Refactor 🔨

Signed-off-by: makaveli10 <vineet.suryan@collabora.com>
This commit is contained in:
makaveli10
2025-03-24 16:48:33 +05:30
parent f5bea0a693
commit c1ac71ada0
10 changed files with 788 additions and 737 deletions
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# SPDX-FileCopyrightText: Copyright (c) 2022-2023 NVIDIA CORPORATION & AFFILIATES. All rights reserved.
# SPDX-License-Identifier: Apache-2.0
#
# Licensed under the Apache License, Version 2.0 (the "License");
# you may not use this file except in compliance with the License.
# You may obtain a copy of the License at
#
# http://www.apache.org/licenses/LICENSE-2.0
#
# Unless required by applicable law or agreed to in writing, software
# distributed under the License is distributed on an "AS IS" BASIS,
# WITHOUT WARRANTIES OR CONDITIONS OF ANY KIND, either express or implied.
# See the License for the specific language governing permissions and
# limitations under the License.
import logging
import os
from collections import defaultdict
from functools import lru_cache
from pathlib import Path
from subprocess import CalledProcessError, run
from typing import Dict, Iterable, List, Optional, TextIO, Tuple, Union
import kaldialign
import numpy as np
import soundfile
import av
import wave
import torch
import torch.nn.functional as F
from whisper_live.utils import resample
Pathlike = Union[str, Path]
SAMPLE_RATE = 16000
N_FFT = 400
HOP_LENGTH = 160
CHUNK_LENGTH = 30
N_SAMPLES = CHUNK_LENGTH * SAMPLE_RATE # 480000 samples in a 30-second chunk
def load_audio(file: str, sr: int = 16000):
"""
Open an audio file, resample it, and read as a mono waveform.
Parameters
----------
file: str
The audio file to open.
sr: int
The sample rate to resample the audio if necessary.
Returns
-------
A NumPy array containing the audio waveform, in float32 dtype.
"""
resampled_file = resample(file, sr)
with wave.open(resampled_file, "rb") as wav_file:
num_frames = wav_file.getnframes()
raw_data = wav_file.readframes(num_frames)
audio_data = np.frombuffer(raw_data, dtype=np.int16)
audio_data = audio_data.astype(np.float32) / 32768.0
return audio_data
def load_audio_wav_format(wav_path):
# make sure audio in .wav format
assert wav_path.endswith(
'.wav'), f"Only support .wav format, but got {wav_path}"
waveform, sample_rate = soundfile.read(wav_path)
assert sample_rate == 16000, f"Only support 16k sample rate, but got {sample_rate}"
return waveform, sample_rate
def pad_or_trim(array, length: int = N_SAMPLES, *, axis: int = -1):
"""
Pad or trim the audio array to N_SAMPLES, as expected by the encoder.
"""
if torch.is_tensor(array):
if array.shape[axis] > length:
array = array.index_select(dim=axis,
index=torch.arange(length,
device=array.device))
if array.shape[axis] < length:
pad_widths = [(0, 0)] * array.ndim
pad_widths[axis] = (0, length - array.shape[axis])
array = F.pad(array,
[pad for sizes in pad_widths[::-1] for pad in sizes])
else:
if array.shape[axis] > length:
array = array.take(indices=range(length), axis=axis)
if array.shape[axis] < length:
pad_widths = [(0, 0)] * array.ndim
pad_widths[axis] = (0, length - array.shape[axis])
array = np.pad(array, pad_widths)
return array
@lru_cache(maxsize=None)
def mel_filters(device,
n_mels: int,
mel_filters_dir: str = None) -> torch.Tensor:
"""
load the mel filterbank matrix for projecting STFT into a Mel spectrogram.
