mirror of https://github.com/commaai/tinygrad.git
353 lines
18 KiB
Python
353 lines
18 KiB
Python
# thanks to https://github.com/openai/whisper for a good chunk of MIT licensed code
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import sys
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import pathlib
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import base64
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import multiprocessing
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import numpy as np
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from typing import Optional, Union, Literal, List
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from tinygrad.features.jit import TinyJit
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from tinygrad.nn.state import torch_load, load_state_dict
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from tinygrad.helpers import getenv, DEBUG, CI, fetch
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import tinygrad.nn as nn
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from tinygrad.shape.symbolic import Variable
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from tinygrad.tensor import Tensor
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import itertools
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import librosa
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class MultiHeadAttention:
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def __init__(self, n_state, n_head, kv_caching: Literal['cross', 'self']=None, max_self_attn_cache_len=None):
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self.n_head = n_head
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self.query = nn.Linear(n_state, n_state)
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self.key = nn.Linear(n_state, n_state, bias=False)
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self.value = nn.Linear(n_state, n_state)
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self.out = nn.Linear(n_state, n_state)
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self.kv_caching = kv_caching
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self.max_self_attn_cache_len = max_self_attn_cache_len
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def __call__(self, x:Tensor, xa:Optional[Tensor]=None, mask:Optional[Tensor]=None, len: Union[Variable,int]=None):
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if self.kv_caching == 'cross':
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if xa is not None:
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k, v = self.key(xa), self.value(xa)
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if not hasattr(self, 'cache_k'):
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self.cache_k, self.cache_v = k, v
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else:
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# see test_jitted_read_assign in test_jit.py. more context https://github.com/tinygrad/tinygrad/pull/2360#issuecomment-1817989994
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self.cache_k.assign(k+1-1).realize()
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self.cache_v.assign(v+1-1).realize()
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else:
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k, v = self.cache_k, self.cache_v
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else:
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k, v = self.key(x), self.value(x)
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if self.kv_caching == 'self':
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if not hasattr(self, 'cache_k'):
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self.cache_k = Tensor.zeros(x.shape[0], self.max_self_attn_cache_len, x.shape[2])
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self.cache_v = Tensor.zeros(x.shape[0], self.max_self_attn_cache_len, x.shape[2])
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k = self.cache_k.shrink((None, (0, len), None)).cat(k, dim=1)
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v = self.cache_v.shrink((None, (0, len), None)).cat(v, dim=1)
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padding = self.max_self_attn_cache_len-len-x.shape[1]
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self.cache_k.assign(k.pad((None, (0, padding), None)).contiguous()).realize()
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self.cache_v.assign(v.pad((None, (0, padding), None)).contiguous()).realize()
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q = self.query(x)
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n_ctx = q.shape[1]
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assert(q.shape[-1] == k.shape[-1] == v.shape[-1])
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head_dim = q.shape[-1] // self.n_head
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q = q.reshape(*q.shape[:2], self.n_head, head_dim).permute(0, 2, 1, 3)
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k = k.reshape(*k.shape[:2], self.n_head, head_dim).permute(0, 2, 1, 3)
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v = v.reshape(*v.shape[:2], self.n_head, head_dim).permute(0, 2, 1, 3)
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attn = Tensor.scaled_dot_product_attention(q, k, v, mask[:n_ctx,:n_ctx] if mask is not None else None)
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wv = attn.permute(0, 2, 1, 3).flatten(start_dim=2)
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return self.out(wv)
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class ResidualAttentionBlock:
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def __init__(self, n_state, n_head, is_decoder_block=False, max_self_attn_cache_len=None):
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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)
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self.attn_ln = nn.LayerNorm(n_state)
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self.cross_attn = MultiHeadAttention(n_state, n_head, kv_caching='cross') if is_decoder_block else None
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self.cross_attn_ln = nn.LayerNorm(n_state) if is_decoder_block else None
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self.mlp = [nn.Linear(n_state, n_state*4), Tensor.gelu, nn.Linear(n_state*4, n_state)]
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self.mlp_ln = nn.LayerNorm(n_state)
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def __call__(self, x, xa=None, mask=None, len: Union[Variable, int]=None):
