Merge pull request #9421 from timchinowsky/rawsample-doublebuffer

Add double buffering to RawSample

locale/circuitpython.pot

@@ -1177,6 +1177,7 @@ msgid "Interrupted by output function"
 msgstr ""
 
 #: ports/espressif/common-hal/espulp/ULP.c
+#: ports/espressif/common-hal/microcontroller/Processor.c
 #: ports/mimxrt10xx/common-hal/audiobusio/__init__.c
 #: ports/mimxrt10xx/common-hal/pwmio/PWMOut.c
 #: ports/raspberrypi/bindings/picodvi/Framebuffer.c
@@ -1284,6 +1285,10 @@ msgstr ""
 msgid "Layer must be a Group or TileGrid subclass"
 msgstr ""
 
+#: shared-bindings/audiocore/RawSample.c
+msgid "Length of %q must be an even multiple of channel_count * type_size"
+msgstr ""
+
 #: ports/espressif/common-hal/espidf/__init__.c
 msgid "MAC address was invalid"
 msgstr ""

shared-bindings/audiocore/RawSample.c

@@ -17,7 +17,12 @@
 //|     """A raw audio sample buffer in memory"""
 //|
 //|     def __init__(
-//|         self, buffer: ReadableBuffer, *, channel_count: int = 1, sample_rate: int = 8000
+//|         self,
+//|         buffer: ReadableBuffer,
+//|         *,
+//|         channel_count: int = 1,
+//|         sample_rate: int = 8000,
+//|         single_buffer: bool = True
 //|     ) -> None:
 //|         """Create a RawSample based on the given buffer of values. If channel_count is more than
 //|         1 then each channel's samples should alternate. In other words, for a two channel buffer, the
@@ -27,34 +32,58 @@
 //|         :param ~circuitpython_typing.ReadableBuffer buffer: A buffer with samples
 //|         :param int channel_count: The number of channels in the buffer
 //|         :param int sample_rate: The desired playback sample rate
+//|         :param bool single_buffer: Selects single buffered or double buffered transfer mode.  This affects
+//|                                    what happens if the sample buffer is changed while the sample is playing.
+//|                                    In single buffered transfers, a change in buffer contents will not affect active playback.
+//|                                    In double buffered transfers, changed buffer contents will
+//|                                    be played back when the transfer reaches the next half-buffer point.
 //|
-//|         Simple 8ksps 440 Hz sin wave::
+//|         Playing 8ksps 440 Hz and 880 Hz sine waves::
 //|
+//|           import analogbufio
+//|           import array
 //|           import audiocore
-//|           import audioio
+//|           import audiopwmio
 //|           import board
-//|           import array
-//|           import time
 //|           import math
+//|           import time
 //|
-//|           # Generate one period of sine wav.
+//|           # Generate one period of sine wave.
 //|           length = 8000 // 440
 //|           sine_wave = array.array("h", [0] * length)
 //|           for i in range(length):
 //|               sine_wave[i] = int(math.sin(math.pi * 2 * i / length) * (2 ** 15))
+//|           pwm = audiopwmio.PWMAudioOut(left_channel=board.D12, right_channel=board.D13)
 //|
-//|           dac = audioio.AudioOut(board.SPEAKER)
-//|           sine_wave = audiocore.RawSample(sine_wave)
-//|           dac.play(sine_wave, loop=True)
+//|           # Play single-buffered
+//|           sample = audiocore.RawSample(sine_wave)
+//|           pwm.play(sample, loop=True)
+//|           time.sleep(3)
+//|           # changing the wave has no effect
+//|           for i in range(length):
+//|                sine_wave[i] = int(math.sin(math.pi * 4 * i / length) * (2 ** 15))
+//|           time.sleep(3)
+//|           pwm.stop()
 //|           time.sleep(1)
-//|           dac.stop()"""
+//|
+//|           # Play double-buffered
+//|           sample = audiocore.RawSample(sine_wave, single_buffer=False)
+//|           pwm.play(sample, loop=True)
+//|           time.sleep(3)
+//|           # changing the wave takes effect almost immediately
+//|           for i in range(length):
+//|               sine_wave[i] = int(math.sin(math.pi * 2 * i / length) * (2 ** 15))
+//|           time.sleep(3)
+//|           pwm.stop()
+//|           pwm.deinit()"""
 //|         ...
 static mp_obj_t audioio_rawsample_make_new(const mp_obj_type_t *type, size_t n_args, size_t n_kw, const mp_obj_t *all_args) {
-    enum { ARG_buffer, ARG_channel_count, ARG_sample_rate };
+    enum { ARG_buffer, ARG_channel_count, ARG_sample_rate, ARG_single_buffer };
     static const mp_arg_t allowed_args[] = {
         { MP_QSTR_buffer, MP_ARG_OBJ | MP_ARG_REQUIRED, {.u_obj = MP_OBJ_NULL } },
         { MP_QSTR_channel_count, MP_ARG_INT | MP_ARG_KW_ONLY, {.u_int = 1 } },
         { MP_QSTR_sample_rate, MP_ARG_INT | MP_ARG_KW_ONLY, {.u_int = 8000} },
+        { MP_QSTR_single_buffer, MP_ARG_BOOL | MP_ARG_KW_ONLY, {.u_bool = true} },
     };
     mp_arg_val_t args[MP_ARRAY_SIZE(allowed_args)];
     mp_arg_parse_all_kw_array(n_args, n_kw, all_args, MP_ARRAY_SIZE(allowed_args), allowed_args, args);
@@ -69,9 +98,12 @@ static mp_obj_t audioio_rawsample_make_new(const mp_obj_type_t *type, size_t n_a
     } else if (bufinfo.typecode != 'b' && bufinfo.typecode != 'B' && bufinfo.typecode != BYTEARRAY_TYPECODE) {
         mp_raise_ValueError_varg(MP_ERROR_TEXT("%q must be a bytearray or array of type 'h', 'H', 'b', or 'B'"), MP_QSTR_buffer);
     }
+    if (!args[ARG_single_buffer].u_bool && bufinfo.len % (bytes_per_sample * args[ARG_channel_count].u_int * 2) != 0) {
+        mp_raise_ValueError_varg(MP_ERROR_TEXT("Length of %q must be an even multiple of channel_count * type_size"), MP_QSTR_buffer);
+    }
     common_hal_audioio_rawsample_construct(self, ((uint8_t *)bufinfo.buf), bufinfo.len,
         bytes_per_sample, signed_samples, args[ARG_channel_count].u_int,
-        args[ARG_sample_rate].u_int);
+        args[ARG_sample_rate].u_int, args[ARG_single_buffer].u_bool);
 
