#!/usr/bin/env python3

"""Library for performing speech recognition, with support for several engines and APIs, online and offline."""

import io
import os
import sys
import subprocess
import wave
import aifc
import math
import audioop
import collections
import json
import base64
import threading
import platform
import stat
import hashlib
import hmac
import time
import uuid

__author__ = "Anthony Zhang (Uberi)"
__version__ = "3.8.1"
__license__ = "BSD"

try:  # attempt to use the Python 2 modules
    from urllib import urlencode
    from urllib2 import Request, urlopen, URLError, HTTPError
except ImportError:  # use the Python 3 modules
    from urllib.parse import urlencode
    from urllib.request import Request, urlopen
    from urllib.error import URLError, HTTPError


class WaitTimeoutError(Exception): pass


class RequestError(Exception): pass


class UnknownValueError(Exception): pass


class AudioSource(object):
    def __init__(self):
        raise NotImplementedError("this is an abstract class")

    def __enter__(self):
        raise NotImplementedError("this is an abstract class")

    def __exit__(self, exc_type, exc_value, traceback):
        raise NotImplementedError("this is an abstract class")


class Microphone(AudioSource):
    """
    Creates a new ``Microphone`` instance, which represents a physical microphone on the computer. Subclass of ``AudioSource``.

    This will throw an ``AttributeError`` if you don't have PyAudio 0.2.11 or later installed.

    If ``device_index`` is unspecified or ``None``, the default microphone is used as the audio source. Otherwise, ``device_index`` should be the index of the device to use for audio input.

    A device index is an integer between 0 and ``pyaudio.get_device_count() - 1`` (assume we have used ``import pyaudio`` beforehand) inclusive. It represents an audio device such as a microphone or speaker. See the `PyAudio documentation <http://people.csail.mit.edu/hubert/pyaudio/docs/>`__ for more details.

    The microphone audio is recorded in chunks of ``chunk_size`` samples, at a rate of ``sample_rate`` samples per second (Hertz). If not specified, the value of ``sample_rate`` is determined automatically from the system's microphone settings.

    Higher ``sample_rate`` values result in better audio quality, but also more bandwidth (and therefore, slower recognition). Additionally, some CPUs, such as those in older Raspberry Pi models, can't keep up if this value is too high.

    Higher ``chunk_size`` values help avoid triggering on rapidly changing ambient noise, but also makes detection less sensitive. This value, generally, should be left at its default.
    """
    def __init__(self, device_index=None, sample_rate=None, chunk_size=1024):
        assert device_index is None or isinstance(device_index, int), "Device index must be None or an integer"
        assert sample_rate is None or (isinstance(sample_rate, int) and sample_rate > 0), "Sample rate must be None or a positive integer"
        assert isinstance(chunk_size, int) and chunk_size > 0, "Chunk size must be a positive integer"

        # set up PyAudio
        self.pyaudio_module = self.get_pyaudio()
        audio = self.pyaudio_module.PyAudio()
        try:
            count = audio.get_device_count()  # obtain device count
            if device_index is not None:  # ensure device index is in range
                assert 0 <= device_index < count, "Device index out of range ({} devices available; device index should be between 0 and {} inclusive)".format(count, count - 1)
            if sample_rate is None:  # automatically set the sample rate to the hardware's default sample rate if not specified
                device_info = audio.get_device_info_by_index(device_index) if device_index is not None else audio.get_default_input_device_info()
                assert isinstance(device_info.get("defaultSampleRate"), (float, int)) and device_info["defaultSampleRate"] > 0, "Invalid device info returned from PyAudio: {}".format(device_info)
                sample_rate = int(device_info["defaultSampleRate"])
        except Exception:
            audio.terminate()
            raise

        self.device_index = device_index
        self.format = self.pyaudio_module.paInt16  # 16-bit int sampling
        self.SAMPLE_WIDTH = self.pyaudio_module.get_sample_size(self.format)  # size of each sample
        self.SAMPLE_RATE = sample_rate  # sampling rate in Hertz
        self.CHUNK = chunk_size  # number of frames stored in each buffer

        self.audio = None
        self.stream = None

    @staticmethod
    def get_pyaudio():
        """
        Imports the pyaudio module and checks its version. Throws exceptions if pyaudio can't be found or a wrong version is installed
        """
        try:
            import pyaudio
        except ImportError:
            raise AttributeError("Could not find PyAudio; check installation")
        from distutils.version import LooseVersion
        if LooseVersion(pyaudio.__version__) < LooseVersion("0.2.11"):
            raise AttributeError("PyAudio 0.2.11 or later is required (found version {})".format(pyaudio.__version__))
        return pyaudio

    @staticmethod
    def list_microphone_names():
        """
        Returns a list of the names of all available microphones. For microphones where the name can't be retrieved, the list entry contains ``None`` instead.

        The index of each microphone's name is the same as its device index when creating a ``Microphone`` instance - indices in this list can be used as values of ``device_index``.
        """
        audio = Microphone.get_pyaudio().PyAudio()
        try:
            result = []
            for i in range(audio.get_device_count()):
                device_info = audio.get_device_info_by_index(i)
                result.append(device_info.get("name"))
        finally:
            audio.terminate()
        return result

    def __enter__(self):
        assert self.stream is None, "This audio source is already inside a context manager"
        self.audio = self.pyaudio_module.PyAudio()
        try:
            self.stream = Microphone.MicrophoneStream(
                self.audio.open(
                    input_device_index=self.device_index, channels=1,
                    format=self.format, rate=self.SAMPLE_RATE, frames_per_buffer=self.CHUNK,
                    input=True,  # stream is an input stream
                )
            )
        except Exception:
            self.audio.terminate()
            raise
        return self

    def __exit__(self, exc_type, exc_value, traceback):
        try:
            self.stream.close()
        finally:
            self.stream = None
            self.audio.terminate()

    class MicrophoneStream(object):
        def __init__(self, pyaudio_stream):
            self.pyaudio_stream = pyaudio_stream

        def read(self, size):
            return self.pyaudio_stream.read(size, exception_on_overflow=False)

        def close(self):
            try:
                # sometimes, if the stream isn't stopped, closing the stream throws an exception
                if not self.pyaudio_stream.is_stopped():
                    self.pyaudio_stream.stop_stream()
            finally:
                self.pyaudio_stream.close()


class AudioFile(AudioSource):
    """
    Creates a new ``AudioFile`` instance given a WAV/AIFF/FLAC audio file ``filename_or_fileobject``. Subclass of ``AudioSource``.

    If ``filename_or_fileobject`` is a string, then it is interpreted as a path to an audio file on the filesystem. Otherwise, ``filename_or_fileobject`` should be a file-like object such as ``io.BytesIO`` or similar.

    Note that functions that read from the audio (such as ``recognizer_instance.record`` or ``recognizer_instance.listen``) will move ahead in the stream. For example, if you execute ``recognizer_instance.record(audiofile_instance, duration=10)`` twice, the first time it will return the first 10 seconds of audio, and the second time it will return the 10 seconds of audio right after that. This is always reset to the beginning when entering an ``AudioFile`` context.

    WAV files must be in PCM/LPCM format; WAVE_FORMAT_EXTENSIBLE and compressed WAV are not supported and may result in undefined behaviour.

    Both AIFF and AIFF-C (compressed AIFF) formats are supported.

    FLAC files must be in native FLAC format; OGG-FLAC is not supported and may result in undefined behaviour.
    """

    def __init__(self, filename_or_fileobject):
        assert isinstance(filename_or_fileobject, (type(""), type(u""))) or hasattr(filename_or_fileobject, "read"), "Given audio file must be a filename string or a file-like object"
        self.filename_or_fileobject = filename_or_fileobject
        self.stream = None
        self.DURATION = None

        self.audio_reader = None
        self.little_endian = False
        self.SAMPLE_RATE = None
        self.CHUNK = None
        self.FRAME_COUNT = None

    def __enter__(self):
        assert self.stream is None, "This audio source is already inside a context manager"
        try:
            # attempt to read the file as WAV
            self.audio_reader = wave.open(self.filename_or_fileobject, "rb")
            self.little_endian = True  # RIFF WAV is a little-endian format (most ``audioop`` operations assume that the frames are stored in little-endian form)
        except (wave.Error, EOFError):
            try:
                # attempt to read the file as AIFF
                self.audio_reader = aifc.open(self.filename_or_fileobject, "rb")
                self.little_endian = False  # AIFF is a big-endian format
            except (aifc.Error, EOFError):
                # attempt to read the file as FLAC
                if hasattr(self.filename_or_fileobject, "read"):
                    flac_data = self.filename_or_fileobject.read()
                else:
                    with open(self.filename_or_fileobject, "rb") as f: flac_data = f.read()

