diff --git a/lib/pimoroni_yukon/devices/audio.py b/lib/pimoroni_yukon/devices/audio.py
index c547c28..f47c5d4 100644
--- a/lib/pimoroni_yukon/devices/audio.py
+++ b/lib/pimoroni_yukon/devices/audio.py
@@ -1,11 +1,10 @@
-# SPDX-FileCopyrightText: 2023 Christopher Parrott for Pimoroni Ltd
+# SPDX-FileCopyrightText: 2024 Christopher Parrott for Pimoroni Ltd
 #
 # SPDX-License-Identifier: MIT
 
-import os
 import math
 import struct
-from machine import I2S
+from machine import I2S, Pin
 
 """
 A class for playing Wav files out of an I2S audio amp. It can also play pure tones.
@@ -14,6 +13,70 @@
 """
 
 
+class WavReader:
+    def __init__(self, file):
+        self.wav_file = open(file, "rb")
+        self._parse(self.wav_file)
+
+    def _parse(self, wav_file):
+        chunk_ID = wav_file.read(4)
+        if chunk_ID != b"RIFF":
+            raise ValueError("WAV chunk ID invalid")
+        _ = wav_file.read(4)                            # chunk_size
+        if wav_file.read(4) != b"WAVE":
+            raise ValueError("WAV format invalid")
+        sub_chunk1_ID = wav_file.read(4)
+        if sub_chunk1_ID != b"fmt ":
+            raise ValueError("WAV sub chunk 1 ID invalid")
+        _ = wav_file.read(4)                            # sub_chunk1_size
+        _ = struct.unpack("<H", wav_file.read(2))[0]    # audio_format
+        num_channels = struct.unpack("<H", wav_file.read(2))[0]
+
+        if num_channels == 1:
+            self.format = I2S.MONO
+        else:
+            self.format = I2S.STEREO
+
+        self.sample_rate = struct.unpack("<I", wav_file.read(4))[0]
+        # if sample_rate != 44_100 and sample_rate != 48_000:
+        #    raise ValueError(f"WAV sample rate of {sample_rate} invalid. Only 44.1KHz or 48KHz audio are supported")
+
+        _ = struct.unpack("<I", wav_file.read(4))[0]    # byte_rate
+        _ = struct.unpack("<H", wav_file.read(2))[0]    # block_align
+        self.bits_per_sample = struct.unpack("<H", wav_file.read(2))[0]
+
+        # usually the sub chunk2 ID ("data") comes next, but
+        # some online MP3->WAV converters add
+        # binary data before "data".  So, read a fairly large
+        # block of bytes and search for "data".
+
+        binary_block = wav_file.read(200)
+        offset = binary_block.find(b"data")
+        if offset == -1:
+            raise ValueError("WAV sub chunk 2 ID not found")
+
+        self.offset = offset + 44
+
+        wav_file.seek(offset + 40)
+        self.size = struct.unpack("<I", wav_file.read(4))[0]
+
+        wav_file.seek(self.offset)
+
+    def seek(self, pos):
+        return self.wav_file.seek(pos + self.offset)
+
+    def tell(self):
+        return self.wav_file.tell() - self.offset
+
+    def readinto(self, buf):
+        max_bytes = self.size - self.tell()
+        max_bytes = max(0, min(len(buf), max_bytes))
+        return self.wav_file.readinto(buf[:max_bytes])
+
+    def close(self):
+        self.wav_file.close()
+
+
 class WavPlayer:
     # Internal states
     PLAY = 0
@@ -25,21 +88,32 @@ class WavPlayer:
     MODE_WAV = 0
     MODE_TONE = 1
 
+    TONE_SINE = 0
+    TONE_SQUARE = 1
+    TONE_TRIANGLE = 2
+
     # Default buffer length
-    SILENCE_BUFFER_LENGTH = 1000
-    WAV_BUFFER_LENGTH = 10000
-    INTERNAL_BUFFER_LENGTH = 20000
+    SILENCE_BUFFER_LENGTH = 1024
+    WAV_BUFFER_LENGTH = 1024
+    INTERNAL_BUFFER_LENGTH = WAV_BUFFER_LENGTH * 2
 
