Source code for madmom.audio.ffmpeg

# encoding: utf-8
# pylint: disable=no-member
# pylint: disable=invalid-name
# pylint: disable=too-many-arguments
"""
This module contains audio handling via ffmpeg functionality.

"""

from __future__ import absolute_import, division, print_function

import tempfile
import subprocess
import os
import sys
import numpy as np


[docs]def decode_to_disk(infile, fmt='f32le', sample_rate=None, num_channels=1, skip=None, max_len=None, outfile=None, tmp_dir=None, tmp_suffix=None, cmd='ffmpeg'): """ Decodes the given audio file, optionally down-mixes it to mono and writes it to another file as a sequence of samples. Returns the file name of the output file. Parameters ---------- infile : str Name of the audio sound file to decode. fmt : {'f32le', 's16le'}, optional Format of the samples: - 'f32le' for float32, little-endian, - 's16le' for signed 16-bit int, little-endian. sample_rate : int, optional Sample rate to re-sample the signal to (if set) [Hz]. num_channels : int, optional Number of channels to reduce the signal to. skip : float, optional Number of seconds to skip at beginning of file. max_len : float, optional Maximum length in seconds to decode. outfile : str, optional The file to decode the sound file to; if not given, a temporary file will be created. tmp_dir : str, optional The directory to create the temporary file in (if no `outfile` is given). tmp_suffix : str, optional The file suffix for the temporary file if no `outfile` is given; e.g. ".pcm" (including the dot). cmd : {'ffmpeg', 'avconv'}, optional Decoding command (defaults to ffmpeg, alternatively supports avconv). Returns ------- outfile : str The output file name. """ # check input file type if not isinstance(infile, str): raise ValueError("only file names are supported as `infile`, not %s." % infile) # create temp file if no outfile is given if outfile is None: # looks stupid, but is recommended over tempfile.mktemp() f = tempfile.NamedTemporaryFile(delete=False, dir=tmp_dir, suffix=tmp_suffix) f.close() outfile = f.name delete_on_fail = True else: delete_on_fail = False # check output file type if not isinstance(outfile, str): raise ValueError("only file names are supported as `outfile`, not %s." % outfile) # call ffmpeg (throws exception on error) try: call = _assemble_ffmpeg_call(infile, outfile, fmt, sample_rate, num_channels, skip, max_len, cmd) subprocess.check_call(call) except Exception: if delete_on_fail: os.unlink(outfile) raise return outfile
[docs]def decode_to_memory(infile, fmt='f32le', sample_rate=None, num_channels=1, skip=None, max_len=None, cmd='ffmpeg'): """ Decodes the given audio file, down-mixes it to mono and returns it as a binary string of a sequence of samples. Parameters ---------- infile : str Name of the audio sound file to decode. fmt : {'f32le', 's16le'}, optional Format of the samples: - 'f32le' for float32, little-endian, - 's16le' for signed 16-bit int, little-endian. sample_rate : int, optional Sample rate to re-sample the signal to (if set) [Hz]. num_channels : int, optional Number of channels to reduce the signal to. skip : float, optional Number of seconds to skip at beginning of file. max_len : float, optional Maximum length in seconds to decode. cmd : {'ffmpeg', 'avconv'}, optional Decoding command (defaults to ffmpeg, alternatively supports avconv). Returns ------- samples : str a binary string of samples """ # check input file type if not isinstance(infile, str): raise ValueError("only file names are supported as `infile`, not %s." % infile) # assemble ffmpeg call call = _assemble_ffmpeg_call(infile, "pipe:1", fmt, sample_rate, num_channels, skip, max_len, cmd) if hasattr(subprocess, 'check_output'): # call ffmpeg (throws exception on error) signal = subprocess.check_output(call) else: # this is an old version of Python, do subprocess.check_output manually proc = subprocess.Popen(call, stdout=subprocess.PIPE, bufsize=-1) signal, _ = proc.communicate() if proc.returncode != 0: raise subprocess.CalledProcessError(proc.returncode, call) return signal
