This commit is contained in:
游雁
2024-03-01 14:58:36 +08:00
parent 5c8af9c7e5
commit cc41a9ee88
26 changed files with 216 additions and 101926 deletions
@@ -0,0 +1,13 @@
#!/usr/bin/env python3
# -*- encoding: utf-8 -*-
# Copyright FunASR (https://github.com/alibaba-damo-academy/FunASR). All Rights Reserved.
# MIT License (https://opensource.org/licenses/MIT)
from funasr import AutoModel
model = AutoModel(model="iic/speech_whisper-large_asr_multilingual",
model_revision="v2.0.4",
)
res = model.generate(input="https://isv-data.oss-cn-hangzhou.aliyuncs.com/ics/MaaS/ASR/test_audio/asr_example_zh.wav")
print(res)
@@ -0,0 +1,21 @@
# Copyright FunASR (https://github.com/alibaba-damo-academy/FunASR). All Rights Reserved.
# MIT License (https://opensource.org/licenses/MIT)
# method1, inference from model hub
# for more input type, please ref to readme.md
input="https://isv-data.oss-cn-hangzhou.aliyuncs.com/ics/MaaS/ASR/test_audio/asr_example_zh.wav"
output_dir="./outputs/debug"
model="iic/speech_whisper-large_asr_multilingual"
model_revision="v2.0.4"
device="cuda:0" # "cuda:0" for gpu0, "cuda:1" for gpu1, "cpu"
python -m funasr.bin.inference \
++model=${model} \
++model_revision=${model_revision} \
++input="${input}" \
++output_dir="${output_dir}" \
++device="${device}" \
@@ -0,0 +1,34 @@
# Copyright FunASR (https://github.com/alibaba-damo-academy/FunASR). All Rights Reserved.
# MIT License (https://opensource.org/licenses/MIT)
# method2, inference from local model
# for more input type, please ref to readme.md
input="https://isv-data.oss-cn-hangzhou.aliyuncs.com/ics/MaaS/ASR/test_audio/asr_example_zh.wav"
output_dir="./outputs/debug"
workspace=`pwd`
# download model
local_path_root=${workspace}/modelscope_models
mkdir -p ${local_path_root}
local_path=${local_path_root}/speech_whisper-large_asr_multilingual
git clone https://www.modelscope.cn/iic/speech_whisper-large_asr_multilingual.git ${local_path}
device="cuda:0" # "cuda:0" for gpu0, "cuda:1" for gpu1, "cpu"
config="config.yaml"
init_param="${local_path}/large-v2.pt"
python -m funasr.bin.inference \
--config-path "${local_path}" \
--config-name "${config}" \
++init_param="${init_param}" \
++input="${input}" \
++output_dir="${output_dir}" \
++device="${device}" \
+3 -2
View File
@@ -165,17 +165,18 @@ class AutoModel:
kwargs["token_list"] = tokenizer.token_list if hasattr(tokenizer, "token_list") else None
kwargs["token_list"] = tokenizer.get_vocab() if hasattr(tokenizer, "get_vocab") else kwargs["token_list"]
vocab_size = len(kwargs["token_list"])
vocab_size = len(kwargs["token_list"]) if kwargs["token_list"] is not None else -1
else:
vocab_size = -1
# build frontend
frontend = kwargs.get("frontend", None)
kwargs["input_size"] = None
if frontend is not None:
frontend_class = tables.frontend_classes.get(frontend)
frontend = frontend_class(**kwargs["frontend_conf"])
kwargs["frontend"] = frontend
kwargs["input_size"] = frontend.output_size()
kwargs["input_size"] = frontend.output_size() if hasattr(frontend, "output_size") else None
# build model
model_class = tables.model_classes.get(kwargs["model"])
+4 -2
View File
@@ -18,14 +18,16 @@ def download_from_ms(**kwargs):
model_or_path = name_maps_ms[model_or_path]
model_revision = kwargs.get("model_revision")
if not os.path.exists(model_or_path):
model_or_path = get_or_download_model_dir(model_or_path, model_revision, is_training=kwargs.get("is_training"), check_latest=kwargs.get("kwargs", True))
model_or_path = get_or_download_model_dir(model_or_path, model_revision, is_training=kwargs.get("is_training"), check_latest=kwargs.get("check_latest", True))
kwargs["model_path"] = model_or_path
if os.path.exists(os.path.join(model_or_path, "configuration.json")):
with open(os.path.join(model_or_path, "configuration.json"), 'r', encoding='utf-8') as f:
conf_json = json.load(f)
cfg = {}
add_file_root_path(model_or_path, conf_json["file_path_metas"], cfg)
if "file_path_metas" in conf_json:
add_file_root_path(model_or_path, conf_json["file_path_metas"], cfg)
cfg.update(kwargs)
config = OmegaConf.load(cfg["config"])
kwargs = OmegaConf.merge(config, cfg)
+4 -4
View File
@@ -17,8 +17,9 @@ class WhisperFrontend(nn.Module):
def __init__(
self,
fs: int = 16000,
whisper_model: str = "large-v3",
whisper_model: str = None,
do_pad_trim: bool = True,
n_mels: int = 80,
):
super().__init__()
assert fs == 16000
@@ -30,17 +31,16 @@ class WhisperFrontend(nn.Module):
self.pad_samples = N_SAMPLES
self.frame_shift = self.hop_length
self.lfr_n = 1
self.n_mels = n_mels
if whisper_model == "large-v3" or whisper_model == "large":
self.n_mels = 128
else:
self.n_mels = 80
self.mel_filters = whisper.audio.mel_filters
self.do_pad_trim = do_pad_trim
if do_pad_trim:
self.pad_or_trim = whisper.pad_or_trim
assert whisper_model in whisper.available_models()
# assert whisper_model in whisper.available_models()
def output_size(self) -> int:
return self.n_mels
+64 -252
View File
@@ -1,273 +1,85 @@
from dataclasses import dataclass
from typing import Dict
from typing import Iterable, Optional
import time
import numpy as np
import torch
import torch.nn.functional as F
from torch import Tensor
from torch import nn
import whisper
from funasr.utils.load_utils import load_audio_text_image_video, extract_fbank
from funasr.models.whisper.utils.decoding import detect_language as detect_language_function, decode as decode_function
from funasr.register import tables
@dataclass
class ModelDimensions:
n_mels: int
n_audio_ctx: int
n_audio_state: int
n_audio_head: int
n_audio_layer: int
n_vocab: int
n_text_ctx: int
n_text_state: int
n_text_head: int
n_text_layer: int
class LayerNorm(nn.LayerNorm):
def forward(self, x: Tensor) -> Tensor:
return super().forward(x.float()).type(x.dtype)
class Linear(nn.Linear):
def forward(self, x: Tensor) -> Tensor:
return F.linear(
x, self.weight.to(x.dtype), None if self.bias is None else self.bias.to(x.dtype)
)
class Conv1d(nn.Conv1d):
def _conv_forward(self, x: Tensor, weight: Tensor, bias: Optional[Tensor]) -> Tensor:
return super()._conv_forward(
x, weight.to(x.dtype), None if bias is None else bias.to(x.dtype)
)
def sinusoids(length, channels, max_timescale=10000):
"""Returns sinusoids for positional embedding"""
assert channels % 2 == 0
log_timescale_increment = np.log(max_timescale) / (channels // 2 - 1)
inv_timescales = torch.exp(-log_timescale_increment * torch.arange(channels // 2))
scaled_time = torch.arange(length)[:, np.newaxis] * inv_timescales[np.newaxis, :]
return torch.cat([torch.sin(scaled_time), torch.cos(scaled_time)], dim=1)
class MultiHeadAttention(nn.Module):
def __init__(self, n_state: int, n_head: int):
@tables.register("model_classes", "WhisperWarp")
class WhisperWarp(nn.Module):
def __init__(self, whisper_dims: dict, **kwargs):
super().__init__()
self.n_head = n_head
self.query = Linear(n_state, n_state)
self.key = Linear(n_state, n_state, bias=False)
self.value = Linear(n_state, n_state)
self.out = Linear(n_state, n_state)
def forward(
self,
x: Tensor,
xa: Optional[Tensor] = None,
mask: Optional[Tensor] = None,
kv_cache: Optional[dict] = None,
):
q = self.query(x)
if kv_cache is None or xa is None or self.key not in kv_cache:
# hooks, if installed (i.e. kv_cache is not None), will prepend the cached kv tensors;
# otherwise, perform key/value projections for self- or cross-attention as usual.
k = self.key(x if xa is None else xa)
v = self.value(x if xa is None else xa)
hub = kwargs.get("hub", "funasr")
if hub == "openai":
init_param_path = kwargs.get("init_param_path", "large-v3")
model = whisper.load_model(init_param_path)
else:
