def set_input_format(self, file_type=None, rate=None, bits=None,<EOL>channels=None, encoding=None, ignore_length=False):
if file_type not in VALID_FORMATS + [None]:<EOL><INDENT>raise ValueError(<EOL>'<STR_LIT>'.format(VALID_FORMATS)<EOL>)<EOL><DEDENT>if not is_number(rate) and rate is not None:<EOL><INDENT>raise ValueError('<STR_LIT>')<EOL><DEDENT>if rate is not None and rate <= <NUM_LIT:0>:<EOL><INDENT>raise ValueError('<STR_LIT>')<EOL><DEDENT>if not isinstance(bits, int) and bits is not None:<EOL><INDENT>raise ValueError('<STR_LIT>')<EOL><DEDENT>if bits is not None and bits <= <NUM_LIT:0>:<EOL><INDENT>raise ValueError('<STR_LIT>')<EOL><DEDENT>if not isinstance(channels, int) and channels is not None:<EOL><INDENT>raise ValueError('<STR_LIT>')<EOL><DEDENT>if channels is not None and channels <= <NUM_LIT:0>:<EOL><INDENT>raise ValueError('<STR_LIT>')<EOL><DEDENT>if encoding not in ENCODING_VALS + [None]:<EOL><INDENT>raise ValueError(<EOL>'<STR_LIT>'.format(ENCODING_VALS)<EOL>)<EOL><DEDENT>if not isinstance(ignore_length, bool):<EOL><INDENT>raise ValueError('<STR_LIT>')<EOL><DEDENT>input_format = []<EOL>if file_type is not None:<EOL><INDENT>input_format.extend(['<STR_LIT>', '<STR_LIT:{}>'.format(file_type)])<EOL><DEDENT>if rate is not None:<EOL><INDENT>input_format.extend(['<STR_LIT>', '<STR_LIT>'.format(rate)])<EOL><DEDENT>if bits is not None:<EOL><INDENT>input_format.extend(['<STR_LIT>', '<STR_LIT:{}>'.format(bits)])<EOL><DEDENT>if channels is not None:<EOL><INDENT>input_format.extend(['<STR_LIT:-c>', '<STR_LIT:{}>'.format(channels)])<EOL><DEDENT>if encoding is not None:<EOL><INDENT>input_format.extend(['<STR_LIT>', '<STR_LIT:{}>'.format(encoding)])<EOL><DEDENT>if ignore_length:<EOL><INDENT>input_format.append('<STR_LIT>')<EOL><DEDENT>self.input_format = input_format<EOL>return self<EOL>
Sets input file format arguments. This is primarily useful when
        dealing with audio files without a file extension. Overwrites any
        previously set input file arguments.

        If this function is not explicity called the input format is inferred
        from the file extension or the file's header.

        Parameters
        ----------
        file_type : str or None, default=None
            The file type of the input audio file. Should be the same as what
            the file extension would be, for ex. 'mp3' or 'wav'.
        rate : float or None, default=None
            The sample rate of the input audio file. If None the sample rate
            is inferred.
        bits : int or None, default=None
            The number of bits per sample. If None, the number of bits per
            sample is inferred.
        channels : int or None, default=None
            The number of channels in the audio file. If None the number of
            channels is inferred.
        encoding : str or None, default=None
            The audio encoding type. Sometimes needed with file-types that
            support more than one encoding type. One of:
                * signed-integer : PCM data stored as signed (‘two’s
                    complement’) integers. Commonly used with a 16 or 24−bit
                    encoding size. A value of 0 represents minimum signal
                    power.
                * unsigned-integer : PCM data stored as unsigned integers.
                    Commonly used with an 8-bit encoding size. A value of 0
                    represents maximum signal power.
                * floating-point : PCM data stored as IEEE 753 single precision
                    (32-bit) or double precision (64-bit) floating-point
                    (‘real’) numbers. A value of 0 represents minimum signal
                    power.
                * a-law : International telephony standard for logarithmic
                    encoding to 8 bits per sample. It has a precision
                    equivalent to roughly 13-bit PCM and is sometimes encoded
                    with reversed bit-ordering.
                * u-law : North American telephony standard for logarithmic
                    encoding to 8 bits per sample. A.k.a. μ-law. It has a
                    precision equivalent to roughly 14-bit PCM and is sometimes
                    encoded with reversed bit-ordering.
                * oki-adpcm : OKI (a.k.a. VOX, Dialogic, or Intel) 4-bit ADPCM;
                    it has a precision equivalent to roughly 12-bit PCM. ADPCM
                    is a form of audio compression that has a good compromise
                    between audio quality and encoding/decoding speed.
                * ima-adpcm : IMA (a.k.a. DVI) 4-bit ADPCM; it has a precision
                    equivalent to roughly 13-bit PCM.
                * ms-adpcm : Microsoft 4-bit ADPCM; it has a precision
                    equivalent to roughly 14-bit PCM.
                * gsm-full-rate : GSM is currently used for the vast majority
                    of the world’s digital wireless telephone calls. It
                    utilises several audio formats with different bit-rates and
                    associated speech quality. SoX has support for GSM’s
                    original 13kbps ‘Full Rate’ audio format. It is usually
                    CPU-intensive to work with GSM audio.
        ignore_length : bool, default=False
            If True, overrides an (incorrect) audio length given in an audio
            file’s header. If this option is given then SoX will keep reading
            audio until it reaches the end of the input file.