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Python bitstream.BitstreamRecorder类代码示例

本文整理汇总了Python中audiotools.bitstream.BitstreamRecorder的典型用法代码示例。如果您正苦于以下问题:Python BitstreamRecorder类的具体用法?Python BitstreamRecorder怎么用?Python BitstreamRecorder使用的例子?那么恭喜您, 这里精选的类代码示例或许可以为您提供帮助。


在下文中一共展示了BitstreamRecorder类的15个代码示例,这些例子默认根据受欢迎程度排序。您可以为喜欢或者感觉有用的代码点赞,您的评价将有助于系统推荐出更棒的Python代码示例。

示例1: encode_flac

def encode_flac(filename, pcmreader,
                block_size=4096,
                max_lpc_order=8,
                adaptive_mid_side=False,
                mid_side=True,
                exhaustive_model_search=False,
                max_residual_partition_order=5):

    options = Encoding_Options(block_size,
                               max_lpc_order,
                               adaptive_mid_side,
                               mid_side,
                               exhaustive_model_search,
                               max_residual_partition_order,
                               14 if pcmreader.bits_per_sample <= 16 else 30)

    streaminfo = STREAMINFO(block_size, block_size,
                            2 ** 32, 0,
                            pcmreader.sample_rate,
                            pcmreader.channels,
                            pcmreader.bits_per_sample,
                            0, md5())

    pcmreader = BufferedPCMReader(pcmreader)
    output_file = open(filename, "wb")
    writer = BitstreamWriter(output_file, 0)

    #write placeholder metadata blocks
    writer.write_bytes("fLaC")
    writer.build("1u 7u 24u", [1, 0, 34])
    streaminfo.write(writer)

    #walk through PCM reader's FrameLists
    frame_number = 0
    frame = pcmreader.read(block_size *
                           (pcmreader.bits_per_sample / 8) *
                           pcmreader.channels)

    flac_frame = BitstreamRecorder(0)

    while (len(frame) > 0):

        streaminfo.input_update(frame)

        flac_frame.reset()
        encode_flac_frame(flac_frame, pcmreader, options, frame_number, frame)

        streaminfo.output_update(flac_frame)

        flac_frame.copy(writer)

        frame_number += 1
        frame = pcmreader.read(block_size *
                               (pcmreader.bits_per_sample / 8) *
                               pcmreader.channels)

    #return to beginning of file and rewrite STREAMINFO block
    output_file.seek(8, 0)
    streaminfo.write(writer)
    writer.close()
开发者ID:XiaonuoGantan,项目名称:python-audio-tools,代码行数:60,代码来源:flac.py

示例2: calculate_lpc_coefficients

def calculate_lpc_coefficients(pcmreader, options, sample_size, channel):
    windowed = [s * t for s, t in zip(channel, tukey_window(len(channel), 0.5))]

    autocorrelated = [sum([s1 * s2 for s1, s2 in zip(windowed, windowed[lag:])]) for lag in range(0, 9)]

    assert len(autocorrelated) == 9

    if autocorrelated[0] != 0.0:
        lp_coefficients = compute_lp_coefficients(autocorrelated)

        assert len(lp_coefficients) == 8

        qlp_coefficients4 = quantize_coefficients(lp_coefficients, 4)
        qlp_coefficients8 = quantize_coefficients(lp_coefficients, 8)

        residuals4 = compute_residuals(sample_size, qlp_coefficients4[:], channel)
        residuals8 = compute_residuals(sample_size, qlp_coefficients8[:], channel)

        residual_block4 = BitstreamRecorder(0)
        residual_block8 = BitstreamRecorder(0)

        encode_residuals(residual_block4, options, sample_size, residuals4)
        encode_residuals(residual_block8, options, sample_size, residuals8)

        if residual_block4.bits() < residual_block8.bits():
            return (qlp_coefficients4, residual_block4)
        else:
            return (qlp_coefficients8, residual_block8)
    else:
        qlp_coefficients = [0, 0, 0, 0]
        residuals = compute_residuals(sample_size, qlp_coefficients[:], channel)
        residual_block = BitstreamRecorder(0)
        encode_residuals(residual_block, options, sample_size, residuals)
        return (qlp_coefficients, residual_block)
开发者ID:ryechus,项目名称:python-audio-tools,代码行数:34,代码来源:alac.py