# for cross-attention, calculate keys and values once and reuse in subsequent calls.
k = kv_cache[self.key]
v = kv_cache[self.value]
dims = whisper.model.ModelDimensions(**whisper_dims)
model = whisper.model.Whisper(dims=dims)
self.model = model
def forward(self, ):
pass
def inference(self,
data_in,
data_lengths=None,
key: list = None,
tokenizer=None,
frontend=None,
**kwargs,
):
if kwargs.get("batch_size", 1) > 1:
raise NotImplementedError("batch decoding is not implemented")
wv, qk = self.qkv_attention(q, k, v, mask)
return self.out(wv), qk
meta_data = {}
if isinstance(data_in, torch.Tensor) and kwargs.get("data_type", "sound") == "fbank": # fbank
speech, speech_lengths = data_in, data_lengths
if len(speech.shape) < 3:
speech = speech[None, :, :]
if speech_lengths is None:
speech_lengths = speech.shape[1]
else:
# extract fbank feats
time1 = time.perf_counter()
audio_sample_list = load_audio_text_image_video(data_in, fs=frontend.fs, audio_fs=kwargs.get("fs", 16000),
data_type=kwargs.get("data_type", "sound"),
tokenizer=tokenizer)
time2 = time.perf_counter()
meta_data["load_data"] = f"{time2 - time1:0.3f}"
speech, speech_lengths = extract_fbank(audio_sample_list, data_type=kwargs.get("data_type", "sound"),
frontend=frontend)
time3 = time.perf_counter()
meta_data["extract_feat"] = f"{time3 - time2:0.3f}"
frame_shift = frontend.frame_shift if hasattr(frontend, "frame_shift") else 10
lfr_n = frontend.lfr_n if hasattr(frontend, "lfr_n") else 1
meta_data["batch_data_time"] = speech_lengths.sum().item() * frame_shift * lfr_n / 1000
def qkv_attention(self, q: Tensor, k: Tensor, v: Tensor, mask: Optional[Tensor] = None):
n_batch, n_ctx, n_state = q.shape
scale = (n_state // self.n_head) ** -0.25
q = q.view(*q.shape[:2], self.n_head, -1).permute(0, 2, 1, 3) * scale
k = k.view(*k.shape[:2], self.n_head, -1).permute(0, 2, 3, 1) * scale
v = v.view(*v.shape[:2], self.n_head, -1).permute(0, 2, 1, 3)
speech = speech.to(device=kwargs["device"])[0, :, :]
speech_lengths = speech_lengths.to(device=kwargs["device"])
qk = q @ k
if mask is not None:
qk = qk + mask[:n_ctx, :n_ctx]
qk = qk.float()
# detect the spoken language
_, probs = self.model.detect_language(speech)
print(f"Detected language: {max(probs, key=probs.get)}")
w = F.softmax(qk, dim=-1).to(q.dtype)
return (w @ v).permute(0, 2, 1, 3).flatten(start_dim=2), qk.detach()
# decode the audio
options = whisper.DecodingOptions(language=kwargs.get("language", None), fp16=False)
result = whisper.decode(self.model, speech, options)
results = []
result_i = {"key": key[0], "text": result.text}
class ResidualAttentionBlock(nn.Module):
def __init__(self, n_state: int, n_head: int, cross_attention: bool = False):
super().__init__()
self.attn = MultiHeadAttention(n_state, n_head)
self.attn_ln = LayerNorm(n_state)
self.cross_attn = MultiHeadAttention(n_state, n_head) if cross_attention else None
self.cross_attn_ln = LayerNorm(n_state) if cross_attention else None
n_mlp = n_state * 4
self.mlp = nn.Sequential(Linear(n_state, n_mlp), nn.GELU(), Linear(n_mlp, n_state))
self.mlp_ln = LayerNorm(n_state)
def forward(
self,
x: Tensor,
xa: Optional[Tensor] = None,
mask: Optional[Tensor] = None,
kv_cache: Optional[dict] = None,
):
x = x + self.attn(self.attn_ln(x), mask=mask, kv_cache=kv_cache)[0]
if self.cross_attn:
x = x + self.cross_attn(self.cross_attn_ln(x), xa, kv_cache=kv_cache)[0]
x = x + self.mlp(self.mlp_ln(x))
return x
@tables.register("encoder_classes", "WhisperEncoder")
class AudioEncoder(nn.Module):
def __init__(self, n_mels: int, n_ctx: int, n_state: int, n_head: int, n_layer: int):
super().__init__()
self.conv1 = Conv1d(n_mels, n_state, kernel_size=3, padding=1)
self.conv2 = Conv1d(n_state, n_state, kernel_size=3, stride=2, padding=1)
self.register_buffer("positional_embedding", sinusoids(n_ctx, n_state))
self.blocks: Iterable[ResidualAttentionBlock] = nn.ModuleList(
[ResidualAttentionBlock(n_state, n_head) for _ in range(n_layer)]
)
self.ln_post = LayerNorm(n_state)
def forward(self, x: Tensor):
"""
x : torch.Tensor, shape = (batch_size, n_mels, n_ctx)
the mel spectrogram of the audio
"""
x = F.gelu(self.conv1(x))
x = F.gelu(self.conv2(x))
x = x.permute(0, 2, 1)
assert x.shape[1:] == self.positional_embedding.shape, "incorrect audio shape"
x = (x + self.positional_embedding).to(x.dtype)
for block in self.blocks:
x = block(x)
x = self.ln_post(x)
return x
@tables.register("decoder_classes", "WhisperDecoder")
class TextDecoder(nn.Module):
def __init__(self, n_vocab: int, n_ctx: int, n_state: int, n_head: int, n_layer: int):
super().__init__()
self.token_embedding = nn.Embedding(n_vocab, n_state)
self.positional_embedding = nn.Parameter(torch.empty(n_ctx, n_state))
self.blocks: Iterable[ResidualAttentionBlock] = nn.ModuleList(
[ResidualAttentionBlock(n_state, n_head, cross_attention=True) for _ in range(n_layer)]
)
self.ln = LayerNorm(n_state)
mask = torch.empty(n_ctx, n_ctx).fill_(-np.inf).triu_(1)
self.register_buffer("mask", mask, persistent=False)
def forward(self, x: Tensor, xa: Tensor, kv_cache: Optional[dict] = None):
"""
x : torch.LongTensor, shape = (batch_size, <= n_ctx)
the text tokens
xa : torch.Tensor, shape = (batch_size, n_mels, n_audio_ctx)
the encoded audio features to be attended on
"""
offset = next(iter(kv_cache.values())).shape[1] if kv_cache else 0
x = self.token_embedding(x) + self.positional_embedding[offset : offset + x.shape[-1]]
x = x.to(xa.dtype)
for block in self.blocks:
x = block(x, xa, mask=self.mask, kv_cache=kv_cache)
x = self.ln(x)
logits = (x @ torch.transpose(self.token_embedding.weight.to(x.dtype), 0, 1)).float()
return logits
@tables.register("model_classes", "Whisper")
class Whisper(nn.Module):
def __init__(self, dims: dict):
super().__init__()
dims = ModelDimensions(**dims)
self.dims = dims
self.sos = 1
self.eos = 1
self.encoder = AudioEncoder(
self.dims.n_mels,
self.dims.n_audio_ctx,
self.dims.n_audio_state,
self.dims.n_audio_head,
self.dims.n_audio_layer,
)
self.decoder = TextDecoder(
self.dims.n_vocab,
self.dims.n_text_ctx,
self.dims.n_text_state,
self.dims.n_text_head,
self.dims.n_text_layer,
)
def embed_audio(self, mel: torch.Tensor):
return self.encoder(mel)
def logits(self, tokens: torch.Tensor, audio_features: torch.Tensor):
return self.decoder(tokens, audio_features)
def forward(self, mel: torch.Tensor, tokens: torch.Tensor) -> Dict[str, torch.Tensor]:
return self.decoder(tokens, self.encoder(mel))
@property
def device(self):
return next(self.parameters()).device
@property
def is_multilingual(self):
return self.dims.n_vocab == 51865
def install_kv_cache_hooks(self, cache: Optional[dict] = None):
"""
The `MultiHeadAttention` module optionally accepts `kv_cache` which stores the key and value
tensors calculated for the previous positions. This method returns a dictionary that stores
all caches, and the necessary hooks for the key and value projection modules that save the
intermediate tensors to be reused during later calculations.
Returns
-------
cache : Dict[nn.Module, torch.Tensor]
A dictionary object mapping the key/value projection modules to its cache
hooks : List[RemovableHandle]
List of PyTorch RemovableHandle objects to stop the hooks to be called
"""
cache = {**cache} if cache is not None else {}
hooks = []
def save_to_cache(module, _, output):
if module not in cache or output.shape[1] > self.decoder.positional_embedding.shape[0]:
cache[module] = output # save as-is, for the first token or cross attention
else:
cache[module] = torch.cat([cache[module], output], dim=1).detach()
return cache[module]
def install_hooks(layer: nn.Module):
if isinstance(layer, MultiHeadAttention):
hooks.append(layer.key.register_forward_hook(save_to_cache))
hooks.append(layer.value.register_forward_hook(save_to_cache))
self.decoder.apply(install_hooks)
return cache, hooks
detect_language = detect_language_function
decode = decode_function
results.append(result_i)
return results, meta_data
+46
View File
@@ -0,0 +1,46 @@
# This is an example that demonstrates how to configure a model file.
# You can modify the configuration according to your own requirements.
# to print the register_table:
# from funasr.register import tables
# tables.print()
# network architecture
model: WhisperWarp
model_conf:
lsm_weight: 0.1
length_normalized_loss: true
hub: funasr # openai
init_param_path: null # large-v2 or large-v3 if hub == "openai"
# only use for hub == funasr,
# if hub == openai, whisper_dims is automaticall download
whisper_dims:
'n_mels': 80
'n_vocab': 51865
'n_audio_ctx': 1500
'n_audio_state': 1280
'n_audio_head': 20
'n_audio_layer': 32
'n_text_ctx': 448
'n_text_state': 1280
'n_text_head': 20
'n_text_layer': 32
# frontend related
frontend: WhisperFrontend
frontend_conf:
fs: 16000
n_mels: 80
do_pad_trim: true
tokenizer: WhisperTokenizer
tokenizer_conf:
language: null
task: transcribe
is_multilingual: true
num_languages: 99
scope_map: ['none', "model."]
File diff suppressed because it is too large Load Diff
@@ -1 +0,0 @@
{"bos_token": "<|endoftext|>", "eos_token": "<|endoftext|>", "unk_token": "<|endoftext|>"}
@@ -1 +0,0 @@
{"unk_token": "<|endoftext|>", "bos_token": "<|endoftext|>", "eos_token": "<|endoftext|>", "add_prefix_space": false, "model_max_length": 1024, "special_tokens_map_file": null, "name_or_path": "gpt2", "tokenizer_class": "GPT2Tokenizer"}
File diff suppressed because one or more lines are too long
Binary file not shown.
