remove custom_speech_recognition
This commit is contained in:
misyaguziya
File diff suppressed because it is too large
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@@ -1,24 +0,0 @@
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import custom_speech_recognition as sr
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r = sr.Recognizer()
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m = sr.Microphone()
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try:
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print("A moment of silence, please...")
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with m as source: r.adjust_for_ambient_noise(source)
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print("Set minimum energy threshold to {}".format(r.energy_threshold))
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while True:
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print("Say something!")
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with m as source: audio = r.listen(source)
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print("Got it! Now to recognize it...")
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try:
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# recognize speech using Google Speech Recognition
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value = r.recognize_google(audio)
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print("You said {}".format(value))
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except sr.UnknownValueError:
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print("Oops! Didn't catch that")
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except sr.RequestError as e:
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print("Uh oh! Couldn't request results from Google Speech Recognition service; {0}".format(e))
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except KeyboardInterrupt:
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pass
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@@ -1,317 +0,0 @@
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import aifc
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import audioop
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import io
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import os
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import platform
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import stat
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import subprocess
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import sys
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import wave
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class AudioData(object):
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"""
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Creates a new ``AudioData`` instance, which represents mono audio data.
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The raw audio data is specified by ``frame_data``, which is a sequence of bytes representing audio samples. This is the frame data structure used by the PCM WAV format.
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The width of each sample, in bytes, is specified by ``sample_width``. Each group of ``sample_width`` bytes represents a single audio sample.
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The audio data is assumed to have a sample rate of ``sample_rate`` samples per second (Hertz).
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Usually, instances of this class are obtained from ``recognizer_instance.record`` or ``recognizer_instance.listen``, or in the callback for ``recognizer_instance.listen_in_background``, rather than instantiating them directly.
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"""
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def __init__(self, frame_data, sample_rate, sample_width):
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assert sample_rate > 0, "Sample rate must be a positive integer"
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assert (
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sample_width % 1 == 0 and 1 <= sample_width <= 4
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), "Sample width must be between 1 and 4 inclusive"
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self.frame_data = frame_data
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self.sample_rate = sample_rate
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self.sample_width = int(sample_width)
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def get_segment(self, start_ms=None, end_ms=None):
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"""
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Returns a new ``AudioData`` instance, trimmed to a given time interval. In other words, an ``AudioData`` instance with the same audio data except starting at ``start_ms`` milliseconds in and ending ``end_ms`` milliseconds in.
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If not specified, ``start_ms`` defaults to the beginning of the audio, and ``end_ms`` defaults to the end.
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"""
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assert (
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start_ms is None or start_ms >= 0
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), "``start_ms`` must be a non-negative number"
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assert end_ms is None or end_ms >= (
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0 if start_ms is None else start_ms
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), "``end_ms`` must be a non-negative number greater or equal to ``start_ms``"
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if start_ms is None:
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start_byte = 0
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else:
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start_byte = int(
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(start_ms * self.sample_rate * self.sample_width) // 1000
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)
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if end_ms is None:
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end_byte = len(self.frame_data)
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else:
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end_byte = int(
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(end_ms * self.sample_rate * self.sample_width) // 1000
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)
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return AudioData(
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self.frame_data[start_byte:end_byte],
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self.sample_rate,
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self.sample_width,
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)
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def get_raw_data(self, convert_rate=None, convert_width=None):
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"""
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Returns a byte string representing the raw frame data for the audio represented by the ``AudioData`` instance.
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If ``convert_rate`` is specified and the audio sample rate is not ``convert_rate`` Hz, the resulting audio is resampled to match.
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If ``convert_width`` is specified and the audio samples are not ``convert_width`` bytes each, the resulting audio is converted to match.
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Writing these bytes directly to a file results in a valid `RAW/PCM audio file <https://en.wikipedia.org/wiki/Raw_audio_format>`__.