                # run the FLAC converter with the FLAC data to get the AIFF data
                flac_converter = get_flac_converter()
                if os.name == "nt":  # on Windows, specify that the process is to be started without showing a console window
                    startup_info = subprocess.STARTUPINFO()
                    startup_info.dwFlags |= subprocess.STARTF_USESHOWWINDOW  # specify that the wShowWindow field of `startup_info` contains a value
                    startup_info.wShowWindow = subprocess.SW_HIDE  # specify that the console window should be hidden
                else:
                    startup_info = None  # default startupinfo
                process = subprocess.Popen([
                    flac_converter,
                    "--stdout", "--totally-silent",  # put the resulting AIFF file in stdout, and make sure it's not mixed with any program output
                    "--decode", "--force-aiff-format",  # decode the FLAC file into an AIFF file
                    "-",  # the input FLAC file contents will be given in stdin
                ], stdin=subprocess.PIPE, stdout=subprocess.PIPE, startupinfo=startup_info)
                aiff_data, _ = process.communicate(flac_data)
                aiff_file = io.BytesIO(aiff_data)
                try:
                    self.audio_reader = aifc.open(aiff_file, "rb")
                except (aifc.Error, EOFError):
                    raise ValueError("Audio file could not be read as PCM WAV, AIFF/AIFF-C, or Native FLAC; check if file is corrupted or in another format")
                self.little_endian = False  # AIFF is a big-endian format
        assert 1 <= self.audio_reader.getnchannels() <= 2, "Audio must be mono or stereo"
        self.SAMPLE_WIDTH = self.audio_reader.getsampwidth()

        # 24-bit audio needs some special handling for old Python versions (workaround for https://bugs.python.org/issue12866)
        samples_24_bit_pretending_to_be_32_bit = False
        if self.SAMPLE_WIDTH == 3:  # 24-bit audio
            try: audioop.bias(b"", self.SAMPLE_WIDTH, 0)  # test whether this sample width is supported (for example, ``audioop`` in Python 3.3 and below don't support sample width 3, while Python 3.4+ do)
            except audioop.error:  # this version of audioop doesn't support 24-bit audio (probably Python 3.3 or less)
                samples_24_bit_pretending_to_be_32_bit = True  # while the ``AudioFile`` instance will outwardly appear to be 32-bit, it will actually internally be 24-bit
                self.SAMPLE_WIDTH = 4  # the ``AudioFile`` instance should present itself as a 32-bit stream now, since we'll be converting into 32-bit on the fly when reading

        self.SAMPLE_RATE = self.audio_reader.getframerate()
        self.CHUNK = 4096
        self.FRAME_COUNT = self.audio_reader.getnframes()
        self.DURATION = self.FRAME_COUNT / float(self.SAMPLE_RATE)
        self.stream = AudioFile.AudioFileStream(self.audio_reader, self.little_endian, samples_24_bit_pretending_to_be_32_bit)
        return self

    def __exit__(self, exc_type, exc_value, traceback):
        if not hasattr(self.filename_or_fileobject, "read"):  # only close the file if it was opened by this class in the first place (if the file was originally given as a path)
            self.audio_reader.close()
        self.stream = None
        self.DURATION = None

    class AudioFileStream(object):
        def __init__(self, audio_reader, little_endian, samples_24_bit_pretending_to_be_32_bit):
            self.audio_reader = audio_reader  # an audio file object (e.g., a `wave.Wave_read` instance)
            self.little_endian = little_endian  # whether the audio data is little-endian (when working with big-endian things, we'll have to convert it to little-endian before we process it)
            self.samples_24_bit_pretending_to_be_32_bit = samples_24_bit_pretending_to_be_32_bit  # this is true if the audio is 24-bit audio, but 24-bit audio isn't supported, so we have to pretend that this is 32-bit audio and convert it on the fly

        def read(self, size=-1):
            buffer = self.audio_reader.readframes(self.audio_reader.getnframes() if size == -1 else size)
            if not isinstance(buffer, bytes): buffer = b""  # workaround for https://bugs.python.org/issue24608

            sample_width = self.audio_reader.getsampwidth()
            if not self.little_endian:  # big endian format, convert to little endian on the fly
                if hasattr(audioop, "byteswap"):  # ``audioop.byteswap`` was only added in Python 3.4 (incidentally, that also means that we don't need to worry about 24-bit audio being unsupported, since Python 3.4+ always has that functionality)
                    buffer = audioop.byteswap(buffer, sample_width)
                else:  # manually reverse the bytes of each sample, which is slower but works well enough as a fallback
                    buffer = buffer[sample_width - 1::-1] + b"".join(buffer[i + sample_width:i:-1] for i in range(sample_width - 1, len(buffer), sample_width))

            # workaround for https://bugs.python.org/issue12866
            if self.samples_24_bit_pretending_to_be_32_bit:  # we need to convert samples from 24-bit to 32-bit before we can process them with ``audioop`` functions
                buffer = b"".join(b"\x00" + buffer[i:i + sample_width] for i in range(0, len(buffer), sample_width))  # since we're in little endian, we prepend a zero byte to each 24-bit sample to get a 32-bit sample
                sample_width = 4  # make sure we thread the buffer as 32-bit audio now, after converting it from 24-bit audio
            if self.audio_reader.getnchannels() != 1:  # stereo audio
                buffer = audioop.tomono(buffer, sample_width, 1, 1)  # convert stereo audio data to mono
            return buffer


class AudioData(object):
    """
    Creates a new ``AudioData`` instance, which represents mono audio data.

    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.

    The width of each sample, in bytes, is specified by ``sample_width``. Each group of ``sample_width`` bytes represents a single audio sample.

    The audio data is assumed to have a sample rate of ``sample_rate`` samples per second (Hertz).

    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.
    """
    def __init__(self, frame_data, sample_rate, sample_width):
        assert sample_rate > 0, "Sample rate must be a positive integer"
        assert sample_width % 1 == 0 and 1 <= sample_width <= 4, "Sample width must be between 1 and 4 inclusive"
        self.frame_data = frame_data
        self.sample_rate = sample_rate
        self.sample_width = int(sample_width)

    def get_segment(self, start_ms=None, end_ms=None):
        """
        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.

        If not specified, ``start_ms`` defaults to the beginning of the audio, and ``end_ms`` defaults to the end.
        """
        assert start_ms is None or start_ms >= 0, "``start_ms`` must be a non-negative number"
        assert end_ms is None or end_ms >= (0 if start_ms is None else start_ms), "``end_ms`` must be a non-negative number greater or equal to ``start_ms``"
        if start_ms is None:
            start_byte = 0
        else:
            start_byte = int((start_ms * self.sample_rate * self.sample_width) // 1000)
        if end_ms is None:
            end_byte = len(self.frame_data)
        else:
            end_byte = int((end_ms * self.sample_rate * self.sample_width) // 1000)
        return AudioData(self.frame_data[start_byte:end_byte], self.sample_rate, self.sample_width)

    def get_raw_data(self, convert_rate=None, convert_width=None):
        """
        Returns a byte string representing the raw frame data for the audio represented by the ``AudioData`` instance.

        If ``convert_rate`` is specified and the audio sample rate is not ``convert_rate`` Hz, the resulting audio is resampled to match.

        If ``convert_width`` is specified and the audio samples are not ``convert_width`` bytes each, the resulting audio is converted to match.

        Writing these bytes directly to a file results in a valid `RAW/PCM audio file <https://en.wikipedia.org/wiki/Raw_audio_format>`__.
        """
        assert convert_rate is None or convert_rate > 0, "Sample rate to convert to must be a positive integer"
        assert convert_width is None or (convert_width % 1 == 0 and 1 <= convert_width <= 4), "Sample width to convert to must be between 1 and 4 inclusive"

        raw_data = self.frame_data

        # make sure unsigned 8-bit audio (which uses unsigned samples) is handled like higher sample width audio (which uses signed samples)
        if self.sample_width == 1:
            raw_data = audioop.bias(raw_data, 1, -128)  # subtract 128 from every sample to make them act like signed samples

        # resample audio at the desired rate if specified
        if convert_rate is not None and self.sample_rate != convert_rate:
            raw_data, _ = audioop.ratecv(raw_data, self.sample_width, 1, self.sample_rate, convert_rate, None)

        # convert samples to desired sample width if specified
        if convert_width is not None and self.sample_width != convert_width:
            if convert_width == 3:  # we're converting the audio into 24-bit (workaround for https://bugs.python.org/issue12866)
                raw_data = audioop.lin2lin(raw_data, self.sample_width, 4)  # convert audio into 32-bit first, which is always supported
                try: audioop.bias(b"", 3, 0)  # 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)
                except audioop.error:  # this version of audioop doesn't support 24-bit audio (probably Python 3.3 or less)
                    raw_data = b"".join(raw_data[i + 1:i + 4] for i in range(0, len(raw_data), 4))  # since we're in little endian, we discard the first byte from each 32-bit sample to get a 24-bit sample
                else:  # 24-bit audio fully supported, we don't need to shim anything
                    raw_data = audioop.lin2lin(raw_data, self.sample_width, convert_width)
            else:
                raw_data = audioop.lin2lin(raw_data, self.sample_width, convert_width)

        # if the output is 8-bit audio with unsigned samples, convert the samples we've been treating as signed to unsigned again
        if convert_width == 1:
            raw_data = audioop.bias(raw_data, 1, 128)  # add 128 to every sample to make them act like unsigned samples again

        return raw_data

    def get_wav_data(self, convert_rate=None, convert_width=None):
        """
        Returns a byte string representing the contents of a WAV file containing the audio represented by the ``AudioData`` instance.

