mirror of
https://github.com/tinygrad/tinygrad.git
synced 2026-02-15 17:15:48 -05:00
* whisper: support batch inference, add librispeech WER test, add kv caching and JIT * remove JIT_SUPPORTED_DEVICE --------- Co-authored-by: George Hotz <72895+geohot@users.noreply.github.com>
341 lines
16 KiB
Python
341 lines
16 KiB
Python
# thanks to https://github.com/openai/whisper for a good chunk of MIT licensed code
|
|
|
|
import sys
|
|
import pathlib
|
|
import base64
|
|
import multiprocessing
|
|
import numpy as np
|
|
from typing import Optional, Union, Literal
|
|
from extra.utils import download_file
|
|
from tinygrad.jit import TinyJit
|
|
from tinygrad.nn.state import torch_load, load_state_dict
|
|
from tinygrad.helpers import getenv, DEBUG, CI
|
|
import tinygrad.nn as nn
|
|
from tinygrad.shape.symbolic import Variable
|
|
from tinygrad.tensor import Tensor
|
|
import itertools
|
|
import librosa
|
|
|
|
class MultiHeadAttention:
|
|
def __init__(self, n_state, n_head, kv_caching: Literal['cross', 'self']=None, max_self_attn_cache_len=None):
|
|
self.n_head = n_head
|
|
self.query = nn.Linear(n_state, n_state)
|
|
self.key = nn.Linear(n_state, n_state, bias=False)
|
|
self.value = nn.Linear(n_state, n_state)
|
|
self.out = nn.Linear(n_state, n_state)
|
|
|
|
self.kv_caching = kv_caching
|
|
self.max_self_attn_cache_len = max_self_attn_cache_len
|
|
|
|
def __call__(self, x:Tensor, xa:Optional[Tensor]=None, mask:Optional[Tensor]=None, len: Union[Variable,int]=None):
|
|
if self.kv_caching == 'cross':
|
|
if xa is not None:
|
|
k, v = self.key(xa), self.value(xa)
|
|
if not hasattr(self, 'cache_k'):
|
|
self.cache_k, self.cache_v = k, v
|
|
else:
|
|
# see test_jitted_read_assign in test_jit.py
|
|
self.cache_k.assign(k+1-1).realize()
|
|
self.cache_v.assign(v+1-1).realize()
|
|
else:
|
|
k, v = self.cache_k, self.cache_v
|
|
else:
|
|
k, v = self.key(x), self.value(x)
|
|
if self.kv_caching == 'self':
|
|
if not hasattr(self, 'cache_k'):
|
|
self.cache_k = Tensor.zeros(x.shape[0], self.max_self_attn_cache_len, x.shape[2])
|
|
self.cache_v = Tensor.zeros(x.shape[0], self.max_self_attn_cache_len, x.shape[2])
|
|
k = self.cache_k.shrink((None, (0, len), None)).cat(k, dim=1)
|
|
v = self.cache_v.shrink((None, (0, len), None)).cat(v, dim=1)
|
|
padding = self.max_self_attn_cache_len-len-x.shape[1]
|
|
self.cache_k.assign(k.pad((None, (0, padding), None)).contiguous()).realize()
|
|
self.cache_v.assign(v.pad((None, (0, padding), None)).contiguous()).realize()
|
|
|
|
q = self.query(x)
|
|
n_ctx = q.shape[1]
|
|
assert(q.shape[-1] == k.shape[-1] == v.shape[-1])
|
|
head_dim = q.shape[-1] // self.n_head
|
|
q = q.reshape(*q.shape[:2], self.n_head, head_dim).permute(0, 2, 1, 3)
|
|
k = k.reshape(*k.shape[:2], self.n_head, head_dim).permute(0, 2, 1, 3)
|
|
v = v.reshape(*v.shape[:2], self.n_head, head_dim).permute(0, 2, 1, 3)
|
|
attn = Tensor.scaled_dot_product_attention(q, k, v, mask[:n_ctx,:n_ctx] if mask is not None else None)
|
|
wv = attn.permute(0, 2, 1, 3).flatten(start_dim=2)
|
|
return self.out(wv)
|
|
|
|
|
|
