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# Copyright 2023 The HuggingFace Team. All rights reserved. | |
import datetime | |
import platform | |
import subprocess | |
from typing import Optional, Tuple, Union | |
import numpy as np | |
def ffmpeg_read(bpayload: bytes, sampling_rate: int) -> np.array: | |
""" | |
Helper function to read an audio file through ffmpeg. | |
""" | |
ar = f"{sampling_rate}" | |
ac = "1" | |
format_for_conversion = "f32le" | |
ffmpeg_command = [ | |
"ffmpeg", | |
"-i", | |
"pipe:0", | |
"-ac", | |
ac, | |
"-ar", | |
ar, | |
"-f", | |
format_for_conversion, | |
"-hide_banner", | |
"-loglevel", | |
"quiet", | |
"pipe:1", | |
] | |
try: | |
with subprocess.Popen(ffmpeg_command, stdin=subprocess.PIPE, stdout=subprocess.PIPE) as ffmpeg_process: | |
output_stream = ffmpeg_process.communicate(bpayload) | |
except FileNotFoundError as error: | |
raise ValueError("ffmpeg was not found but is required to load audio files from filename") from error | |
out_bytes = output_stream[0] | |
audio = np.frombuffer(out_bytes, np.float32) | |
if audio.shape[0] == 0: | |
raise ValueError( | |
"Soundfile is either not in the correct format or is malformed. Ensure that the soundfile has " | |
"a valid audio file extension (e.g. wav, flac or mp3) and is not corrupted. If reading from a remote " | |
"URL, ensure that the URL is the full address to **download** the audio file." | |
) | |
return audio | |
def ffmpeg_microphone( | |
sampling_rate: int, | |
chunk_length_s: float, | |
format_for_conversion: str = "f32le", | |
): | |
""" | |
Helper function ro read raw microphone data. | |
""" | |
ar = f"{sampling_rate}" | |
ac = "1" | |
if format_for_conversion == "s16le": | |
size_of_sample = 2 | |
elif format_for_conversion == "f32le": | |
size_of_sample = 4 | |
else: | |
raise ValueError(f"Unhandled format `{format_for_conversion}`. Please use `s16le` or `f32le`") | |
system = platform.system() | |
if system == "Linux": | |
format_ = "alsa" | |
input_ = "default" | |
elif system == "Darwin": | |
format_ = "avfoundation" | |
input_ = ":0" | |
elif system == "Windows": | |
format_ = "dshow" | |
input_ = "default" | |
ffmpeg_command = [ | |
"ffmpeg", | |
"-f", | |
format_, | |
"-i", | |
input_, | |
"-ac", | |
ac, | |
"-ar", | |
ar, | |
"-f", | |
format_for_conversion, | |
"-fflags", | |
"nobuffer", | |
"-hide_banner", | |
"-loglevel", | |
"quiet", | |
"pipe:1", | |
] | |
chunk_len = int(round(sampling_rate * chunk_length_s)) * size_of_sample | |
iterator = _ffmpeg_stream(ffmpeg_command, chunk_len) | |
for item in iterator: | |
yield item | |
def ffmpeg_microphone_live( | |
sampling_rate: int, | |
chunk_length_s: float, | |
stream_chunk_s: Optional[int] = None, | |
stride_length_s: Optional[Union[Tuple[float, float], float]] = None, | |
format_for_conversion: str = "f32le", | |
): | |
""" | |
Helper function to read audio from the microphone file through ffmpeg. This will output `partial` overlapping | |
chunks starting from `stream_chunk_s` (if it is defined) until `chunk_length_s` is reached. It will make use of | |
striding to avoid errors on the "sides" of the various chunks. | |
Arguments: | |
sampling_rate (`int`): | |
The sampling_rate to use when reading the data from the microphone. Try using the model's sampling_rate to | |
avoid resampling later. | |
chunk_length_s (`float` or `int`): | |
The length of the maximum chunk of audio to be sent returned. This includes the eventual striding. | |
stream_chunk_s (`float` or `int`) | |
The length of the minimal temporary audio to be returned. | |
stride_length_s (`float` or `int` or `(float, float)`, *optional*, defaults to `None`) | |
The length of the striding to be used. Stride is used to provide context to a model on the (left, right) of | |
an audio sample but without using that part to actually make the prediction. Setting this does not change | |
the length of the chunk. | |