        If ``convert_width`` is specified and the audio samples are not ``convert_width`` bytes each, the resulting audio is converted to match.

        If ``convert_rate`` is specified and the audio sample rate is not ``convert_rate`` Hz, the resulting audio is resampled to match.

        Writing these bytes directly to a file results in a valid `WAV file <https://en.wikipedia.org/wiki/WAV>`__.
        """
        raw_data = self.get_raw_data(convert_rate, convert_width)
        sample_rate = self.sample_rate if convert_rate is None else convert_rate
        sample_width = self.sample_width if convert_width is None else convert_width

        # generate the WAV file contents
        with io.BytesIO() as wav_file:
            wav_writer = wave.open(wav_file, "wb")
            try:  # note that we can't use context manager, since that was only added in Python 3.4
                wav_writer.setframerate(sample_rate)
                wav_writer.setsampwidth(sample_width)
                wav_writer.setnchannels(1)
                wav_writer.writeframes(raw_data)
                wav_data = wav_file.getvalue()
            finally:  # make sure resources are cleaned up
                wav_writer.close()
        return wav_data

    def get_aiff_data(self, convert_rate=None, convert_width=None):
        """
        Returns a byte string representing the contents of an AIFF-C file containing the audio represented by the ``AudioData`` instance.

        If ``convert_width`` is specified and the audio samples are not ``convert_width`` bytes each, the resulting audio is converted to match.

        If ``convert_rate`` is specified and the audio sample rate is not ``convert_rate`` Hz, the resulting audio is resampled to match.

        Writing these bytes directly to a file results in a valid `AIFF-C file <https://en.wikipedia.org/wiki/Audio_Interchange_File_Format>`__.
        """
        raw_data = self.get_raw_data(convert_rate, convert_width)
        sample_rate = self.sample_rate if convert_rate is None else convert_rate
        sample_width = self.sample_width if convert_width is None else convert_width

        # the AIFF format is big-endian, so we need to covnert the little-endian raw data to big-endian
        if hasattr(audioop, "byteswap"):  # ``audioop.byteswap`` was only added in Python 3.4
            raw_data = audioop.byteswap(raw_data, sample_width)
        else:  # manually reverse the bytes of each sample, which is slower but works well enough as a fallback
            raw_data = raw_data[sample_width - 1::-1] + b"".join(raw_data[i + sample_width:i:-1] for i in range(sample_width - 1, len(raw_data), sample_width))

        # generate the AIFF-C file contents
        with io.BytesIO() as aiff_file:
            aiff_writer = aifc.open(aiff_file, "wb")
            try:  # note that we can't use context manager, since that was only added in Python 3.4
                aiff_writer.setframerate(sample_rate)
                aiff_writer.setsampwidth(sample_width)
                aiff_writer.setnchannels(1)
                aiff_writer.writeframes(raw_data)
                aiff_data = aiff_file.getvalue()
            finally:  # make sure resources are cleaned up
                aiff_writer.close()
        return aiff_data

    def get_flac_data(self, convert_rate=None, convert_width=None):
        """
        Returns a byte string representing the contents of a FLAC file containing the audio represented by the ``AudioData`` instance.

        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.

        If ``convert_rate`` is specified and the audio sample rate is not ``convert_rate`` Hz, the resulting audio is resampled to match.

        If ``convert_width`` is specified and the audio samples are not ``convert_width`` bytes each, the resulting audio is converted to match.

        Writing these bytes directly to a file results in a valid `FLAC file <https://en.wikipedia.org/wiki/FLAC>`__.
        """
        assert convert_width is None or (convert_width % 1 == 0 and 1 <= convert_width <= 3), "Sample width to convert to must be between 1 and 3 inclusive"

        if self.sample_width > 3 and convert_width is None:  # resulting WAV data would be 32-bit, which is not convertable to FLAC using our encoder
            convert_width = 3  # the largest supported sample width is 24-bit, so we'll limit the sample width to that

        # run the FLAC converter with the WAV data to get the FLAC data
        wav_data = self.get_wav_data(convert_rate, convert_width)
        flac_converter = get_flac_converter()
        if os.name == "nt":  # on Windows, specify that the process is to be started without showing a console window
            startup_info = subprocess.STARTUPINFO()
            startup_info.dwFlags |= subprocess.STARTF_USESHOWWINDOW  # specify that the wShowWindow field of `startup_info` contains a value
            startup_info.wShowWindow = subprocess.SW_HIDE  # specify that the console window should be hidden
        else:
            startup_info = None  # default startupinfo
        process = subprocess.Popen([
            flac_converter,
            "--stdout", "--totally-silent",  # put the resulting FLAC file in stdout, and make sure it's not mixed with any program output
            "--best",  # highest level of compression available
            "-",  # the input FLAC file contents will be given in stdin
        ], stdin=subprocess.PIPE, stdout=subprocess.PIPE, startupinfo=startup_info)
        flac_data, stderr = process.communicate(wav_data)
        return flac_data


class Recognizer(AudioSource):
    def __init__(self):
        """
        Creates a new ``Recognizer`` instance, which represents a collection of speech recognition functionality.
        """
        self.energy_threshold = 300  # minimum audio energy to consider for recording
        self.dynamic_energy_threshold = True
        self.dynamic_energy_adjustment_damping = 0.15
        self.dynamic_energy_ratio = 1.5
        self.pause_threshold = 0.8  # seconds of non-speaking audio before a phrase is considered complete
        self.operation_timeout = None  # seconds after an internal operation (e.g., an API request) starts before it times out, or ``None`` for no timeout

        self.phrase_threshold = 0.3  # minimum seconds of speaking audio before we consider the speaking audio a phrase - values below this are ignored (for filtering out clicks and pops)
        self.non_speaking_duration = 0.5  # seconds of non-speaking audio to keep on both sides of the recording

    def record(self, source, duration=None, offset=None):
        """
        Records up to ``duration`` seconds of audio from ``source`` (an ``AudioSource`` instance) starting at ``offset`` (or at the beginning if not specified) into an ``AudioData`` instance, which it returns.

        If ``duration`` is not specified, then it will record until there is no more audio input.
        """
        assert isinstance(source, AudioSource), "Source must be an audio source"
        assert source.stream is not None, "Audio source must be entered before recording, see documentation for ``AudioSource``; are you using ``source`` outside of a ``with`` statement?"

        frames = io.BytesIO()
        seconds_per_buffer = (source.CHUNK + 0.0) / source.SAMPLE_RATE
        elapsed_time = 0
        offset_time = 0
        offset_reached = False
        while True:  # loop for the total number of chunks needed
            if offset and not offset_reached:
                offset_time += seconds_per_buffer
                if offset_time > offset:
                    offset_reached = True

            buffer = source.stream.read(source.CHUNK)
            if len(buffer) == 0: break

            if offset_reached or not offset:
                elapsed_time += seconds_per_buffer
                if duration and elapsed_time > duration: break

                frames.write(buffer)

        frame_data = frames.getvalue()
        frames.close()
        return AudioData(frame_data, source.SAMPLE_RATE, source.SAMPLE_WIDTH)

    def adjust_for_ambient_noise(self, source, duration=1):
        """
        Adjusts the energy threshold dynamically using audio from ``source`` (an ``AudioSource`` instance) to account for ambient noise.

        Intended to calibrate the energy threshold with the ambient energy level. Should be used on periods of audio without speech - will stop early if any speech is detected.