@@ -1 +0,0 @@
{"<|endoftext|>": 50257}
File diff suppressed because it is too large Load Diff
@@ -1 +0,0 @@
{"bos_token": "<|endoftext|>", "eos_token": "<|endoftext|>", "unk_token": "<|endoftext|>"}
@@ -1 +0,0 @@
{"unk_token": {"content": "<|endoftext|>", "single_word": false, "lstrip": false, "rstrip": false, "normalized": true, "__type": "AddedToken"}, "bos_token": {"content": "<|endoftext|>", "single_word": false, "lstrip": false, "rstrip": false, "normalized": true, "__type": "AddedToken"}, "eos_token": {"content": "<|endoftext|>", "single_word": false, "lstrip": false, "rstrip": false, "normalized": true, "__type": "AddedToken"}, "add_prefix_space": false, "model_max_length": 1024, "special_tokens_map_file": null, "name_or_path": "multilingual", "errors": "replace", "tokenizer_class": "GPT2Tokenizer"}
File diff suppressed because one or more lines are too long
-127
View File
@@ -1,127 +0,0 @@
import os
from functools import lru_cache
from typing import Union
try:
import ffmpeg
except:
print("Please Requires the ffmpeg CLI and `ffmpeg-python` package to be installed.")
import numpy as np
import torch
import torch.nn.functional as F
from funasr.models.whisper.utils.utils import exact_div
# hard-coded audio hyperparameters
SAMPLE_RATE = 16000
N_FFT = 400
N_MELS = 80
HOP_LENGTH = 160
CHUNK_LENGTH = 30
N_SAMPLES = CHUNK_LENGTH * SAMPLE_RATE # 480000: number of samples in a chunk
N_FRAMES = exact_div(N_SAMPLES, HOP_LENGTH) # 3000: number of frames in a mel spectrogram input
def load_audio(file: str, sr: int = SAMPLE_RATE):
"""
Open an audio file and read as mono waveform, resampling as necessary
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.
"""
try:
# This launches a subprocess to decode audio while down-mixing and resampling as necessary.
# Requires the ffmpeg CLI and `ffmpeg-python` package to be installed.
out, _ = (
ffmpeg.input(file, threads=0)
.output("-", format="s16le", acodec="pcm_s16le", ac=1, ar=sr)
.run(cmd=["ffmpeg", "-nostdin"], capture_stdout=True, capture_stderr=True)
)
except ffmpeg.Error as e:
raise RuntimeError(f"Failed to load audio: {e.stderr.decode()}") from e
return np.frombuffer(out, np.int16).flatten().astype(np.float32) / 32768.0
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 = N_MELS) -> 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 == 80, f"Unsupported n_mels: {n_mels}"
with np.load(os.path.join(os.path.dirname(__file__), "assets", "mel_filters.npz")) 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 = N_MELS):
"""
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 is supported
Returns
-------
torch.Tensor, shape = (80, n_frames)
A Tensor that contains the Mel spectrogram
"""
if not torch.is_tensor(audio):
if isinstance(audio, str):
audio = load_audio(audio)
audio = torch.from_numpy(audio)
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_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
return log_spec
-710
View File
@@ -1,710 +0,0 @@
from dataclasses import dataclass, field
from typing import Dict, List, Tuple, Iterable, Optional, Sequence, Union, TYPE_CHECKING
import numpy as np
import torch
import torch.nn.functional as F
from torch import Tensor
from torch.distributions import Categorical
from funasr.models.whisper.utils.audio import CHUNK_LENGTH
from funasr.models.whisper.utils.tokenizer import Tokenizer, get_tokenizer
from funasr.models.whisper.utils.utils import compression_ratio
if TYPE_CHECKING:
from funasr.models.whisper_models.model import Whisper
@torch.no_grad()
def detect_language(model: "Whisper", mel: Tensor, tokenizer: Tokenizer = None) -> Tuple[Tensor, List[dict]]:
"""
Detect the spoken language in the audio, and return them as list of strings, along with the ids
of the most probable language tokens and the probability distribution over all language tokens.
This is performed outside the main decode loop in order to not interfere with kv-caching.
Returns
-------
language_tokens : Tensor, shape = (n_audio,)
ids of the most probable language tokens, which appears after the startoftranscript token.
language_probs : List[Dict[str, float]], length = n_audio
list of dictionaries containing the probability distribution over all languages.
"""
if tokenizer is None:
tokenizer = get_tokenizer(model.is_multilingual)
if tokenizer.language is None or tokenizer.language_token not in tokenizer.sot_sequence:
raise ValueError(f"This model doesn't have language tokens so it can't perform lang id")
single = mel.ndim == 2
if single:
mel = mel.unsqueeze(0)
# skip encoder forward pass if already-encoded audio features were given
if mel.shape[-2:] != (model.dims.n_audio_ctx, model.dims.n_audio_state):
mel = model.encoder(mel)
# forward pass using a single token, startoftranscript
n_audio = mel.shape[0]
x = torch.tensor([[tokenizer.sot]] * n_audio).to(mel.device) # [n_audio, 1]
logits = model.logits(x, mel)[:, 0]
# collect detected languages; suppress all non-language tokens
mask = torch.ones(logits.shape[-1], dtype=torch.bool)
mask[list(tokenizer.all_language_tokens)] = False
logits[:, mask] = -np.inf
language_tokens = logits.argmax(dim=-1)
language_token_probs = logits.softmax(dim=-1).cpu()
language_probs = [
{
c: language_token_probs[i, j].item()
for j, c in zip(tokenizer.all_language_tokens, tokenizer.all_language_codes)
}
for i in range(n_audio)
]
if single:
language_tokens = language_tokens[0]
language_probs = language_probs[0]
return language_tokens, language_probs
@dataclass(frozen=True)
class DecodingOptions:
task: str = "transcribe" # whether to perform X->X "transcribe" or X->English "translate"
language: Optional[str] = None # language that the audio is in; uses detected language if None
# sampling-related options
temperature: float = 0.0
sample_len: Optional[int] = None # maximum number of tokens to sample
best_of: Optional[int] = None # number of independent samples to collect, when t > 0
beam_size: Optional[int] = None # number of beams in beam search, when t == 0
patience: Optional[float] = None # patience in beam search (https://arxiv.org/abs/2204.05424)
# options for ranking generations (either beams or best-of-N samples)
length_penalty: Optional[float] = None # "alpha" in Google NMT, None defaults to length norm
# prompt, prefix, and token suppression
prompt: Optional[Union[str, List[int]]] = None # text or tokens for the previous context
prefix: Optional[Union[str, List[int]]] = None # text or tokens to prefix the current context
suppress_blank: bool = True # this will suppress blank outputs
# list of tokens ids (or comma-separated token ids) to suppress
# "-1" will suppress a set of symbols as defined in `tokenizer.non_speech_tokens()`
suppress_tokens: Optional[Union[str, Iterable[int]]] = "-1"
# timestamp sampling options
without_timestamps: bool = False # use <|notimestamps|> to sample text tokens only
max_initial_timestamp: Optional[float] = 1.0 # the initial timestamp cannot be later than this
# implementation details
fp16: bool = True # use fp16 for most of the calculation
@dataclass(frozen=True)
class DecodingResult:
audio_features: Tensor
language: str
language_probs: Optional[Dict[str, float]] = None
tokens: List[int] = field(default_factory=list)
text: str = ""
avg_logprob: float = np.nan
no_speech_prob: float = np.nan
temperature: float = np.nan
compression_ratio: float = np.nan
class Inference:
def logits(self, tokens: Tensor, audio_features: Tensor) -> Tensor:
"""Perform a forward pass on the decoder and return per-token logits"""
raise NotImplementedError
def rearrange_kv_cache(self, source_indices) -> None:
"""Update the key-value cache according to the updated beams"""
raise NotImplementedError
def cleanup_caching(self) -> None:
"""Clean up any resources or hooks after decoding is finished"""
pass
class PyTorchInference(Inference):
def __init__(self, model: "Whisper", initial_token_length: int):
self.model: "Whisper" = model
self.initial_token_length = initial_token_length
self.kv_cache = {}
self.hooks = []
def logits(self, tokens: Tensor, audio_features: Tensor) -> Tensor:
if not self.kv_cache:
self.kv_cache, self.hooks = self.model.install_kv_cache_hooks()
if tokens.shape[-1] > self.initial_token_length:
# only need to use the last token except in the first forward pass
tokens = tokens[:, -1:]
return self.model.decoder(tokens, audio_features, kv_cache=self.kv_cache)
def cleanup_caching(self):
for hook in self.hooks:
hook.remove()
self.kv_cache = {}
self.hooks = []
def rearrange_kv_cache(self, source_indices):
for module, tensor in self.kv_cache.items():
# update the key/value cache to contain the selected sequences
self.kv_cache[module] = tensor[source_indices].detach()
class SequenceRanker:
def rank(self, tokens: List[List[Tensor]], sum_logprobs: List[List[float]]) -> List[int]:
"""
Given a list of groups of samples and their cumulative log probabilities,
return the indices of the samples in each group to select as the final result
"""
raise NotImplementedError
class MaximumLikelihoodRanker(SequenceRanker):
"""
Select the sample with the highest log probabilities, penalized using either
a simple length normalization or Google NMT paper's length penalty
"""
def __init__(self, length_penalty: Optional[float]):