class ResidualAttentionBlock:
|
|
def __init__(self, n_state, n_head, is_decoder_block=False, max_self_attn_cache_len=None):
|
|
self.attn = MultiHeadAttention(n_state, n_head, kv_caching='self' if is_decoder_block else None, max_self_attn_cache_len=max_self_attn_cache_len)
|
|
self.attn_ln = nn.LayerNorm(n_state)
|
|
|
|
self.cross_attn = MultiHeadAttention(n_state, n_head, kv_caching='cross') if is_decoder_block else None
|
|
self.cross_attn_ln = nn.LayerNorm(n_state) if is_decoder_block else None
|
|
|
|
self.mlp = [nn.Linear(n_state, n_state*4), Tensor.gelu, nn.Linear(n_state*4, n_state)]
|
|
self.mlp_ln = nn.LayerNorm(n_state)
|
|
|
|
def __call__(self, x, xa=None, mask=None, len: Union[Variable, int]=None):
|
|
x = x + self.attn(self.attn_ln(x), mask=mask, len=len)
|
|
if self.cross_attn: x = x + self.cross_attn(self.cross_attn_ln(x), xa)
|
|
x = x + self.mlp_ln(x).sequential(self.mlp)
|
|
return x.realize()
|
|
|
|
class AudioEncoder:
|
|
def __init__(self, n_mels, n_audio_ctx, n_audio_state, n_audio_head, n_audio_layer, **_):
|
|
self.conv1 = nn.Conv1d(n_mels, n_audio_state, kernel_size=3, padding=1)
|
|
self.conv2 = nn.Conv1d(n_audio_state, n_audio_state, kernel_size=3, stride=2, padding=1)
|
|
self.blocks = [ResidualAttentionBlock(n_audio_state, n_audio_head) for _ in range(n_audio_layer)]
|
|
self.ln_post = nn.LayerNorm(n_audio_state)
|
|
self.positional_embedding = Tensor.empty(n_audio_ctx, n_audio_state)
|
|
self.encode = TinyJit(self.__call__)
|
|
|
|
def __call__(self, x):
|
|
x = self.conv1(x).gelu()
|
|
x = self.conv2(x).gelu()
|
|
x = x.permute(0, 2, 1)
|
|
x = x + self.positional_embedding[:x.shape[1]]
|
|
x = x.sequential(self.blocks)
|
|
x = self.ln_post(x)
|
|
return x.realize()
|
|
|
|
class TextDecoder:
|
|
def __init__(self, n_vocab, n_text_ctx, n_text_state, n_text_head, n_text_layer, **_):
|
|
self.max_tokens_to_sample = n_text_ctx // 2
|
|
self.max_self_attn_cache_len = self.max_tokens_to_sample * 2 # double the size as an extra buffer for prefix/start tokens
|
|
|
|
self.token_embedding = nn.Embedding(n_vocab, n_text_state)
|
|
self.positional_embedding = Tensor.empty(n_text_ctx, n_text_state)
|
|
self.blocks = [ResidualAttentionBlock(n_text_state, n_text_head, is_decoder_block=True, max_self_attn_cache_len=self.max_self_attn_cache_len) for _ in range(n_text_layer)]
|
|
self.ln = nn.LayerNorm(n_text_state)
|
|
self.mask = Tensor.full((n_text_ctx, n_text_ctx), -np.inf).triu(1).realize()
|
|
self.blocks_start_tok = [TinyJit(block.__call__) for block in self.blocks]
|
|
self.blocks_after_start_tok = [TinyJit(block.__call__) for block in self.blocks]
|
|
self.start_output_tok = TinyJit(self.output_tok)
|
|
self.after_start_output_tok = TinyJit(self.output_tok)
|
|
|
|
def __call__(self, x: Tensor, pos: int, encoded_audio: Tensor, streaming=False):
|
|
seqlen = x.shape[-1]
|
|
x = self.token_embedding(x) + self.positional_embedding[pos:pos+seqlen]
|
|
if pos == 0:
|
|
for block in (self.blocks if streaming else self.blocks_start_tok):
|
|
x = block(x, xa=encoded_audio, mask=self.mask, len=0) # pass xa for cross attn kv caching