        The ``duration`` parameter is the maximum number of seconds that it will dynamically adjust the threshold for before returning. This value should be at least 0.5 in order to get a representative sample of the ambient noise.
        """
        assert isinstance(source, AudioSource), "Source must be an audio source"
        assert source.stream is not None, "Audio source must be entered before adjusting, see documentation for ``AudioSource``; are you using ``source`` outside of a ``with`` statement?"
        assert self.pause_threshold >= self.non_speaking_duration >= 0

        seconds_per_buffer = (source.CHUNK + 0.0) / source.SAMPLE_RATE
        elapsed_time = 0

        # adjust energy threshold until a phrase starts
        while True:
            elapsed_time += seconds_per_buffer
            if elapsed_time > duration: break
            buffer = source.stream.read(source.CHUNK)
            energy = audioop.rms(buffer, source.SAMPLE_WIDTH)  # energy of the audio signal

            # dynamically adjust the energy threshold using asymmetric weighted average
            damping = self.dynamic_energy_adjustment_damping ** seconds_per_buffer  # account for different chunk sizes and rates
            target_energy = energy * self.dynamic_energy_ratio
            self.energy_threshold = self.energy_threshold * damping + target_energy * (1 - damping)

    def snowboy_wait_for_hot_word(self, snowboy_location, snowboy_hot_word_files, source, timeout=None):
        # load snowboy library (NOT THREAD SAFE)
        sys.path.append(snowboy_location)
        import snowboydetect
        sys.path.pop()

        detector = snowboydetect.SnowboyDetect(
            resource_filename=os.path.join(snowboy_location, "resources", "common.res").encode(),
            model_str=",".join(snowboy_hot_word_files).encode()
        )
        detector.SetAudioGain(1.0)
        detector.SetSensitivity(",".join(["0.4"] * len(snowboy_hot_word_files)).encode())
        snowboy_sample_rate = detector.SampleRate()

        elapsed_time = 0
        seconds_per_buffer = float(source.CHUNK) / source.SAMPLE_RATE
        resampling_state = None

        # buffers capable of holding 5 seconds of original and resampled audio
        five_seconds_buffer_count = int(math.ceil(5 / seconds_per_buffer))
        frames = collections.deque(maxlen=five_seconds_buffer_count)
        resampled_frames = collections.deque(maxlen=five_seconds_buffer_count)
        while True:
            elapsed_time += seconds_per_buffer
            if timeout and elapsed_time > timeout:
                raise WaitTimeoutError("listening timed out while waiting for hotword to be said")

            buffer = source.stream.read(source.CHUNK)
            if len(buffer) == 0: break  # reached end of the stream
            frames.append(buffer)

            # resample audio to the required sample rate
            resampled_buffer, resampling_state = audioop.ratecv(buffer, source.SAMPLE_WIDTH, 1, source.SAMPLE_RATE, snowboy_sample_rate, resampling_state)
            resampled_frames.append(resampled_buffer)

            # run Snowboy on the resampled audio
            snowboy_result = detector.RunDetection(b"".join(resampled_frames))
            assert snowboy_result != -1, "Error initializing streams or reading audio data"
            if snowboy_result > 0: break  # wake word found

        return b"".join(frames), elapsed_time

    def listen(self, source, timeout=None, phrase_time_limit=None, snowboy_configuration=None):
        """
        Records a single phrase from ``source`` (an ``AudioSource`` instance) into an ``AudioData`` instance, which it returns.

        This is done by waiting until the audio has an energy above ``recognizer_instance.energy_threshold`` (the user has started speaking), and then recording until it encounters ``recognizer_instance.pause_threshold`` seconds of non-speaking or there is no more audio input. The ending silence is not included.

        The ``timeout`` parameter is the maximum number of seconds that this will wait for a phrase to start before giving up and throwing an ``speech_recognition.WaitTimeoutError`` exception. If ``timeout`` is ``None``, there will be no wait timeout.

        The ``phrase_time_limit`` parameter is the maximum number of seconds that this will allow a phrase to continue before stopping and returning the part of the phrase processed before the time limit was reached. The resulting audio will be the phrase cut off at the time limit. If ``phrase_timeout`` is ``None``, there will be no phrase time limit.

        The ``snowboy_configuration`` parameter allows integration with `Snowboy <https://snowboy.kitt.ai/>`__, an offline, high-accuracy, power-efficient hotword recognition engine. When used, this function will pause until Snowboy detects a hotword, after which it will unpause. This parameter should either be ``None`` to turn off Snowboy support, or a tuple of the form ``(SNOWBOY_LOCATION, LIST_OF_HOT_WORD_FILES)``, where ``SNOWBOY_LOCATION`` is the path to the Snowboy root directory, and ``LIST_OF_HOT_WORD_FILES`` is a list of paths to Snowboy hotword configuration files (`*.pmdl` or `*.umdl` format).

        This operation will always complete within ``timeout + phrase_timeout`` seconds if both are numbers, either by returning the audio data, or by raising a ``speech_recognition.WaitTimeoutError`` exception.
        """
        assert isinstance(source, AudioSource), "Source must be an audio source"
        assert source.stream is not None, "Audio source must be entered before listening, see documentation for ``AudioSource``; are you using ``source`` outside of a ``with`` statement?"
        assert self.pause_threshold >= self.non_speaking_duration >= 0
        if snowboy_configuration is not None:
            assert os.path.isfile(os.path.join(snowboy_configuration[0], "snowboydetect.py")), "``snowboy_configuration[0]`` must be a Snowboy root directory containing ``snowboydetect.py``"
            for hot_word_file in snowboy_configuration[1]:
                assert os.path.isfile(hot_word_file), "``snowboy_configuration[1]`` must be a list of Snowboy hot word configuration files"

        seconds_per_buffer = float(source.CHUNK) / source.SAMPLE_RATE
        pause_buffer_count = int(math.ceil(self.pause_threshold / seconds_per_buffer))  # number of buffers of non-speaking audio during a phrase, before the phrase should be considered complete
        phrase_buffer_count = int(math.ceil(self.phrase_threshold / seconds_per_buffer))  # minimum number of buffers of speaking audio before we consider the speaking audio a phrase
        non_speaking_buffer_count = int(math.ceil(self.non_speaking_duration / seconds_per_buffer))  # maximum number of buffers of non-speaking audio to retain before and after a phrase

        # read audio input for phrases until there is a phrase that is long enough
        elapsed_time = 0  # number of seconds of audio read
        buffer = b""  # an empty buffer means that the stream has ended and there is no data left to read
        while True:
            frames = collections.deque()

            if snowboy_configuration is None:
                # store audio input until the phrase starts
                while True:
                    # handle waiting too long for phrase by raising an exception
                    elapsed_time += seconds_per_buffer
                    if timeout and elapsed_time > timeout:
                        raise WaitTimeoutError("listening timed out while waiting for phrase to start")

                    buffer = source.stream.read(source.CHUNK)
                    if len(buffer) == 0: break  # reached end of the stream
                    frames.append(buffer)
                    if len(frames) > non_speaking_buffer_count:  # ensure we only keep the needed amount of non-speaking buffers
                        frames.popleft()

                    # detect whether speaking has started on audio input
                    energy = audioop.rms(buffer, source.SAMPLE_WIDTH)  # energy of the audio signal
                    if energy > self.energy_threshold: break

                    # dynamically adjust the energy threshold using asymmetric weighted average
                    if self.dynamic_energy_threshold:
                        damping = self.dynamic_energy_adjustment_damping ** seconds_per_buffer  # account for different chunk sizes and rates
                        target_energy = energy * self.dynamic_energy_ratio
                        self.energy_threshold = self.energy_threshold * damping + target_energy * (1 - damping)
            else:
                # read audio input until the hotword is said
                snowboy_location, snowboy_hot_word_files = snowboy_configuration
                buffer, delta_time = self.snowboy_wait_for_hot_word(snowboy_location, snowboy_hot_word_files, source, timeout)
                elapsed_time += delta_time
                if len(buffer) == 0: break  # reached end of the stream
                frames.append(buffer)

            # read audio input until the phrase ends
            pause_count, phrase_count = 0, 0
            phrase_start_time = elapsed_time
            while True:
                # handle phrase being too long by cutting off the audio
                elapsed_time += seconds_per_buffer
                if phrase_time_limit and elapsed_time - phrase_start_time > phrase_time_limit:
                    break

                buffer = source.stream.read(source.CHUNK)
                if len(buffer) == 0: break  # reached end of the stream
                frames.append(buffer)
                phrase_count += 1

                # check if speaking has stopped for longer than the pause threshold on the audio input
                energy = audioop.rms(buffer, source.SAMPLE_WIDTH)  # unit energy of the audio signal within the buffer
                if energy > self.energy_threshold:
                    pause_count = 0
                else:
                    pause_count += 1
                if pause_count > pause_buffer_count:  # end of the phrase
                    break

            # check how long the detected phrase is, and retry listening if the phrase is too short
            phrase_count -= pause_count  # exclude the buffers for the pause before the phrase
            if phrase_count >= phrase_buffer_count or len(buffer) == 0: break  # phrase is long enough or we've reached the end of the stream, so stop listening

        # obtain frame data
        for i in range(pause_count - non_speaking_buffer_count): frames.pop()  # remove extra non-speaking frames at the end
        frame_data = b"".join(frames)

        return AudioData(frame_data, source.SAMPLE_RATE, source.SAMPLE_WIDTH)

    def listen_in_background(self, source, callback, phrase_time_limit=None):
        """
        Spawns a thread to repeatedly record phrases from ``source`` (an ``AudioSource`` instance) into an ``AudioData`` instance and call ``callback`` with that ``AudioData`` instance as soon as each phrase are detected.

        Returns a function object that, when called, requests that the background listener thread stop. The background thread is a daemon and will not stop the program from exiting if there are no other non-daemon threads. The function accepts one parameter, ``wait_for_stop``: if truthy, the function will wait for the background listener to stop before returning, otherwise it will return immediately and the background listener thread might still be running for a second or two afterwards. Additionally, if you are using a truthy value for ``wait_for_stop``, you must call the function from the same thread you originally called ``listen_in_background`` from.

