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@ -78,6 +78,194 @@ pretrained_models = {
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},
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}
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}
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# ASR server connection process class
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class PaddleASRConnectionHanddler:
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def __init__(self, asr_engine):
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super().__init__()
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self.config = asr_engine.config
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self.model_config = asr_engine.executor.config
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self.asr_engine = asr_engine
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self.init()
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self.reset()
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def init(self):
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self.model_type = self.asr_engine.executor.model_type
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if "deepspeech2online" in self.model_type or "deepspeech2offline" in self.model_type:
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pass
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elif "conformer" in self.model_type or "transformer" in self.model_type or "wenetspeech" in self.model_type:
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self.sample_rate = self.asr_engine.executor.sample_rate
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# acoustic model
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self.model = self.asr_engine.executor.model
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# tokens to text
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self.text_feature = self.asr_engine.executor.text_feature
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# ctc decoding
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self.ctc_decode_config = self.asr_engine.executor.config.decode
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self.searcher = CTCPrefixBeamSearch(self.ctc_decode_config)
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# extract fbank
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self.preprocess_conf = self.model_config.preprocess_config
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self.preprocess_args = {"train": False}
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self.preprocessing = Transformation(self.preprocess_conf)
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self.win_length = self.preprocess_conf.process[0]['win_length']
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self.n_shift = self.preprocess_conf.process[0]['n_shift']
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def extract_feat(self, samples):
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if "deepspeech2online" in self.model_type:
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pass
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elif "conformer2online" in self.model_type:
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logger.info("Online ASR extract the feat")
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samples = np.frombuffer(samples, dtype=np.int16)
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assert samples.ndim == 1
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logger.info(f"This package receive {samples.shape[0]} pcm data")
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self.num_samples += samples.shape[0]
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# self.reamined_wav stores all the samples,
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# include the original remained_wav and this package samples
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if self.remained_wav is None:
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self.remained_wav = samples
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else:
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assert self.remained_wav.ndim == 1
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self.remained_wav = np.concatenate([self.remained_wav, samples])
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logger.info(
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f"The connection remain the audio samples: {self.remained_wav.shape}"
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)
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if len(self.remained_wav) < self.win_length:
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return 0
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# fbank
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x_chunk = self.preprocessing(self.remained_wav, **self.preprocess_args)
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x_chunk = paddle.to_tensor(
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x_chunk, dtype="float32").unsqueeze(axis=0)
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if self.cached_feat is None:
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self.cached_feat = x_chunk
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else:
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self.cached_feat = paddle.concat([self.cached_feat, x_chunk], axis=1)
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num_frames = x_chunk.shape[1]
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self.num_frames += num_frames
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self.remained_wav = self.remained_wav[self.n_shift * num_frames:]
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logger.info(
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f"process the audio feature success, the connection feat shape: {self.cached_feat.shape}"
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)
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logger.info(
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f"After extract feat, the connection remain the audio samples: {self.remained_wav.shape}"
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)
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# logger.info(f"accumulate samples: {self.num_samples}")
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def reset(self):
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self.subsampling_cache = None
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self.elayers_output_cache = None
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self.conformer_cnn_cache = None
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self.encoder_outs_ = None
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self.cached_feat = None
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self.remained_wav = None
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self.offset = 0
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self.num_samples = 0
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self.num_frames = 0
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self.global_frame_offset = 0
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self.result = []
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def decode(self, is_finished=False):
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if "deepspeech2online" in self.model_type:
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pass
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elif "conformer" in self.model_type or "transformer" in self.model_type:
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try:
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logger.info(
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f"we will use the transformer like model : {self.model_type}"
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)
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self.advance_decoding(is_finished)
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# self.update_result()
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# return self.result_transcripts[0]
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except Exception as e:
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logger.exception(e)
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else:
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raise Exception("invalid model name")
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def advance_decoding(self, is_finished=False):
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logger.info("start to decode with advanced_decoding method")
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cfg = self.ctc_decode_config
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decoding_chunk_size = cfg.decoding_chunk_size
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num_decoding_left_chunks = cfg.num_decoding_left_chunks
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assert decoding_chunk_size > 0
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subsampling = self.model.encoder.embed.subsampling_rate
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context = self.model.encoder.embed.right_context + 1
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stride = subsampling * decoding_chunk_size
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# decoding window for model
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decoding_window = (decoding_chunk_size - 1) * subsampling + context
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num_frames = self.cached_feat.shape[1]
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logger.info(f"Required decoding window {decoding_window} frames, and the connection has {num_frames} frames")
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# the cached feat must be larger decoding_window
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if num_frames < decoding_window and not is_finished:
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return None, None
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# logger.info("start to do model forward")
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# required_cache_size = decoding_chunk_size * num_decoding_left_chunks
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# outputs = []
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# # num_frames - context + 1 ensure that current frame can get context window
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# if is_finished:
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# # if get the finished chunk, we need process the last context
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# left_frames = context
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# else:
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# # we only process decoding_window frames for one chunk
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# left_frames = decoding_window
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# logger.info(f"")
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# end = None
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# for cur in range(0, num_frames - left_frames + 1, stride):
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# end = min(cur + decoding_window, num_frames)
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# print(f"cur: {cur}, end: {end}")
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# chunk_xs = self.cached_feat[:, cur:end, :]
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# (y, self.subsampling_cache, self.elayers_output_cache,
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# self.conformer_cnn_cache) = self.model.encoder.forward_chunk(
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# chunk_xs, self.offset, required_cache_size,
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# self.subsampling_cache, self.elayers_output_cache,
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# self.conformer_cnn_cache)
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# outputs.append(y)
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# update the offset
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# self.offset += y.shape[1]
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# self.cached_feat = self.cached_feat[end:]
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# ys = paddle.cat(outputs, 1)
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# masks = paddle.ones([1, ys.shape[1]], dtype=paddle.bool)
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# masks = masks.unsqueeze(1)
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# # get the ctc probs
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# ctc_probs = self.model.ctc.log_softmax(ys) # (1, maxlen, vocab_size)
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# ctc_probs = ctc_probs.squeeze(0)
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# # self.searcher.search(xs, ctc_probs, xs.place)
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# self.searcher.search(None, ctc_probs, self.cached_feat.place)
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# self.hyps = self.searcher.get_one_best_hyps()
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# ys for rescoring
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# return ys, masks
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def update_result(self):
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logger.info("update the final result")
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hyps = self.hyps
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self.result_transcripts = [
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self.text_feature.defeaturize(hyp) for hyp in hyps
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]
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self.result_tokenids = [hyp for hyp in hyps]
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def rescoring(self):
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pass
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class ASRServerExecutor(ASRExecutor):
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class ASRServerExecutor(ASRExecutor):
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def __init__(self):
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def __init__(self):
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@ -492,7 +680,7 @@ class ASRServerExecutor(ASRExecutor):
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if sample_rate != self.sample_rate:
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if sample_rate != self.sample_rate:
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logger.info(f"audio sample rate {sample_rate} is not match,"
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logger.info(f"audio sample rate {sample_rate} is not match,"
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"the model sample_rate is {self.sample_rate}")
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"the model sample_rate is {self.sample_rate}")
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logger.info("ASR Engine use the {self.model_type} to process")
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logger.info(f"ASR Engine use the {self.model_type} to process")
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logger.info("Create the preprocess instance")
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logger.info("Create the preprocess instance")
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preprocess_conf = self.config.preprocess_config
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preprocess_conf = self.config.preprocess_config
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preprocess_args = {"train": False}
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preprocess_args = {"train": False}
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