|
|
return self.output_tok(x) if streaming else self.start_output_tok(x)
|
|
else:
|
|
for block in self.blocks_after_start_tok:
|
|
len_v = Variable("self_attn_cache_len", 1, self.max_self_attn_cache_len).bind(pos)
|
|
x = block(x, mask=self.mask, len=len_v)
|
|
return self.after_start_output_tok(x)
|
|
|
|
def output_tok(self, x):
|
|
return (self.ln(x) @ self.token_embedding.weight.T).realize()
|
|
|
|
class Whisper:
|
|
def __init__(self, dims, batch_size=1):
|
|
self.encoder = AudioEncoder(**dims)
|
|
self.decoder = TextDecoder(**dims)
|
|
self.is_multilingual = dims["n_vocab"] == 51865
|
|
self.batch_size = batch_size
|
|
|
|
|
|
RATE = 16000
|
|
CHUNK = 1600
|
|
RECORD_SECONDS = 10
|
|
|
|
def prep_audio(waveform, batch_size) -> np.ndarray:
|
|
assert waveform is not None
|
|
|
|
def pad_or_trim(arr, target_len=480000):
|
|
curr_len = len(arr)
|
|
if curr_len == target_len:
|
|
return arr
|
|
elif curr_len < target_len:
|
|
return np.pad(arr, (0, target_len - curr_len), 'constant')
|
|
else:
|
|
return arr[:target_len]
|
|
|
|
waveform = np.array(list(map(pad_or_trim, waveform)))
|
|
assert waveform.shape[0] <= batch_size
|
|
# pad the waveform to match the model's batch_size to avoid JIT shape mismatch errors.
|
|
# if operations like conv in the AudioEncoder could support symbolic shapes, then we wouldn't need to do this here
|
|
if waveform.shape[0] < batch_size:
|
|
waveform = np.pad(waveform, pad_width=((0, batch_size - waveform.shape[0]), (0, 0)))
|
|
|
|
N_FFT = 400
|
|
HOP_LENGTH = 160
|
|
N_MELS = 80
|
|
|
|
stft = librosa.stft(waveform, n_fft=N_FFT, hop_length=HOP_LENGTH, window='hann', dtype=np.csingle)
|
|
magnitudes = np.absolute(stft[..., :-1]) ** 2
|
|
mel_spec = librosa.filters.mel(sr=RATE, n_fft=N_FFT, n_mels=N_MELS) @ magnitudes
|
|
|
|
log_spec = np.log10(np.clip(mel_spec, 1e-10, None))
|
|
log_spec = np.maximum(log_spec, log_spec.max() - 8.0)
|
|
log_spec = (log_spec + 4.0) / 4.0
|
|
|
|
return log_spec
|
|
|
|
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",
|
|
}
|
|
|
|
BASE = pathlib.Path(__file__).parents[1] / "weights"
|
|
def get_encoding(encoding_name):
|
|
filename = encoding_name + ".tiktoken"
|
|
download_file("https://raw.githubusercontent.com/openai/whisper/main/whisper/assets/" + filename, BASE / filename)
|
|
with open(BASE / filename) as f:
|
|
ranks = {base64.b64decode(token): int(rank) for token, rank in (line.split() for line in f if line)}
|
|
n_vocab = len(ranks)
|
|
specials = [
|
|
"<|endoftext|>",
|
|
"<|startoftranscript|>",
|
|
*[f"<|{lang}|>" for lang in LANGUAGES.keys()],
|
|
"<|translate|>",
|
|
"<|transcribe|>",
|
|
"<|startoflm|>",
|
|
"<|startofprev|>",
|
|
"<|nospeech|>",
|
|
"<|notimestamps|>",
|
|
*[f"<|{i * 0.02:.2f}|>" for i in range(1501)],
|
|
]
|
|
special_tokens = dict(zip(specials, itertools.count(n_vocab)))
|
|
n_vocab += len(specials)
|
|
import tiktoken
|
|
return tiktoken.Encoding(
|
|
name=encoding_name,
|
|
explicit_n_vocab=n_vocab,
|
|
pat_str=r"""'s|'t|'re|'ve|'m|'ll|'d| ?\p{L}+| ?\p{N}+| ?[^\s\p{L}\p{N}]+|\s+(?!\S)|\s+""",