        Phrase recognition uses the exact same mechanism as ``recognizer_instance.listen(source)``. The ``phrase_time_limit`` parameter works in the same way as the ``phrase_time_limit`` parameter for ``recognizer_instance.listen(source)``, as well.

        The ``callback`` parameter is a function that should accept two parameters - the ``recognizer_instance``, and an ``AudioData`` instance representing the captured audio. Note that ``callback`` function will be called from a non-main thread.
        """
        assert isinstance(source, AudioSource), "Source must be an audio source"
        running = [True]

        def threaded_listen():
            with source as s:
                while running[0]:
                    try:  # listen for 1 second, then check again if the stop function has been called
                        audio = self.listen(s, 1, phrase_time_limit)
                    except WaitTimeoutError:  # listening timed out, just try again
                        pass
                    else:
                        if running[0]: callback(self, audio)

        def stopper(wait_for_stop=True):
            running[0] = False
            if wait_for_stop:
                listener_thread.join()  # block until the background thread is done, which can take around 1 second

        listener_thread = threading.Thread(target=threaded_listen)
        listener_thread.daemon = True
        listener_thread.start()
        return stopper

    def recognize_sphinx(self, audio_data, language="en-US", keyword_entries=None, grammar=None, show_all=False):
        """
        Performs speech recognition on ``audio_data`` (an ``AudioData`` instance), using CMU Sphinx.

        The recognition language is determined by ``language``, an RFC5646 language tag like ``"en-US"`` or ``"en-GB"``, defaulting to US English. Out of the box, only ``en-US`` is supported. See `Notes on using `PocketSphinx <https://github.com/Uberi/speech_recognition/blob/master/reference/pocketsphinx.rst>`__ for information about installing other languages. This document is also included under ``reference/pocketsphinx.rst``. The ``language`` parameter can also be a tuple of filesystem paths, of the form ``(acoustic_parameters_directory, language_model_file, phoneme_dictionary_file)`` - this allows you to load arbitrary Sphinx models.

        If specified, the keywords to search for are determined by ``keyword_entries``, an iterable of tuples of the form ``(keyword, sensitivity)``, where ``keyword`` is a phrase, and ``sensitivity`` is how sensitive to this phrase the recognizer should be, on a scale of 0 (very insensitive, more false negatives) to 1 (very sensitive, more false positives) inclusive. If not specified or ``None``, no keywords are used and Sphinx will simply transcribe whatever words it recognizes. Specifying ``keyword_entries`` is more accurate than just looking for those same keywords in non-keyword-based transcriptions, because Sphinx knows specifically what sounds to look for.

        Sphinx can also handle FSG or JSGF grammars. The parameter ``grammar`` expects a path to the grammar file. Note that if a JSGF grammar is passed, an FSG grammar will be created at the same location to speed up execution in the next run. If ``keyword_entries`` are passed, content of ``grammar`` will be ignored.

        Returns the most likely transcription if ``show_all`` is false (the default). Otherwise, returns the Sphinx ``pocketsphinx.pocketsphinx.Decoder`` object resulting from the recognition.

        Raises a ``speech_recognition.UnknownValueError`` exception if the speech is unintelligible. Raises a ``speech_recognition.RequestError`` exception if there are any issues with the Sphinx installation.
        """
        assert isinstance(audio_data, AudioData), "``audio_data`` must be audio data"
        assert isinstance(language, str) or (isinstance(language, tuple) and len(language) == 3), "``language`` must be a string or 3-tuple of Sphinx data file paths of the form ``(acoustic_parameters, language_model, phoneme_dictionary)``"
        assert keyword_entries is None or all(isinstance(keyword, (type(""), type(u""))) and 0 <= sensitivity <= 1 for keyword, sensitivity in keyword_entries), "``keyword_entries`` must be ``None`` or a list of pairs of strings and numbers between 0 and 1"

        # import the PocketSphinx speech recognition module
        try:
            from pocketsphinx import pocketsphinx, Jsgf, FsgModel

        except ImportError:
            raise RequestError("missing PocketSphinx module: ensure that PocketSphinx is set up correctly.")
        except ValueError:
            raise RequestError("bad PocketSphinx installation; try reinstalling PocketSphinx version 0.0.9 or better.")
        if not hasattr(pocketsphinx, "Decoder") or not hasattr(pocketsphinx.Decoder, "default_config"):
            raise RequestError("outdated PocketSphinx installation; ensure you have PocketSphinx version 0.0.9 or better.")

        if isinstance(language, str):  # directory containing language data
            language_directory = os.path.join(os.path.dirname(os.path.realpath(__file__)), "pocketsphinx-data", language)
            if not os.path.isdir(language_directory):
                raise RequestError("missing PocketSphinx language data directory: \"{}\"".format(language_directory))
            acoustic_parameters_directory = os.path.join(language_directory, "acoustic-model")
            language_model_file = os.path.join(language_directory, "language-model.lm.bin")
            phoneme_dictionary_file = os.path.join(language_directory, "pronounciation-dictionary.dict")
        else:  # 3-tuple of Sphinx data file paths
            acoustic_parameters_directory, language_model_file, phoneme_dictionary_file = language
        if not os.path.isdir(acoustic_parameters_directory):
            raise RequestError("missing PocketSphinx language model parameters directory: \"{}\"".format(acoustic_parameters_directory))
        if not os.path.isfile(language_model_file):
            raise RequestError("missing PocketSphinx language model file: \"{}\"".format(language_model_file))
        if not os.path.isfile(phoneme_dictionary_file):
            raise RequestError("missing PocketSphinx phoneme dictionary file: \"{}\"".format(phoneme_dictionary_file))

        # create decoder object
        config = pocketsphinx.Decoder.default_config()
        config.set_string("-hmm", acoustic_parameters_directory)  # set the path of the hidden Markov model (HMM) parameter files
        config.set_string("-lm", language_model_file)
        config.set_string("-dict", phoneme_dictionary_file)
        config.set_string("-logfn", os.devnull)  # disable logging (logging causes unwanted output in terminal)
        decoder = pocketsphinx.Decoder(config)

        # obtain audio data
        raw_data = audio_data.get_raw_data(convert_rate=16000, convert_width=2)  # the included language models require audio to be 16-bit mono 16 kHz in little-endian format

        # obtain recognition results
        if keyword_entries is not None:  # explicitly specified set of keywords
            with PortableNamedTemporaryFile("w") as f:
                # generate a keywords file - Sphinx documentation recommendeds sensitivities between 1e-50 and 1e-5
                f.writelines("{} /1e{}/\n".format(keyword, 100 * sensitivity - 110) for keyword, sensitivity in keyword_entries)
                f.flush()

                # perform the speech recognition with the keywords file (this is inside the context manager so the file isn;t deleted until we're done)
                decoder.set_kws("keywords", f.name)
                decoder.set_search("keywords")
                decoder.start_utt()  # begin utterance processing
                decoder.process_raw(raw_data, False, True)  # process audio data with recognition enabled (no_search = False), as a full utterance (full_utt = True)
                decoder.end_utt()  # stop utterance processing
        elif grammar is not None:  # a path to a FSG or JSGF grammar
            if not os.path.exists(grammar):
                raise ValueError("Grammar '{0}' does not exist.".format(grammar))
            grammar_path = os.path.abspath(os.path.dirname(grammar))
            grammar_name = os.path.splitext(os.path.basename(grammar))[0]
            fsg_path = "{0}/{1}.fsg".format(grammar_path, grammar_name)
            if not os.path.exists(fsg_path):  # create FSG grammar if not available
                jsgf = Jsgf(grammar)
                rule = jsgf.get_rule("{0}.{0}".format(grammar_name))
                fsg = jsgf.build_fsg(rule, decoder.get_logmath(), 7.5)
                fsg.writefile(fsg_path)
            else:
                fsg = FsgModel(fsg_path, decoder.get_logmath(), 7.5)
            decoder.set_fsg(grammar_name, fsg)
            decoder.set_search(grammar_name)
            decoder.start_utt()
            decoder.process_raw(raw_data, False, True)  # process audio data with recognition enabled (no_search = False), as a full utterance (full_utt = True)
            decoder.end_utt()  # stop utterance processing
        else:  # no keywords, perform freeform recognition
            decoder.start_utt()  # begin utterance processing
            decoder.process_raw(raw_data, False, True)  # process audio data with recognition enabled (no_search = False), as a full utterance (full_utt = True)
            decoder.end_utt()  # stop utterance processing

        if show_all: return decoder

        # return results
        hypothesis = decoder.hyp()
        if hypothesis is not None: return hypothesis.hypstr
        raise UnknownValueError()  # no transcriptions available

    def recognize_google(self, audio_data, key=None, language="en-US", show_all=False):
        """
        Performs speech recognition on ``audio_data`` (an ``AudioData`` instance), using the Google Speech Recognition API.

        The Google Speech Recognition API key is specified by ``key``. If not specified, it uses a generic key that works out of the box. This should generally be used for personal or testing purposes only, as it **may be revoked by Google at any time**.