[docs]def decode_to_pipe(infile, fmt='f32le', sample_rate=None, num_channels=1, skip=None, max_len=None, buf_size=-1, cmd='ffmpeg'): """ Decodes the given audio file, down-mixes it to mono and returns a file-like object for reading the samples, as well as a process object. To stop decoding the file, call close() on the returned file-like object, then call wait() on the returned process object. Parameters ---------- infile : str Name of the audio sound file to decode. fmt : {'f32le', 's16le'}, optional Format of the samples: - 'f32le' for float32, little-endian, - 's16le' for signed 16-bit int, little-endian. sample_rate : int, optional Sample rate to re-sample the signal to (if set) [Hz]. num_channels : int, optional Number of channels to reduce the signal to. skip : float, optional Number of seconds to skip at beginning of file. max_len : float, optional Maximum length in seconds to decode. buf_size : int, optional Size of buffer for the file-like object: - '-1' means OS default (default), - '0' means unbuffered, - '1' means line-buffered, any other value is the buffer size in bytes. cmd : {'ffmpeg','avconv'}, optional Decoding command (defaults to ffmpeg, alternatively supports avconv). Returns ------- pipe : file-like object File-like object for reading the decoded samples. proc : process object Process object for the decoding process. """ # check input file type if not isinstance(infile, str): raise ValueError("only file names are supported as `infile`, not %s." % infile) # Note: closing the file-like object only stops decoding because ffmpeg # reacts on that. A cleaner solution would be calling # proc.terminate explicitly, but this is only available in # Python 2.6+. proc.wait needs to be called in any case. call = _assemble_ffmpeg_call(infile, "pipe:1", fmt, sample_rate, num_channels, skip, max_len, cmd) # redirect stdout to a pipe and buffer as requested proc = subprocess.Popen(call, stdout=subprocess.PIPE, bufsize=buf_size) return proc.stdout, proc
def _assemble_ffmpeg_call(infile, output, fmt='f32le', sample_rate=None, num_channels=1, skip=None, max_len=None, cmd='ffmpeg'): """ Internal function. Creates a sequence of strings indicating the application (ffmpeg) to be called as well as the parameters necessary to decode the given input file to the given format, at the given offset and for the given length to the given output. Parameters ---------- infile : str Name of the audio sound file to decode. output : str Where to decode to. fmt : {'f32le', 's16le'}, optional Format of the samples: - 'f32le' for float32, little-endian, - 's16le' for signed 16-bit int, little-endian. sample_rate : int, optional Sample rate to re-sample the signal to (if set) [Hz]. num_channels : int, optional Number of channels to reduce the signal to. skip : float, optional Number of seconds to skip at beginning of file. max_len : float, optional Maximum length in seconds to decode. cmd : {'ffmpeg','avconv'}, optional Decoding command (defaults to ffmpeg, alternatively supports avconv). Returns ------- list Assembled ffmpeg call. Notes ----- 'avconv' rounds decoding positions and decodes in blocks of 4096 length resulting in incorrect start and stop positions. Thus it should only be used to decode complete files. """ # Note: avconv rounds decoding positions and decodes in blocks of 4096 # length resulting in incorrect start and stop positions if cmd == 'avconv' and skip is not None and max_len is not None: raise RuntimeError('avconv has a bug, which results in wrong audio ' 'slices! Decode the audio files to .wav first or ' 'use ffmpeg.') if isinstance(infile, str): infile = infile.encode(sys.getfilesystemencoding()) else: infile = str(infile) # general options call = [cmd, "-v", "quiet"] # infile options if skip is not None: # use "%f" to avoid e-05 and the like call.extend(["-ss", "%f" % float(skip)]) call.extend(["-i", infile, "-y", "-f", str(fmt)]) if max_len is not None: # use "%f" to avoid e-05 and the like call.extend(["-t", "%f" % float(max_len)]) # output options if num_channels is not None: call.extend(["-ac", str(int(num_channels))]) if sample_rate is not None: call.extend(["-ar", str(int(sample_rate))]) call.append(output) return call