示例3: build

    def build(self, writer):
        """outputs an APEv2 tag to writer"""

        from audiotools.bitstream import BitstreamRecorder

        tags = BitstreamRecorder(1)

        for tag in self.tags:
            tag.build(tags)

        if (self.contains_header):
            writer.build(ApeTag.HEADER_FORMAT,
                         ("APETAGEX",                # preamble
                          2000,                      # version
                          tags.bytes() + 32,         # tag size
                          len(self.tags),            # item count
                          0,                         # read only
                          0,                         # encoding
                          1,                         # is header
                          not self.contains_footer,  # no footer
                          self.contains_header))     # has header

        tags.copy(writer)
        if (self.contains_footer):
            writer.build(ApeTag.HEADER_FORMAT,
                         ("APETAGEX",                # preamble
                          2000,                      # version
                          tags.bytes() + 32,         # tag size
                          len(self.tags),            # item count
                          0,                         # read only
                          0,                         # encoding
                          0,                         # is header
                          not self.contains_footer,  # no footer
                          self.contains_header))     # has header
开发者ID:ryechus,项目名称:python-audio-tools,代码行数:34,代码来源:ape.py

示例4: aiff_header

def aiff_header(sample_rate,
                channels,
                bits_per_sample,
                total_pcm_frames):
    """given a set of integer stream attributes,
    returns header string of everything before an AIFF's PCM data

    may raise ValueError if the total size of the file is too large"""

    from audiotools.bitstream import (BitstreamRecorder, format_size)

    header = BitstreamRecorder(False)

    data_size = (bits_per_sample // 8) * channels * total_pcm_frames
    total_size = ((format_size("4b" + "4b 32u" +
                               "16u 32u 16u 1u 15u 64U" +
                               "4b 32u 32u 32u") // 8) +
                  data_size + (data_size % 2))

    if total_size < (2 ** 32):
        header.build("4b 32u 4b", (b"FORM", total_size, b"AIFF"))
        header.build("4b 32u", (b"COMM", 0x12))
        header.build("16u 32u 16u", (channels,
                                     total_pcm_frames,
                                     bits_per_sample))
        build_ieee_extended(header, sample_rate)
        header.build("4b 32u 32u 32u", (b"SSND", data_size + 8, 0, 0))

        return header.data()
    else:
        raise ValueError("total size too large for aiff file")
开发者ID:brigittebigi,项目名称:sppas,代码行数:31,代码来源:aiff.py

示例5: encode_residual_partition

def encode_residual_partition(rice_parameter, residuals):
    partition = BitstreamRecorder(0)
    for residual in residuals:
        if residual >= 0:
            unsigned = residual << 1
        else:
            unsigned = ((-residual - 1) << 1) | 1
        MSB = unsigned >> rice_parameter
        LSB = unsigned - (MSB << rice_parameter)
        partition.unary(1, MSB)
        partition.write(rice_parameter, LSB)

    return partition
开发者ID:brigittebigi,项目名称:sppas,代码行数:13,代码来源:flac.py

示例6: update_metadata

    def update_metadata(self, metadata):
        """takes this track's current MetaData object
        as returned by get_metadata() and sets this track's metadata
        with any fields updated in that object

        raises IOError if unable to write the file
        """

        from audiotools import transfer_data, TemporaryFile
        from audiotools.id3 import ID3v22Comment
        from audiotools.bitstream import BitstreamRecorder
        from audiotools.text import ERR_FOREIGN_METADATA
        import os

        if metadata is None:
            return
        elif not isinstance(metadata, ID3v22Comment):
            raise ValueError(ERR_FOREIGN_METADATA)
        elif not os.access(self.filename, os.W_OK):
            raise IOError(self.filename)

        # turn our ID3v2.2 tag into a raw binary chunk
        id3_chunk = BitstreamRecorder(0)
        metadata.build(id3_chunk)