Allows decoupling librosa dependency; saved using:
np.savez_compressed(
"mel_filters.npz",
mel_80=librosa.filters.mel(sr=16000, n_fft=400, n_mels=80),
)
"""
assert n_mels in {80, 128}, f"Unsupported n_mels: {n_mels}"
if mel_filters_dir is None:
mel_filters_path = os.path.join(os.path.dirname(__file__), "assets",
"mel_filters.npz")
else:
mel_filters_path = os.path.join(mel_filters_dir, "mel_filters.npz")
with np.load(mel_filters_path) as f:
return torch.from_numpy(f[f"mel_{n_mels}"]).to(device)
def log_mel_spectrogram(
audio: Union[str, np.ndarray, torch.Tensor],
n_mels: int,
padding: int = 0,
device: Optional[Union[str, torch.device]] = None,
return_duration: bool = False,
mel_filters_dir: str = None,
):
"""
Compute the log-Mel spectrogram of
Parameters
----------
audio: Union[str, np.ndarray, torch.Tensor], shape = (*)
The path to audio or either a NumPy array or Tensor containing the audio waveform in 16 kHz
n_mels: int
The number of Mel-frequency filters, only 80 and 128 are supported
padding: int
Number of zero samples to pad to the right
device: Optional[Union[str, torch.device]]
If given, the audio tensor is moved to this device before STFT
Returns
-------
torch.Tensor, shape = (80 or 128, n_frames)
A Tensor that contains the Mel spectrogram
"""
if not torch.is_tensor(audio):
if isinstance(audio, str):
if audio.endswith('.wav'):
audio, _ = load_audio_wav_format(audio)
else:
audio = load_audio(audio)
assert isinstance(audio,
np.ndarray), f"Unsupported audio type: {type(audio)}"
duration = audio.shape[-1] / SAMPLE_RATE
audio = pad_or_trim(audio, N_SAMPLES)
audio = audio.astype(np.float32)
audio = torch.from_numpy(audio)
if device is not None:
audio = audio.to(device)
if padding > 0:
audio = F.pad(audio, (0, padding))
window = torch.hann_window(N_FFT).to(audio.device)
stft = torch.stft(audio,
N_FFT,
HOP_LENGTH,
window=window,
return_complex=True)
magnitudes = stft[..., :-1].abs()**2
filters = mel_filters(audio.device, n_mels, mel_filters_dir)
mel_spec = filters @ magnitudes
log_spec = torch.clamp(mel_spec, min=1e-10).log10()
log_spec = torch.maximum(log_spec, log_spec.max() - 8.0)
log_spec = (log_spec + 4.0) / 4.0
if return_duration:
return log_spec, duration
else:
return log_spec
def store_transcripts(filename: Pathlike, texts: Iterable[Tuple[str, str,
str]]) -> None:
"""Save predicted results and reference transcripts to a file.
https://github.com/k2-fsa/icefall/blob/master/icefall/utils.py
Args:
filename:
File to save the results to.
texts:
An iterable of tuples. The first element is the cur_id, the second is
the reference transcript and the third element is the predicted result.
Returns:
Return None.
"""
with open(filename, "w") as f:
for cut_id, ref, hyp in texts:
print(f"{cut_id}:\tref={ref}", file=f)
print(f"{cut_id}:\thyp={hyp}", file=f)
def write_error_stats( # noqa: C901
f: TextIO,
test_set_name: str,
results: List[Tuple[str, str]],
enable_log: bool = True,
) -> float:
"""Write statistics based on predicted results and reference transcripts.
https://github.com/k2-fsa/icefall/blob/master/icefall/utils.py
It will write the following to the given file:
- WER
- number of insertions, deletions, substitutions, corrects and total
reference words. For example::
Errors: 23 insertions, 57 deletions, 212 substitutions, over 2606
reference words (2337 correct)
- The difference between the reference transcript and predicted result.
An instance is given below::
THE ASSOCIATION OF (EDISON->ADDISON) ILLUMINATING COMPANIES
The above example shows that the reference word is `EDISON`,
but it is predicted to `ADDISON` (a substitution error).