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"""
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assert (
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convert_rate is None or convert_rate > 0
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), "Sample rate to convert to must be a positive integer"
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assert convert_width is None or (
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convert_width % 1 == 0 and 1 <= convert_width <= 4
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), "Sample width to convert to must be between 1 and 4 inclusive"
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raw_data = self.frame_data
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# make sure unsigned 8-bit audio (which uses unsigned samples) is handled like higher sample width audio (which uses signed samples)
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if self.sample_width == 1:
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raw_data = audioop.bias(
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raw_data, 1, -128
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) # subtract 128 from every sample to make them act like signed samples
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# resample audio at the desired rate if specified
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if convert_rate is not None and self.sample_rate != convert_rate:
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raw_data, _ = audioop.ratecv(
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raw_data,
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self.sample_width,
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1,
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self.sample_rate,
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convert_rate,
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None,
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)
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# convert samples to desired sample width if specified
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if convert_width is not None and self.sample_width != convert_width:
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if (
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convert_width == 3
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): # we're converting the audio into 24-bit (workaround for https://bugs.python.org/issue12866)
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raw_data = audioop.lin2lin(
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raw_data, self.sample_width, 4
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) # convert audio into 32-bit first, which is always supported
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try:
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audioop.bias(
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b"", 3, 0
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) # test whether 24-bit audio is supported (for example, ``audioop`` in Python 3.3 and below don't support sample width 3, while Python 3.4+ do)
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except (
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audioop.error
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): # this version of audioop doesn't support 24-bit audio (probably Python 3.3 or less)
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raw_data = b"".join(
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raw_data[i + 1 : i + 4]
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for i in range(0, len(raw_data), 4)
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) # since we're in little endian, we discard the first byte from each 32-bit sample to get a 24-bit sample
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else: # 24-bit audio fully supported, we don't need to shim anything
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raw_data = audioop.lin2lin(
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raw_data, self.sample_width, convert_width
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)
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else:
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raw_data = audioop.lin2lin(
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raw_data, self.sample_width, convert_width
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)
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# if the output is 8-bit audio with unsigned samples, convert the samples we've been treating as signed to unsigned again
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if convert_width == 1:
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raw_data = audioop.bias(
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raw_data, 1, 128
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) # add 128 to every sample to make them act like unsigned samples again
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return raw_data
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def get_wav_data(self, convert_rate=None, convert_width=None, nchannels = 1):
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"""
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Returns a byte string representing the contents of a WAV file containing the audio represented by the ``AudioData`` instance.
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If ``convert_width`` is specified and the audio samples are not ``convert_width`` bytes each, the resulting audio is converted to match.
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If ``convert_rate`` is specified and the audio sample rate is not ``convert_rate`` Hz, the resulting audio is resampled to match.
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Writing these bytes directly to a file results in a valid `WAV file <https://en.wikipedia.org/wiki/WAV>`__.
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"""
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raw_data = self.get_raw_data(convert_rate, convert_width)
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sample_rate = (
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self.sample_rate if convert_rate is None else convert_rate
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)
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sample_width = (
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self.sample_width if convert_width is None else convert_width
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)
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# generate the WAV file contents
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with io.BytesIO() as wav_file:
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wav_writer = wave.open(wav_file, "wb")
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try: # note that we can't use context manager, since that was only added in Python 3.4
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wav_writer.setframerate(sample_rate)
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wav_writer.setsampwidth(sample_width)
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wav_writer.setnchannels(nchannels)
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wav_writer.writeframes(raw_data)
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wav_data = wav_file.getvalue()
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finally: # make sure resources are cleaned up
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wav_writer.close()
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return wav_data
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def get_aiff_data(self, convert_rate=None, convert_width=None):
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"""
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Returns a byte string representing the contents of an AIFF-C file containing the audio represented by the ``AudioData`` instance.
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If ``convert_width`` is specified and the audio samples are not ``convert_width`` bytes each, the resulting audio is converted to match.
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If ``convert_rate`` is specified and the audio sample rate is not ``convert_rate`` Hz, the resulting audio is resampled to match.
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Writing these bytes directly to a file results in a valid `AIFF-C file <https://en.wikipedia.org/wiki/Audio_Interchange_File_Format>`__.