     TONE_SAMPLE_RATE = 44_100
     TONE_BITS_PER_SAMPLE = 16
     TONE_FULL_WAVES = 2
 
-    def __init__(self, id, sck_pin, ws_pin, sd_pin, ibuf_len=INTERNAL_BUFFER_LENGTH, root="/"):
+    def __init__(self, id, sck_pin, ws_pin, sd_pin, amp_enable=None, ibuf_len=INTERNAL_BUFFER_LENGTH, root="/"):
         self.__id = id
         self.__sck_pin = sck_pin
         self.__ws_pin = ws_pin
         self.__sd_pin = sd_pin
         self.__ibuf_len = ibuf_len
+        self.__enable = None
+
+        # Manually tweak the tone amplitude for equal loudness of sine/square/triangle
+        self.__amplitude_scale = [1.0, 0.2, 0.5]
+
+        if amp_enable is not None:
+            self.__enable = Pin(amp_enable, Pin.OUT)
 
         # Set the directory to search for files in
         self.set_root(root)
@@ -48,7 +122,6 @@ def __init__(self, id, sck_pin, ws_pin, sd_pin, ibuf_len=INTERNAL_BUFFER_LENGTH,
         self.__mode = WavPlayer.MODE_WAV
         self.__wav_file = None
         self.__loop_wav = False
-        self.__first_sample_offset = None
         self.__flush_count = 0
         self.__audio_out = None
 
@@ -66,44 +139,60 @@ def set_root(self, root):
         self.__root = root.rstrip("/") + "/"
 
     def play_wav(self, wav_file, loop=False):
-        if os.listdir(self.__root).count(wav_file) == 0:
-            raise ValueError(f"'{wav_file}' not found")
-
         self.__stop_i2s()                                       # Stop any active playback and terminate the I2S instance
 
-        self.__wav_file = open(self.__root + wav_file, "rb")    # Open the chosen WAV file in read-only, binary mode
+        try:
+            self.__wav_file = open(self.__root + wav_file, "rb")    # Open the chosen WAV file in read-only, binary mode
+        except OSError:
+            raise ValueError(f"'{wav_file}' not found")
         self.__loop_wav = loop                                  # Record if the user wants the file to loop
+        self.__loop_count = 0                                    # Count loops for debugging purposes
 
         # Parse the WAV file, returning the necessary parameters to initialise I2S communication
-        format, sample_rate, bits_per_sample, self.__first_sample_offset = WavPlayer.__parse_wav(self.__wav_file)
+        self.__wav_file = WavReader(self.__root + wav_file)
 
-        self.__wav_file.seek(self.__first_sample_offset)        # Advance to first byte of sample data
-
-        self.__start_i2s(bits=bits_per_sample,
-                         format=format,
-                         rate=sample_rate,
+        self.__start_i2s(bits=self.__wav_file.bits_per_sample,
+                         format=self.__wav_file.format,
+                         rate=self.__wav_file.sample_rate,
                          state=WavPlayer.PLAY,
                          mode=WavPlayer.MODE_WAV)
 
-    def play_tone(self, frequency, amplitude):
+    def play_tone(self, frequency, amplitude, shape=TONE_SINE):
         if frequency < 20.0 or frequency > 20_000:
             raise ValueError("frequency out of range. Expected between 20Hz and 20KHz")
 
         if amplitude < 0.0 or amplitude > 1.0:
             raise ValueError("amplitude out of range. Expected 0.0 to 1.0")
 