Another example is::
FOR THE FIRST DAY (SIR->*) I THINK
The reference word `SIR` is missing in the predicted
results (a deletion error).
results:
An iterable of tuples. The first element is the cur_id, the second is
the reference transcript and the third element is the predicted result.
enable_log:
If True, also print detailed WER to the console.
Otherwise, it is written only to the given file.
Returns:
Return None.
"""
subs: Dict[Tuple[str, str], int] = defaultdict(int)
ins: Dict[str, int] = defaultdict(int)
dels: Dict[str, int] = defaultdict(int)
# `words` stores counts per word, as follows:
# corr, ref_sub, hyp_sub, ins, dels
words: Dict[str, List[int]] = defaultdict(lambda: [0, 0, 0, 0, 0])
num_corr = 0
ERR = "*"
for cut_id, ref, hyp in results:
ali = kaldialign.align(ref, hyp, ERR)
for ref_word, hyp_word in ali:
if ref_word == ERR:
ins[hyp_word] += 1
words[hyp_word][3] += 1
elif hyp_word == ERR:
dels[ref_word] += 1
words[ref_word][4] += 1
elif hyp_word != ref_word:
subs[(ref_word, hyp_word)] += 1
words[ref_word][1] += 1
words[hyp_word][2] += 1
else:
words[ref_word][0] += 1
num_corr += 1
ref_len = sum([len(r) for _, r, _ in results])
sub_errs = sum(subs.values())
ins_errs = sum(ins.values())
del_errs = sum(dels.values())
tot_errs = sub_errs + ins_errs + del_errs
tot_err_rate = "%.2f" % (100.0 * tot_errs / ref_len)
if enable_log:
logging.info(f"[{test_set_name}] %WER {tot_errs / ref_len:.2%} "
f"[{tot_errs} / {ref_len}, {ins_errs} ins, "
f"{del_errs} del, {sub_errs} sub ]")
print(f"%WER = {tot_err_rate}", file=f)
print(
f"Errors: {ins_errs} insertions, {del_errs} deletions, "
f"{sub_errs} substitutions, over {ref_len} reference "
f"words ({num_corr} correct)",
file=f,
)
print(
"Search below for sections starting with PER-UTT DETAILS:, "
"SUBSTITUTIONS:, DELETIONS:, INSERTIONS:, PER-WORD STATS:",
file=f,
)
print("", file=f)
print("PER-UTT DETAILS: corr or (ref->hyp) ", file=f)
for cut_id, ref, hyp in results:
ali = kaldialign.align(ref, hyp, ERR)
combine_successive_errors = True
if combine_successive_errors:
ali = [[[x], [y]] for x, y in ali]
for i in range(len(ali) - 1):
if ali[i][0] != ali[i][1] and ali[i + 1][0] != ali[i + 1][1]:
ali[i + 1][0] = ali[i][0] + ali[i + 1][0]
ali[i + 1][1] = ali[i][1] + ali[i + 1][1]
ali[i] = [[], []]
ali = [[
list(filter(lambda a: a != ERR, x)),
list(filter(lambda a: a != ERR, y)),
] for x, y in ali]
ali = list(filter(lambda x: x != [[], []], ali))
ali = [[
ERR if x == [] else " ".join(x),
ERR if y == [] else " ".join(y),
] for x, y in ali]
print(
f"{cut_id}:\t" + " ".join((ref_word if ref_word == hyp_word else
f"({ref_word}->{hyp_word})"
for ref_word, hyp_word in ali)),
file=f,
)
print("", file=f)
print("SUBSTITUTIONS: count ref -> hyp", file=f)
for count, (ref, hyp) in sorted([(v, k) for k, v in subs.items()],
reverse=True):
print(f"{count} {ref} -> {hyp}", file=f)
print("", file=f)
print("DELETIONS: count ref", file=f)
for count, ref in sorted([(v, k) for k, v in dels.items()], reverse=True):
print(f"{count} {ref}", file=f)
print("", file=f)
print("INSERTIONS: count hyp", file=f)
