audio_io_tutorial.py 15 KB
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# -*- coding: utf-8 -*-
"""
Audio I/O
=========

``torchaudio`` integrates ``libsox`` and provides a rich set of audio I/O.
"""

# When running this tutorial in Google Colab, install the required packages
# with the following.
# !pip install torchaudio boto3

import torch
import torchaudio

print(torch.__version__)
print(torchaudio.__version__)

######################################################################
# Preparing data and utility functions (skip this section)
# --------------------------------------------------------
#

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# @title Prepare data and utility functions. {display-mode: "form"}
# @markdown
# @markdown You do not need to look into this cell.
# @markdown Just execute once and you are good to go.
# @markdown
# @markdown In this tutorial, we will use a speech data from [VOiCES dataset](https://iqtlabs.github.io/voices/),
# @markdown which is licensed under Creative Commos BY 4.0.
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import io
import os
import tarfile

import boto3
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import matplotlib.pyplot as plt
import requests
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from botocore import UNSIGNED
from botocore.config import Config
from IPython.display import Audio, display


_SAMPLE_DIR = "_assets"
SAMPLE_WAV_URL = "https://pytorch-tutorial-assets.s3.amazonaws.com/steam-train-whistle-daniel_simon.wav"
SAMPLE_WAV_PATH = os.path.join(_SAMPLE_DIR, "steam.wav")

SAMPLE_MP3_URL = "https://pytorch-tutorial-assets.s3.amazonaws.com/steam-train-whistle-daniel_simon.mp3"
SAMPLE_MP3_PATH = os.path.join(_SAMPLE_DIR, "steam.mp3")

SAMPLE_GSM_URL = "https://pytorch-tutorial-assets.s3.amazonaws.com/steam-train-whistle-daniel_simon.gsm"
SAMPLE_GSM_PATH = os.path.join(_SAMPLE_DIR, "steam.gsm")

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SAMPLE_WAV_SPEECH_URL = "https://pytorch-tutorial-assets.s3.amazonaws.com/VOiCES_devkit/source-16k/train/sp0307/Lab41-SRI-VOiCES-src-sp0307-ch127535-sg0042.wav"  # noqa: E501
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SAMPLE_WAV_SPEECH_PATH = os.path.join(_SAMPLE_DIR, "speech.wav")

SAMPLE_TAR_URL = "https://pytorch-tutorial-assets.s3.amazonaws.com/VOiCES_devkit.tar.gz"
SAMPLE_TAR_PATH = os.path.join(_SAMPLE_DIR, "sample.tar.gz")
SAMPLE_TAR_ITEM = "VOiCES_devkit/source-16k/train/sp0307/Lab41-SRI-VOiCES-src-sp0307-ch127535-sg0042.wav"

S3_BUCKET = "pytorch-tutorial-assets"
S3_KEY = "VOiCES_devkit/source-16k/train/sp0307/Lab41-SRI-VOiCES-src-sp0307-ch127535-sg0042.wav"

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def _fetch_data():
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    os.makedirs(_SAMPLE_DIR, exist_ok=True)
    uri = [
        (SAMPLE_WAV_URL, SAMPLE_WAV_PATH),
        (SAMPLE_MP3_URL, SAMPLE_MP3_PATH),
        (SAMPLE_GSM_URL, SAMPLE_GSM_PATH),
        (SAMPLE_WAV_SPEECH_URL, SAMPLE_WAV_SPEECH_PATH),
        (SAMPLE_TAR_URL, SAMPLE_TAR_PATH),
    ]
    for url, path in uri:
        with open(path, "wb") as file_:
            file_.write(requests.get(url).content)

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_fetch_data()

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def print_stats(waveform, sample_rate=None, src=None):
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    if src:
        print("-" * 10)
        print("Source:", src)
        print("-" * 10)
    if sample_rate:
        print("Sample Rate:", sample_rate)
    print("Shape:", tuple(waveform.shape))
    print("Dtype:", waveform.dtype)
    print(f" - Max:     {waveform.max().item():6.3f}")
    print(f" - Min:     {waveform.min().item():6.3f}")
    print(f" - Mean:    {waveform.mean().item():6.3f}")
    print(f" - Std Dev: {waveform.std().item():6.3f}")
    print()
    print(waveform)
    print()