self.length_penalty = length_penalty
def rank(self, tokens: List[List[Tensor]], sum_logprobs: List[List[float]]):
def scores(logprobs, lengths):
result = []
for logprob, length in zip(logprobs, lengths):
if self.length_penalty is None:
penalty = length
else:
# from the Google NMT paper
penalty = ((5 + length) / 6) ** self.length_penalty
result.append(logprob / penalty)
return result
# get the sequence with the highest score
lengths = [[len(t) for t in s] for s in tokens]
return [np.argmax(scores(p, l)) for p, l in zip(sum_logprobs, lengths)]
class TokenDecoder:
def reset(self):
"""Initialize any stateful variables for decoding a new sequence"""
def update(self, tokens: Tensor, logits: Tensor, sum_logprobs: Tensor) -> Tuple[Tensor, bool]:
"""Specify how to select the next token, based on the current trace and logits
Parameters
----------
tokens : Tensor, shape = (n_batch, current_sequence_length)
all tokens in the context so far, including the prefix and sot_sequence tokens
logits : Tensor, shape = (n_batch, vocab_size)
per-token logits of the probability distribution at the current step
sum_logprobs : Tensor, shape = (n_batch)
cumulative log probabilities for each sequence
Returns
-------
tokens : Tensor, shape = (n_batch, current_sequence_length + 1)
the tokens, appended with the selected next token
completed : bool
True if all sequences has reached the end of text
"""
raise NotImplementedError
def finalize(
self, tokens: Tensor, sum_logprobs: Tensor
) -> Tuple[Sequence[Sequence[Tensor]], List[List[float]]]:
"""Finalize search and return the final candidate sequences
Parameters
----------
tokens : Tensor, shape = (n_audio, n_group, current_sequence_length)
all tokens in the context so far, including the prefix and sot_sequence
sum_logprobs : Tensor, shape = (n_audio, n_group)
cumulative log probabilities for each sequence
Returns
-------
tokens : Sequence[Sequence[Tensor]], length = n_audio
sequence of Tensors containing candidate token sequences, for each audio input
sum_logprobs : List[List[float]], length = n_audio
sequence of cumulative log probabilities corresponding to the above
"""
raise NotImplementedError
class GreedyDecoder(TokenDecoder):
def __init__(self, temperature: float, eot: int):
self.temperature = temperature
self.eot = eot
def update(self, tokens: Tensor, logits: Tensor, sum_logprobs: Tensor) -> Tuple[Tensor, bool]:
temperature = self.temperature
if temperature == 0:
next_tokens = logits.argmax(dim=-1)
else:
next_tokens = Categorical(logits=logits / temperature).sample()
logprobs = F.log_softmax(logits.float(), dim=-1)
current_logprobs = logprobs[torch.arange(logprobs.shape[0]), next_tokens]
sum_logprobs += current_logprobs * (tokens[:, -1] != self.eot)
next_tokens[tokens[:, -1] == self.eot] = self.eot
tokens = torch.cat([tokens, next_tokens[:, None]], dim=-1)
completed = (tokens[:, -1] == self.eot).all()
return tokens, completed
def finalize(self, tokens: Tensor, sum_logprobs: Tensor):
# make sure each sequence has at least one EOT token at the end
tokens = F.pad(tokens, (0, 1), value=self.eot)
return tokens, sum_logprobs.tolist()
class BeamSearchDecoder(TokenDecoder):
def __init__(self, beam_size: int, eot: int, inference: Inference, patience: Optional[float] = None):
self.beam_size = beam_size
self.eot = eot
self.inference = inference
self.patience = patience or 1.0
self.max_candidates: int = round(beam_size * self.patience)
self.finished_sequences = None
assert self.max_candidates > 0, f"Invalid beam size ({beam_size}) or patience ({patience})"
def reset(self):
self.finished_sequences = None
def update(self, tokens: Tensor, logits: Tensor, sum_logprobs: Tensor) -> Tuple[Tensor, bool]:
if tokens.shape[0] % self.beam_size != 0:
raise ValueError(f"{tokens.shape}[0] % {self.beam_size} != 0")
n_audio = tokens.shape[0] // self.beam_size
if self.finished_sequences is None: # for the first update
self.finished_sequences = [{} for _ in range(n_audio)]
logprobs = F.log_softmax(logits.float(), dim=-1)
next_tokens, source_indices, finished_sequences = [], [], []
for i in range(n_audio):
scores, sources, finished = {}, {}, {}
# STEP 1: calculate the cumulative log probabilities for possible candidates
for j in range(self.beam_size):
idx = i * self.beam_size + j
prefix = tokens[idx].tolist()
for logprob, token in zip(*logprobs[idx].topk(self.beam_size + 1)):
new_logprob = (sum_logprobs[idx] + logprob).item()
sequence = tuple(prefix + [token.item()])
scores[sequence] = new_logprob
sources[sequence] = idx
# STEP 2: rank the candidates and keep the top beam_size sequences for each audio
saved = 0
for sequence in sorted(scores, key=scores.get, reverse=True):
if sequence[-1] == self.eot:
finished[sequence] = scores[sequence]
else:
sum_logprobs[len(next_tokens)] = scores[sequence]
next_tokens.append(sequence)
source_indices.append(sources[sequence])
saved += 1
if saved == self.beam_size:
break
finished_sequences.append(finished)
tokens = torch.tensor(next_tokens, device=tokens.device)
self.inference.rearrange_kv_cache(source_indices)
# add newly finished sequences to self.finished_sequences
assert len(self.finished_sequences) == len(finished_sequences)
for previously_finished, newly_finished in zip(self.finished_sequences, finished_sequences):
for seq in sorted(newly_finished, key=newly_finished.get, reverse=True):
if len(previously_finished) >= self.max_candidates:
break # the candidate list is full
previously_finished[seq] = newly_finished[seq]
# mark as completed if all audio has enough number of samples
completed = all(
len(sequences) >= self.max_candidates for sequences in self.finished_sequences
)
return tokens, completed
def finalize(self, preceding_tokens: Tensor, sum_logprobs: Tensor):
# collect all finished sequences, including patience, and add unfinished ones if not enough
sum_logprobs = sum_logprobs.cpu()
for i, sequences in enumerate(self.finished_sequences):
if len(sequences) < self.beam_size: # when not enough sequences are finished
for j in list(np.argsort(sum_logprobs[i]))[::-1]:
sequence = preceding_tokens[i, j].tolist() + [self.eot]
sequences[tuple(sequence)] = sum_logprobs[i][j].item()
if len(sequences) >= self.beam_size:
break
tokens: List[List[Tensor]] = [
[torch.tensor(seq) for seq in sequences.keys()] for sequences in self.finished_sequences
]
sum_logprobs: List[List[float]] = [
list(sequences.values()) for sequences in self.finished_sequences
]
return tokens, sum_logprobs
class LogitFilter:
def apply(self, logits: Tensor, tokens: Tensor) -> None:
"""Apply any filtering or masking to logits in-place
Parameters
----------
logits : Tensor, shape = (n_batch, vocab_size)
per-token logits of the probability distribution at the current step
tokens : Tensor, shape = (n_batch, current_sequence_length)
all tokens in the context so far, including the prefix and sot_sequence tokens
"""
raise NotImplementedError
class SuppressBlank(LogitFilter):
def __init__(self, tokenizer: Tokenizer, sample_begin: int):
self.tokenizer = tokenizer
self.sample_begin = sample_begin
def apply(self, logits: Tensor, tokens: Tensor):
if tokens.shape[1] == self.sample_begin:
logits[:, self.tokenizer.encode(" ") + [self.tokenizer.eot]] = -np.inf
class SuppressTokens(LogitFilter):
def __init__(self, suppress_tokens: Sequence[int]):
self.suppress_tokens = list(suppress_tokens)
def apply(self, logits: Tensor, tokens: Tensor):
logits[:, self.suppress_tokens] = -np.inf
class ApplyTimestampRules(LogitFilter):
def __init__(
self, tokenizer: Tokenizer, sample_begin: int, max_initial_timestamp_index: Optional[int]
):
self.tokenizer = tokenizer
self.sample_begin = sample_begin
self.max_initial_timestamp_index = max_initial_timestamp_index
def apply(self, logits: Tensor, tokens: Tensor):
# suppress <|notimestamps|> which is handled by without_timestamps
if self.tokenizer.no_timestamps is not None:
logits[:, self.tokenizer.no_timestamps] = -np.inf
# timestamps have to appear in pairs, except directly before EOT; mask logits accordingly
for k in range(tokens.shape[0]):
seq = [t for t in tokens[k, self.sample_begin :].tolist()]
last_was_timestamp = len(seq) >= 1 and seq[-1] >= self.tokenizer.timestamp_begin
penultimate_was_timestamp = len(seq) < 2 or seq[-2] >= self.tokenizer.timestamp_begin
if last_was_timestamp:
if penultimate_was_timestamp: # has to be non-timestamp
logits[k, self.tokenizer.timestamp_begin :] = -np.inf
else: # cannot be normal text tokens
logits[k, : self.tokenizer.eot] = -np.inf
if tokens.shape[1] == self.sample_begin:
# suppress generating non-timestamp tokens at the beginning
logits[:, : self.tokenizer.timestamp_begin] = -np.inf
# apply the `max_initial_timestamp` option
if self.max_initial_timestamp_index is not None:
last_allowed = self.tokenizer.timestamp_begin + self.max_initial_timestamp_index
logits[:, last_allowed + 1 :] = -np.inf
# if sum of probability over timestamps is above any other token, sample timestamp
logprobs = F.log_softmax(logits.float(), dim=-1)
for k in range(tokens.shape[0]):
timestamp_logprob = logprobs[k, self.tokenizer.timestamp_begin :].logsumexp(dim=-1)
max_text_token_logprob = logprobs[k, : self.tokenizer.timestamp_begin].max()
if timestamp_logprob > max_text_token_logprob:
