ecoute/speech_recognition/audio.py

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import aifc
import audioop
import io
import os
import platform
import stat
import subprocess
import sys
import wave
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
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def get_wav_data(self, convert_rate=None, convert_width=None, nchannels = 1):
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"""
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)
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wav_writer.setnchannels(nchannels)
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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 convert 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
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