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"""
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raw_data = self.get_raw_data(convert_rate, convert_width)
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sample_rate = (
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self.sample_rate if convert_rate is None else convert_rate
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)
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sample_width = (
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self.sample_width if convert_width is None else convert_width
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)
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# the AIFF format is big-endian, so we need to convert the little-endian raw data to big-endian
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if hasattr(
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audioop, "byteswap"
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): # ``audioop.byteswap`` was only added in Python 3.4
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raw_data = audioop.byteswap(raw_data, sample_width)
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else: # manually reverse the bytes of each sample, which is slower but works well enough as a fallback
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raw_data = raw_data[sample_width - 1 :: -1] + b"".join(
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raw_data[i + sample_width : i : -1]
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for i in range(sample_width - 1, len(raw_data), sample_width)
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)
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# generate the AIFF-C file contents
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with io.BytesIO() as aiff_file:
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aiff_writer = aifc.open(aiff_file, "wb")
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try: # note that we can't use context manager, since that was only added in Python 3.4
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aiff_writer.setframerate(sample_rate)
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aiff_writer.setsampwidth(sample_width)
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aiff_writer.setnchannels(1)
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aiff_writer.writeframes(raw_data)
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aiff_data = aiff_file.getvalue()
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finally: # make sure resources are cleaned up
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aiff_writer.close()
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return aiff_data
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def get_flac_data(self, convert_rate=None, convert_width=None):
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"""
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Returns a byte string representing the contents of a FLAC file containing the audio represented by the ``AudioData`` instance.
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Note that 32-bit FLAC is not supported. If the audio data is 32-bit and ``convert_width`` is not specified, then the resulting FLAC will be a 24-bit FLAC.
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If ``convert_rate`` is specified and the audio sample rate is not ``convert_rate`` Hz, the resulting audio is resampled to match.
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If ``convert_width`` is specified and the audio samples are not ``convert_width`` bytes each, the resulting audio is converted to match.
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Writing these bytes directly to a file results in a valid `FLAC file <https://en.wikipedia.org/wiki/FLAC>`__.
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"""
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assert convert_width is None or (
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convert_width % 1 == 0 and 1 <= convert_width <= 3
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), "Sample width to convert to must be between 1 and 3 inclusive"
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if (
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self.sample_width > 3 and convert_width is None
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): # resulting WAV data would be 32-bit, which is not convertable to FLAC using our encoder
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convert_width = 3 # the largest supported sample width is 24-bit, so we'll limit the sample width to that
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# run the FLAC converter with the WAV data to get the FLAC data
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wav_data = self.get_wav_data(convert_rate, convert_width)
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flac_converter = get_flac_converter()
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if (
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os.name == "nt"
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): # on Windows, specify that the process is to be started without showing a console window
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startup_info = subprocess.STARTUPINFO()
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startup_info.dwFlags |= (
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subprocess.STARTF_USESHOWWINDOW
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) # specify that the wShowWindow field of `startup_info` contains a value
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startup_info.wShowWindow = (
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subprocess.SW_HIDE
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) # specify that the console window should be hidden
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else:
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startup_info = None # default startupinfo
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process = subprocess.Popen(
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[
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flac_converter,
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"--stdout",
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"--totally-silent", # put the resulting FLAC file in stdout, and make sure it's not mixed with any program output
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"--best", # highest level of compression available
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"-", # the input FLAC file contents will be given in stdin
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],
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stdin=subprocess.PIPE,
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stdout=subprocess.PIPE,
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startupinfo=startup_info,
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)
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flac_data, stderr = process.communicate(wav_data)
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return flac_data
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def get_flac_converter():
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"""Returns the absolute path of a FLAC converter executable, or raises an OSError if none can be found."""