        # generate a temporary AIFF file in which our new ID3v2.2 chunk
        # replaces the existing ID3v2.2 chunk
        new_aiff = TemporaryFile(self.filename)

        self.__class__.aiff_from_chunks(
            new_aiff,
            [(chunk if chunk.id != b"ID3 " else
              AIFF_Chunk(b"ID3 ",
                         id3_chunk.bytes(),
                         id3_chunk.data())) for chunk in self.chunks()])

        new_aiff.close()
开发者ID:brigittebigi,项目名称:sppas,代码行数:37,代码来源:aiff.py

示例7: set_metadata

    def set_metadata(self, metadata):
        """takes a MetaData object and sets this track's metadata

        this metadata includes track name, album name, and so on
        raises IOError if unable to write the file"""

        from audiotools.id3 import ID3v22Comment

        if metadata is None:
            return self.delete_metadata()
        elif self.get_metadata() is not None:
            # current file has metadata, so replace it with new metadata
            self.update_metadata(ID3v22Comment.converted(metadata))
        else:
            # current file has no metadata, so append new ID3 block
            import os
            from audiotools.bitstream import BitstreamRecorder
            from audiotools import transfer_data, TemporaryFile

            if not os.access(self.filename, os.W_OK):
                raise IOError(self.filename)

            # turn our ID3v2.2 tag into a raw binary chunk
            id3_chunk = BitstreamRecorder(0)
            ID3v22Comment.converted(metadata).build(id3_chunk)

            # generate a temporary AIFF file in which our new ID3v2.2 chunk
            # is appended to the file's set of chunks
            new_aiff = TemporaryFile(self.filename)
            self.__class__.aiff_from_chunks(
                new_aiff,
                [c for c in self.chunks()] + [AIFF_Chunk(b"ID3 ",
                                                         id3_chunk.bytes(),
                                                         id3_chunk.data())])

            new_aiff.close()
开发者ID:brigittebigi,项目名称:sppas,代码行数:36,代码来源:aiff.py

示例8: encode_wavpack

def encode_wavpack(filename, pcmreader, block_size, correlation_passes=0, wave_header=None, wave_footer=None):
    pcmreader = BufferedPCMReader(pcmreader)
    output_file = open(filename, "wb")
    writer = BitstreamWriter(output_file, 1)
    context = EncoderContext(
        pcmreader,
        block_parameters(pcmreader.channels, pcmreader.channel_mask, correlation_passes),
        wave_header,
        wave_footer,
    )

    block_index = 0

    # walk through PCM reader's FrameLists
    frame = pcmreader.read(block_size * (pcmreader.bits_per_sample / 8) * pcmreader.channels)
    while len(frame) > 0:
        context.total_frames += frame.frames
        context.md5sum.update(frame.to_bytes(False, pcmreader.bits_per_sample >= 16))

        c = 0
        for parameters in context.block_parameters:
            if parameters.channel_count == 1:
                channel_data = [list(frame.channel(c))]
            else:
                channel_data = [list(frame.channel(c)), list(frame.channel(c + 1))]

            first_block = parameters is context.block_parameters[0]
            last_block = parameters is context.block_parameters[-1]

            context.block_offsets.append(output_file.tell())
            write_block(writer, context, channel_data, block_index, first_block, last_block, parameters)

            c += parameters.channel_count

        block_index += frame.frames
        frame = pcmreader.read(block_size * (pcmreader.bits_per_sample / 8) * pcmreader.channels)

    # write MD5 sum and optional Wave footer in final block
    sub_blocks = BitstreamRecorder(1)
    sub_block = BitstreamRecorder(1)

    sub_block.reset()
    sub_block.write_bytes(context.md5sum.digest())
    write_sub_block(sub_blocks, WV_MD5, 1, sub_block)

    # write Wave footer in final block, if present
    if context.wave_footer is not None:
        sub_block.reset()
        sub_block.write_bytes(context.wave_footer)
        write_sub_block(sub_blocks, WV_WAVE_FOOTER, 1, sub_block)

    write_block_header(
        writer,
        sub_blocks.bytes(),
        0xFFFFFFFF,
        0,
        pcmreader.bits_per_sample,
        1,
        0,
        0,
        0,
        1,
        1,
        0,
        pcmreader.sample_rate,
        0,
        0xFFFFFFFF,
    )
    sub_blocks.copy(writer)