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x = x + self.attn(self.attn_ln(x), mask=mask, len=len)
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if self.cross_attn: x = x + self.cross_attn(self.cross_attn_ln(x), xa)
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x = x + self.mlp_ln(x).sequential(self.mlp)
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return x.realize()
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class AudioEncoder:
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def __init__(self, n_mels, n_audio_ctx, n_audio_state, n_audio_head, n_audio_layer, **_):
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self.conv1 = nn.Conv1d(n_mels, n_audio_state, kernel_size=3, padding=1)
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self.conv2 = nn.Conv1d(n_audio_state, n_audio_state, kernel_size=3, stride=2, padding=1)
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self.blocks = [ResidualAttentionBlock(n_audio_state, n_audio_head) for _ in range(n_audio_layer)]
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self.ln_post = nn.LayerNorm(n_audio_state)
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self.positional_embedding = Tensor.empty(n_audio_ctx, n_audio_state)
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self.encode = TinyJit(self.__call__)
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def __call__(self, x):
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x = self.conv1(x).gelu()
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x = self.conv2(x).gelu()
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x = x.permute(0, 2, 1)
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x = x + self.positional_embedding[:x.shape[1]]
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x = x.sequential(self.blocks)
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x = self.ln_post(x)
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return x.realize()
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class TextDecoder:
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def __init__(self, n_vocab, n_text_ctx, n_text_state, n_text_head, n_text_layer, **_):
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self.max_tokens_to_sample = n_text_ctx // 2
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self.max_self_attn_cache_len = self.max_tokens_to_sample * 2 + 5 # roughly prompt + start toks + max_tokens_to_sample
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self.token_embedding = nn.Embedding(n_vocab, n_text_state)
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self.positional_embedding = Tensor.empty(n_text_ctx, n_text_state)
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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)]
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self.ln = nn.LayerNorm(n_text_state)
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self.mask = Tensor.full((n_text_ctx, n_text_ctx), -np.inf).triu(1).realize()
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self.blocks_start_tok = [TinyJit(block.__call__) for block in self.blocks]
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self.blocks_after_start_tok = [TinyJit(block.__call__) for block in self.blocks]
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self.start_output_tok = TinyJit(self.output_tok)
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self.after_start_output_tok = TinyJit(self.output_tok)
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# if layernorm supported symbolic shapes, we wouldn't need this hacky 'streaming' param (which should be called something more descriptive like 'x_is_start_toks_only')
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def __call__(self, x: Tensor, pos: int, encoded_audio: Tensor, streaming=False):
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seqlen = x.shape[-1]
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x = self.token_embedding(x) + self.positional_embedding[pos:pos+seqlen]
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if pos == 0:
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for block in (self.blocks if streaming else self.blocks_start_tok):
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x = block(x, xa=encoded_audio, mask=self.mask, len=0) # pass xa for cross attn kv caching
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return self.output_tok(x) if streaming else self.start_output_tok(x)
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else:
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for block in self.blocks_after_start_tok:
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len_v = Variable("self_attn_cache_len", 1, self.max_self_attn_cache_len).bind(pos)
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x = block(x, mask=self.mask, len=len_v)
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return self.after_start_output_tok(x)
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def output_tok(self, x):
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return (self.ln(x) @ self.token_embedding.weight.T).realize()
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class Whisper:
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def __init__(self, dims, batch_size=1):
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self.encoder = AudioEncoder(**dims)
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self.decoder = TextDecoder(**dims)
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self.is_multilingual = dims["n_vocab"] == 51865
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self.batch_size = batch_size
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RATE = 16000
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SEGMENT_SECONDS=30
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SAMPLES_PER_SEGMENT = RATE * SEGMENT_SECONDS # 480000
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N_FFT = 400
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HOP_LENGTH = 160
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N_MELS = 80
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FRAMES_PER_SEGMENT = SAMPLES_PER_SEGMENT // HOP_LENGTH # 3000
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def prep_audio(waveforms: List[np.ndarray], batch_size: int, truncate=False) -> np.ndarray:
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"""
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:param waveforms: A list of possibly variable length 16000Hz audio samples
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:param batch_size: The batch_size associated with the Whisper model being used to transcribe the audio.