|
|
mergeable_ranks=ranks,
|
|
special_tokens=special_tokens)
|
|
|
|
def img(x):
|
|
import matplotlib.pyplot as plt
|
|
plt.imshow(x.numpy())
|
|
plt.show()
|
|
|
|
def listener(q):
|
|
import pyaudio
|
|
p = pyaudio.PyAudio()
|
|
stream = p.open(format=pyaudio.paInt16, channels=1, rate=RATE, input=True, frames_per_buffer=CHUNK)
|
|
print("listening")
|
|
for _ in range(0, int(RATE / CHUNK * RECORD_SECONDS)):
|
|
data = stream.read(CHUNK)
|
|
waveform = ((np.frombuffer(data, np.int16)/32768).astype(np.float32)*3)
|
|
q.put(waveform)
|
|
print("done listening")
|
|
|
|
|
|
MODEL_URLS = {
|
|
"tiny.en": "https://openaipublic.azureedge.net/main/whisper/models/d3dd57d32accea0b295c96e26691aa14d8822fac7d9d27d5dc00b4ca2826dd03/tiny.en.pt",
|
|
"tiny": "https://openaipublic.azureedge.net/main/whisper/models/65147644a518d12f04e32d6f3b26facc3f8dd46e5390956a9424a650c0ce22b9/tiny.pt",
|
|
"base.en": "https://openaipublic.azureedge.net/main/whisper/models/25a8566e1d0c1e2231d1c762132cd20e0f96a85d16145c3a00adf5d1ac670ead/base.en.pt",
|
|
"base": "https://openaipublic.azureedge.net/main/whisper/models/ed3a0b6b1c0edf879ad9b11b1af5a0e6ab5db9205f891f668f8b0e6c6326e34e/base.pt",
|
|
"small.en": "https://openaipublic.azureedge.net/main/whisper/models/f953ad0fd29cacd07d5a9eda5624af0f6bcf2258be67c92b79389873d91e0872/small.en.pt",
|
|
"small": "https://openaipublic.azureedge.net/main/whisper/models/9ecf779972d90ba49c06d968637d720dd632c55bbf19d441fb42bf17a411e794/small.pt",
|
|
"medium.en": "https://openaipublic.azureedge.net/main/whisper/models/d7440d1dc186f76616474e0ff0b3b6b879abc9d1a4926b7adfa41db2d497ab4f/medium.en.pt",
|
|
"medium": "https://openaipublic.azureedge.net/main/whisper/models/345ae4da62f9b3d59415adc60127b97c714f32e89e936602e85993674d08dcb1/medium.pt",
|
|
"large-v1": "https://openaipublic.azureedge.net/main/whisper/models/e4b87e7e0bf463eb8e6956e646f1e277e901512310def2c24bf0e11bd3c28e9a/large-v1.pt",
|
|
"large-v2": "https://openaipublic.azureedge.net/main/whisper/models/81f7c96c852ee8fc832187b0132e569d6c3065a3252ed18e56effd0b6a73e524/large-v2.pt",
|
|
"large": "https://openaipublic.azureedge.net/main/whisper/models/81f7c96c852ee8fc832187b0132e569d6c3065a3252ed18e56effd0b6a73e524/large-v2.pt",
|
|
}
|
|
def init_whisper(model_name="tiny.en", batch_size=1):
|
|
assert MODEL_URLS[model_name] is not None
|
|
|
|
filename = BASE / "whisper-{}.pt".format(model_name)
|
|
download_file(MODEL_URLS[model_name], filename)
|
|
state = torch_load(filename)
|
|
model = Whisper(state['dims'], batch_size)
|
|
load_state_dict(model, state['model_state_dict'], strict=False)
|
|
enc = get_encoding("multilingual" if model.is_multilingual else "gpt2")
|
|
return model, enc
|
|
|
|
def load_file_waveform(filename):
|
|
waveform, _ = librosa.load(filename, sr=RATE)
|
|
return waveform
|
|
|
|
def transcribe_file(model, enc, filename):
|
|
return transcribe_waveform(model, enc, [load_file_waveform(filename)])
|
|
|
|
def transcribe_waveform(model, enc, waveforms):
|
|
"""
|
|
