kopia lustrzana https://github.com/pimoroni/pimoroni-pico
277 wiersze
11 KiB
Python
277 wiersze
11 KiB
Python
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# SPDX-FileCopyrightText: 2023 Christopher Parrott for Pimoroni Ltd
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#
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# SPDX-License-Identifier: MIT
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import os
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import math
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import struct
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from machine import I2S, Pin
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"""
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A class for playing Wav files out of an I2S audio amp. It can also play pure tones.
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This code is based heavily on the work of Mike Teachman, at:
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https://github.com/miketeachman/micropython-i2s-examples/blob/master/examples/wavplayer.py
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"""
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class WavPlayer:
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# Internal states
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PLAY = 0
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PAUSE = 1
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FLUSH = 2
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STOP = 3
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NONE = 4
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MODE_WAV = 0
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MODE_TONE = 1
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# Default buffer length
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SILENCE_BUFFER_LENGTH = 1000
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WAV_BUFFER_LENGTH = 10000
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INTERNAL_BUFFER_LENGTH = 20000
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TONE_SAMPLE_RATE = 44_100
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TONE_BITS_PER_SAMPLE = 16
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TONE_FULL_WAVES = 2
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def __init__(self, id, sck_pin, ws_pin, sd_pin, amp_enable=None, ibuf_len=INTERNAL_BUFFER_LENGTH, root="/"):
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self.__id = id
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self.__sck_pin = sck_pin
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self.__ws_pin = ws_pin
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self.__sd_pin = sd_pin
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self.__ibuf_len = ibuf_len
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self.__enable = None
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if amp_enable is not None:
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self.__enable = Pin(amp_enable, Pin.OUT)
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# Set the directory to search for files in
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self.set_root(root)
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self.__state = WavPlayer.NONE
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self.__mode = WavPlayer.MODE_WAV
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self.__wav_file = None
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self.__loop_wav = False
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self.__first_sample_offset = None
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self.__flush_count = 0
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self.__audio_out = None
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# Allocate a small array of blank audio samples used for silence
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self.__silence_samples = bytearray(self.SILENCE_BUFFER_LENGTH)
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# Allocate a larger array for WAV audio samples, using a memoryview for more efficient access
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self.__wav_samples_mv = memoryview(bytearray(self.WAV_BUFFER_LENGTH))
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# Reserve a variable for audio samples used for tones
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self.__tone_samples = None
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self.__queued_samples = None
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def set_root(self, root):
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self.__root = root.rstrip("/") + "/"
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def play_wav(self, wav_file, loop=False):
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if os.listdir(self.__root).count(wav_file) == 0:
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raise ValueError(f"'{wav_file}' not found")
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self.__stop_i2s() # Stop any active playback and terminate the I2S instance
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self.__wav_file = open(self.__root + wav_file, "rb") # Open the chosen WAV file in read-only, binary mode
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self.__loop_wav = loop # Record if the user wants the file to loop
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# Parse the WAV file, returning the necessary parameters to initialise I2S communication
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format, sample_rate, bits_per_sample, self.__first_sample_offset, self.sample_size = WavPlayer.__parse_wav(self.__wav_file)
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# Keep a track of total bytes read from WAV File
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self.total_bytes_read = 0
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self.__wav_file.seek(self.__first_sample_offset) # Advance to first byte of sample data
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self.__start_i2s(bits=bits_per_sample,
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format=format,
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rate=sample_rate,
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state=WavPlayer.PLAY,
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mode=WavPlayer.MODE_WAV)
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def play_tone(self, frequency, amplitude):
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if frequency < 20.0 or frequency > 20_000:
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raise ValueError("frequency out of range. Expected between 20Hz and 20KHz")
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if amplitude < 0.0 or amplitude > 1.0:
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raise ValueError("amplitude out of range. Expected 0.0 to 1.0")
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# Create a buffer containing the pure tone samples
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samples_per_cycle = self.TONE_SAMPLE_RATE // frequency
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sample_size_in_bytes = self.TONE_BITS_PER_SAMPLE // 8
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samples = bytearray(self.TONE_FULL_WAVES * samples_per_cycle * sample_size_in_bytes)
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range = pow(2, self.TONE_BITS_PER_SAMPLE) // 2
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format = "<h" if self.TONE_BITS_PER_SAMPLE == 16 else "<l"
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# Populate the buffer with multiple cycles to avoid it completing too quickly and causing drop outs
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for i in range(samples_per_cycle * self.TONE_FULL_WAVES):
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sample = int((range - 1) * (math.sin(2 * math.pi * i / samples_per_cycle)) * amplitude)
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struct.pack_into(format, samples, i * sample_size_in_bytes, sample)
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# Are we not already playing tones?
