Python audioop.bias方法代码示例

# 需要导入模块: import audioop [as 别名]
# 或者: from audioop import bias [as 别名]
def test_issue7673(self):
        state = None
        for data, size in INVALID_DATA:
            size2 = size
            self.assertRaises(audioop.error, audioop.getsample, data, size, 0)
            self.assertRaises(audioop.error, audioop.max, data, size)
            self.assertRaises(audioop.error, audioop.minmax, data, size)
            self.assertRaises(audioop.error, audioop.avg, data, size)
            self.assertRaises(audioop.error, audioop.rms, data, size)
            self.assertRaises(audioop.error, audioop.avgpp, data, size)
            self.assertRaises(audioop.error, audioop.maxpp, data, size)
            self.assertRaises(audioop.error, audioop.cross, data, size)
            self.assertRaises(audioop.error, audioop.mul, data, size, 1.0)
            self.assertRaises(audioop.error, audioop.tomono, data, size, 0.5, 0.5)
            self.assertRaises(audioop.error, audioop.tostereo, data, size, 0.5, 0.5)
            self.assertRaises(audioop.error, audioop.add, data, data, size)
            self.assertRaises(audioop.error, audioop.bias, data, size, 0)
            self.assertRaises(audioop.error, audioop.reverse, data, size)
            self.assertRaises(audioop.error, audioop.lin2lin, data, size, size2)
            self.assertRaises(audioop.error, audioop.ratecv, data, size, 1, 1, 1, state)
            self.assertRaises(audioop.error, audioop.lin2ulaw, data, size)
            self.assertRaises(audioop.error, audioop.lin2alaw, data, size)
            self.assertRaises(audioop.error, audioop.lin2adpcm, data, size, state) 

# 需要导入模块: import audioop [as 别名]
# 或者: from audioop import bias [as 别名]
def test_string(self):
        data = 'abcd'
        size = 2
        self.assertRaises(TypeError, audioop.getsample, data, size, 0)
        self.assertRaises(TypeError, audioop.max, data, size)
        self.assertRaises(TypeError, audioop.minmax, data, size)
        self.assertRaises(TypeError, audioop.avg, data, size)
        self.assertRaises(TypeError, audioop.rms, data, size)
        self.assertRaises(TypeError, audioop.avgpp, data, size)
        self.assertRaises(TypeError, audioop.maxpp, data, size)
        self.assertRaises(TypeError, audioop.cross, data, size)
        self.assertRaises(TypeError, audioop.mul, data, size, 1.0)
        self.assertRaises(TypeError, audioop.tomono, data, size, 0.5, 0.5)
        self.assertRaises(TypeError, audioop.tostereo, data, size, 0.5, 0.5)
        self.assertRaises(TypeError, audioop.add, data, data, size)
        self.assertRaises(TypeError, audioop.bias, data, size, 0)
        self.assertRaises(TypeError, audioop.reverse, data, size)
        self.assertRaises(TypeError, audioop.lin2lin, data, size, size)
        self.assertRaises(TypeError, audioop.ratecv, data, size, 1, 1, 1, None)
        self.assertRaises(TypeError, audioop.lin2ulaw, data, size)
        self.assertRaises(TypeError, audioop.lin2alaw, data, size)
        self.assertRaises(TypeError, audioop.lin2adpcm, data, size, None) 