     return MP_OBJ_FROM_PTR(self);
 }

shared-bindings/audiocore/RawSample.h

@@ -12,7 +12,7 @@ extern const mp_obj_type_t audioio_rawsample_type;
 
 void common_hal_audioio_rawsample_construct(audioio_rawsample_obj_t *self,
     uint8_t *buffer, uint32_t len, uint8_t bytes_per_sample, bool samples_signed,
-    uint8_t channel_count, uint32_t sample_rate);
+    uint8_t channel_count, uint32_t sample_rate, bool single_buffer);
 
 void common_hal_audioio_rawsample_deinit(audioio_rawsample_obj_t *self);
 bool common_hal_audioio_rawsample_deinited(audioio_rawsample_obj_t *self);

shared-module/audiocore/RawSample.c

@@ -2,6 +2,8 @@
 //
 // SPDX-FileCopyrightText: Copyright (c) 2018 Scott Shawcroft for Adafruit Industries
 //
+// SPDX-FileCopyrightText: Copyright (c) 2024 Tim Chinowsky
+//
 // SPDX-License-Identifier: MIT
 
 #include "shared-bindings/audiocore/RawSample.h"
@@ -16,13 +18,17 @@ void common_hal_audioio_rawsample_construct(audioio_rawsample_obj_t *self,
     uint8_t bytes_per_sample,
     bool samples_signed,
     uint8_t channel_count,
-    uint32_t sample_rate) {
+    uint32_t sample_rate,
+    bool single_buffer) {
+
     self->buffer = buffer;
     self->bits_per_sample = bytes_per_sample * 8;
     self->samples_signed = samples_signed;
     self->len = len;
     self->channel_count = channel_count;
     self->sample_rate = sample_rate;
+    self->single_buffer = single_buffer;
+    self->buffer_index = 0;
 }
 