        To obtain your own API key, simply following the steps on the `API Keys <http://www.chromium.org/developers/how-tos/api-keys>`__ page at the Chromium Developers site. In the Google Developers Console, Google Speech Recognition is listed as "Speech API".

        The recognition language is determined by ``language``, an RFC5646 language tag like ``"en-US"`` (US English) or ``"fr-FR"`` (International French), defaulting to US English. A list of supported language tags can be found in this `StackOverflow answer <http://stackoverflow.com/a/14302134>`__.

        Returns the most likely transcription if ``show_all`` is false (the default). Otherwise, returns the raw API response as a JSON dictionary.

        Raises a ``speech_recognition.UnknownValueError`` exception if the speech is unintelligible. Raises a ``speech_recognition.RequestError`` exception if the speech recognition operation failed, if the key isn't valid, or if there is no internet connection.
        """
        assert isinstance(audio_data, AudioData), "``audio_data`` must be audio data"
        assert key is None or isinstance(key, str), "``key`` must be ``None`` or a string"
        assert isinstance(language, str), "``language`` must be a string"

        flac_data = audio_data.get_flac_data(
            convert_rate=None if audio_data.sample_rate >= 8000 else 8000,  # audio samples must be at least 8 kHz
            convert_width=2  # audio samples must be 16-bit
        )
        if key is None: key = "AIzaSyBOti4mM-6x9WDnZIjIeyEU21OpBXqWBgw"
        url = "http://www.google.com/speech-api/v2/recognize?{}".format(urlencode({
            "client": "chromium",
            "lang": language,
            "key": key,
        }))
        request = Request(url, data=flac_data, headers={"Content-Type": "audio/x-flac; rate={}".format(audio_data.sample_rate)})

        # obtain audio transcription results
        try:
            response = urlopen(request, timeout=self.operation_timeout)
        except HTTPError as e:
            raise RequestError("recognition request failed: {}".format(e.reason))
        except URLError as e:
            raise RequestError("recognition connection failed: {}".format(e.reason))
        response_text = response.read().decode("utf-8")

        # ignore any blank blocks
        actual_result = []
        for line in response_text.split("\n"):
            if not line: continue
            result = json.loads(line)["result"]
            if len(result) != 0:
                actual_result = result[0]
                break

        # return results
        if show_all: return actual_result
        if not isinstance(actual_result, dict) or len(actual_result.get("alternative", [])) == 0: raise UnknownValueError()

        if "confidence" in actual_result["alternative"]:
            # return alternative with highest confidence score
            best_hypothesis = max(actual_result["alternative"], key=lambda alternative: alternative["confidence"])
        else:
            # when there is no confidence available, we arbitrarily choose the first hypothesis.
            best_hypothesis = actual_result["alternative"][0]
        if "transcript" not in best_hypothesis: raise UnknownValueError()
        return best_hypothesis["transcript"]

    def recognize_google_cloud(self, audio_data, credentials_json=None, language="en-US", preferred_phrases=None, show_all=False):
        """
        Performs speech recognition on ``audio_data`` (an ``AudioData`` instance), using the Google Cloud Speech API.

        This function requires a Google Cloud Platform account; see the `Google Cloud Speech API Quickstart <https://cloud.google.com/speech/docs/getting-started>`__ for details and instructions. Basically, create a project, enable billing for the project, enable the Google Cloud Speech API for the project, and set up Service Account Key credentials for the project. The result is a JSON file containing the API credentials. The text content of this JSON file is specified by ``credentials_json``. If not specified, the library will try to automatically `find the default API credentials JSON file <https://developers.google.com/identity/protocols/application-default-credentials>`__.

        The recognition language is determined by ``language``, which is a BCP-47 language tag like ``"en-US"`` (US English). A list of supported language tags can be found in the `Google Cloud Speech API documentation <https://cloud.google.com/speech/docs/languages>`__.

        If ``preferred_phrases`` is an iterable of phrase strings, those given phrases will be more likely to be recognized over similar-sounding alternatives. This is useful for things like keyword/command recognition or adding new phrases that aren't in Google's vocabulary. Note that the API imposes certain `restrictions on the list of phrase strings <https://cloud.google.com/speech/limits#content>`__.

        Returns the most likely transcription if ``show_all`` is False (the default). Otherwise, returns the raw API response as a JSON dictionary.

        Raises a ``speech_recognition.UnknownValueError`` exception if the speech is unintelligible. Raises a ``speech_recognition.RequestError`` exception if the speech recognition operation failed, if the credentials aren't valid, or if there is no Internet connection.
        """
        assert isinstance(audio_data, AudioData), "``audio_data`` must be audio data"
        if credentials_json is not None:
            try: json.loads(credentials_json)
            except Exception: raise AssertionError("``credentials_json`` must be ``None`` or a valid JSON string")
        assert isinstance(language, str), "``language`` must be a string"
        assert preferred_phrases is None or all(isinstance(preferred_phrases, (type(""), type(u""))) for preferred_phrases in preferred_phrases), "``preferred_phrases`` must be a list of strings"

        # See https://cloud.google.com/speech/reference/rest/v1/RecognitionConfig
        flac_data = audio_data.get_flac_data(
            convert_rate=None if 8000 <= audio_data.sample_rate <= 48000 else max(8000, min(audio_data.sample_rate, 48000)),  # audio sample rate must be between 8 kHz and 48 kHz inclusive - clamp sample rate into this range
            convert_width=2  # audio samples must be 16-bit
        )

        try:
            from oauth2client.client import GoogleCredentials
            from googleapiclient.discovery import build
            import googleapiclient.errors

            # cannot simply use 'http = httplib2.Http(timeout=self.operation_timeout)'
            # because discovery.build() says 'Arguments http and credentials are mutually exclusive'
            import socket
            import googleapiclient.http
            if self.operation_timeout and socket.getdefaulttimeout() is None:
                # override constant (used by googleapiclient.http.build_http())
                googleapiclient.http.DEFAULT_HTTP_TIMEOUT_SEC = self.operation_timeout

            if credentials_json is None:
                api_credentials = GoogleCredentials.get_application_default()
            else:
                # the credentials can only be read from a file, so we'll make a temp file and write in the contents to work around that
                with PortableNamedTemporaryFile("w") as f:
                    f.write(credentials_json)
                    f.flush()
                    api_credentials = GoogleCredentials.from_stream(f.name)

            speech_service = build("speech", "v1", credentials=api_credentials, cache_discovery=False)
        except ImportError:
            raise RequestError("missing google-api-python-client module: ensure that google-api-python-client is set up correctly.")

        speech_config = {"encoding": "FLAC", "sampleRateHertz": audio_data.sample_rate, "languageCode": language}
        if preferred_phrases is not None:
            speech_config["speechContext"] = {"phrases": preferred_phrases}
        if show_all:
            speech_config["enableWordTimeOffsets"] = True  # some useful extra options for when we want all the output
        request = speech_service.speech().recognize(body={"audio": {"content": base64.b64encode(flac_data).decode("utf8")}, "config": speech_config})

        try:
            response = request.execute()
        except googleapiclient.errors.HttpError as e:
            raise RequestError(e)
        except URLError as e:
            raise RequestError("recognition connection failed: {0}".format(e.reason))

        if show_all: return response
        if "results" not in response or len(response["results"]) == 0: raise UnknownValueError()
        transcript = ""
        for result in response["results"]:
            transcript += result["alternatives"][0]["transcript"].strip() + " "

        return transcript

    def recognize_wit(self, audio_data, key, show_all=False):
        """
        Performs speech recognition on ``audio_data`` (an ``AudioData`` instance), using the Wit.ai API.

        The Wit.ai API key is specified by ``key``. Unfortunately, these are not available without `signing up for an account <https://wit.ai/>`__ and creating an app. You will need to add at least one intent to the app before you can see the API key, though the actual intent settings don't matter.

        To get the API key for a Wit.ai app, go to the app's overview page, go to the section titled "Make an API request", and look for something along the lines of ``Authorization: Bearer XXXXXXXXXXXXXXXXXXXXXXXXXXXXXXXX``; ``XXXXXXXXXXXXXXXXXXXXXXXXXXXXXXXX`` is the API key. Wit.ai API keys are 32-character uppercase alphanumeric strings.

        The recognition language is configured in the Wit.ai app settings.

        Returns the most likely transcription if ``show_all`` is false (the default). Otherwise, returns the `raw API response <https://wit.ai/docs/http/20141022#get-intent-via-text-link>`__ as a JSON dictionary.