    # update Wave header's "data" chunk size, if generated
    if context.wave_header is None:
        output_file.seek(32 + 2)
        if context.wave_footer is None:
            write_wave_header(writer, context.pcmreader, context.total_frames, 0)
        else:
            write_wave_header(writer, context.pcmreader, context.total_frames, len(context.wave_footer))

    # go back and populate block headers with total samples
    for block_offset in context.block_offsets:
        output_file.seek(block_offset + 12, 0)
        writer.write(32, block_index)

    writer.close()
开发者ID:hzlf,项目名称:python-audio-tools,代码行数:84,代码来源:wavpack.py

示例9: write_block

def write_block(writer, context, channels, block_index, first_block, last_block, parameters):
    """writer is a BitstreamWriter-compatible object
    context is an EncoderContext object
    channels[c][s] is sample "s" in channel "c"
    block_index is an integer of the block's offset in PCM frames
    first_block and last_block are flags indicating the block's sequence
    parameters is an EncodingParameters object
    """

    assert (len(channels) == 1) or (len(channels) == 2)

    if (len(channels) == 1) or (channels[0] == channels[1]):
        # 1 channel block or equivalent
        if len(channels) == 1:
            false_stereo = 0
        else:
            false_stereo = 1

        # calculate maximum magnitude of channel_0
        magnitude = max(map(bits, channels[0]))

        # determine wasted bits
        wasted = min(map(wasted_bps, channels[0]))
        if wasted == INFINITY:
            # all samples are 0
            wasted = 0

        # if wasted bits, remove them from channel_0
        if (wasted > 0) and (wasted != INFINITY):
            shifted = [[s >> wasted for s in channels[0]]]
        else:
            shifted = [channels[0]]

        # calculate CRC of shifted_0
        crc = calculate_crc(shifted)
    else:
        # 2 channel block
        false_stereo = 0

        # calculate maximum magnitude of channel_0/channel_1
        magnitude = max(max(map(bits, channels[0])), max(map(bits, channels[1])))

        # determine wasted bits
        wasted = min(min(map(wasted_bps, channels[0])), min(map(wasted_bps, channels[1])))
        if wasted == INFINITY:
            # all samples are 0
            wasted = 0

        # if wasted bits, remove them from channel_0/channel_1
        if wasted > 0:
            shifted = [[s >> wasted for s in channels[0]], [s >> wasted for s in channels[1]]]
        else:
            shifted = channels

        # calculate CRC of shifted_0/shifted_1
        crc = calculate_crc(shifted)

        # joint stereo conversion of shifted_0/shifted_1 to mid/side channels
        mid_side = joint_stereo(shifted[0], shifted[1])

    sub_blocks = BitstreamRecorder(1)
    sub_block = BitstreamRecorder(1)

    # if first block in file, write Wave header
    if not context.first_block_written:
        sub_block.reset()
        if context.wave_header is None:
            if context.wave_footer is None:
                write_wave_header(sub_block, context.pcmreader, 0, 0)
            else:
                write_wave_header(sub_block, context.pcmreader, 0, len(context.wave_footer))
        else:
            sub_block.write_bytes(context.wave_header)
        write_sub_block(sub_blocks, WV_WAVE_HEADER, 1, sub_block)
        context.first_block_written = True

    # if correlation passes, write three sub blocks of pass data
    if parameters.correlation_passes > 0:
        sub_block.reset()
        write_correlation_terms(
            sub_block,
            [p.term for p in parameters.correlation_parameters(false_stereo)],
            [p.delta for p in parameters.correlation_parameters(false_stereo)],
        )
        write_sub_block(sub_blocks, WV_TERMS, 0, sub_block)

        sub_block.reset()
        write_correlation_weights(sub_block, [p.weights for p in parameters.correlation_parameters(false_stereo)])
        write_sub_block(sub_blocks, WV_WEIGHTS, 0, sub_block)

        sub_block.reset()
        write_correlation_samples(
            sub_block,
            [p.term for p in parameters.correlation_parameters(false_stereo)],
            [p.samples for p in parameters.correlation_parameters(false_stereo)],
            2 if ((len(channels) == 2) and (not false_stereo)) else 1,
        )
        write_sub_block(sub_blocks, WV_SAMPLES, 0, sub_block)