 void common_hal_audioio_rawsample_deinit(audioio_rawsample_obj_t *self) {
@@ -56,19 +62,33 @@ audioio_get_buffer_result_t audioio_rawsample_get_buffer(audioio_rawsample_obj_t
     uint8_t channel,
     uint8_t **buffer,
     uint32_t *buffer_length) {
-    *buffer_length = self->len;
-    if (single_channel_output) {
-        *buffer = self->buffer + (channel % self->channel_count) * (self->bits_per_sample / 8);
+
+    if (self->single_buffer) {
+        *buffer_length = self->len;
+        if (single_channel_output) {
+            *buffer = self->buffer + (channel % self->channel_count) * (self->bits_per_sample / 8);
+        } else {
+            *buffer = self->buffer;
+        }
+        return GET_BUFFER_DONE;
     } else {
-        *buffer = self->buffer;
+        *buffer_length = self->len / 2;
+        if (single_channel_output) {
+            *buffer = self->buffer + (channel % self->channel_count) * (self->bits_per_sample / 8) + \
+                self->len / 2 * self->buffer_index;
+        } else {
+            *buffer = self->buffer + self->len / 2 * self->buffer_index;
+        }
+        self->buffer_index = 1 - self->buffer_index;
+        return GET_BUFFER_DONE;
     }
-    return GET_BUFFER_DONE;
 }
 
 void audioio_rawsample_get_buffer_structure(audioio_rawsample_obj_t *self, bool single_channel_output,
     bool *single_buffer, bool *samples_signed,
     uint32_t *max_buffer_length, uint8_t *spacing) {
-    *single_buffer = true;
+
+    *single_buffer = self->single_buffer;
     *samples_signed = self->samples_signed;
     *max_buffer_length = self->len;
     if (single_channel_output) {

shared-module/audiocore/RawSample.h

@@ -18,6 +18,8 @@ typedef struct {
     bool samples_signed;
     uint8_t channel_count;
     uint32_t sample_rate;
+    bool single_buffer;
+    uint8_t buffer_index;
 } audioio_rawsample_obj_t;
 
 

tests/circuitpython-manual/live_audio/mix.py

@@ -0,0 +1,67 @@
+import audiocore
+import audiopwmio
+import audiomixer
+import board
+import array
+import time
+import math
+
+CHANNELS = 2
+RATE = 8000
+SAMPLE_TYPE = "H"
+OFFSET = 2**15 - 1
+BUFFER_SIZE = 640
+SINGLE_BUFFER = True
+LOOP = True
+
+# (frequency, amp_left, amp_right)
+VOICES = ((200, 1, 0), (400, 0, 1), (100, 1, 1))
+
+
+def play(
+    voices=VOICES,
+    channels=CHANNELS,
+    rate=RATE,
+    sample_type=SAMPLE_TYPE,
+    offset=OFFSET,
+    buffer_size=BUFFER_SIZE,
+    single_buffer=SINGLE_BUFFER,
+    loop=LOOP,
+):
+    waves = []
+    samples = []
+    for v in voices:
+        print(v)
+        sample_length = int(rate // v[0])
+        wave = array.array(sample_type, [offset] * sample_length * channels)
+        for i in range(0, sample_length):
+            if channels == 1:
+                wave[i] = int(
+                    math.sin(math.pi * 2 * i / sample_length) * v[1] * (2**15 - 1) + offset
+                )
+            else:
+                wave[2 * i] = int(
+                    math.sin(math.pi * 2 * i / sample_length) * v[1] * (2**15 - 1) + offset
+                )
+                wave[2 * i + 1] = int(
+                    math.sin(math.pi * 2 * i / sample_length) * v[2] * (2**15 - 1) + offset
+                )
+        waves.append(wave)
+        samples.append(
+            audiocore.RawSample(
+                wave, sample_rate=rate, channel_count=channels, single_buffer=single_buffer
+            )
+        )
+    mixer = audiomixer.Mixer(
+        voice_count=len(voices),
+        sample_rate=rate,
+        channel_count=channels,
+        bits_per_sample=16,
+        samples_signed=False,
+        buffer_size=buffer_size,
+    )
+    pwm = audiopwmio.PWMAudioOut(left_channel=board.D12, right_channel=board.D13)
+    pwm.play(mixer)
+    for i in range(len(samples)):
+        mixer.voice[i].play(samples[i], loop=loop)
+        mixer.voice[i].level = 0.5