Expects an array of shape (N,S) where N is the number waveforms to transcribe in parallel and S is number of 16000Hz samples
|
|
Returns the transcribed text if a single waveform is provided, or an array of transcriptions if multiple are provided
|
|
"""
|
|
N_audio = len(waveforms)
|
|
log_spec = prep_audio(waveforms, model.batch_size)
|
|
encoded_audio = model.encoder.encode(Tensor(log_spec))
|
|
# print('encoded audio', np.sum(encoded_audio.numpy()))
|
|
|
|
start_tokens = [enc._special_tokens["<|startoftranscript|>"]]
|
|
if model.is_multilingual:
|
|
# TODO detect language
|
|
language_token = enc._special_tokens["<|startoftranscript|>"] + 1 + tuple(LANGUAGES.keys()).index("en")
|
|
start_tokens.append(language_token)
|
|
start_tokens.append(enc._special_tokens["<|transcribe|>"])
|
|
start_tokens.append(enc._special_tokens["<|notimestamps|>"])
|
|
|
|
transcription_start_index = len(start_tokens)
|
|
eot = enc._special_tokens["<|endoftext|>"]
|
|
tokens = np.tile(start_tokens, (log_spec.shape[0], 1))
|
|
|
|
pos = 0
|
|
for i in range(model.decoder.max_tokens_to_sample):
|
|
out = model.decoder(Tensor(tokens if i == 0 else tokens[:, -1:]), pos, encoded_audio)
|
|
next_tokens = out[:, -1].argmax(axis=-1).numpy().astype(np.int32)
|
|
next_tokens[tokens[:, -1] == eot] = eot
|
|
tokens = np.concatenate((tokens, next_tokens.reshape(-1, 1)), axis=1)
|
|
pos = tokens.shape[-1] - 1
|
|
if DEBUG >= 1: print(i, list(map(lambda tokens: enc.decode(tokens), tokens)))
|
|
if (tokens[:, -1] == eot).all():
|
|
break
|
|
|
|
transcriptions = []
|
|
for t in tokens:
|
|
eot_index = np.where(t == eot)[0]
|
|
eot_index = None if len(eot_index) == 0 else eot_index[0]
|
|
transcriptions.append(enc.decode(t[transcription_start_index:eot_index]).strip())
|
|
return transcriptions[:N_audio] if N_audio > 1 else transcriptions[0]
|
|
|
|
|
|
if __name__ == "__main__":
|
|
model, enc = init_whisper("small.en" if getenv("SMALL") else "tiny.en", batch_size=1)
|
|
|
|
if len(sys.argv) > 1:
|
|
print(transcribe_file(model, enc, sys.argv[1]))
|
|
else:
|
|
# online
|
|
|
|
q = multiprocessing.Queue()
|
|
p = multiprocessing.Process(target=listener, args=(q,))
|
|
p.daemon = True
|
|
p.start()
|
|
|
|
lst = [enc._special_tokens["<|startoftranscript|>"], enc._special_tokens["<|notimestamps|>"]]
|
|
total = None
|
|
did_read = False
|
|
for i in range(0, int(RATE / CHUNK * RECORD_SECONDS)):
|
|
while not q.empty() or total is None:
|
|
waveform = q.get()
|
|
if total is None: total = waveform
|
|
else: total = np.concatenate([total, waveform])
|
|
did_read = True
|
|
if did_read:
|
|
log_spec = prep_audio(total.reshape(1, -1), 1)
|
|
encoded_audio = model.encoder.encode(Tensor(log_spec))
|
|
# pass the previously inferred tokens as 'prefix' - https://github.com/openai/whisper/discussions/117#discussioncomment-3727051
|
|
out = model.decoder(Tensor([lst]), 0, encoded_audio, streaming=True).realize()
|
|
idx = int(out[0,-1].argmax().numpy().item())
|
|
lst.append(idx)
|
|
dec = enc.decode(lst)
|
|
print(dec) # DO NOT REMOVE PRINT. IT'S VERY IMPORTANT
|
|
if dec.endswith("<|endoftext|>"):
|
|
lst.pop()
|