for count, hyp in sorted([(v, k) for k, v in ins.items()], reverse=True):
print(f"{count} {hyp}", file=f)
print("", file=f)
print("PER-WORD STATS: word corr tot_errs count_in_ref count_in_hyp",
file=f)
for _, word, counts in sorted([(sum(v[1:]), k, v)
for k, v in words.items()],
reverse=True):
(corr, ref_sub, hyp_sub, ins, dels) = counts
tot_errs = ref_sub + hyp_sub + ins + dels
ref_count = corr + ref_sub + dels
hyp_count = corr + hyp_sub + ins
print(f"{word} {corr} {tot_errs} {ref_count} {hyp_count}", file=f)
return float(tot_err_rate)
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import json
import re
import math
from collections import OrderedDict
from pathlib import Path
from typing import Union
import torch
import numpy as np
import torch.nn.functional as F
from whisper.tokenizer import get_tokenizer
from whisper_live.transcriber.tensorrt_utils import (
mel_filters,
load_audio_wav_format,
pad_or_trim,
load_audio
)
import tensorrt_llm
import tensorrt_llm.logger as logger
from tensorrt_llm._utils import (str_dtype_to_torch, str_dtype_to_trt,
trt_dtype_to_torch)
from tensorrt_llm.bindings import GptJsonConfig, KVCacheType
from tensorrt_llm.runtime import PYTHON_BINDINGS, ModelConfig, SamplingConfig
from tensorrt_llm.runtime.session import Session, TensorInfo
SAMPLE_RATE = 16000
N_FFT = 400
HOP_LENGTH = 160
CHUNK_LENGTH = 30
N_SAMPLES = CHUNK_LENGTH * SAMPLE_RATE # 480000 samples in a 30-second chunk
def read_config(component, engine_dir):
config_path = engine_dir / component / 'config.json'
with open(config_path, 'r') as f:
config = json.load(f)
model_config = OrderedDict()
model_config.update(config['pretrained_config'])
model_config.update(config['build_config'])
return model_config
def remove_tensor_padding(input_tensor,
input_tensor_lengths=None,
pad_value=None):
if pad_value:
assert input_tensor_lengths is None, "input_tensor_lengths should be None when pad_value is provided"
# Text tensor case: batch, seq_len
assert torch.all(
input_tensor[:, 0] != pad_value
), "First token in each sequence should not be pad_value"
assert input_tensor_lengths is None
# Create a mask for all non-pad tokens
mask = input_tensor != pad_value
# Apply the mask to input_tensor to remove pad tokens
output_tensor = input_tensor[mask].view(1, -1)
else:
# Audio tensor case: batch, seq_len, feature_len
# position_ids case: batch, seq_len
assert input_tensor_lengths is not None, "input_tensor_lengths must be provided for 3D input_tensor"
# Initialize a list to collect valid sequences
valid_sequences = []
for i in range(input_tensor.shape[0]):
valid_length = input_tensor_lengths[i]
valid_sequences.append(input_tensor[i, :valid_length])
# Concatenate all valid sequences along the batch dimension
output_tensor = torch.cat(valid_sequences, dim=0)
return output_tensor
class WhisperEncoding:
def __init__(self, engine_dir):
self.session = self.get_session(engine_dir)
config = read_config('encoder', engine_dir)
self.n_mels = config['n_mels']
self.dtype = config['dtype']
self.num_languages = config['num_languages']
self.encoder_config = config
def get_session(self, engine_dir):