logits[k, : self.tokenizer.timestamp_begin] = -np.inf
class DecodingTask:
inference: Inference
sequence_ranker: SequenceRanker
decoder: TokenDecoder
logit_filters: List[LogitFilter]
def __init__(self, model: "Whisper", options: DecodingOptions):
self.model = model
language = options.language or "en"
tokenizer = get_tokenizer(model.is_multilingual, language=language, task=options.task)
self.tokenizer: Tokenizer = tokenizer
self.options: DecodingOptions = self._verify_options(options)
self.n_group: int = options.beam_size or options.best_of or 1
self.n_ctx: int = model.dims.n_text_ctx
self.sample_len: int = options.sample_len or model.dims.n_text_ctx // 2
self.sot_sequence: Tuple[int] = tokenizer.sot_sequence
if self.options.without_timestamps:
self.sot_sequence = tokenizer.sot_sequence_including_notimestamps
self.initial_tokens: Tuple[int] = self._get_initial_tokens()
self.sample_begin: int = len(self.initial_tokens)
self.sot_index: int = self.initial_tokens.index(tokenizer.sot)
# inference: implements the forward pass through the decoder, including kv caching
self.inference = PyTorchInference(model, len(self.initial_tokens))
# sequence ranker: implements how to rank a group of sampled sequences
self.sequence_ranker = MaximumLikelihoodRanker(options.length_penalty)
# decoder: implements how to select the next tokens, given the autoregressive distribution
if options.beam_size is not None:
self.decoder = BeamSearchDecoder(
options.beam_size, tokenizer.eot, self.inference, options.patience
)
else:
self.decoder = GreedyDecoder(options.temperature, tokenizer.eot)
# logit filters: applies various rules to suppress or penalize certain tokens
self.logit_filters = []
if self.options.suppress_blank:
self.logit_filters.append(SuppressBlank(self.tokenizer, self.sample_begin))
if self.options.suppress_tokens:
self.logit_filters.append(SuppressTokens(self._get_suppress_tokens()))
if not options.without_timestamps:
precision = CHUNK_LENGTH / model.dims.n_audio_ctx # usually 0.02 seconds
max_initial_timestamp_index = None
if options.max_initial_timestamp:
max_initial_timestamp_index = round(self.options.max_initial_timestamp / precision)
self.logit_filters.append(
ApplyTimestampRules(tokenizer, self.sample_begin, max_initial_timestamp_index)
)
def _verify_options(self, options: DecodingOptions) -> DecodingOptions:
if options.beam_size is not None and options.best_of is not None:
raise ValueError("beam_size and best_of can't be given together")
if options.temperature == 0:
if options.best_of is not None:
raise ValueError("best_of with greedy sampling (T=0) is not compatible")
if options.patience is not None and options.beam_size is None:
raise ValueError("patience requires beam_size to be given")
if options.length_penalty is not None and not (0 <= options.length_penalty <= 1):
raise ValueError("length_penalty (alpha) should be a value between 0 and 1")
return options
def _get_initial_tokens(self) -> Tuple[int]:
tokens = list(self.sot_sequence)
prefix = self.options.prefix
prompt = self.options.prompt
if prefix:
prefix_tokens = (
self.tokenizer.encode(" " + prefix.strip()) if isinstance(prefix, str) else prefix
)
if self.sample_len is not None:
max_prefix_len = self.n_ctx // 2 - self.sample_len
prefix_tokens = prefix_tokens[-max_prefix_len:]
tokens = tokens + prefix_tokens
if prompt:
prompt_tokens = (
self.tokenizer.encode(" " + prompt.strip()) if isinstance(prompt, str) else prompt
)
tokens = [self.tokenizer.sot_prev] + prompt_tokens[-(self.n_ctx // 2 - 1) :] + tokens
return tuple(tokens)
def _get_suppress_tokens(self) -> Tuple[int]:
suppress_tokens = self.options.suppress_tokens
if isinstance(suppress_tokens, str):
suppress_tokens = [int(t) for t in suppress_tokens.split(",")]
if -1 in suppress_tokens:
suppress_tokens = [t for t in suppress_tokens if t >= 0]
suppress_tokens.extend(self.tokenizer.non_speech_tokens)
elif suppress_tokens is None or len(suppress_tokens) == 0:
suppress_tokens = [] # interpret empty string as an empty list
else:
assert isinstance(suppress_tokens, list), "suppress_tokens must be a list"
suppress_tokens.extend(
[self.tokenizer.sot, self.tokenizer.sot_prev, self.tokenizer.sot_lm]
)
if self.tokenizer.no_speech is not None:
# no-speech probability is collected separately
suppress_tokens.append(self.tokenizer.no_speech)
return tuple(sorted(set(suppress_tokens)))
def _get_audio_features(self, mel: Tensor):
if self.options.fp16:
mel = mel.half()
if mel.shape[-2:] == (self.model.dims.n_audio_ctx, self.model.dims.n_audio_state):
# encoded audio features are given; skip audio encoding
audio_features = mel
else:
audio_features = self.model.encoder(mel)
if audio_features.dtype != (torch.float16 if self.options.fp16 else torch.float32):
return TypeError(f"audio_features has an incorrect dtype: {audio_features.dtype}")
return audio_features
def _detect_language(self, audio_features: Tensor, tokens: Tensor):
languages = [self.options.language] * audio_features.shape[0]
lang_probs = None
if self.options.language is None or self.options.task == "lang_id":
lang_tokens, lang_probs = self.model.detect_language(audio_features, self.tokenizer)
languages = [max(probs, key=probs.get) for probs in lang_probs]
if self.options.language is None:
tokens[:, self.sot_index + 1] = lang_tokens # write language tokens
return languages, lang_probs
def _main_loop(self, audio_features: Tensor, tokens: Tensor):
assert audio_features.shape[0] == tokens.shape[0]
n_batch = tokens.shape[0]
sum_logprobs: Tensor = torch.zeros(n_batch, device=audio_features.device)
no_speech_probs = [np.nan] * n_batch
try:
for i in range(self.sample_len):
logits = self.inference.logits(tokens, audio_features)
if i == 0 and self.tokenizer.no_speech is not None: # save no_speech_probs
probs_at_sot = logits[:, self.sot_index].float().softmax(dim=-1)
no_speech_probs = probs_at_sot[:, self.tokenizer.no_speech].tolist()
# now we need to consider the logits at the last token only
logits = logits[:, -1]
# apply the logit filters, e.g. for suppressing or applying penalty to
for logit_filter in self.logit_filters:
logit_filter.apply(logits, tokens)
# expand the tokens tensor with the selected next tokens
tokens, completed = self.decoder.update(tokens, logits, sum_logprobs)
if completed or tokens.shape[-1] > self.n_ctx:
break
finally:
self.inference.cleanup_caching()
return tokens, sum_logprobs, no_speech_probs
@torch.no_grad()
def run(self, mel: Tensor) -> List[DecodingResult]:
self.decoder.reset()
tokenizer: Tokenizer = self.tokenizer
n_audio: int = mel.shape[0]
audio_features: Tensor = self._get_audio_features(mel) # encoder forward pass
tokens: Tensor = torch.tensor([self.initial_tokens]).repeat(n_audio, 1)
# detect language if requested, overwriting the language token
languages, language_probs = self._detect_language(audio_features, tokens)
if self.options.task == "lang_id":
return [
DecodingResult(audio_features=features, language=language, language_probs=probs)
for features, language, probs in zip(audio_features, languages, language_probs)
]
# repeat the audio & text tensors by the group size, for beam search or best-of-n sampling
audio_features = audio_features.repeat_interleave(self.n_group, dim=0)
tokens = tokens.repeat_interleave(self.n_group, dim=0).to(audio_features.device)
# call the main sampling loop
tokens, sum_logprobs, no_speech_probs = self._main_loop(audio_features, tokens)
# reshape the tensors to have (n_audio, n_group) as the first two dimensions
audio_features = audio_features[:: self.n_group]
no_speech_probs = no_speech_probs[:: self.n_group]
assert audio_features.shape[0] == len(no_speech_probs) == n_audio
tokens = tokens.reshape(n_audio, self.n_group, -1)
sum_logprobs = sum_logprobs.reshape(n_audio, self.n_group)
# get the final candidates for each group, and slice between the first sampled token and EOT
tokens, sum_logprobs = self.decoder.finalize(tokens, sum_logprobs)
tokens: List[List[Tensor]] = [
[t[self.sample_begin : (t == tokenizer.eot).nonzero()[0, 0]] for t in s] for s in tokens
]
# select the top-ranked sample in each group
selected = self.sequence_ranker.rank(tokens, sum_logprobs)
tokens: List[List[int]] = [t[i].tolist() for i, t in zip(selected, tokens)]
texts: List[str] = [tokenizer.decode(t).strip() for t in tokens]
sum_logprobs: List[float] = [lp[i] for i, lp in zip(selected, sum_logprobs)]
avg_logprobs: List[float] = [lp / (len(t) + 1) for t, lp in zip(tokens, sum_logprobs)]
fields = (texts, languages, tokens, audio_features, avg_logprobs, no_speech_probs)
if len(set(map(len, fields))) != 1:
raise RuntimeError(f"inconsistent result lengths: {list(map(len, fields))}")
return [
DecodingResult(
audio_features=features,
language=language,
tokens=tokens,
text=text,
avg_logprob=avg_logprob,
no_speech_prob=no_speech_prob,
temperature=self.options.temperature,
compression_ratio=compression_ratio(text),
)
for text, language, tokens, features, avg_logprob, no_speech_prob in zip(*fields)
]
@torch.no_grad()
def decode(model: "Whisper", mel: Tensor, options: DecodingOptions = DecodingOptions()) -> Union[DecodingResult, List[DecodingResult]]:
"""
Performs decoding of 30-second audio segment(s), provided as Mel spectrogram(s).
