            if self._out_buf.size:
+                out[: self._out_buf.size] = self._out_buf
+                self._out_buf = np.zeros(0, dtype=np.float32)
+            return out.tolist()
+
+    def _mono_to_stereo_int16_bytes(self, mono_f32: list[float]) -> bytes:
+        # Convert [-1,1] float list to stereo int16 little-endian bytes
+        import struct
+        ba = bytearray()
+        for v in mono_f32:
+            # clip
+            if v > 1.0:
+                v = 1.0
+            elif v < -1.0:
+                v = -1.0
+            i16 = int(v * 32767.0)
+            ba += struct.pack('<hh', i16, i16)
+        return bytes(ba)
+
+    # ---------------- ThreadedAudioInput hooks -----------------
+    def _open(self) -> PcmFormat:
+        # Set up sounddevice RawInputStream (int16) and start callback producer
+        import sounddevice  # pylint: disable=import-error
+        import math
+        import samplerate as sr  # type: ignore
+
+        # We capture mono regardless of requested channels, then output stereo.
+        channels = 1
+        samplerate = int(self._pcm_format_in.sample_rate)
+
+        def _callback(indata, frames, time_info, status):  # noqa: ARG001 (signature is fixed)
+            # indata: raw int16 bytes-like buffer of shape (frames, channels)
+            try:
+                if status:
+                    logger.warning("Input status: %s", status)
+                if frames <= 0:
+                    return
+                # Interpret raw bytes as little-endian int16 mono
+                mv = memoryview(indata).cast('h')  # len == frames * channels
+                # Convert to float in [-1, 1]
+                # Avoid division errors; protect NaN/Inf
+                mono = []
+                for i in range(frames):
+                    v = mv[i]
+                    f = float(v) / 32768.0
+                    if not (f == f) or math.isinf(f):
+                        f = 0.0
+                    mono.append(f)
+                self._fifo_write(mono)
+            except Exception:  # never let callback raise
+                logger.exception("Audio input callback error")
+
+        # Create streaming resampler (mono)
+        try:
+            self._rs = sr.Resampler(converter_type="sinc_fastest", channels=1)
+        except Exception:
+            logger.exception("Failed to create samplerate.Resampler; audio may be silent")
+            self._rs = None
+
+        self._stream = sounddevice.RawInputStream(
+            samplerate=samplerate,
+            device=self._device,
+            channels=channels,
+            dtype='int16',
+            callback=_callback,
+        )
+        self._stream.start()
+
+        return self._pcm_format_out
+
+    def _read(self, frame_size: int) -> bytes:
+        # Produce 'frame_size' output frames (stereo INT16)
+        if frame_size <= 0:
+            return b''
+        # Update resampling ratio based on FIFO level
+        try:
+            self._update_ratio()
+        except Exception:
+            # keep going even if update failed
+            pass
+        # Process mono float32
+        mono = self._process(frame_size)
+        # Convert to stereo int16 LE bytes
+        return self._mono_to_stereo_int16_bytes(mono)
+
+    def _close(self) -> None:
+        try:
+            if self._stream is not None:
+                self._stream.stop()
+                self._stream.close()
+        except Exception:
+            logger.exception('Error closing input stream')
+        finally:
+            self._stream = None

examples/README.md

@@ -82,3 +82,7 @@ services and characteristics.
 
 # `run_scanner.py`
 Run a host application connected to a 'real' BLE controller over a UART HCI to a dev board running Zephyr in HCI mode (could be any other UART BLE controller, or BlueZ over a virtual UART), that starts scanning and prints out the scan results.
+
+
+# run auracast with usb stick
+bumble-auracast transmit 'serial:/dev/serial/by-id/usb-ZEPHYR_Zephyr_HCI_UART_sample_CC69A2912F84AE5E-if00,1000000,rtscts' --input device