serialize_path = engine_dir / 'encoder' / 'rank0.engine'
with open(serialize_path, 'rb') as f:
session = Session.from_serialized_engine(f.read())
return session
def get_audio_features(self,
mel,
mel_input_lengths,
encoder_downsampling_factor=2):
if isinstance(mel, list):
longest_mel = max([f.shape[-1] for f in mel])
mel = [
torch.nn.functional.pad(f, (0, longest_mel - f.shape[-1]),
mode='constant') for f in mel
]
mel = torch.cat(mel, dim=0).type(
str_dtype_to_torch("float16")).contiguous()
bsz, seq_len = mel.shape[0], mel.shape[2]
position_ids = torch.arange(
math.ceil(seq_len / encoder_downsampling_factor),
dtype=torch.int32,
device=mel.device).expand(bsz, -1).contiguous()
if self.encoder_config['plugin_config']['remove_input_padding']:
# mel B,D,T -> B,T,D -> BxT, D
mel = mel.transpose(1, 2)
mel = remove_tensor_padding(mel, mel_input_lengths)
position_ids = remove_tensor_padding(
position_ids, mel_input_lengths // encoder_downsampling_factor)
inputs = OrderedDict()
inputs['input_features'] = mel
inputs['input_lengths'] = mel_input_lengths
inputs['position_ids'] = position_ids
output_list = [
TensorInfo('input_features', str_dtype_to_trt(self.dtype),
mel.shape),
TensorInfo('input_lengths', str_dtype_to_trt('int32'),
mel_input_lengths.shape),
TensorInfo('position_ids', str_dtype_to_trt('int32'),
inputs['position_ids'].shape)
]
output_info = (self.session).infer_shapes(output_list)
logger.debug(f'output info {output_info}')
outputs = {
t.name: torch.empty(tuple(t.shape),
dtype=trt_dtype_to_torch(t.dtype),
device='cuda')
for t in output_info
}
stream = torch.cuda.current_stream()
ok = self.session.run(inputs=inputs,
outputs=outputs,
stream=stream.cuda_stream)
assert ok, 'Engine execution failed'
stream.synchronize()
encoder_output = outputs['encoder_output']
encoder_output_lengths = mel_input_lengths // encoder_downsampling_factor
return encoder_output, encoder_output_lengths
class WhisperDecoding:
def __init__(self, engine_dir, runtime_mapping, debug_mode=False):
self.decoder_config = read_config('decoder', engine_dir)
self.decoder_generation_session = self.get_session(
engine_dir, runtime_mapping, debug_mode)
def get_session(self, engine_dir, runtime_mapping, debug_mode=False):
serialize_path = engine_dir / 'decoder' / 'rank0.engine'
with open(serialize_path, "rb") as f:
decoder_engine_buffer = f.read()
decoder_model_config = ModelConfig(
max_batch_size=self.decoder_config['max_batch_size'],
max_beam_width=self.decoder_config['max_beam_width'],
num_heads=self.decoder_config['num_attention_heads'],
num_kv_heads=self.decoder_config['num_attention_heads'],
hidden_size=self.decoder_config['hidden_size'],
vocab_size=self.decoder_config['vocab_size'],
cross_attention=True,
num_layers=self.decoder_config['num_hidden_layers'],
gpt_attention_plugin=self.decoder_config['plugin_config']
['gpt_attention_plugin'],
remove_input_padding=self.decoder_config['plugin_config']
['remove_input_padding'],
kv_cache_type=KVCacheType.PAGED
if self.decoder_config['plugin_config']['paged_kv_cache'] == True
else KVCacheType.CONTINUOUS,
has_position_embedding=self.