Parameters
----------
model: Whisper
the Whisper model instance
mel: torch.Tensor, shape = (80, 3000) or (*, 80, 3000)
A tensor containing the Mel spectrogram(s)
options: DecodingOptions
A dataclass that contains all necessary options for decoding 30-second segments
Returns
-------
result: Union[DecodingResult, List[DecodingResult]]
The result(s) of decoding contained in `DecodingResult` dataclass instance(s)
"""
single = mel.ndim == 2
if single:
mel = mel.unsqueeze(0)
result = DecodingTask(model, options).run(mel)
if single:
result = result[0]
return result
-339
View File
@@ -1,339 +0,0 @@
import os
from dataclasses import dataclass
from functools import lru_cache
from typing import List, Optional, Tuple, Union
import numpy as np
import torch
try:
from transformers import GPT2TokenizerFast
except ImportError:
raise ImportError(
"transformers was not installed. Please install transformers first."
)
LANGUAGES = {
"en": "english",
"zh": "chinese",
"de": "german",
"es": "spanish",
"ru": "russian",
"ko": "korean",
"fr": "french",
"ja": "japanese",
"pt": "portuguese",
"tr": "turkish",
"pl": "polish",
"ca": "catalan",
"nl": "dutch",
"ar": "arabic",
"sv": "swedish",
"it": "italian",
"id": "indonesian",
"hi": "hindi",
"fi": "finnish",
"vi": "vietnamese",
"he": "hebrew",
"uk": "ukrainian",
"el": "greek",
"ms": "malay",
"cs": "czech",
"ro": "romanian",
"da": "danish",
"hu": "hungarian",
"ta": "tamil",
"no": "norwegian",
"th": "thai",
"ur": "urdu",
"hr": "croatian",
"bg": "bulgarian",
"lt": "lithuanian",
"la": "latin",
"mi": "maori",
"ml": "malayalam",
"cy": "welsh",
"sk": "slovak",
"te": "telugu",
"fa": "persian",
"lv": "latvian",
"bn": "bengali",
"sr": "serbian",
"az": "azerbaijani",
"sl": "slovenian",
"kn": "kannada",
"et": "estonian",
"mk": "macedonian",
"br": "breton",
"eu": "basque",
"is": "icelandic",
"hy": "armenian",
"ne": "nepali",
"mn": "mongolian",
"bs": "bosnian",
"kk": "kazakh",
"sq": "albanian",
"sw": "swahili",
"gl": "galician",
"mr": "marathi",
"pa": "punjabi",
"si": "sinhala",
"km": "khmer",
"sn": "shona",
"yo": "yoruba",
"so": "somali",
"af": "afrikaans",
"oc": "occitan",
"ka": "georgian",
"be": "belarusian",
"tg": "tajik",
"sd": "sindhi",
"gu": "gujarati",
"am": "amharic",
"yi": "yiddish",
"lo": "lao",
"uz": "uzbek",
"fo": "faroese",
"ht": "haitian creole",
"ps": "pashto",
"tk": "turkmen",
"nn": "nynorsk",
"mt": "maltese",
"sa": "sanskrit",
"lb": "luxembourgish",
"my": "myanmar",
"bo": "tibetan",
"tl": "tagalog",
"mg": "malagasy",
"as": "assamese",
"tt": "tatar",
"haw": "hawaiian",
"ln": "lingala",
"ha": "hausa",
"ba": "bashkir",
"jw": "javanese",
"su": "sundanese",
}
# language code lookup by name, with a few language aliases
TO_LANGUAGE_CODE = {
**{language: code for code, language in LANGUAGES.items()},
"burmese": "my",
"valencian": "ca",
"flemish": "nl",
"haitian": "ht",
"letzeburgesch": "lb",
"pushto": "ps",
"panjabi": "pa",
"moldavian": "ro",
"moldovan": "ro",
"sinhalese": "si",
"castilian": "es",
}
@dataclass(frozen=True)
class Tokenizer:
"""A thin wrapper around `GPT2TokenizerFast` providing quick access to special tokens"""
tokenizer: "GPT2TokenizerFast"
language: Optional[str]
sot_sequence: Tuple[int]
def encode(self, text, **kwargs):
return self.tokenizer.encode(text, **kwargs)
def decode(self, token_ids: Union[int, List[int], np.ndarray, torch.Tensor], **kwargs):
return self.tokenizer.decode(token_ids, **kwargs)
def decode_with_timestamps(self, tokens) -> str:
"""
Timestamp tokens are above the special tokens' id range and are ignored by `decode()`.
This method decodes given tokens with timestamps tokens annotated, e.g. "<|1.08|>".
"""
outputs = [[]]
for token in tokens:
if token >= self.timestamp_begin:
timestamp = f"<|{(token - self.timestamp_begin) * 0.02:.2f}|>"
outputs.append(timestamp)
outputs.append([])
else:
outputs[-1].append(token)
outputs = [s if isinstance(s, str) else self.tokenizer.decode(s) for s in outputs]
return "".join(outputs)
@property
@lru_cache()
def eot(self) -> int:
return self.tokenizer.eos_token_id
@property
@lru_cache()
def sot(self) -> int:
return self._get_single_token_id("<|startoftranscript|>")
@property
@lru_cache()
def sot_lm(self) -> int:
return self._get_single_token_id("<|startoflm|>")
@property
@lru_cache()
def sot_prev(self) -> int:
return self._get_single_token_id("<|startofprev|>")
@property
@lru_cache()
def no_speech(self) -> int:
return self._get_single_token_id("<|nospeech|>")
@property
@lru_cache()
def no_timestamps(self) -> int:
return self._get_single_token_id("<|notimestamps|>")
@property
@lru_cache()
def timestamp_begin(self) -> int:
return self.tokenizer.all_special_ids[-1] + 1
@property
@lru_cache()
def language_token(self) -> int:
"""Returns the token id corresponding to the value of the `language` field"""
if self.language is None:
raise ValueError(f"This tokenizer does not have language token configured")
additional_tokens = dict(
zip(
self.tokenizer.additional_special_tokens,
self.tokenizer.additional_special_tokens_ids,
)
)
candidate = f"<|{self.language}|>"
if candidate in additional_tokens:
return additional_tokens[candidate]
raise KeyError(f"Language {self.language} not found in tokenizer.")
@property
@lru_cache()
def all_language_tokens(self) -> Tuple[int]:
result = []
for token, token_id in zip(
self.tokenizer.additional_special_tokens,
self.tokenizer.additional_special_tokens_ids,
):
if token.strip("<|>") in LANGUAGES:
result.append(token_id)
return tuple(result)
@property
@lru_cache()
def all_language_codes(self) -> Tuple[str]:
return tuple(self.decode([l]).strip("<|>") for l in self.all_language_tokens)
@property
@lru_cache()
def sot_sequence_including_notimestamps(self) -> Tuple[int]:
return tuple(list(self.sot_sequence) + [self.no_timestamps])
@property
@lru_cache()
def non_speech_tokens(self) -> Tuple[int]:
"""
Returns the list of tokens to suppress in order to avoid any speaker tags or non-speech
annotations, to prevent sampling texts that are not actually spoken in the audio, e.g.
- ♪♪♪
- ( SPEAKING FOREIGN LANGUAGE )
- [DAVID] Hey there,
keeping basic punctuations like commas, periods, question marks, exclamation points, etc.
"""
symbols = list("\"#()*+/:;<=>@[\\]^_`{|}~「」『』")
symbols += "<< >> <<< >>> -- --- -( -[ (' (\" (( )) ((( ))) [[ ]] {{ }} ♪♪ ♪♪♪".split()
# symbols that may be a single token or multiple tokens depending on the tokenizer.
# In case they're multiple tokens, suppress the first token, which is safe because:
# These are between U+2640 and U+267F miscellaneous symbols that are okay to suppress
# in generations, and in the 3-byte UTF-8 representation they share the first two bytes.
miscellaneous = set("♩♪♫♬♭♮♯")
assert all(0x2640 <= ord(c) <= 0x267F for c in miscellaneous)
# allow hyphens "-" and single quotes "'" between words, but not at the beginning of a word
result = {self.tokenizer.encode(" -")[0], self.tokenizer.encode(" '")[0]}
for symbol in symbols + list(miscellaneous):
for tokens in [self.tokenizer.encode(symbol), self.tokenizer.encode(" " + symbol)]:
if len(tokens) == 1 or symbol in miscellaneous:
result.add(tokens[0])
return tuple(sorted(result))
def _get_single_token_id(self, text) -> int:
tokens = self.tokenizer.encode(text)
assert len(tokens) == 1, f"{text} is not encoded as a single token"
return tokens[0]
@lru_cache(maxsize=None)
def build_tokenizer(name: str = "gpt2", resource_path: str = None):
os.environ["TOKENIZERS_PARALLELISM"] = "false"
if resource_path is not None:
path = os.path.join(resource_path, name)
else:
path = os.path.join(os.path.dirname(__file__), "assets", name)
tokenizer = GPT2TokenizerFast.from_pretrained(path)
specials = [
"<|startoftranscript|>",
*[f"<|{lang}|>" for lang in LANGUAGES.keys()],
"<|translate|>",
"<|transcribe|>",
"<|startoflm|>",
"<|startofprev|>",
"<|nospeech|>",
"<|notimestamps|>",
]
tokenizer.add_special_tokens(dict(additional_special_tokens=specials))
return tokenizer
@lru_cache(maxsize=None)
def get_tokenizer(
multilingual: bool,
*,
task: Optional[str] = None, # Literal["transcribe", "translate", None]
language: Optional[str] = None,
) -> Tokenizer:
if language is not None:
language = language.lower()
if language not in LANGUAGES:
if language in TO_LANGUAGE_CODE:
language = TO_LANGUAGE_CODE[language]
else:
raise ValueError(f"Unsupported language: {language}")
if multilingual:
tokenizer_name = "multilingual"
task = task or "transcribe"
language = language or "en"
else:
tokenizer_name = "gpt2"
task = None
language = None
tokenizer = build_tokenizer(name=tokenizer_name)
all_special_ids: List[int] = tokenizer.all_special_ids
sot: int = all_special_ids[1]
translate: int = all_special_ids[-6]
transcribe: int = all_special_ids[-5]
langs = tuple(LANGUAGES.keys())
sot_sequence = [sot]
if language is not None:
sot_sequence.append(sot + 1 + langs.index(language))
if task is not None:
sot_sequence.append(transcribe if task == "transcribe" else translate)
return Tokenizer(tokenizer=tokenizer, language=language, sot_sequence=tuple(sot_sequence))
-316
View File
@@ -1,316 +0,0 @@
import argparse
import os
import warnings
from typing import Optional, Tuple, Union, TYPE_CHECKING
import numpy as np
import torch
import tqdm
from funasr.models.whisper.utils.audio import SAMPLE_RATE, N_FRAMES, HOP_LENGTH, pad_or_trim, log_mel_spectrogram
from funasr.models.whisper.utils.decoding import DecodingOptions, DecodingResult
from funasr.models.whisper.utils.tokenizer import LANGUAGES, TO_LANGUAGE_CODE, get_tokenizer
from funasr.models.whisper.utils.utils import exact_div, format_timestamp, make_safe, optional_int, optional_float, str2bool, get_writer
if TYPE_CHECKING:
from .model import Whisper
def transcribe(
model: "Whisper",
audio: Union[str, np.ndarray, torch.Tensor],
*,
verbose: Optional[bool] = None,
temperature: Union[float, Tuple[float, ...]] = (0.0, 0.2, 0.4, 0.6, 0.8, 1.0),
compression_ratio_threshold: Optional[float] = 2.4,
logprob_threshold: Optional[float] = -1.0,
no_speech_threshold: Optional[float] = 0.6,
condition_on_previous_text: bool = True,
**decode_options,
):
"""
Transcribe an audio file using Whisper
Parameters
----------
model: Whisper
The Whisper model instance
audio: Union[str, np.ndarray, torch.Tensor]
The path to the audio file to open, or the audio waveform
verbose: bool
Whether to display the text being decoded to the console. If True, displays all the details,
If False, displays minimal details. If None, does not display anything
temperature: Union[float, Tuple[float, ...]]