    # if wasted bits, write extended integers sub block
#.........这里部分代码省略.........
开发者ID:hzlf,项目名称:python-audio-tools,代码行数:101,代码来源:wavpack.py

示例10: wave_header_footer

    def wave_header_footer(self):
        """returns a pair of data strings before and after PCM data

        the first contains all data before the PCM content of the data chunk
        the second containing all data after the data chunk
        for example:

        >>> w = audiotools.open("input.wav")
        >>> (head, tail) = w.wave_header_footer()
        >>> f = open("output.wav", "wb")
        >>> f.write(head)
        >>> audiotools.transfer_framelist_data(w.to_pcm(), f.write)
        >>> f.write(tail)
        >>> f.close()

        should result in "output.wav" being identical to "input.wav"
        """

        from audiotools.bitstream import BitstreamReader
        from audiotools.bitstream import BitstreamRecorder

        head = BitstreamRecorder(1)
        tail = BitstreamRecorder(1)
        current_block = head
        fmt_found = False

        with BitstreamReader(open(self.filename, 'rb'), 1) as wave_file:
            # transfer the 12-byte "RIFFsizeWAVE" header to head
            (riff, size, wave) = wave_file.parse("4b 32u 4b")
            if riff != b'RIFF':
                from audiotools.text import ERR_WAV_NOT_WAVE
                raise ValueError(ERR_WAV_NOT_WAVE)
            elif wave != b'WAVE':
                from audiotools.text import ERR_WAV_INVALID_WAVE
                raise ValueError(ERR_WAV_INVALID_WAVE)
            else:
                current_block.build("4b 32u 4b", (riff, size, wave))
                total_size = size - 4

            while total_size > 0:
                # transfer each chunk header
                (chunk_id, chunk_size) = wave_file.parse("4b 32u")
                if not frozenset(chunk_id).issubset(self.PRINTABLE_ASCII):
                    from audiotools.text import ERR_WAV_INVALID_CHUNK
                    raise ValueError(ERR_WAV_INVALID_CHUNK)
                else:
                    current_block.build("4b 32u", (chunk_id, chunk_size))
                    total_size -= 8

                # and transfer the full content of non-audio chunks
                if chunk_id != b"data":
                    if chunk_id == b"fmt ":
                        if not fmt_found:
                            fmt_found = True
                        else:
                            from audiotools.text import ERR_WAV_MULTIPLE_FMT
                            raise ValueError(ERR_WAV_MULTIPLE_FMT)

                    if chunk_size % 2:
                        current_block.write_bytes(
                            wave_file.read_bytes(chunk_size + 1))
                        total_size -= (chunk_size + 1)
                    else:
                        current_block.write_bytes(
                            wave_file.read_bytes(chunk_size))
                        total_size -= chunk_size
                else:
                    wave_file.skip_bytes(chunk_size)
                    current_block = tail

                    if chunk_size % 2:
                        current_block.write_bytes(wave_file.read_bytes(1))
                        total_size -= (chunk_size + 1)
                    else:
                        total_size -= chunk_size

            if fmt_found:
                return (head.data(), tail.data())
            else:
                from audiotools.text import ERR_WAV_NO_FMT_CHUNK
                raise ValueError(ERR_WAV_NO_FMT_CHUNK)
开发者ID:KristoforMaynard,项目名称:python-audio-tools,代码行数:81,代码来源:wav.py

示例11: encode_subframe

def encode_subframe(writer, options, bits_per_sample, samples):
    def all_identical(l):
        if (len(l) == 1):
            return True
        else:
            for i in l[1:]:
                if (i != l[0]):
                    return False
            else:
                return True

    def wasted(s):
        w = 0
        while ((s & 1) == 0):
            w += 1
            s >>= 1
        return w

    if (all_identical(samples)):
        encode_constant_subframe(writer, bits_per_sample, samples[0])
    else:
        #account for wasted BPS, if any
        wasted_bps = 2 ** 32
        for sample in samples:
            if (sample != 0):
                wasted_bps = min(wasted_bps, wasted(sample))
                if (wasted_bps == 0):
                    break