decoder_config['has_position_embedding'],
dtype=self.decoder_config['dtype'],
has_token_type_embedding=False,
)
decoder_generation_session = tensorrt_llm.runtime.GenerationSession(
decoder_model_config,
decoder_engine_buffer,
runtime_mapping,
debug_mode=debug_mode)
return decoder_generation_session
def generate(self,
decoder_input_ids,
encoder_outputs,
encoder_max_input_length,
encoder_input_lengths,
eot_id,
max_new_tokens=40,
num_beams=1):
batch_size = decoder_input_ids.shape[0]
decoder_input_lengths = torch.tensor([
decoder_input_ids.shape[-1]
for _ in range(decoder_input_ids.shape[0])
],
dtype=torch.int32,
device='cuda')
decoder_max_input_length = torch.max(decoder_input_lengths).item()
cross_attention_mask = torch.ones([
batch_size, decoder_max_input_length + max_new_tokens,
encoder_max_input_length
]).int().cuda()
# generation config
sampling_config = SamplingConfig(end_id=eot_id,
pad_id=eot_id,
num_beams=num_beams)
self.decoder_generation_session.setup(
decoder_input_lengths.size(0),
decoder_max_input_length,
max_new_tokens,
beam_width=num_beams,
encoder_max_input_length=encoder_max_input_length)
torch.cuda.synchronize()
decoder_input_ids = decoder_input_ids.type(torch.int32).cuda()
if self.decoder_config['plugin_config']['remove_input_padding']:
# 50256 is the index of <pad> for all whisper models' decoder
WHISPER_PAD_TOKEN_ID = 50256
decoder_input_ids = remove_tensor_padding(
decoder_input_ids, pad_value=WHISPER_PAD_TOKEN_ID)
if encoder_outputs.dim() == 3:
encoder_output_lens = torch.full((encoder_outputs.shape[0], ),
encoder_outputs.shape[1],
dtype=torch.int32,
device='cuda')
encoder_outputs = remove_tensor_padding(encoder_outputs,
encoder_output_lens)
output_ids = self.decoder_generation_session.decode(
decoder_input_ids,
decoder_input_lengths,
sampling_config,
encoder_output=encoder_outputs,
encoder_input_lengths=encoder_input_lengths,
cross_attention_mask=cross_attention_mask,
)
torch.cuda.synchronize()
# get the list of int from output_ids tensor
output_ids = output_ids.cpu().numpy().tolist()
return output_ids
class WhisperTRTLLM(object):
def __init__(self, engine_dir, assets_dir=None, device=None, is_multilingual=False,
language="en", task="transcribe"):
world_size = 1
runtime_rank = tensorrt_llm.mpi_rank()
runtime_mapping = tensorrt_llm.Mapping(world_size, runtime_rank)
torch.cuda.set_device(runtime_rank % runtime_mapping.gpus_per_node)
engine_dir = Path(engine_dir)
encoder_config = read_config('encoder', engine_dir)
decoder_config = read_config('decoder', engine_dir)
self.n_mels = encoder_config['n_mels']
self.num_languages = encoder_config['num_languages']
is_multilingual = (decoder_config['vocab_size'] >= 51865)
self.encoder = WhisperEncoding(engine_dir)
self.decoder = WhisperDecoding(engine_dir,
runtime_mapping,
debug_mode=False)
self.n_mels = self.encoder.n_mels
# self.tokenizer = get_tokenizer(num_languages=self.encoder.num_languages,
# tokenizer_dir=assets_dir)
self.device = device
self.tokenizer = get_tokenizer(
is_multilingual,
num_languages=self.num_languages,
language=language,
task=task,
)
self.filters = mel_filters(self.device, self.encoder.n_mels, assets_dir)
def log_mel_spectrogram(
self,
audio: Union[str, np.ndarray, torch.Tensor],
padding: int = 0,
return_duration=True
):
"""
Compute the log-Mel spectrogram of
Parameters
----------
audio: Union[str, np.ndarray, torch.Tensor], shape = (*)
The path to audio or either a NumPy array or Tensor containing the audio waveform in 16 kHz
n_mels: int