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Used to prevent JIT mismatch errors since the encoder does not accept symbolic shapes
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:param truncate: If true, truncates (or pads) audio to exactly 30s for a single encoder pass
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:return: mel spectrogram of the given waveforms
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"""
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def pad_or_trim(arr, target_len):
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curr_len = len(arr)
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if curr_len == target_len:
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return arr
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elif curr_len < target_len:
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return np.pad(arr, (0, target_len - curr_len), 'constant')
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else:
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return arr[:target_len]
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max_len = SAMPLES_PER_SEGMENT if truncate else max(len(wav) for wav in waveforms)
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if (r := max_len % SAMPLES_PER_SEGMENT) > 0: max_len += SAMPLES_PER_SEGMENT - r
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waveforms = np.array(list(map(lambda w: pad_or_trim(w, max_len), waveforms)))
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assert waveforms.shape[0] <= batch_size
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if waveforms.shape[0] < batch_size:
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# we could have a symbolic batch_size dim instead of manually padding here if conv/layernorm supported symbolic shapes
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waveforms = np.pad(waveforms, pad_width=((0, batch_size - waveforms.shape[0]), (0, 0)))
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stft = librosa.stft(waveforms, n_fft=N_FFT, hop_length=HOP_LENGTH, window='hann', dtype=np.csingle)
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magnitudes = np.absolute(stft[..., :-1]) ** 2
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mel_spec = librosa.filters.mel(sr=RATE, n_fft=N_FFT, n_mels=N_MELS) @ magnitudes
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log_spec = np.log10(np.clip(mel_spec, 1e-10, None))
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log_spec = np.maximum(log_spec, log_spec.max() - 8.0)
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log_spec = (log_spec + 4.0) / 4.0
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return log_spec
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LANGUAGES = {
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"en": "english", "zh": "chinese", "de": "german", "es": "spanish", "ru": "russian", "ko": "korean", "fr": "french", "ja": "japanese", "pt": "portuguese", "tr": "turkish",
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"pl": "polish", "ca": "catalan", "nl": "dutch", "ar": "arabic", "sv": "swedish", "it": "italian", "id": "indonesian", "hi": "hindi", "fi": "finnish", "vi": "vietnamese",
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"he": "hebrew", "uk": "ukrainian", "el": "greek", "ms": "malay", "cs": "czech", "ro": "romanian", "da": "danish", "hu": "hungarian", "ta": "tamil", "no": "norwegian",
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"th": "thai", "ur": "urdu", "hr": "croatian", "bg": "bulgarian", "lt": "lithuanian", "la": "latin", "mi": "maori", "ml": "malayalam", "cy": "welsh", "sk": "slovak", "te": "telugu",
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"fa": "persian", "lv": "latvian", "bn": "bengali", "sr": "serbian", "az": "azerbaijani", "sl": "slovenian", "kn": "kannada", "et": "estonian", "mk": "macedonian",
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"br": "breton", "eu": "basque", "is": "icelandic", "hy": "armenian", "ne": "nepali", "mn": "mongolian", "bs": "bosnian", "kk": "kazakh", "sq": "albanian", "sw": "swahili",
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"gl": "galician", "mr": "marathi", "pa": "punjabi", "si": "sinhala", "km": "khmer", "sn": "shona", "yo": "yoruba", "so": "somali", "af": "afrikaans", "oc": "occitan", "ka": "georgian",
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"be": "belarusian", "tg": "tajik", "sd": "sindhi", "gu": "gujarati", "am": "amharic", "yi": "yiddish", "lo": "lao", "uz": "uzbek", "fo": "faroese", "ht": "haitian creole",
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"ps": "pashto", "tk": "turkmen", "nn": "nynorsk", "mt": "maltese", "sa": "sanskrit", "lb": "luxembourgish", "my": "myanmar", "bo": "tibetan", "tl": "tagalog", "mg": "malagasy",
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"as": "assamese", "tt": "tatar", "haw": "hawaiian", "ln": "lingala", "ha": "hausa", "ba": "bashkir", "jw": "javanese", "su": "sundanese",
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}
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def get_encoding(encoding_name):
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with fetch(f"https://raw.githubusercontent.com/openai/whisper/main/whisper/assets/{encoding_name}.tiktoken").open() as f:
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ranks = {base64.b64decode(token): int(rank) for token, rank in (line.split() for line in f if line)}