# 需要导入模块: import audioop [as 别名]
# 或者: from audioop import bias [as 别名]
def test_bias(self):
        for w in 1, 2, 4:
            for bias in 0, 1, -1, 127, -128, 0x7fffffff, -0x80000000:
                self.assertEqual(audioop.bias(b'', w, bias), b'')
        self.assertEqual(audioop.bias(datas[1], 1, 1),
                         b'\x01\x13\x46\xbc\x80\x81\x00')
        self.assertEqual(audioop.bias(datas[1], 1, -1),
                         b'\xff\x11\x44\xba\x7e\x7f\xfe')
        self.assertEqual(audioop.bias(datas[1], 1, 0x7fffffff),
                         b'\xff\x11\x44\xba\x7e\x7f\xfe')
        self.assertEqual(audioop.bias(datas[1], 1, -0x80000000),
                         datas[1])
        self.assertEqual(audioop.bias(datas[2], 2, 1),
                packs[2](1, 0x1235, 0x4568, -0x4566, -0x8000, -0x7fff, 0))
        self.assertEqual(audioop.bias(datas[2], 2, -1),
                packs[2](-1, 0x1233, 0x4566, -0x4568, 0x7ffe, 0x7fff, -2))
        self.assertEqual(audioop.bias(datas[2], 2, 0x7fffffff),
                packs[2](-1, 0x1233, 0x4566, -0x4568, 0x7ffe, 0x7fff, -2))
        self.assertEqual(audioop.bias(datas[2], 2, -0x80000000),
                datas[2])
        self.assertEqual(audioop.bias(datas[4], 4, 1),
                packs[4](1, 0x12345679, 0x456789ac, -0x456789aa,
                         -0x80000000, -0x7fffffff, 0))
        self.assertEqual(audioop.bias(datas[4], 4, -1),
                packs[4](-1, 0x12345677, 0x456789aa, -0x456789ac,
                         0x7ffffffe, 0x7fffffff, -2))
        self.assertEqual(audioop.bias(datas[4], 4, 0x7fffffff),
                packs[4](0x7fffffff, -0x6dcba989, -0x3a987656, 0x3a987654,
                         -2, -1, 0x7ffffffe))
        self.assertEqual(audioop.bias(datas[4], 4, -0x80000000),
                packs[4](-0x80000000, -0x6dcba988, -0x3a987655, 0x3a987655,
                         -1, 0, 0x7fffffff)) 

# 需要导入模块: import audioop [as 别名]
# 或者: from audioop import bias [as 别名]
def bias(self, bias: int) -> 'Sample':
        """Add a bias constant to each sample value."""
        if self.__locked:
            raise RuntimeError("cannot modify a locked sample")
        self.__frames = audioop.bias(self.__frames, self.__samplewidth, bias)
        return self 

# 需要导入模块: import audioop [as 别名]
# 或者: from audioop import bias [as 别名]
def testbias(data):
    if verbose:
        print 'bias'
    # Note: this test assumes that avg() works
    d1 = audioop.bias(data[0], 1, 100)
    d2 = audioop.bias(data[1], 2, 100)
    d4 = audioop.bias(data[2], 4, 100)
    if audioop.avg(d1, 1) != 101 or \
              audioop.avg(d2, 2) != 101 or \
              audioop.avg(d4, 4) != 101:
        return 0
    return 1 

# 需要导入模块: import audioop [as 别名]
# 或者: from audioop import bias [as 别名]
def convert_sampwidth(fragment, sampwidth_in, sampwidth_out):
    """
    Convert the sampwidth (byte width) of the input fragment between 1-, 2-, 3-, 4-byte formats.

    Parameters
    ----------
    fragment : bytes object
        Specifies the original fragment.
    sampwidth_in : int
        Specifies the fragment's original sampwidth.
    sampwidth_out : int
        Specifies the fragment's desired sampwidth.

    Returns
    -------
    bytes
        Converted audio with the desired sampwidth 'sampwidth_out'.

    """
    if sampwidth_in == sampwidth_out:
        return fragment

    # In .wav files, 16, 24, and 32 bit samples are signed, 8 bit samples are unsigned.
    # So when converting from 8 bit wide samples, you need to also subtract 128 from the sample.
    # Similarly, when converting to 8 bit wide samples, you need to also add 128 to the result.
    if sampwidth_in == 1:
        new_fragment = audioop.bias(fragment, 1, -128)
    else:
        new_fragment = fragment
    new_fragment = audioop.lin2lin(new_fragment, sampwidth_in, sampwidth_out)
    if sampwidth_out == 1:
        new_fragment = audioop.bias(new_fragment, 1, 128)
    return new_fragment 