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if not (self.__mode == WavPlayer.MODE_TONE and (self.__state == WavPlayer.PLAY or self.__state == WavPlayer.PAUSE)):
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self.__stop_i2s() # Stop any active playback and terminate the I2S instance
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self.__tone_samples = samples
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self.__start_i2s(bits=self.TONE_BITS_PER_SAMPLE,
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format=I2S.MONO,
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rate=self.TONE_SAMPLE_RATE,
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state=WavPlayer.PLAY,
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mode=WavPlayer.MODE_TONE)
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else:
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self.__queued_samples = samples
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self.__state = WavPlayer.PLAY
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def pause(self):
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if self.__state == WavPlayer.PLAY:
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self.__state = WavPlayer.PAUSE # Enter the pause state on the next callback
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def resume(self):
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if self.__state == WavPlayer.PAUSE:
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self.__state = WavPlayer.PLAY # Enter the play state on the next callback
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def stop(self):
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if self.__state == WavPlayer.PLAY or self.__state == WavPlayer.PAUSE:
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if self.__mode == WavPlayer.MODE_WAV:
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# Enter the flush state on the next callback and close the file
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# It is done in this order to prevent the callback entering the play
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# state after we close the file but before we change the state)
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self.__state = WavPlayer.FLUSH
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self.__wav_file.close()
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else:
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self.__state = WavPlayer.STOP
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def is_playing(self):
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return self.__state != WavPlayer.NONE and self.__state != WavPlayer.STOP
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def is_paused(self):
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return self.__state == WavPlayer.PAUSE
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def __start_i2s(self, bits=16, format=I2S.MONO, rate=44_100, state=STOP, mode=MODE_WAV):
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import gc
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gc.collect()
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self.__audio_out = I2S(
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self.__id,
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sck=self.__sck_pin,
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ws=self.__ws_pin,
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sd=self.__sd_pin,
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mode=I2S.TX,
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bits=bits,
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format=format,
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rate=rate,
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ibuf=self.__ibuf_len,
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)
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self.__state = state
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self.__mode = mode
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self.__flush_count = self.__ibuf_len // self.SILENCE_BUFFER_LENGTH + 1
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self.__audio_out.irq(self.__i2s_callback)
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self.__audio_out.write(self.__silence_samples)
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if self.__enable is not None:
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self.__enable.on()
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def __stop_i2s(self):
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self.stop() # Stop any active playback
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while self.is_playing(): # and wait for it to complete
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pass
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if self.__enable is not None:
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self.__enable.off()
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if self.__audio_out is not None:
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self.__audio_out.deinit() # Deinit any active I2S comms
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self.__state == WavPlayer.NONE # Return to the none state
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def __i2s_callback(self, arg):
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# PLAY
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if self.__state == WavPlayer.PLAY:
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if self.__mode == WavPlayer.MODE_WAV:
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num_read = self.__wav_file.readinto(self.__wav_samples_mv) # Read the next section of the WAV file
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self.total_bytes_read += num_read
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# Have we reached the end of the file?
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if num_read == 0:
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# Do we want to loop the WAV playback?
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if self.__loop_wav:
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_ = self.__wav_file.seek(self.__first_sample_offset) # Play again, so advance to first byte of sample data
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else:
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self.__wav_file.close() # Stop playing, so close the file
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self.__state = WavPlayer.FLUSH # and enter the flush state on the next callback
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self.__audio_out.write(self.__silence_samples) # In both cases play silence to end this callback
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else:
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if num_read > 0 and num_read < self.WAV_BUFFER_LENGTH:
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num_read = num_read - (self.total_bytes_read - self.sample_size)
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self.__audio_out.write(self.__wav_samples_mv[: num_read]) # We are within the file, so write out the next audio samples
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else:
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if self.__queued_samples is not None:
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self.__tone_samples = self.__queued_samples
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self.__queued_samples = None
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self.__audio_out.write(self.__tone_samples)
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# PAUSE or STOP
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elif self.__state == WavPlayer.PAUSE or self.__state == WavPlayer.STOP:
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self.__audio_out.write(self.__silence_samples) # Play silence
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# FLUSH
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elif self.__state == WavPlayer.FLUSH:
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# Flush is used to allow the residual audio samples in the internal buffer to be written
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# to the I2S peripheral. This step avoids part of the sound file from being cut off
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if self.__flush_count > 0:
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self.__flush_count -= 1
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else:
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self.__state = WavPlayer.STOP # Enter the stop state on the next callback
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self.__audio_out.write(self.__silence_samples) # Play silence
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# NONE
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elif self.__state == WavPlayer.NONE:
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pass
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@staticmethod
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def __parse_wav(wav_file):
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chunk_ID = wav_file.read(4)
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if chunk_ID != b"RIFF":
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raise ValueError("WAV chunk ID invalid")
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_ = wav_file.read(4) # chunk_size
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format = wav_file.read(4)
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if format != b"WAVE":
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raise ValueError("WAV format invalid")
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sub_chunk1_ID = wav_file.read(4)
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if sub_chunk1_ID != b"fmt ":
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raise ValueError("WAV sub chunk 1 ID invalid")
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_ = wav_file.read(4) # sub_chunk1_size
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_ = struct.unpack("<H", wav_file.read(2))[0] # audio_format
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num_channels = struct.unpack("<H", wav_file.read(2))[0]
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if num_channels == 1:
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format = I2S.MONO
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else:
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format = I2S.STEREO
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sample_rate = struct.unpack("<I", wav_file.read(4))[0]
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# if sample_rate != 44_100 and sample_rate != 48_000:
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# raise ValueError(f"WAV sample rate of {sample_rate} invalid. Only 44.1KHz or 48KHz audio are supported")
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_ = struct.unpack("<I", wav_file.read(4))[0] # byte_rate
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_ = struct.unpack("<H", wav_file.read(2))[0] # block_align
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bits_per_sample = struct.unpack("<H", wav_file.read(2))[0]
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# usually the sub chunk2 ID ("data") comes next, but
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# some online MP3->WAV converters add
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# binary data before "data". So, read a fairly large
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# block of bytes and search for "data".
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binary_block = wav_file.read(200)
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offset = binary_block.find(b"data")
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if offset == -1:
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raise ValueError("WAV sub chunk 2 ID not found")
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wav_file.seek(40)
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sub_chunk2_size = struct.unpack("<I", wav_file.read(4))[0]
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return (format, sample_rate, bits_per_sample, 44 + offset, sub_chunk2_size)
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