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def plot_waveform(waveform, sample_rate, title="Waveform", xlim=None, ylim=None):
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    waveform = waveform.numpy()

    num_channels, num_frames = waveform.shape
    time_axis = torch.arange(0, num_frames) / sample_rate

    figure, axes = plt.subplots(num_channels, 1)
    if num_channels == 1:
        axes = [axes]
    for c in range(num_channels):
        axes[c].plot(time_axis, waveform[c], linewidth=1)
        axes[c].grid(True)
        if num_channels > 1:
            axes[c].set_ylabel(f"Channel {c+1}")
        if xlim:
            axes[c].set_xlim(xlim)
        if ylim:
            axes[c].set_ylim(ylim)
    figure.suptitle(title)
    plt.show(block=False)

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def plot_specgram(waveform, sample_rate, title="Spectrogram", xlim=None):
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    waveform = waveform.numpy()

    num_channels, num_frames = waveform.shape

    figure, axes = plt.subplots(num_channels, 1)
    if num_channels == 1:
        axes = [axes]
    for c in range(num_channels):
        axes[c].specgram(waveform[c], Fs=sample_rate)
        if num_channels > 1:
            axes[c].set_ylabel(f"Channel {c+1}")
        if xlim:
            axes[c].set_xlim(xlim)
    figure.suptitle(title)
    plt.show(block=False)

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def play_audio(waveform, sample_rate):
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    waveform = waveform.numpy()

    num_channels, num_frames = waveform.shape
    if num_channels == 1:
        display(Audio(waveform[0], rate=sample_rate))
    elif num_channels == 2:
        display(Audio((waveform[0], waveform[1]), rate=sample_rate))
    else:
        raise ValueError("Waveform with more than 2 channels are not supported.")
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def _get_sample(path, resample=None):
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    effects = [["remix", "1"]]
    if resample:
        effects.extend(
            [
                ["lowpass", f"{resample // 2}"],
                ["rate", f"{resample}"],
            ]
        )
    return torchaudio.sox_effects.apply_effects_file(path, effects=effects)

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def get_sample(*, resample=None):
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    return _get_sample(SAMPLE_WAV_PATH, resample=resample)

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def inspect_file(path):
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    print("-" * 10)
    print("Source:", path)
    print("-" * 10)
    print(f" - File size: {os.path.getsize(path)} bytes")
    print(f" - {torchaudio.info(path)}")

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######################################################################
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# Querying audio metadata
# -----------------------
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#
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# Function :py:func:`torchaudio.info` fetches audio metadata.
# You can provide a path-like object or file-like object.
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#

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metadata = torchaudio.info(SAMPLE_WAV_PATH)
print(metadata)

######################################################################
# Where
#
# -  ``sample_rate`` is the sampling rate of the audio
# -  ``num_channels`` is the number of channels
# -  ``num_frames`` is the number of frames per channel
# -  ``bits_per_sample`` is bit depth
# -  ``encoding`` is the sample coding format
#
# ``encoding`` can take on one of the following values:
#
# -  ``"PCM_S"``: Signed integer linear PCM
# -  ``"PCM_U"``: Unsigned integer linear PCM
# -  ``"PCM_F"``: Floating point linear PCM
# -  ``"FLAC"``: Flac, `Free Lossless Audio
#    Codec <https://xiph.org/flac/>`__
# -  ``"ULAW"``: Mu-law,
#    [`wikipedia <https://en.wikipedia.org/wiki/%CE%9C-law_algorithm>`__]
# -  ``"ALAW"``: A-law
#    [`wikipedia <https://en.wikipedia.org/wiki/A-law_algorithm>`__]
# -  ``"MP3"`` : MP3, MPEG-1 Audio Layer III
# -  ``"VORBIS"``: OGG Vorbis [`xiph.org <https://xiph.org/vorbis/>`__]
# -  ``"AMR_NB"``: Adaptive Multi-Rate
#    [`wikipedia <https://en.wikipedia.org/wiki/Adaptive_Multi-Rate_audio_codec>`__]
# -  ``"AMR_WB"``: Adaptive Multi-Rate Wideband
#    [`wikipedia <https://en.wikipedia.org/wiki/Adaptive_Multi-Rate_Wideband>`__]
# -  ``"OPUS"``: Opus [`opus-codec.org <https://opus-codec.org/>`__]
# -  ``"GSM"``: GSM-FR
#    [`wikipedia <https://en.wikipedia.org/wiki/Full_Rate>`__]
# -  ``"UNKNOWN"`` None of above
#