        if (wasted_bps == 2 ** 32):
            #all samples are 0
            wasted_bps = 0
        elif (wasted_bps > 0):
            samples = [s >> wasted_bps for s in samples]

        fixed_subframe = BitstreamRecorder(0)
        encode_fixed_subframe(fixed_subframe,
                              options,
                              wasted_bps,
                              bits_per_sample,
                              samples)

        if (options.max_lpc_order > 0):
            (lpc_order,
             qlp_coeffs,
             qlp_shift_needed) = compute_lpc_coefficients(options,
                                                          wasted_bps,
                                                          bits_per_sample,
                                                          samples)

            lpc_subframe = BitstreamRecorder(0)
            encode_lpc_subframe(lpc_subframe,
                                options,
                                wasted_bps,
                                bits_per_sample,
                                lpc_order,
                                options.qlp_precision,
                                qlp_shift_needed,
                                qlp_coeffs,
                                samples)

            if (((bits_per_sample * len(samples)) <
                 min(fixed_subframe.bits(), lpc_subframe.bits()))):
                encode_verbatim_subframe(writer, wasted_bps,
                                         bits_per_sample, samples)
            elif (fixed_subframe.bits() < lpc_subframe.bits()):
                fixed_subframe.copy(writer)
            else:
                lpc_subframe.copy(writer)
        else:
            if ((bits_per_sample * len(samples)) < fixed_subframe.bits()):
                encode_verbatim_subframe(writer, wasted_bps,
                                         bits_per_sample, samples)
            else:
                fixed_subframe.copy(writer)
开发者ID:wesclemens,项目名称:python-audio-tools,代码行数:75,代码来源:flac.py

示例12: encode_flac_frame

def encode_flac_frame(writer, pcmreader, options, frame_number, frame):
    crc16 = CRC16()
    writer.add_callback(crc16.update)

    if ((pcmreader.channels == 2) and (options.adaptive_mid_side or
                                       options.mid_side)):
        # calculate average/difference
        average = [(c0 + c1) // 2 for c0, c1 in zip(frame.channel(0),
                                                    frame.channel(1))]
        difference = [c0 - c1 for c0, c1 in zip(frame.channel(0),
                                                frame.channel(1))]

        # try different subframes based on encoding options
        left_subframe = BitstreamRecorder(0)
        encode_subframe(left_subframe, options,
                        pcmreader.bits_per_sample, list(frame.channel(0)))

        right_subframe = BitstreamRecorder(0)
        encode_subframe(right_subframe, options,
                        pcmreader.bits_per_sample, list(frame.channel(1)))

        average_subframe = BitstreamRecorder(0)
        encode_subframe(average_subframe, options,
                        pcmreader.bits_per_sample, average)

        difference_subframe = BitstreamRecorder(0)
        encode_subframe(difference_subframe, options,
                        pcmreader.bits_per_sample + 1, difference)

        # write best header/subframes to disk
        if options.mid_side:
            if ((left_subframe.bits() + right_subframe.bits()) <
                min(left_subframe.bits() + difference_subframe.bits(),
                    difference_subframe.bits() + right_subframe.bits(),
                    average_subframe.bits() + difference_subframe.bits())):
                write_frame_header(writer, pcmreader, frame_number, frame, 0x1)
                left_subframe.copy(writer)
                right_subframe.copy(writer)
            elif (left_subframe.bits() <
                  min(right_subframe.bits(), difference_subframe.bits())):
                write_frame_header(writer, pcmreader, frame_number, frame, 0x8)
                left_subframe.copy(writer)
                difference_subframe.copy(writer)
            elif right_subframe.bits() < average_subframe.bits():
                write_frame_header(writer, pcmreader, frame_number, frame, 0x9)
                difference_subframe.copy(writer)
                right_subframe.copy(writer)
            else:
                write_frame_header(writer, pcmreader, frame_number, frame, 0xA)
                average_subframe.copy(writer)
                difference_subframe.copy(writer)
        else:
            if (((left_subframe.bits() + right_subframe.bits()) <
                 (average_subframe.bits() + difference_subframe.bits()))):
                write_frame_header(writer, pcmreader, frame_number, frame, 0x1)
                left_subframe.copy(writer)
                right_subframe.copy(writer)
            else:
                write_frame_header(writer, pcmreader, frame_number, frame, 0xA)
                average_subframe.copy(writer)
                difference_subframe.copy(writer)
    else:
        write_frame_header(writer, pcmreader, frame_number, frame,
                           pcmreader.channels - 1)