Temperature for sampling. It can be a tuple of temperatures, which will be successively used
upon failures according to either `compression_ratio_threshold` or `logprob_threshold`.
compression_ratio_threshold: float
If the gzip compression ratio is above this value, treat as failed
logprob_threshold: float
If the average log probability over sampled tokens is below this value, treat as failed
no_speech_threshold: float
If the no_speech probability is higher than this value AND the average log probability
over sampled tokens is below `logprob_threshold`, consider the segment as silent
condition_on_previous_text: bool
if True, the previous output of the model is provided as a prompt for the next window;
disabling may make the text inconsistent across windows, but the model becomes less prone to
getting stuck in a failure loop, such as repetition looping or timestamps going out of sync.
decode_options: dict
Keyword arguments to construct `DecodingOptions` instances
Returns
-------
A dictionary containing the resulting text ("text") and segment-level details ("segments"), and
the spoken language ("language"), which is detected when `decode_options["language"]` is None.
"""
dtype = torch.float16 if decode_options.get("fp16", True) else torch.float32
if model.device == torch.device("cpu"):
if torch.cuda.is_available():
warnings.warn("Performing inference on CPU when CUDA is available")
if dtype == torch.float16:
warnings.warn("FP16 is not supported on CPU; using FP32 instead")
dtype = torch.float32
if dtype == torch.float32:
decode_options["fp16"] = False
mel = log_mel_spectrogram(audio)
if decode_options.get("language", None) is None:
if not model.is_multilingual:
decode_options["language"] = "en"
else:
if verbose:
print("Detecting language using up to the first 30 seconds. Use `--language` to specify the language")
segment = pad_or_trim(mel, N_FRAMES).to(model.device).to(dtype)
_, probs = model.detect_language(segment)
decode_options["language"] = max(probs, key=probs.get)
if verbose is not None:
print(f"Detected language: {LANGUAGES[decode_options['language']].title()}")
language = decode_options["language"]
task = decode_options.get("task", "transcribe")
tokenizer = get_tokenizer(model.is_multilingual, language=language, task=task)
def decode_with_fallback(segment: torch.Tensor) -> DecodingResult:
temperatures = [temperature] if isinstance(temperature, (int, float)) else temperature
decode_result = None
for t in temperatures:
kwargs = {**decode_options}
if t > 0:
# disable beam_size and patience when t > 0
kwargs.pop("beam_size", None)
kwargs.pop("patience", None)
else:
# disable best_of when t == 0
kwargs.pop("best_of", None)
options = DecodingOptions(**kwargs, temperature=t)
decode_result = model.decode(segment, options)
needs_fallback = False
if compression_ratio_threshold is not None and decode_result.compression_ratio > compression_ratio_threshold:
needs_fallback = True # too repetitive
if logprob_threshold is not None and decode_result.avg_logprob < logprob_threshold:
needs_fallback = True # average log probability is too low
if not needs_fallback:
break
return decode_result
seek = 0
input_stride = exact_div(
N_FRAMES, model.dims.n_audio_ctx
) # mel frames per output token: 2
time_precision = (
input_stride * HOP_LENGTH / SAMPLE_RATE
) # time per output token: 0.02 (seconds)
all_tokens = []
all_segments = []
prompt_reset_since = 0
initial_prompt = decode_options.pop("initial_prompt", None) or []
if initial_prompt:
initial_prompt = tokenizer.encode(" " + initial_prompt.strip())
all_tokens.extend(initial_prompt)
def add_segment(
*, start: float, end: float, text_tokens: torch.Tensor, result: DecodingResult
):
text = tokenizer.decode([token for token in text_tokens if token < tokenizer.eot])
if len(text.strip()) == 0: # skip empty text output
return
all_segments.append(
{
"id": len(all_segments),
"seek": seek,
"start": start,
"end": end,
"text": text,
"tokens": text_tokens.tolist(),
"temperature": result.temperature,
"avg_logprob": result.avg_logprob,
"compression_ratio": result.compression_ratio,
"no_speech_prob": result.no_speech_prob,
}
)
if verbose:
print(make_safe(f"[{format_timestamp(start)} --> {format_timestamp(end)}] {text}"))
# show the progress bar when verbose is False (otherwise the transcribed text will be printed)
num_frames = mel.shape[-1]
previous_seek_value = seek
with tqdm.tqdm(total=num_frames, unit='frames', disable=verbose is not False) as pbar:
while seek < num_frames:
timestamp_offset = float(seek * HOP_LENGTH / SAMPLE_RATE)
segment = pad_or_trim(mel[:, seek:], N_FRAMES).to(model.device).to(dtype)
segment_duration = segment.shape[-1] * HOP_LENGTH / SAMPLE_RATE
decode_options["prompt"] = all_tokens[prompt_reset_since:]
result: DecodingResult = decode_with_fallback(segment)
tokens = torch.tensor(result.tokens)
if no_speech_threshold is not None:
# no voice activity check
should_skip = result.no_speech_prob > no_speech_threshold
if logprob_threshold is not None and result.avg_logprob > logprob_threshold:
# don't skip if the logprob is high enough, despite the no_speech_prob
should_skip = False
if should_skip:
seek += segment.shape[-1] # fast-forward to the next segment boundary
continue
timestamp_tokens: torch.Tensor = tokens.ge(tokenizer.timestamp_begin)
consecutive = torch.where(timestamp_tokens[:-1] & timestamp_tokens[1:])[0].add_(1)
if len(consecutive) > 0: # if the output contains two consecutive timestamp tokens
last_slice = 0
for current_slice in consecutive:
sliced_tokens = tokens[last_slice:current_slice]
start_timestamp_position = (
sliced_tokens[0].item() - tokenizer.timestamp_begin
)
end_timestamp_position = (
sliced_tokens[-1].item() - tokenizer.timestamp_begin
)
add_segment(
start=timestamp_offset + start_timestamp_position * time_precision,
end=timestamp_offset + end_timestamp_position * time_precision,
text_tokens=sliced_tokens[1:-1],
result=result,
)
last_slice = current_slice
last_timestamp_position = (
tokens[last_slice - 1].item() - tokenizer.timestamp_begin
)
seek += last_timestamp_position * input_stride
all_tokens.extend(tokens[: last_slice + 1].tolist())
else:
duration = segment_duration
timestamps = tokens[timestamp_tokens.nonzero().flatten()]
if len(timestamps) > 0 and timestamps[-1].item() != tokenizer.timestamp_begin:
# no consecutive timestamps but it has a timestamp; use the last one.
# single timestamp at the end means no speech after the last timestamp.
last_timestamp_position = timestamps[-1].item() - tokenizer.timestamp_begin
duration = last_timestamp_position * time_precision
add_segment(
start=timestamp_offset,
end=timestamp_offset + duration,
text_tokens=tokens,
result=result,
)
seek += segment.shape[-1]
all_tokens.extend(tokens.tolist())
if not condition_on_previous_text or result.temperature > 0.5:
# do not feed the prompt tokens if a high temperature was used
prompt_reset_since = len(all_tokens)
# update progress bar
pbar.update(min(num_frames, seek) - previous_seek_value)
previous_seek_value = seek
return dict(text=tokenizer.decode(all_tokens[len(initial_prompt):]), segments=all_segments, language=language)
def cli():
from . import available_models
parser = argparse.ArgumentParser(formatter_class=argparse.ArgumentDefaultsHelpFormatter)
parser.add_argument("audio", nargs="+", type=str, help="audio file(s) to transcribe")
parser.add_argument("--model", default="small", choices=available_models(), help="name of the Whisper model to use")
parser.add_argument("--model_dir", type=str, default=None, help="the path to save model files; uses ~/.cache/whisper by default")
parser.add_argument("--device", default="cuda" if torch.cuda.is_available() else "cpu", help="device to use for PyTorch inference")
parser.add_argument("--output_dir", "-o", type=str, default=".", help="directory to save the outputs")
parser.add_argument("--output_format", "-f", type=str, default="all", choices=["txt", "vtt", "srt", "tsv", "json", "all"], help="format of the output file; if not specified, all available formats will be produced")
parser.add_argument("--verbose", type=str2bool, default=True, help="whether to print out the progress and debug messages")
parser.add_argument("--task", type=str, default="transcribe", choices=["transcribe", "translate"], help="whether to perform X->X speech recognition ('transcribe') or X->English translation ('translate')")
parser.add_argument("--language", type=str, default=None, choices=sorted(LANGUAGES.keys()) + sorted([k.title() for k in TO_LANGUAGE_CODE.keys()]), help="language spoken in the audio, specify None to perform language detection")
parser.add_argument("--temperature", type=float, default=0, help="temperature to use for sampling")
parser.add_argument("--best_of", type=optional_int, default=5, help="number of candidates when sampling with non-zero temperature")
parser.add_argument("--beam_size", type=optional_int, default=5, help="number of beams in beam search, only applicable when temperature is zero")
parser.add_argument("--patience", type=float, default=None, help="optional patience value to use in beam decoding, as in https://arxiv.org/abs/2204.05424, the default (1.0) is equivalent to conventional beam search")
parser.add_argument("--length_penalty", type=float, default=None, help="optional token length penalty coefficient (alpha) as in https://arxiv.org/abs/1609.08144, uses simple length normalization by default")
parser.add_argument("--suppress_tokens", type=str, default="-1", help="comma-separated list of token ids to suppress during sampling; '-1' will suppress most special characters except common punctuations")
parser.add_argument("--initial_prompt", type=str, default=None, help="optional text to provide as a prompt for the first window.")
parser.add_argument("--condition_on_previous_text", type=str2bool, default=True, help="if True, provide the previous output of the model as a prompt for the next window; disabling may make the text inconsistent across windows, but the model becomes less prone to getting stuck in a failure loop")
parser.add_argument("--fp16", type=str2bool, default=True, help="whether to perform inference in fp16; True by default")
parser.add_argument("--temperature_increment_on_fallback", type=optional_float, default=0.2, help="temperature to increase when falling back when the decoding fails to meet either of the thresholds below")
parser.add_argument("--compression_ratio_threshold", type=optional_float, default=2.4, help="if the gzip compression ratio is higher than this value, treat the decoding as failed")
parser.add_argument("--logprob_threshold", type=optional_float, default=-1.0, help="if the average log probability is lower than this value, treat the decoding as failed")
parser.add_argument("--no_speech_threshold", type=optional_float, default=0.6, help="if the probability of the <|nospeech|> token is higher than this value AND the decoding has failed due to `logprob_threshold`, consider the segment as silence")
parser.add_argument("--threads", type=optional_int, default=0, help="number of threads used by torch for CPU inference; supercedes MKL_NUM_THREADS/OMP_NUM_THREADS")
args = parser.parse_args().__dict__
model_name: str = args.pop("model")
model_dir: str = args.pop("model_dir")
output_dir: str = args.pop("output_dir")
output_format: str = args.pop("output_format")
device: str = args.pop("device")
os.makedirs(output_dir, exist_ok=True)
if model_name.endswith(".en") and args["language"] not in {"en", "English"}:
if args["language"] is not None:
warnings.warn(f"{model_name} is an English-only model but receipted '{args['language']}'; using English instead.")
args["language"] = "en"
temperature = args.pop("temperature")
temperature_increment_on_fallback = args.pop("temperature_increment_on_fallback")
if temperature_increment_on_fallback is not None:
temperature = tuple(np.arange(temperature, 1.0 + 1e-6, temperature_increment_on_fallback))
else:
temperature = [temperature]
threads = args.pop("threads")
if threads > 0:
torch.set_num_threads(threads)
from . import load_model
model = load_model(model_name, device=device, download_root=model_dir)
writer = get_writer(output_format, output_dir)
for audio_path in args.pop("audio"):
result = transcribe(model, audio_path, temperature=temperature, **args)
writer(result, audio_path)
if __name__ == '__main__':
cli()
-163
View File
@@ -1,163 +0,0 @@
import json
import os
import sys
import zlib
from typing import Callable, TextIO
system_encoding = sys.getdefaultencoding()
if system_encoding != "utf-8":
def make_safe(string):
# replaces any character not representable using the system default encoding with an '?',
# avoiding UnicodeEncodeError (https://github.com/openai/whisper/discussions/729).
return string.encode(system_encoding, errors="replace").decode(system_encoding)
else:
def make_safe(string):
# utf-8 can encode any Unicode code point, so no need to do the round-trip encoding
return string
def exact_div(x, y):
assert x % y == 0
return x // y
def str2bool(string):
str2val = {"True": True, "False": False}
if string in str2val:
return str2val[string]
else:
raise ValueError(f"Expected one of {set(str2val.keys())}, got {string}")
def optional_int(string):
return None if string == "None" else int(string)
def optional_float(string):
return None if string == "None" else float(string)
def compression_ratio(text) -> float:
text_bytes = text.encode("utf-8")
return len(text_bytes) / len(zlib.compress(text_bytes))
def format_timestamp(seconds: float, always_include_hours: bool = False, decimal_marker: str = '.'):
assert seconds >= 0, "non-negative timestamp expected"
milliseconds = round(seconds * 1000.0)
hours = milliseconds // 3_600_000
milliseconds -= hours * 3_600_000
minutes = milliseconds // 60_000
milliseconds -= minutes * 60_000
seconds = milliseconds // 1_000
milliseconds -= seconds * 1_000
hours_marker = f"{hours:02d}:" if always_include_hours or hours > 0 else ""
return f"{hours_marker}{minutes:02d}:{seconds:02d}{decimal_marker}{milliseconds:03d}"
class ResultWriter:
extension: str
def __init__(self, output_dir: str):
self.output_dir = output_dir
def __call__(self, result: dict, audio_path: str):
audio_basename = os.path.basename(audio_path)
output_path = os.path.join(self.output_dir, audio_basename + "." + self.extension)
with open(output_path, "w", encoding="utf-8") as f:
self.write_result(result, file=f)
def write_result(self, result: dict, file: TextIO):
raise NotImplementedError
class WriteTXT(ResultWriter):
extension: str = "txt"
def write_result(self, result: dict, file: TextIO):
for segment in result["segments"]:
print(segment['text'].strip(), file=file, flush=True)
class WriteVTT(ResultWriter):
extension: str = "vtt"
def write_result(self, result: dict, file: TextIO):
print("WEBVTT\n", file=file)
for segment in result["segments"]:
print(
f"{format_timestamp(segment['start'])} --> {format_timestamp(segment['end'])}\n"
f"{segment['text'].strip().replace('-->', '->')}\n",
file=file,
flush=True,
)
class WriteSRT(ResultWriter):
extension: str = "srt"
def write_result(self, result: dict, file: TextIO):
for i, segment in enumerate(result["segments"], start=1):
# write srt lines
print(
f"{i}\n"
f"{format_timestamp(segment['start'], always_include_hours=True, decimal_marker=',')} --> "
f"{format_timestamp(segment['end'], always_include_hours=True, decimal_marker=',')}\n"
f"{segment['text'].strip().replace('-->', '->')}\n",
file=file,
flush=True,
)
class WriteTSV(ResultWriter):
"""
Write a transcript to a file in TSV (tab-separated values) format containing lines like:
<start time in integer milliseconds>\t<end time in integer milliseconds>\t<transcript text>
Using integer milliseconds as start and end times means there's no chance of interference from
an environment setting a language encoding that causes the decimal in a floating point number
to appear as a comma; also is faster and more efficient to parse & store, e.g., in C++.
"""
extension: str = "tsv"
def write_result(self, result: dict, file: TextIO):
print("start", "end", "text", sep="\t", file=file)
for segment in result["segments"]:
print(round(1000 * segment['start']), file=file, end="\t")
print(round(1000 * segment['end']), file=file, end="\t")
print(segment['text'].strip().replace("\t", " "), file=file, flush=True)
class WriteJSON(ResultWriter):
extension: str = "json"
def write_result(self, result: dict, file: TextIO):
json.dump(result, file)
def get_writer(output_format: str, output_dir: str) -> Callable[[dict, TextIO], None]:
writers = {
"txt": WriteTXT,
"vtt": WriteVTT,
"srt": WriteSRT,
"tsv": WriteTSV,
"json": WriteJSON,
}
if output_format == "all":
all_writers = [writer(output_dir) for writer in writers.values()]
def write_all(result: dict, file: TextIO):
for writer in all_writers:
writer(result, file)
return write_all
return writers[output_format](output_dir)
+24
View File
@@ -0,0 +1,24 @@
try:
from whisper.tokenizer import get_tokenizer
except:
print("If you want to use hugging, please `pip install -U transformers`")
from funasr.register import tables
@tables.register("tokenizer_classes", "WhisperTokenizer")
def WhisperTokenizer(**kwargs):
language = kwargs.get("language", None)
task = kwargs.get("task", "transcribe")
is_multilingual = kwargs.get("is_multilingual", True)
num_languages = kwargs.get("num_languages", 99)
tokenizer = get_tokenizer(
multilingual=is_multilingual,
num_languages=num_languages,
language=language,
task=task,
)
return tokenizer
+3 -3
View File
@@ -68,9 +68,9 @@ def load_pretrained_model(
else:
buffer = BytesIO(oss_bucket.get_object(path).read())
src_state = torch.load(buffer, map_location=map_location)
if "state_dict" in src_state:
src_state = src_state["state_dict"]
src_state = src_state["state_dict"] if "state_dict" in src_state else src_state
src_state = src_state["model_state_dict"] if "model_state_dict" in src_state else src_state
src_state = src_state["model"] if "model" in src_state else src_state
if isinstance(scope_map, str):