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n_vocab = len(ranks)
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specials = [
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"<|endoftext|>",
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"<|startoftranscript|>",
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*[f"<|{lang}|>" for lang in LANGUAGES.keys()],
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"<|translate|>",
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"<|transcribe|>",
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"<|startoflm|>",
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"<|startofprev|>",
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"<|nospeech|>",
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"<|notimestamps|>",
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*[f"<|{i * 0.02:.2f}|>" for i in range(1501)],
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]
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special_tokens = dict(zip(specials, itertools.count(n_vocab)))
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n_vocab += len(specials)
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import tiktoken
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return tiktoken.Encoding(
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name=encoding_name,
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explicit_n_vocab=n_vocab,
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pat_str=r"""'s|'t|'re|'ve|'m|'ll|'d| ?\p{L}+| ?\p{N}+| ?[^\s\p{L}\p{N}]+|\s+(?!\S)|\s+""",
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mergeable_ranks=ranks,
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special_tokens=special_tokens)
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MODEL_URLS = {
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"tiny.en": "https://openaipublic.azureedge.net/main/whisper/models/d3dd57d32accea0b295c96e26691aa14d8822fac7d9d27d5dc00b4ca2826dd03/tiny.en.pt",
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"tiny": "https://openaipublic.azureedge.net/main/whisper/models/65147644a518d12f04e32d6f3b26facc3f8dd46e5390956a9424a650c0ce22b9/tiny.pt",
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"base.en": "https://openaipublic.azureedge.net/main/whisper/models/25a8566e1d0c1e2231d1c762132cd20e0f96a85d16145c3a00adf5d1ac670ead/base.en.pt",
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"base": "https://openaipublic.azureedge.net/main/whisper/models/ed3a0b6b1c0edf879ad9b11b1af5a0e6ab5db9205f891f668f8b0e6c6326e34e/base.pt",
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"small.en": "https://openaipublic.azureedge.net/main/whisper/models/f953ad0fd29cacd07d5a9eda5624af0f6bcf2258be67c92b79389873d91e0872/small.en.pt",
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"small": "https://openaipublic.azureedge.net/main/whisper/models/9ecf779972d90ba49c06d968637d720dd632c55bbf19d441fb42bf17a411e794/small.pt",
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"medium.en": "https://openaipublic.azureedge.net/main/whisper/models/d7440d1dc186f76616474e0ff0b3b6b879abc9d1a4926b7adfa41db2d497ab4f/medium.en.pt",
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"medium": "https://openaipublic.azureedge.net/main/whisper/models/345ae4da62f9b3d59415adc60127b97c714f32e89e936602e85993674d08dcb1/medium.pt",
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"large-v1": "https://openaipublic.azureedge.net/main/whisper/models/e4b87e7e0bf463eb8e6956e646f1e277e901512310def2c24bf0e11bd3c28e9a/large-v1.pt",
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"large-v2": "https://openaipublic.azureedge.net/main/whisper/models/81f7c96c852ee8fc832187b0132e569d6c3065a3252ed18e56effd0b6a73e524/large-v2.pt",
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"large": "https://openaipublic.azureedge.net/main/whisper/models/81f7c96c852ee8fc832187b0132e569d6c3065a3252ed18e56effd0b6a73e524/large-v2.pt",
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}
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def init_whisper(model_name="tiny.en", batch_size=1):
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assert MODEL_URLS[model_name] is not None
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filename = fetch(MODEL_URLS[model_name])
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state = torch_load(filename)
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model = Whisper(state['dims'], batch_size)
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load_state_dict(model, state['model_state_dict'], strict=False)
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enc = get_encoding("multilingual" if model.is_multilingual else "gpt2")
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return model, enc
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def load_file_waveform(filename):
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waveform, _ = librosa.load(filename, sr=RATE)
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return waveform
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def transcribe_file(model, enc, filename):
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return transcribe_waveform(model, enc, [load_file_waveform(filename)])
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def transcribe_waveform(model, enc, waveforms, truncate=False):