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flac_converter = shutil_which("flac") # check for installed version first
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if flac_converter is None: # flac utility is not installed
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base_path = os.path.dirname(
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os.path.abspath(__file__)
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) # directory of the current module file, where all the FLAC bundled binaries are stored
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system, machine = platform.system(), platform.machine()
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if system == "Windows" and machine in {
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"i686",
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"i786",
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"x86",
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"x86_64",
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"AMD64",
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}:
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flac_converter = os.path.join(base_path, "flac-win32.exe")
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elif system == "Darwin" and machine in {
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"i686",
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"i786",
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"x86",
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"x86_64",
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"AMD64",
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}:
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flac_converter = os.path.join(base_path, "flac-mac")
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elif system == "Linux" and machine in {"i686", "i786", "x86"}:
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flac_converter = os.path.join(base_path, "flac-linux-x86")
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elif system == "Linux" and machine in {"x86_64", "AMD64"}:
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flac_converter = os.path.join(base_path, "flac-linux-x86_64")
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else: # no FLAC converter available
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raise OSError(
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"FLAC conversion utility not available - consider installing the FLAC command line application by running `apt-get install flac` or your operating system's equivalent"
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)
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# mark FLAC converter as executable if possible
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try:
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# handle known issue when running on docker:
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# run executable right after chmod() may result in OSError "Text file busy"
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# fix: flush FS with sync
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if not os.access(flac_converter, os.X_OK):
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stat_info = os.stat(flac_converter)
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os.chmod(flac_converter, stat_info.st_mode | stat.S_IEXEC)
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if "Linux" in platform.system():
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os.sync() if sys.version_info >= (3, 3) else os.system("sync")
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except OSError:
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pass
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return flac_converter
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def shutil_which(pgm):
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"""Python 2 compatibility: backport of ``shutil.which()`` from Python 3"""
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path = os.getenv("PATH")
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for p in path.split(os.path.pathsep):
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p = os.path.join(p, pgm)
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if os.path.exists(p) and os.access(p, os.X_OK):
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return p
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@@ -1,22 +0,0 @@
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class SetupError(Exception):
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pass
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class WaitTimeoutError(Exception):
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pass
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class RequestError(Exception):
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pass
|
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|
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|
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class UnknownValueError(Exception):
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pass
|
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|
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|
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class TranscriptionNotReady(Exception):
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pass
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|
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class TranscriptionFailed(Exception):
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pass
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@@ -1,31 +0,0 @@
|
||||
Copyright (c) 1999-2015 Carnegie Mellon University. All rights
|
||||
reserved.
|
||||
|
||||
Redistribution and use in source and binary forms, with or without
|
||||
modification, are permitted provided that the following conditions
|
||||
are met:
|
||||
|
||||
1. Redistributions of source code must retain the above copyright
|
||||
notice, this list of conditions and the following disclaimer.
|
||||
|
||||
2. Redistributions in binary form must reproduce the above copyright
|
||||
notice, this list of conditions and the following disclaimer in
|
||||
the documentation and/or other materials provided with the
|
||||
distribution.
|
||||
|
||||
This work was supported in part by funding from the Defense Advanced
|
||||
Research Projects Agency and the National Science Foundation of the
|
||||
United States of America, and the CMU Sphinx Speech Consortium.
|
||||
|
||||
THIS SOFTWARE IS PROVIDED BY CARNEGIE MELLON UNIVERSITY ``AS IS'' AND
|
||||
ANY EXPRESSED OR IMPLIED WARRANTIES, INCLUDING, BUT NOT LIMITED TO,
|
||||
THE IMPLIED WARRANTIES OF MERCHANTABILITY AND FITNESS FOR A PARTICULAR
|
||||
PURPOSE ARE DISCLAIMED. IN NO EVENT SHALL CARNEGIE MELLON UNIVERSITY
|
||||
NOR ITS EMPLOYEES BE LIABLE FOR ANY DIRECT, INDIRECT, INCIDENTAL,
|
||||
SPECIAL, EXEMPLARY, OR CONSEQUENTIAL DAMAGES (INCLUDING, BUT NOT
|
||||
LIMITED TO, PROCUREMENT OF SUBSTITUTE GOODS OR SERVICES; LOSS OF USE,
|
||||
DATA, OR PROFITS; OR BUSINESS INTERRUPTION) HOWEVER CAUSED AND ON ANY
|
||||
THEORY OF LIABILITY, WHETHER IN CONTRACT, STRICT LIABILITY, OR TORT
|
||||
(INCLUDING NEGLIGENCE OR OTHERWISE) ARISING IN ANY WAY OUT OF THE USE
|
||||
OF THIS SOFTWARE, EVEN IF ADVISED OF THE POSSIBILITY OF SUCH DAMAGE.
|
||||
|
||||
@@ -1,34 +0,0 @@
|
||||
/* ====================================================================
|
||||
* Copyright (c) 2015 Alpha Cephei Inc. All rights
|
||||
* reserved.
|
||||
*
|
||||
* Redistribution and use in source and binary forms, with or without
|
||||
* modification, are permitted provided that the following conditions
|
||||
* are met:
|
||||
*
|
||||
* 1. Redistributions of source code must retain the above copyright
|
||||
* notice, this list of conditions and the following disclaimer.
|
||||
*
|
||||
* 2. Redistributions in binary form must reproduce the above copyright
|
||||
* notice, this list of conditions and the following disclaimer in
|
||||
* the documentation and/or other materials provided with the
|
||||
* distribution.