[docs]def get_file_info(infile, cmd='ffprobe'): """ Extract and return information about audio files. Parameters ---------- infile : str Name of the audio file. cmd : {'ffprobe', 'avprobe'}, optional Probing command (defaults to ffprobe, alternatively supports avprobe). Returns ------- dict Audio file information. """ # check input file type if not isinstance(infile, str): raise ValueError("only file names are supported as `infile`, not %s." % infile) # init dictionary info = {'num_channels': None, 'sample_rate': None} # call ffprobe output = subprocess.check_output([cmd, "-v", "quiet", "-show_streams", infile]) # parse information for line in output.split(): if line.startswith(b'channels='): info['num_channels'] = int(line[len('channels='):]) if line.startswith(b'sample_rate='): # the int(float(...)) conversion is necessary because # avprobe returns sample_rate as floating point number # which int() can't handle. info['sample_rate'] = int(float(line[len('sample_rate='):])) # return the dictionary return info
[docs]def load_ffmpeg_file(filename, sample_rate=None, num_channels=None, start=None, stop=None, dtype=None, cmd_decode='ffmpeg', cmd_probe='ffprobe'): """ Load the audio data from the given file and return it as a numpy array. This uses ffmpeg (or avconv) and thus supports a lot of different file formats, resampling and channel conversions. The file will be fully decoded into memory if no start and stop positions are given. Parameters ---------- filename : str Name of the audio sound file to load. sample_rate : int, optional Sample rate to re-sample the signal to [Hz]; 'None' returns the signal in its original rate. num_channels : int, optional Reduce or expand the signal to `num_channels` channels; 'None' returns the signal with its original channels. start : float, optional Start position [seconds]. stop : float, optional Stop position [seconds]. dtype : numpy dtype, optional Numpy dtype to return the signal in (supports signed and unsigned 8/16/32-bit integers, and single and double precision floats, each in little or big endian). If 'None', np.int16 is used. cmd_decode : {'ffmpeg', 'avconv'}, optional Decoding command (defaults to ffmpeg, alternatively supports avconv). cmd_probe : {'ffprobe', 'avprobe'}, optional Probing command (defaults to ffprobe, alternatively supports avprobe). Returns ------- signal : numpy array Audio samples. sample_rate : int Sample rate of the audio samples. """ # convert dtype to sample type # (all ffmpeg PCM sample types: ffmpeg -formats | grep PCM) if dtype is None: dtype = np.int16 dtype = np.dtype(dtype) # - unsigned int, signed int, floating point: sample_type = {'u': 'u', 'i': 's', 'f': 'f'}.get(dtype.kind) # - sample size in bits: sample_type += str(8 * dtype.itemsize) # - little endian or big endian: if dtype.byteorder == '=': sample_type += sys.byteorder[0] + 'e' else: sample_type += {'|': '', '<': 'le', '>': 'be'}.get(dtype.byteorder) # start and stop position if start is None: start = 0 max_len = None if stop is not None: max_len = stop - start # convert the audio signal using ffmpeg signal = np.frombuffer(decode_to_memory(filename, fmt=sample_type, sample_rate=sample_rate, num_channels=num_channels, skip=start, max_len=max_len, cmd=cmd_decode), dtype=dtype) # get the needed information from the file if sample_rate is None or num_channels is None: info = get_file_info(filename, cmd=cmd_probe) if sample_rate is None: sample_rate = info['sample_rate'] if num_channels is None: num_channels = info['num_channels'] # reshape the audio signal if num_channels > 1: signal = signal.reshape((-1, num_channels)) return signal, sample_rate