+        if not isinstance(shape, (list, tuple)):
+            shape = (shape, )
+
         # Create a buffer containing the pure tone samples
         samples_per_cycle = self.TONE_SAMPLE_RATE // frequency
         sample_size_in_bytes = self.TONE_BITS_PER_SAMPLE // 8
         samples = bytearray(self.TONE_FULL_WAVES * samples_per_cycle * sample_size_in_bytes)
-        range = pow(2, self.TONE_BITS_PER_SAMPLE) // 2
+        maximum = (pow(2, self.TONE_BITS_PER_SAMPLE) // 2 - 1) * amplitude
 
         format = "<h" if self.TONE_BITS_PER_SAMPLE == 16 else "<l"
 
         # Populate the buffer with multiple cycles to avoid it completing too quickly and causing drop outs
         for i in range(samples_per_cycle * self.TONE_FULL_WAVES):
-            sample = int((range - 1) * (math.sin(2 * math.pi * i / samples_per_cycle)) * amplitude)
-            struct.pack_into(format, samples, i * sample_size_in_bytes, sample)
+            sample = 0
+            if self.TONE_TRIANGLE in shape:
+                triangle = (i % samples_per_cycle) - (samples_per_cycle // 2)
+                triangle /= samples_per_cycle
+                triangle *= self.__amplitude_scale[self.TONE_TRIANGLE]
+                sample += triangle
+            if self.TONE_SINE in shape:
+                sine = math.sin(2 * math.pi * i / samples_per_cycle)
+                sine *= self.__amplitude_scale[self.TONE_SINE]
+                sample += sine
+            if self.TONE_SQUARE in shape:
+                square = 1 if (i % samples_per_cycle) < (samples_per_cycle // 2) else -1
+                square *= self.__amplitude_scale[self.TONE_SQUARE]
+                sample += square
+            sample = max(-1, min(1, sample))
+            struct.pack_into(format, samples, i * sample_size_in_bytes, int(sample * maximum))
 
         # Are we not already playing tones?
         if not (self.__mode == WavPlayer.MODE_TONE and (self.__state == WavPlayer.PLAY or self.__state == WavPlayer.PAUSE)):
@@ -137,6 +226,9 @@ def stop(self):
             else:
                 self.__state = WavPlayer.STOP
 
+    def deinit(self):
+        self.__stop_i2s()
+
     def is_playing(self):
         return self.__state != WavPlayer.NONE and self.__state != WavPlayer.STOP
 
@@ -164,45 +256,61 @@ def __start_i2s(self, bits=16, format=I2S.MONO, rate=44_100, state=STOP, mode=MO
         self.__audio_out.irq(self.__i2s_callback)
         self.__audio_out.write(self.__silence_samples)
 
+        if self.__enable is not None:
+            self.__enable.on()
+
     def __stop_i2s(self):
         self.stop()                     # Stop any active playback
         while self.is_playing():        # and wait for it to complete
             pass
 
+        if self.__enable is not None:
+            self.__enable.off()
+
         if self.__audio_out is not None:
             self.__audio_out.deinit()   # Deinit any active I2S comms
 
         self.__state == WavPlayer.NONE  # Return to the none state
 
     def __i2s_callback(self, arg):
-        ### PLAY ###
+        # PLAY
         if self.__state == WavPlayer.PLAY:
             if self.__mode == WavPlayer.MODE_WAV:
-                num_read = self.__wav_file.readinto(self.__wav_samples_mv)      # Read the next section of the WAV file
-
-                # Have we reached the end of the file?
-                if num_read == 0:
-                    # Do we want to loop the WAV playback?
-                    if self.__loop_wav:
-                        _ = self.__wav_file.seek(self.__first_sample_offset)    # Play again, so advance to first byte of sample data
+                if self.__loop_wav:  # Looped playback
+                    loop_read = 0
+                    while loop_read < self.WAV_BUFFER_LENGTH:
+                        num_read = self.__wav_file.readinto(self.__wav_samples_mv[loop_read:])      # Read the next section of the WAV file
+                        loop_read += num_read
+                        if num_read == 0:
+                            _ = self.__wav_file.seek(0)    # Play again, so advance to first byte of sample data
+                            self.__loop_count += 1
+                    self.__audio_out.write(self.__wav_samples_mv)
+
+                else:  # Single shot playback
+                    num_read = self.__wav_file.readinto(self.__wav_samples_mv)
+
+                    if num_read:
+                        self.__audio_out.write(self.__wav_samples_mv[: num_read])   # We are within the file, so write out the next audio samples
                     else:
+                        self.__audio_out.write(self.__silence_samples)              # Play silence to end this callback
+
+                    # Have we reached the end of the file? (num_read is either 0 or a short read)
+                    if num_read < self.WAV_BUFFER_LENGTH:
+                        # Do we want to loop the WAV playback?
                         self.__wav_file.close()                                 # Stop playing, so close the file
                         self.__state = WavPlayer.FLUSH                          # and enter the flush state on the next callback
 