format_for_conversion (`str`, defalts to `f32le`) | |
The name of the format of the audio samples to be returned by ffmpeg. The standard is `f32le`, `s16le` | |
could also be used. | |
Return: | |
A generator yielding dictionaries of the following form | |
`{"sampling_rate": int, "raw": np.array(), "partial" bool}` With optionnally a `"stride" (int, int)` key if | |
`stride_length_s` is defined. | |
`stride` and `raw` are all expressed in `samples`, and `partial` is a boolean saying if the current yield item | |
is a whole chunk, or a partial temporary result to be later replaced by another larger chunk. | |
""" | |
if stream_chunk_s is not None: | |
chunk_s = stream_chunk_s | |
else: | |
chunk_s = chunk_length_s | |
microphone = ffmpeg_microphone(sampling_rate, chunk_s, format_for_conversion=format_for_conversion) | |
if format_for_conversion == "s16le": | |
dtype = np.int16 | |
size_of_sample = 2 | |
elif format_for_conversion == "f32le": | |
dtype = np.float32 | |
size_of_sample = 4 | |
else: | |
raise ValueError(f"Unhandled format `{format_for_conversion}`. Please use `s16le` or `f32le`") | |
if stride_length_s is None: | |
stride_length_s = chunk_length_s / 6 | |
chunk_len = int(round(sampling_rate * chunk_length_s)) * size_of_sample | |
if isinstance(stride_length_s, (int, float)): | |
stride_length_s = [stride_length_s, stride_length_s] | |
stride_left = int(round(sampling_rate * stride_length_s[0])) * size_of_sample | |
stride_right = int(round(sampling_rate * stride_length_s[1])) * size_of_sample | |
audio_time = datetime.datetime.now() | |
delta = datetime.timedelta(seconds=chunk_s) | |
for item in chunk_bytes_iter(microphone, chunk_len, stride=(stride_left, stride_right), stream=True): | |
# Put everything back in numpy scale | |
item["raw"] = np.frombuffer(item["raw"], dtype=dtype) | |
item["stride"] = ( | |
item["stride"][0] // size_of_sample, | |
item["stride"][1] // size_of_sample, | |
) | |
item["sampling_rate"] = sampling_rate | |
audio_time += delta | |
if datetime.datetime.now() > audio_time + 10 * delta: | |
# We're late !! SKIP | |
continue | |
yield item | |
def chunk_bytes_iter(iterator, chunk_len: int, stride: Tuple[int, int], stream: bool = False): | |
""" | |
Reads raw bytes from an iterator and does chunks of length `chunk_len`. Optionally adds `stride` to each chunks to | |
get overlaps. `stream` is used to return partial results even if a full `chunk_len` is not yet available. | |
""" | |
acc = b"" | |
stride_left, stride_right = stride | |
if stride_left + stride_right >= chunk_len: | |
raise ValueError( | |
f"Stride needs to be strictly smaller than chunk_len: ({stride_left}, {stride_right}) vs {chunk_len}" | |
) | |
_stride_left = 0 | |
for raw in iterator: | |
acc += raw | |
if stream and len(acc) < chunk_len: | |
stride = (_stride_left, 0) | |
yield {"raw": acc[:chunk_len], "stride": stride, "partial": True} | |
else: | |
while len(acc) >= chunk_len: | |
# We are flushing the accumulator | |
stride = (_stride_left, stride_right) | |
item = {"raw": acc[:chunk_len], "stride": stride} | |
if stream: | |
item["partial"] = False | |
yield item | |
_stride_left = stride_left | |
acc = acc[chunk_len - stride_left - stride_right :] | |
# Last chunk | |
if len(acc) > stride_left: | |
item = {"raw": acc, "stride": (_stride_left, 0)} | |
if stream: | |
item["partial"] = False | |
yield item | |
def _ffmpeg_stream(ffmpeg_command, buflen: int): | |
""" | |
Internal function to create the generator of data through ffmpeg | |
""" | |
bufsize = 2**24 # 16Mo | |
try: | |
with subprocess.Popen(ffmpeg_command, stdout=subprocess.PIPE, bufsize=bufsize) as ffmpeg_process: | |
while True: | |
raw = ffmpeg_process.stdout.read(buflen) | |
if raw == b"": | |
break | |
yield raw | |
except FileNotFoundError as error: | |
raise ValueError("ffmpeg was not found but is required to stream audio files from filename") from error | |