######################################################################
# **Note**
#
# -  ``bits_per_sample`` can be ``0`` for formats with compression and/or
#    variable bit rate (such as MP3).
# -  ``num_frames`` can be ``0`` for GSM-FR format.
#
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# .. code::
#
#    metadata = torchaudio.info(SAMPLE_MP3_PATH)
#    print(metadata)
#
#    metadata = torchaudio.info(SAMPLE_GSM_PATH)
#    print(metadata)
#
#    >>> AudioMetaData(sample_rate=44100, num_frames=110559, num_channels=2, bits_per_sample=0, encoding=MP3)
#    >>> AudioMetaData(sample_rate=8000, num_frames=0, num_channels=1, bits_per_sample=0, encoding=GSM)
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######################################################################
# Querying file-like object
# ~~~~~~~~~~~~~~~~~~~~~~~~~
#
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# :py:func:`torchaudio.info` works on file-like objects.
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#

print("Source:", SAMPLE_WAV_URL)
with requests.get(SAMPLE_WAV_URL, stream=True) as response:
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    metadata = torchaudio.info(response.raw)
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print(metadata)

######################################################################
# **Note** When passing a file-like object, ``info`` does not read
# all of the underlying data; rather, it reads only a portion
# of the data from the beginning.
# Therefore, for a given audio format, it may not be able to retrieve the
# correct metadata, including the format itself.
# The following example illustrates this.
#
# -  Use argument ``format`` to specify the audio format of the input.
# -  The returned metadata has ``num_frames = 0``
#
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# .. code::
#
#    print("Source:", SAMPLE_MP3_URL)
#    with requests.get(SAMPLE_MP3_URL, stream=True) as response:
#        metadata = torchaudio.info(response.raw, format="mp3")
#
#        print(f"Fetched {response.raw.tell()} bytes.")
#    print(metadata)
#
#    >>> Source: https://pytorch-tutorial-assets.s3.amazonaws.com/steam-train-whistle-daniel_simon.mp3
#    >>> Fetched 8192 bytes.
#    >>> AudioMetaData(sample_rate=44100, num_frames=0, num_channels=2, bits_per_sample=0, encoding=MP3)
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######################################################################
# Loading audio data into Tensor
# ------------------------------
#
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# To load audio data, you can use :py:func:`torchaudio.load`.
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#
# This function accepts a path-like object or file-like object as input.
#
# The returned value is a tuple of waveform (``Tensor``) and sample rate
# (``int``).
#
# By default, the resulting tensor object has ``dtype=torch.float32`` and
# its value range is normalized within ``[-1.0, 1.0]``.
#
# For the list of supported format, please refer to `the torchaudio
# documentation <https://pytorch.org/audio>`__.
#

waveform, sample_rate = torchaudio.load(SAMPLE_WAV_SPEECH_PATH)

print_stats(waveform, sample_rate=sample_rate)
plot_waveform(waveform, sample_rate)
plot_specgram(waveform, sample_rate)
play_audio(waveform, sample_rate)


######################################################################
# Loading from file-like object
# ~~~~~~~~~~~~~~~~~~~~~~~~~~~~~
#
# ``torchaudio``\ ’s I/O functions now support file-like objects. This
# allows for fetching and decoding audio data from locations
# within and beyond the local file system.
# The following examples illustrate this.
#

# Load audio data as HTTP request
with requests.get(SAMPLE_WAV_SPEECH_URL, stream=True) as response:
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    waveform, sample_rate = torchaudio.load(response.raw)
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plot_specgram(waveform, sample_rate, title="HTTP datasource")

# Load audio from tar file
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with tarfile.open(SAMPLE_TAR_PATH, mode="r") as tarfile_:
    fileobj = tarfile_.extractfile(SAMPLE_TAR_ITEM)
    waveform, sample_rate = torchaudio.load(fileobj)
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plot_specgram(waveform, sample_rate, title="TAR file")

# Load audio from S3
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client = boto3.client("s3", config=Config(signature_version=UNSIGNED))
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response = client.get_object(Bucket=S3_BUCKET, Key=S3_KEY)
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waveform, sample_rate = torchaudio.load(response["Body"])
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plot_specgram(waveform, sample_rate, title="From S3")