-                    self.__audio_out.write(self.__silence_samples)              # In both cases play silence to end this callback
-                else:
-                    self.__audio_out.write(self.__wav_samples_mv[: num_read])   # We are within the file, so write out the next audio samples
             else:
                 if self.__queued_samples is not None:
                     self.__tone_samples = self.__queued_samples
                     self.__queued_samples = None
                 self.__audio_out.write(self.__tone_samples)
 
-        ### PAUSE or STOP ###
+        # PAUSE or STOP
         elif self.__state == WavPlayer.PAUSE or self.__state == WavPlayer.STOP:
             self.__audio_out.write(self.__silence_samples)                  # Play silence
 
-        ### FLUSH ###
+        # FLUSH
         elif self.__state == WavPlayer.FLUSH:
             # Flush is used to allow the residual audio samples in the internal buffer to be written
             # to the I2S peripheral. This step avoids part of the sound file from being cut off
@@ -212,47 +320,6 @@ def __i2s_callback(self, arg):
                 self.__state = WavPlayer.STOP                               # Enter the stop state on the next callback
             self.__audio_out.write(self.__silence_samples)                  # Play silence
 
-        ### NONE ###
+        # NONE
         elif self.__state == WavPlayer.NONE:
             pass
-
-    @staticmethod
-    def __parse_wav(wav_file):
-        chunk_ID = wav_file.read(4)
-        if chunk_ID != b"RIFF":
-            raise ValueError("WAV chunk ID invalid")
-        _ = wav_file.read(4)                            # chunk_size
-        format = wav_file.read(4)
-        if format != b"WAVE":
-            raise ValueError("WAV format invalid")
-        sub_chunk1_ID = wav_file.read(4)
-        if sub_chunk1_ID != b"fmt ":
-            raise ValueError("WAV sub chunk 1 ID invalid")
-        _ = wav_file.read(4)                            # sub_chunk1_size
-        _ = struct.unpack("<H", wav_file.read(2))[0]    # audio_format
-        num_channels = struct.unpack("<H", wav_file.read(2))[0]
-
-        if num_channels == 1:
-            format = I2S.MONO
-        else:
-            format = I2S.STEREO
-
-        sample_rate = struct.unpack("<I", wav_file.read(4))[0]
-        if sample_rate != 44_100 and sample_rate != 48_000:
-            raise ValueError(f"WAV sample rate of {sample_rate} invalid. Only 44.1KHz or 48KHz audio are supported")
-
-        _ = struct.unpack("<I", wav_file.read(4))[0]    # byte_rate
-        _ = struct.unpack("<H", wav_file.read(2))[0]    # block_align
-        bits_per_sample = struct.unpack("<H", wav_file.read(2))[0]
-
-        # usually the sub chunk2 ID ("data") comes next, but
-        # some online MP3->WAV converters add
-        # binary data before "data".  So, read a fairly large
-        # block of bytes and search for "data".
-
-        binary_block = wav_file.read(200)
-        offset = binary_block.find(b"data")
-        if offset == -1:
-            raise ValueError("WAV sub chunk 2 ID not found")
-
-        return (format, sample_rate, bits_per_sample, 44 + offset)