# 需要导入模块: import audioop [as 别名]
# 或者: from audioop import bias [as 别名]
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 

# 需要导入模块: import audioop [as 别名]
# 或者: from audioop import bias [as 别名]
def test_bias(self):
        for w in 1, 2, 3, 4:
            for bias in 0, 1, -1, 127, -128, 0x7fffffff, -0x80000000:
                self.assertEqual(audioop.bias(b'', w, bias), b'')
                self.assertEqual(audioop.bias(bytearray(), w, bias), b'')
                self.assertEqual(audioop.bias(memoryview(b''), w, bias), b'')
        self.assertEqual(audioop.bias(datas[1], 1, 1),
                         b'\x01\x13\x46\xbc\x80\x81\x00')
        self.assertEqual(audioop.bias(datas[1], 1, -1),
                         b'\xff\x11\x44\xba\x7e\x7f\xfe')
        self.assertEqual(audioop.bias(datas[1], 1, 0x7fffffff),
                         b'\xff\x11\x44\xba\x7e\x7f\xfe')
        self.assertEqual(audioop.bias(datas[1], 1, -0x80000000),
                         datas[1])
        self.assertEqual(audioop.bias(datas[2], 2, 1),
                packs[2](1, 0x1235, 0x4568, -0x4566, -0x8000, -0x7fff, 0))
        self.assertEqual(audioop.bias(datas[2], 2, -1),
                packs[2](-1, 0x1233, 0x4566, -0x4568, 0x7ffe, 0x7fff, -2))
        self.assertEqual(audioop.bias(datas[2], 2, 0x7fffffff),
                packs[2](-1, 0x1233, 0x4566, -0x4568, 0x7ffe, 0x7fff, -2))
        self.assertEqual(audioop.bias(datas[2], 2, -0x80000000),
                datas[2])
        self.assertEqual(audioop.bias(datas[3], 3, 1),
                packs[3](1, 0x123457, 0x45678a, -0x456788,
                         -0x800000, -0x7fffff, 0))
        self.assertEqual(audioop.bias(datas[3], 3, -1),
                packs[3](-1, 0x123455, 0x456788, -0x45678a,
                         0x7ffffe, 0x7fffff, -2))
        self.assertEqual(audioop.bias(datas[3], 3, 0x7fffffff),
                packs[3](-1, 0x123455, 0x456788, -0x45678a,
                         0x7ffffe, 0x7fffff, -2))
        self.assertEqual(audioop.bias(datas[3], 3, -0x80000000),
                datas[3])
        self.assertEqual(audioop.bias(datas[4], 4, 1),
                packs[4](1, 0x12345679, 0x456789ac, -0x456789aa,
                         -0x80000000, -0x7fffffff, 0))
        self.assertEqual(audioop.bias(datas[4], 4, -1),
                packs[4](-1, 0x12345677, 0x456789aa, -0x456789ac,
                         0x7ffffffe, 0x7fffffff, -2))
        self.assertEqual(audioop.bias(datas[4], 4, 0x7fffffff),
                packs[4](0x7fffffff, -0x6dcba989, -0x3a987656, 0x3a987654,
                         -2, -1, 0x7ffffffe))
        self.assertEqual(audioop.bias(datas[4], 4, -0x80000000),
                packs[4](-0x80000000, -0x6dcba988, -0x3a987655, 0x3a987655,
                         -1, 0, 0x7fffffff)) 

# 需要导入模块: import audioop [as 别名]
# 或者: from audioop import bias [as 别名]
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:
            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:
                # 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()
                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)
                aiff_data, stderr = process.communicate(flac_data)
                aiff_file = io.BytesIO(aiff_data)
                try:
                    self.audio_reader = aifc.open(aiff_file, "rb")
                except aifc.Error:
                    assert False, "Audio file could not be read as WAV, AIFF, or FLAC; check if file is corrupted"
                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