######################################################################
# Tips on slicing
# ~~~~~~~~~~~~~~~
#
# Providing ``num_frames`` and ``frame_offset`` arguments restricts
# decoding to the corresponding segment of the input.
#
# The same result can be achieved using vanilla Tensor slicing,
# (i.e. ``waveform[:, frame_offset:frame_offset+num_frames]``). However,
# providing ``num_frames`` and ``frame_offset`` arguments is more
# efficient.
#
# This is because the function will end data acquisition and decoding
# once it finishes decoding the requested frames. This is advantageous
# when the audio data are transferred via network as the data transfer will
# stop as soon as the necessary amount of data is fetched.
#
# The following example illustrates this.
#

# Illustration of two different decoding methods.
# The first one will fetch all the data and decode them, while
# the second one will stop fetching data once it completes decoding.
# The resulting waveforms are identical.

frame_offset, num_frames = 16000, 16000  # Fetch and decode the 1 - 2 seconds

print("Fetching all the data...")
with requests.get(SAMPLE_WAV_SPEECH_URL, stream=True) as response:
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    waveform1, sample_rate1 = torchaudio.load(response.raw)
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    waveform1 = waveform1[:, frame_offset : frame_offset + num_frames]
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    print(f" - Fetched {response.raw.tell()} bytes")
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print("Fetching until the requested frames are available...")
with requests.get(SAMPLE_WAV_SPEECH_URL, stream=True) as response:
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    waveform2, sample_rate2 = torchaudio.load(response.raw, frame_offset=frame_offset, num_frames=num_frames)
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    print(f" - Fetched {response.raw.tell()} bytes")
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print("Checking the resulting waveform ... ", end="")
assert (waveform1 == waveform2).all()
print("matched!")


######################################################################
# Saving audio to file
# --------------------
#
# To save audio data in formats interpretable by common applications,
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# you can use :py:func:`torchaudio.save`.
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#
# This function accepts a path-like object or file-like object.
#
# When passing a file-like object, you also need to provide argument ``format``
# so that the function knows which format it should use. In the
# case of a path-like object, the function will infer the format from
# the extension. If you are saving to a file without an extension, you need
# to provide argument ``format``.
#
# When saving WAV-formatted data, the default encoding for ``float32`` Tensor
# is 32-bit floating-point PCM. You can provide arguments ``encoding`` and
# ``bits_per_sample`` to change this behavior. For example, to save data
# in 16-bit signed integer PCM, you can do the following.
#
# **Note** Saving data in encodings with lower bit depth reduces the
# resulting file size but also precision.
#


waveform, sample_rate = get_sample()
print_stats(waveform, sample_rate=sample_rate)

# Save without any encoding option.
# The function will pick up the encoding which
# the provided data fit
path = f"{_SAMPLE_DIR}/save_example_default.wav"
torchaudio.save(path, waveform, sample_rate)
inspect_file(path)

# Save as 16-bit signed integer Linear PCM
# The resulting file occupies half the storage but loses precision
path = f"{_SAMPLE_DIR}/save_example_PCM_S16.wav"
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torchaudio.save(path, waveform, sample_rate, encoding="PCM_S", bits_per_sample=16)
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inspect_file(path)


######################################################################
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# :py:func`torchaudio.save` can also handle other formats.
# To name a few:
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#

waveform, sample_rate = get_sample(resample=8000)

formats = [
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    "flac",
    "vorbis",
    "sph",
    "amb",
    "amr-nb",
    "gsm",
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]

for format in formats:
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    path = f"{_SAMPLE_DIR}/save_example.{format}"
    torchaudio.save(path, waveform, sample_rate, format=format)
    inspect_file(path)
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######################################################################
# Saving to file-like object
# ~~~~~~~~~~~~~~~~~~~~~~~~~~
#
# Similar to the other I/O functions, you can save audio to file-like
# objects. When saving to a file-like object, argument ``format`` is
# required.
#


waveform, sample_rate = get_sample()

# Saving to bytes buffer
buffer_ = io.BytesIO()
torchaudio.save(buffer_, waveform, sample_rate, format="wav")

buffer_.seek(0)
print(buffer_.read(16))