|
||||
*
|
||||
* THIS SOFTWARE IS PROVIDED BY ALPHA CEPHEI INC. ``AS IS'' AND.
|
||||
* ANY EXPRESSED OR IMPLIED WARRANTIES, INCLUDING, BUT NOT LIMITED TO,.
|
||||
* THE IMPLIED WARRANTIES OF MERCHANTABILITY AND FITNESS FOR A PARTICULAR
|
||||
* PURPOSE ARE DISCLAIMED. IN NO EVENT SHALL ALPHA CEPHEI INC.
|
||||
* NOR ITS EMPLOYEES BE LIABLE FOR ANY DIRECT, INDIRECT, INCIDENTAL,
|
||||
* SPECIAL, EXEMPLARY, OR CONSEQUENTIAL DAMAGES (INCLUDING, BUT NOT.
|
||||
* LIMITED TO, PROCUREMENT OF SUBSTITUTE GOODS OR SERVICES; LOSS OF USE,.
|
||||
* DATA, OR PROFITS; OR BUSINESS INTERRUPTION) HOWEVER CAUSED AND ON ANY.
|
||||
* THEORY OF LIABILITY, WHETHER IN CONTRACT, STRICT LIABILITY, OR TORT.
|
||||
* (INCLUDING NEGLIGENCE OR OTHERWISE) ARISING IN ANY WAY OUT OF THE USE.
|
||||
* OF THIS SOFTWARE, EVEN IF ADVISED OF THE POSSIBILITY OF SUCH DAMAGE.
|
||||
*
|
||||
* ====================================================================
|
||||
*
|
||||
*/
|
||||
|
||||
This directory contains generic US english acoustic model trained with
|
||||
latest sphinxtrain.
|
||||
@@ -1,12 +0,0 @@
|
||||
-lowerf 130
|
||||
-upperf 6800
|
||||
-nfilt 25
|
||||
-transform dct
|
||||
-lifter 22
|
||||
-feat 1s_c_d_dd
|
||||
-svspec 0-12/13-25/26-38
|
||||
-agc none
|
||||
-cmn current
|
||||
-varnorm no
|
||||
-model ptm
|
||||
-cmninit 40,3,-1
|
||||
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|
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|
||||
<s> SIL
|
||||
</s> SIL
|
||||
<sil> SIL
|
||||
[NOISE] +NSN+
|
||||
[SPEECH] +SPN+
|
||||
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Load Diff
@@ -1,42 +0,0 @@
|
||||
from __future__ import annotations
|
||||
|
||||
import os
|
||||
from io import BytesIO
|
||||
|
||||
from custom_speech_recognition.audio import AudioData
|
||||
from custom_speech_recognition.exceptions import SetupError
|
||||
|
||||
|
||||
def recognize_whisper_api(
|
||||
recognizer,
|
||||
audio_data: "AudioData",
|
||||
*,
|
||||
model: str = "whisper-1",
|
||||
api_key: str | None = None,
|
||||
):
|
||||
"""
|
||||
Performs speech recognition on ``audio_data`` (an ``AudioData`` instance), using the OpenAI Whisper API.
|
||||
|
||||
This function requires an OpenAI account; visit https://platform.openai.com/signup, then generate API Key in `User settings <https://platform.openai.com/account/api-keys>`__.
|
||||
|
||||
Detail: https://platform.openai.com/docs/guides/speech-to-text
|
||||
|
||||
Raises a ``speech_recognition.exceptions.SetupError`` exception if there are any issues with the openai installation, or the environment variable is missing.
|
||||
"""
|
||||
if not isinstance(audio_data, AudioData):
|
||||
raise ValueError("``audio_data`` must be an ``AudioData`` instance")
|
||||
if api_key is None and os.environ.get("OPENAI_API_KEY") is None:
|
||||
raise SetupError("Set environment variable ``OPENAI_API_KEY``")
|
||||
|
||||
try:
|
||||
import openai
|
||||
except ImportError:
|
||||
raise SetupError(
|
||||
"missing openai module: ensure that openai is set up correctly."
|
||||
)
|
||||
|
||||
wav_data = BytesIO(audio_data.get_wav_data())
|
||||
wav_data.name = "SpeechRecognition_audio.wav"
|
||||
|
||||
transcript = openai.Audio.transcribe(model, wav_data, api_key=api_key)
|
||||
return transcript["text"]
|
||||
Reference in New Issue
Block a user