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"""
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Expects an array of shape (N,S) where N is the number waveforms to transcribe in parallel and S is number of 16000Hz samples
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Returns the transcribed text if a single waveform is provided, or an array of transcriptions if multiple are provided
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"""
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N_audio = len(waveforms)
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log_spec = prep_audio(waveforms, model.batch_size, truncate)
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if log_spec.shape[-1] > FRAMES_PER_SEGMENT and N_audio > 1:
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# we don't support multi-segment batching because the size of the prompt tokens would be different for each item in the batch
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# if we really want this feature, we can consider padding or trimming prompt tokens of varying lengths to make them consistent
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raise Exception("Multi-segment transcription not supported with batch audio input")
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start_tokens = [enc._special_tokens["<|startoftranscript|>"]]
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if model.is_multilingual:
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# TODO detect language
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language_token = enc._special_tokens["<|startoftranscript|>"] + 1 + tuple(LANGUAGES.keys()).index("en")
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start_tokens.append(language_token)
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start_tokens.append(enc._special_tokens["<|transcribe|>"])
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start_tokens.append(enc._special_tokens["<|notimestamps|>"])
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transcription_start_index = len(start_tokens)
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eot = enc._special_tokens["<|endoftext|>"]
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transcription_tokens = [np.array([], dtype=np.int32)] * log_spec.shape[0]
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for curr_frame in range(0, log_spec.shape[-1], FRAMES_PER_SEGMENT):
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encoded_audio = model.encoder.encode(Tensor(log_spec[:, :, curr_frame:curr_frame + FRAMES_PER_SEGMENT]))
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pos = 0
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curr_segment_tokens = np.tile(start_tokens, (log_spec.shape[0], 1))
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if curr_frame > 0:
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# pass the previously inferred tokens as 'prompt' - https://github.com/openai/whisper/discussions/117#discussioncomment-3727051
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prompt = np.concatenate((
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[enc._special_tokens["<|startofprev|>"]],
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transcription_tokens[0][-model.decoder.max_tokens_to_sample+1:],
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start_tokens))
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curr_segment_tokens = np.tile(prompt, (log_spec.shape[0], 1))
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transcription_start_index = len(curr_segment_tokens[0])
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for i in range(model.decoder.max_tokens_to_sample):
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out = model.decoder(Tensor(curr_segment_tokens if i == 0 else curr_segment_tokens[:, -1:]), pos, encoded_audio, streaming=curr_frame > 0)
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next_tokens = out[:, -1].argmax(axis=-1).numpy().astype(np.int32)
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next_tokens[curr_segment_tokens[:, -1] == eot] = eot
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curr_segment_tokens = np.concatenate((curr_segment_tokens, next_tokens.reshape(-1, 1)), axis=1)
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pos = curr_segment_tokens.shape[-1] - 1
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if DEBUG >= 1: print(i, list(map(lambda tokens: enc.decode(tokens), curr_segment_tokens)))
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if (curr_segment_tokens[:, -1] == eot).all():
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break
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for i, t in enumerate(curr_segment_tokens):
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eot_index = np.where(t == eot)[0]
|
|
eot_index = None if len(eot_index) == 0 else eot_index[0]
|
|
transcription_tokens[i] = np.concatenate((transcription_tokens[i], t[transcription_start_index:eot_index]))
|
|
|
|
transcriptions = list(map(lambda tokens: enc.decode(tokens).strip(), transcription_tokens))
|
|
return transcriptions[:N_audio] if N_audio > 1 else transcriptions[0]
|
|
|
|
CHUNK = 1600
|
|
RECORD_SECONDS = 10
|
|
|
|
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")
|
|
|
|
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), model.batch_size, truncate=True)
|
|
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()
|