        for i in range(frame.channels):
            encode_subframe(writer,
                            options,
                            pcmreader.bits_per_sample,
                            list(frame.channel(i)))

    writer.byte_align()
    writer.pop_callback()
    writer.write(16, int(crc16))
开发者ID:brigittebigi,项目名称:sppas,代码行数:74,代码来源:flac.py

示例13: wave_header

def wave_header(sample_rate,
                channels,
                channel_mask,
                bits_per_sample,
                total_pcm_frames):
    """given a set of integer stream attributes,
    returns header string of everything before a RIFF WAVE's PCM data

    may raise ValueError if the total size of the file is too large"""

    from audiotools.bitstream import (BitstreamRecorder, format_size)

    assert(isinstance(sample_rate, int))
    assert(isinstance(channels, int))
    assert(isinstance(channel_mask, int))
    assert(isinstance(bits_per_sample, int))
    assert(isinstance(total_pcm_frames, int) or
           isinstance(total_pcm_frames, long))

    header = BitstreamRecorder(True)

    avg_bytes_per_second = sample_rate * channels * (bits_per_sample // 8)
    block_align = channels * (bits_per_sample // 8)

    # build a regular or extended fmt chunk
    # based on the reader's attributes
    if ((channels <= 2) and (bits_per_sample <= 16)):
        fmt = "16u 16u 32u 32u 16u 16u"
        fmt_fields = (1,   # compression code
                      channels,
                      sample_rate,
                      avg_bytes_per_second,
                      block_align,
                      bits_per_sample)
    else:
        if channel_mask == 0:
            channel_mask = {1: 0x4,
                            2: 0x3,
                            3: 0x7,
                            4: 0x33,
                            5: 0x37,
                            6: 0x3F}.get(channels, 0)
        fmt = "16u 16u 32u 32u 16u 16u" + "16u 16u 32u 16b"
        fmt_fields = (0xFFFE,   # compression code
                      channels,
                      sample_rate,
                      avg_bytes_per_second,
                      block_align,
                      bits_per_sample,
                      22,       # CB size
                      bits_per_sample,
                      channel_mask,
                      b'\x01\x00\x00\x00\x00\x00\x10\x00' +
                      b'\x80\x00\x00\xaa\x00\x38\x9b\x71'  # sub format
                      )

    data_size = (bits_per_sample // 8) * channels * total_pcm_frames
    total_size = ((format_size("4b" + "4b 32u" + fmt + "4b 32u") // 8) +
                  data_size + (data_size % 2))

    if total_size < (2 ** 32):
        header.build("4b 32u 4b", (b"RIFF", total_size, b"WAVE"))
        header.build("4b 32u", (b"fmt ", format_size(fmt) // 8))
        header.build(fmt, fmt_fields)
        header.build("4b 32u", (b"data", data_size))

        return header.data()
    else:
        raise ValueError("total size too large for wave file")
开发者ID:KristoforMaynard,项目名称:python-audio-tools,代码行数:69,代码来源:wav.py

示例14: aiff_header_footer

    def aiff_header_footer(self):
        """returns (header, footer) tuple of strings
        containing all data before and after the PCM stream

        if self.has_foreign_aiff_chunks() is False,
        may raise ValueError if the file has no header and footer
        for any reason"""

        from audiotools.bitstream import BitstreamReader
        from audiotools.bitstream import BitstreamRecorder
        from audiotools.text import (ERR_AIFF_NOT_AIFF,
                                     ERR_AIFF_INVALID_AIFF,
                                     ERR_AIFF_INVALID_CHUNK_ID)

        head = BitstreamRecorder(0)
        tail = BitstreamRecorder(0)
        current_block = head