        Raises a ``speech_recognition.UnknownValueError`` exception if the speech is unintelligible. Raises a ``speech_recognition.RequestError`` exception if the speech recognition operation failed, if the key isn't valid, or if there is no internet connection.
        """
        assert isinstance(audio_data, AudioData), "Data must be audio data"
        assert isinstance(key, str), "``key`` must be a string"

        wav_data = audio_data.get_wav_data(
            convert_rate=None if audio_data.sample_rate >= 8000 else 8000,  # audio samples must be at least 8 kHz
            convert_width=2  # audio samples should be 16-bit
        )
        url = "https://api.wit.ai/speech?v=20160526"
        request = Request(url, data=wav_data, headers={"Authorization": "Bearer {}".format(key), "Content-Type": "audio/wav"})
        try:
            response = urlopen(request, timeout=self.operation_timeout)
        except HTTPError as e:
            raise RequestError("recognition request failed: {}".format(e.reason))
        except URLError as e:
            raise RequestError("recognition connection failed: {}".format(e.reason))
        response_text = response.read().decode("utf-8")
        result = json.loads(response_text)

        # return results
        if show_all: return result
        if "_text" not in result or result["_text"] is None: raise UnknownValueError()
        return result["_text"]

    def recognize_bing(self, audio_data, key, language="en-US", show_all=False):
        """
        Performs speech recognition on ``audio_data`` (an ``AudioData`` instance), using the Microsoft Bing Speech API.

        The Microsoft Bing Speech API key is specified by ``key``. Unfortunately, these are not available without `signing up for an account <https://azure.microsoft.com/en-ca/pricing/details/cognitive-services/speech-api/>`__ with Microsoft Azure.

        To get the API key, go to the `Microsoft Azure Portal Resources <https://portal.azure.com/>`__ page, go to "All Resources" > "Add" > "See All" > Search "Bing Speech API > "Create", and fill in the form to make a "Bing Speech API" resource. On the resulting page (which is also accessible from the "All Resources" page in the Azure Portal), go to the "Show Access Keys" page, which will have two API keys, either of which can be used for the `key` parameter. Microsoft Bing Speech API keys are 32-character lowercase hexadecimal strings.

        The recognition language is determined by ``language``, a BCP-47 language tag like ``"en-US"`` (US English) or ``"fr-FR"`` (International French), defaulting to US English. A list of supported language values can be found in the `API documentation <https://docs.microsoft.com/en-us/azure/cognitive-services/speech/api-reference-rest/bingvoicerecognition#recognition-language>`__ under "Interactive and dictation mode".

        Returns the most likely transcription if ``show_all`` is false (the default). Otherwise, returns the `raw API response <https://docs.microsoft.com/en-us/azure/cognitive-services/speech/api-reference-rest/bingvoicerecognition#sample-responses>`__ as a JSON dictionary.

        Raises a ``speech_recognition.UnknownValueError`` exception if the speech is unintelligible. Raises a ``speech_recognition.RequestError`` exception if the speech recognition operation failed, if the key isn't valid, or if there is no internet connection.
        """
        assert isinstance(audio_data, AudioData), "Data must be audio data"
        assert isinstance(key, str), "``key`` must be a string"
        assert isinstance(language, str), "``language`` must be a string"

        access_token, expire_time = getattr(self, "bing_cached_access_token", None), getattr(self, "bing_cached_access_token_expiry", None)
        allow_caching = True
        try:
            from time import monotonic  # we need monotonic time to avoid being affected by system clock changes, but this is only available in Python 3.3+
        except ImportError:
            try:
                from monotonic import monotonic  # use time.monotonic backport for Python 2 if available (from https://pypi.python.org/pypi/monotonic)
            except (ImportError, RuntimeError):
                expire_time = None  # monotonic time not available, don't cache access tokens
                allow_caching = False  # don't allow caching, since monotonic time isn't available
        if expire_time is None or monotonic() > expire_time:  # caching not enabled, first credential request, or the access token from the previous one expired
            # get an access token using OAuth
            credential_url = "https://api.cognitive.microsoft.com/sts/v1.0/issueToken"
            credential_request = Request(credential_url, data=b"", headers={
                "Content-type": "application/x-www-form-urlencoded",
                "Content-Length": "0",
                "Ocp-Apim-Subscription-Key": key,
            })

            if allow_caching:
                start_time = monotonic()

            try:
                credential_response = urlopen(credential_request, timeout=60)  # credential response can take longer, use longer timeout instead of default one
            except HTTPError as e:
                raise RequestError("credential request failed: {}".format(e.reason))
            except URLError as e:
                raise RequestError("credential connection failed: {}".format(e.reason))
            access_token = credential_response.read().decode("utf-8")

            if allow_caching:
                # save the token for the duration it is valid for
                self.bing_cached_access_token = access_token
                self.bing_cached_access_token_expiry = start_time + 600  # according to https://docs.microsoft.com/en-us/azure/cognitive-services/speech/api-reference-rest/bingvoicerecognition, the token expires in exactly 10 minutes

        wav_data = audio_data.get_wav_data(
            convert_rate=16000,  # audio samples must be 8kHz or 16 kHz
            convert_width=2  # audio samples should be 16-bit
        )

        url = "https://speech.platform.bing.com/speech/recognition/interactive/cognitiveservices/v1?{}".format(urlencode({
            "language": language,
            "locale": language,
            "requestid": uuid.uuid4(),
        }))

        if sys.version_info >= (3, 6):  # chunked-transfer requests are only supported in the standard library as of Python 3.6+, use it if possible
            request = Request(url, data=io.BytesIO(wav_data), headers={
                "Authorization": "Bearer {}".format(access_token),
                "Content-type": "audio/wav; codec=\"audio/pcm\"; samplerate=16000",
                "Transfer-Encoding": "chunked",
            })
        else:  # fall back on manually formatting the POST body as a chunked request
            ascii_hex_data_length = "{:X}".format(len(wav_data)).encode("utf-8")
            chunked_transfer_encoding_data = ascii_hex_data_length + b"\r\n" + wav_data + b"\r\n0\r\n\r\n"
            request = Request(url, data=chunked_transfer_encoding_data, headers={
                "Authorization": "Bearer {}".format(access_token),
                "Content-type": "audio/wav; codec=\"audio/pcm\"; samplerate=16000",
                "Transfer-Encoding": "chunked",
            })

        try:
            response = urlopen(request, timeout=self.operation_timeout)
        except HTTPError as e:
            raise RequestError("recognition request failed: {}".format(e.reason))
        except URLError as e:
            raise RequestError("recognition connection failed: {}".format(e.reason))
        response_text = response.read().decode("utf-8")
        result = json.loads(response_text)

        # return results
        if show_all: return result
        if "RecognitionStatus" not in result or result["RecognitionStatus"] != "Success" or "DisplayText" not in result: raise UnknownValueError()
        return result["DisplayText"]

    def recognize_houndify(self, audio_data, client_id, client_key, show_all=False):
        """
        Performs speech recognition on ``audio_data`` (an ``AudioData`` instance), using the Houndify API.

        The Houndify client ID and client key are specified by ``client_id`` and ``client_key``, respectively. Unfortunately, these are not available without `signing up for an account <https://www.houndify.com/signup>`__. Once logged into the `dashboard <https://www.houndify.com/dashboard>`__, you will want to select "Register a new client", and fill in the form as necessary. When at the "Enable Domains" page, enable the "Speech To Text Only" domain, and then select "Save & Continue".

        To get the client ID and client key for a Houndify client, go to the `dashboard <https://www.houndify.com/dashboard>`__ and select the client's "View Details" link. On the resulting page, the client ID and client key will be visible. Client IDs and client keys are both Base64-encoded strings.

        Currently, only English is supported as a recognition language.

        Returns the most likely transcription if ``show_all`` is false (the default). Otherwise, returns the raw API response as a JSON dictionary.

        Raises a ``speech_recognition.UnknownValueError`` exception if the speech is unintelligible. Raises a ``speech_recognition.RequestError`` exception if the speech recognition operation failed, if the key isn't valid, or if there is no internet connection.
        """
        assert isinstance(audio_data, AudioData), "Data must be audio data"
        assert isinstance(client_id, str), "``client_id`` must be a string"
        assert isinstance(client_key, str), "``client_key`` must be a string"

        wav_data = audio_data.get_wav_data(
            convert_rate=None if audio_data.sample_rate in [8000, 16000] else 16000,  # audio samples must be 8 kHz or 16 kHz
            convert_width=2  # audio samples should be 16-bit
        )
        url = "https://api.houndify.com/v1/audio"
        user_id, request_id = str(uuid.uuid4()), str(uuid.uuid4())
        request_time = str(int(time.time()))
        request_signature = base64.urlsafe_b64encode(
            hmac.new(
                base64.urlsafe_b64decode(client_key),
                user_id.encode("utf-8") + b";" + request_id.encode("utf-8") + request_time.encode("utf-8"),
                hashlib.sha256
            ).digest()  # get the HMAC digest as bytes
        ).decode("utf-8")
        request = Request(url, data=wav_data, headers={
            "Content-Type": "application/json",
            "Hound-Request-Info": json.dumps({"ClientID": client_id, "UserID": user_id}),
            "Hound-Request-Authentication": "{};{}".format(user_id, request_id),
            "Hound-Client-Authentication": "{};{};{}".format(client_id, request_time, request_signature)
        })
        try:
            response = urlopen(request, timeout=self.operation_timeout)
        except HTTPError as e:
            raise RequestError("recognition request failed: {}".format(e.reason))
        except URLError as e:
            raise RequestError("recognition connection failed: {}".format(e.reason))
        response_text = response.read().decode("utf-8")
        result = json.loads(response_text)

        # return results
        if show_all: return result
        if "Disambiguation" not in result or result["Disambiguation"] is None:
            raise UnknownValueError()
        return result['Disambiguation']['ChoiceData'][0]['Transcription']

    def recognize_ibm(self, audio_data, username, password, language="en-US", show_all=False):
        """
        Performs speech recognition on ``audio_data`` (an ``AudioData`` instance), using the IBM Speech to Text API.