The number of Mel-frequency filters, only 80 and 128 are supported
padding: int
Number of zero samples to pad to the right
device: Optional[Union[str, torch.device]]
If given, the audio tensor is moved to this device before STFT
Returns
-------
torch.Tensor, shape = (80 or 128, n_frames)
A Tensor that contains the Mel spectrogram
"""
if not torch.is_tensor(audio):
if isinstance(audio, str):
if audio.endswith('.wav'):
audio, _ = load_audio_wav_format(audio)
else:
audio = load_audio(audio)
assert isinstance(audio, np.ndarray), f"Unsupported audio type: {type(audio)}"
duration = audio.shape[-1] / SAMPLE_RATE
audio = pad_or_trim(audio, N_SAMPLES)
audio = audio.astype(np.float32)
audio = torch.from_numpy(audio)
if self.device is not None:
audio = audio.to(self.device)
if padding > 0:
audio = F.pad(audio, (0, padding))
window = torch.hann_window(N_FFT).to(audio.device)
stft = torch.stft(audio, N_FFT, HOP_LENGTH, window=window, return_complex=True)
magnitudes = stft[..., :-1].abs()**2
mel_spec = self.filters @ magnitudes
log_spec = torch.clamp(mel_spec, min=1e-10).log10()
log_spec = torch.maximum(log_spec, log_spec.max() - 8.0)
log_spec = (log_spec + 4.0) / 4.0
if return_duration:
return log_spec, duration
else:
return log_spec
def process_batch(
self,
mel,
mel_input_lengths,
text_prefix="<|startoftranscript|><|en|><|transcribe|><|notimestamps|>",
num_beams=1,
max_new_tokens=96):
prompt_id = self.tokenizer.encode(
text_prefix, allowed_special=set(self.tokenizer.special_tokens.keys()))
prompt_id = torch.tensor(prompt_id)
batch_size = mel.shape[0]
decoder_input_ids = prompt_id.repeat(batch_size, 1)
encoder_output, encoder_output_lengths = self.encoder.get_audio_features(mel, mel_input_lengths)
encoder_max_input_length = torch.max(encoder_output_lengths).item()
output_ids = self.decoder.generate(decoder_input_ids,
encoder_output,
encoder_max_input_length,
encoder_output_lengths,
self.tokenizer.eot,
max_new_tokens=max_new_tokens,
num_beams=num_beams)
texts = []
for i in range(len(output_ids)):
text = self.tokenizer.decode(output_ids[i][0]).strip()
texts.append(text)
return texts
def transcribe(
self,
mel,
text_prefix="<|startoftranscript|><|en|><|transcribe|><|notimestamps|>",
dtype='float16',
batch_size=1,
num_beams=1,
padding_strategy="max",
):
mel = mel.type(str_dtype_to_torch(dtype))
mel = mel.unsqueeze(0)
# repeat the mel spectrogram to match the batch size
mel = mel.repeat(batch_size, 1, 1)
if padding_strategy == "longest":
pass
else:
mel = torch.nn.functional.pad(mel, (0, 3000 - mel.shape[2]))
features_input_lengths = torch.full((mel.shape[0], ),
mel.shape[2],
dtype=torch.int32,
device=mel.device)
predictions = self.process_batch(mel, features_input_lengths, text_prefix, num_beams)
prediction = predictions[0]
# remove all special tokens in the prediction
prediction = re.sub(r'<\|.*?\|>', '', prediction)
return prediction.strip()
def decode_wav_file(
model,
mel,
text_prefix="<|startoftranscript|><|en|><|transcribe|><|notimestamps|>",
dtype='float16',
batch_size=1,
num_beams=1,
normalizer=None,
mel_filters_dir=None):
mel = mel.type(str_dtype_to_torch(dtype))
mel = mel.unsqueeze(0)
# repeat the mel spectrogram to match the batch size
mel = mel.repeat(batch_size, 1, 1)
predictions = model.process_batch(mel, text_prefix, num_beams)
prediction = predictions[0]
# remove all special tokens in the prediction
prediction = re.sub(r'<\|.*?\|>', '', prediction)
if normalizer:
prediction = normalizer(prediction)
return prediction.strip()