        with BitstreamReader(open(self.filename, 'rb'), False) as aiff_file:
            # transfer the 12-byte "RIFFsizeWAVE" header to head
            (form, size, aiff) = aiff_file.parse("4b 32u 4b")
            if form != b'FORM':
                raise InvalidAIFF(ERR_AIFF_NOT_AIFF)
            elif aiff != b'AIFF':
                raise InvalidAIFF(ERR_AIFF_INVALID_AIFF)
            else:
                current_block.build("4b 32u 4b", (form, size, aiff))
                total_size = size - 4

            while total_size > 0:
                # transfer each chunk header
                (chunk_id, chunk_size) = aiff_file.parse("4b 32u")
                if not frozenset(chunk_id).issubset(self.PRINTABLE_ASCII):
                    raise InvalidAIFF(ERR_AIFF_INVALID_CHUNK_ID)
                else:
                    current_block.build("4b 32u", (chunk_id, chunk_size))
                    total_size -= 8

                # and transfer the full content of non-audio chunks
                if chunk_id != b"SSND":
                    if chunk_size % 2:
                        current_block.write_bytes(
                            aiff_file.read_bytes(chunk_size + 1))
                        total_size -= (chunk_size + 1)
                    else:
                        current_block.write_bytes(
                            aiff_file.read_bytes(chunk_size))
                        total_size -= chunk_size
                else:
                    # transfer alignment as part of SSND's chunk header
                    align = aiff_file.parse("32u 32u")
                    current_block.build("32u 32u", align)
                    aiff_file.skip_bytes(chunk_size - 8)
                    current_block = tail

                    if chunk_size % 2:
                        current_block.write_bytes(aiff_file.read_bytes(1))
                        total_size -= (chunk_size + 1)
                    else:
                        total_size -= chunk_size

            return (head.data(), tail.data())
开发者ID:brigittebigi,项目名称:sppas,代码行数:62,代码来源:aiff.py

示例15: encode_subframe

def encode_subframe(writer, options, bits_per_sample, samples):
    def all_identical(l):
        if len(l) == 1:
            return True
        else:
            for i in l[1:]:
                if i != l[0]:
                    return False
            else:
                return True

    def wasted(s):
        w = 0
        while (s & 1) == 0:
            w += 1
            s >>= 1
        return w

    if all_identical(samples):
        encode_constant_subframe(writer, bits_per_sample, samples[0])
    else:
        # account for wasted BPS, if any
        wasted_bps = 2 ** 32
        for sample in samples:
            if sample != 0:
                wasted_bps = min(wasted_bps, wasted(sample))
                if wasted_bps == 0:
                    break

        if wasted_bps == 2 ** 32:
            # all samples are 0
            wasted_bps = 0
        elif wasted_bps > 0:
            samples = [s >> wasted_bps for s in samples]

        fixed_subframe = BitstreamRecorder(0)
        encode_fixed_subframe(fixed_subframe,
                              options,
                              wasted_bps,
                              bits_per_sample,
                              samples)

        if options.max_lpc_order > 0:
            lpc_subframe = BitstreamRecorder(0)
            encode_lpc_subframe(lpc_subframe,
                                options,
                                wasted_bps,
                                bits_per_sample,
                                samples)

            if (((bits_per_sample * len(samples)) <
                 min(fixed_subframe.bits(), lpc_subframe.bits()))):
                encode_verbatim_subframe(writer, wasted_bps,
                                         bits_per_sample, samples)
            elif fixed_subframe.bits() < lpc_subframe.bits():
                fixed_subframe.copy(writer)
            else:
                lpc_subframe.copy(writer)
        else:
            if (bits_per_sample * len(samples)) < fixed_subframe.bits():
                encode_verbatim_subframe(writer, wasted_bps,
                                         bits_per_sample, samples)
            else:
                fixed_subframe.copy(writer)
开发者ID:brigittebigi,项目名称:sppas,代码行数:64,代码来源:flac.py


注:本文中的audiotools.bitstream.BitstreamRecorder类示例由纯净天空整理自Github/MSDocs等开源代码及文档管理平台,相关代码片段筛选自各路编程大神贡献的开源项目,源码版权归原作者所有,传播和使用请参考对应项目的License;未经允许,请勿转载。