        The IBM Speech to Text username and password are specified by ``username`` and ``password``, respectively. Unfortunately, these are not available without `signing up for an account <https://console.ng.bluemix.net/registration/>`__. Once logged into the Bluemix console, follow the instructions for `creating an IBM Watson service instance <https://www.ibm.com/watson/developercloud/doc/getting_started/gs-credentials.shtml>`__, where the Watson service is "Speech To Text". IBM Speech to Text usernames are strings of the form XXXXXXXX-XXXX-XXXX-XXXX-XXXXXXXXXXXX, while passwords are mixed-case alphanumeric strings.

        The recognition language is determined by ``language``, an RFC5646 language tag with a dialect like ``"en-US"`` (US English) or ``"zh-CN"`` (Mandarin Chinese), defaulting to US English. The supported language values are listed under the ``model`` parameter of the `audio recognition API documentation <https://www.ibm.com/watson/developercloud/speech-to-text/api/v1/#sessionless_methods>`__, in the form ``LANGUAGE_BroadbandModel``, where ``LANGUAGE`` is the language value.

        Returns the most likely transcription if ``show_all`` is false (the default). Otherwise, returns the `raw API response <https://www.ibm.com/watson/developercloud/speech-to-text/api/v1/#sessionless_methods>`__ as a JSON dictionary.

        Raises a ``speech_recognition.UnknownValueError`` exception if the speech is unintelligible. Raises a ``speech_recognition.RequestError`` exception if the speech recognition operation failed, if the key isn't valid, or if there is no internet connection.
        """
        assert isinstance(audio_data, AudioData), "Data must be audio data"
        assert isinstance(username, str), "``username`` must be a string"
        assert isinstance(password, str), "``password`` must be a string"

        flac_data = audio_data.get_flac_data(
            convert_rate=None if audio_data.sample_rate >= 16000 else 16000,  # audio samples should be at least 16 kHz
            convert_width=None if audio_data.sample_width >= 2 else 2  # audio samples should be at least 16-bit
        )
        url = "https://stream.watsonplatform.net/speech-to-text/api/v1/recognize?{}".format(urlencode({
            "profanity_filter": "false",
            "model": "{}_BroadbandModel".format(language),
            "inactivity_timeout": -1,  # don't stop recognizing when the audio stream activity stops
        }))
        request = Request(url, data=flac_data, headers={
            "Content-Type": "audio/x-flac",
            "X-Watson-Learning-Opt-Out": "true",  # prevent requests from being logged, for improved privacy
        })
        authorization_value = base64.standard_b64encode("{}:{}".format(username, password).encode("utf-8")).decode("utf-8")
        request.add_header("Authorization", "Basic {}".format(authorization_value))
        try:
            response = urlopen(request, timeout=self.operation_timeout)
        except HTTPError as e:
            raise RequestError("recognition request failed: {}".format(e.reason))
        except URLError as e:
            raise RequestError("recognition connection failed: {}".format(e.reason))
        response_text = response.read().decode("utf-8")
        result = json.loads(response_text)

        # return results
        if show_all: return result
        if "results" not in result or len(result["results"]) < 1 or "alternatives" not in result["results"][0]:
            raise UnknownValueError()

        transcription = []
        for utterance in result["results"]:
            if "alternatives" not in utterance: raise UnknownValueError()
            for hypothesis in utterance["alternatives"]:
                if "transcript" in hypothesis:
                    transcription.append(hypothesis["transcript"])
        return "\n".join(transcription)


def get_flac_converter():
    """Returns the absolute path of a FLAC converter executable, or raises an OSError if none can be found."""
    flac_converter = shutil_which("flac")  # check for installed version first
    if flac_converter is None:  # flac utility is not installed
        base_path = os.path.dirname(os.path.abspath(__file__))  # directory of the current module file, where all the FLAC bundled binaries are stored
        system, machine = platform.system(), platform.machine()
        if system == "Windows" and machine in {"i686", "i786", "x86", "x86_64", "AMD64"}:
            flac_converter = os.path.join(base_path, "flac-win32.exe")
        elif system == "Darwin" and machine in {"i686", "i786", "x86", "x86_64", "AMD64"}:
            flac_converter = os.path.join(base_path, "flac-mac")
        elif system == "Linux" and machine in {"i686", "i786", "x86"}:
            flac_converter = os.path.join(base_path, "flac-linux-x86")
        elif system == "Linux" and machine in {"x86_64", "AMD64"}:
            flac_converter = os.path.join(base_path, "flac-linux-x86_64")
        else:  # no FLAC converter available
            raise OSError("FLAC conversion utility not available - consider installing the FLAC command line application by running `apt-get install flac` or your operating system's equivalent")

    # mark FLAC converter as executable if possible
    try:
        # handle known issue when running on docker:
        # run executable right after chmod() may result in OSError "Text file busy"
        # fix: flush FS with sync
        if not os.access(flac_converter, os.X_OK):
            stat_info = os.stat(flac_converter)
            os.chmod(flac_converter, stat_info.st_mode | stat.S_IEXEC)
            if 'Linux' in platform.system():
                os.sync() if sys.version_info >= (3, 3) else os.system('sync')

    except OSError: pass

    return flac_converter


def shutil_which(pgm):
    """Python 2 compatibility: backport of ``shutil.which()`` from Python 3"""
    path = os.getenv('PATH')
    for p in path.split(os.path.pathsep):
        p = os.path.join(p, pgm)
        if os.path.exists(p) and os.access(p, os.X_OK):
            return p


class PortableNamedTemporaryFile(object):
    """Limited replacement for ``tempfile.NamedTemporaryFile``, except unlike ``tempfile.NamedTemporaryFile``, the file can be opened again while it's currently open, even on Windows."""
    def __init__(self, mode="w+b"):
        self.mode = mode

    def __enter__(self):
        # create the temporary file and open it
        import tempfile
        file_descriptor, file_path = tempfile.mkstemp()
        self._file = os.fdopen(file_descriptor, self.mode)

        # the name property is a public field
        self.name = file_path
        return self

    def __exit__(self, exc_type, exc_value, traceback):
        self._file.close()
        os.remove(self.name)

    def write(self, *args, **kwargs):
        return self._file.write(*args, **kwargs)

    def writelines(self, *args, **kwargs):
        return self._file.writelines(*args, **kwargs)

    def flush(self, *args, **kwargs):
        return self._file.flush(*args, **kwargs)


# ===============================
#  backwards compatibility shims
# ===============================

WavFile = AudioFile  # WavFile was renamed to AudioFile in 3.4.1


def recognize_api(self, audio_data, client_access_token, language="en", session_id=None, show_all=False):
    wav_data = audio_data.get_wav_data(convert_rate=16000, convert_width=2)
    url = "https://api.api.ai/v1/query"
    while True:
        boundary = uuid.uuid4().hex
        if boundary.encode("utf-8") not in wav_data: break
    if session_id is None: session_id = uuid.uuid4().hex
    data = b"--" + boundary.encode("utf-8") + b"\r\n" + b"Content-Disposition: form-data; name=\"request\"\r\n" + b"Content-Type: application/json\r\n" + b"\r\n" + b"{\"v\": \"20150910\", \"sessionId\": \"" + session_id.encode("utf-8") + b"\", \"lang\": \"" + language.encode("utf-8") + b"\"}\r\n" + b"--" + boundary.encode("utf-8") + b"\r\n" + b"Content-Disposition: form-data; name=\"voiceData\"; filename=\"audio.wav\"\r\n" + b"Content-Type: audio/wav\r\n" + b"\r\n" + wav_data + b"\r\n" + b"--" + boundary.encode("utf-8") + b"--\r\n"
    request = Request(url, data=data, headers={"Authorization": "Bearer {}".format(client_access_token), "Content-Length": str(len(data)), "Expect": "100-continue", "Content-Type": "multipart/form-data; boundary={}".format(boundary)})
    try: response = urlopen(request, timeout=10)
    except HTTPError as e: raise RequestError("recognition request failed: {}".format(e.reason))
    except URLError as e: raise RequestError("recognition connection failed: {}".format(e.reason))
    response_text = response.read().decode("utf-8")
    result = json.loads(response_text)
    if show_all: return result
    if "status" not in result or "errorType" not in result["status"] or result["status"]["errorType"] != "success":
        raise UnknownValueError()
    return result["result"]["resolvedQuery"]


Recognizer.recognize_api = classmethod(recognize_api)  # API.AI Speech Recognition is deprecated/not recommended as of 3.5.0, and currently is only optionally available for paid plans
