From c3442d9566f5a2011c95b0d2998958a1b5348564 Mon Sep 17 00:00:00 2001
From: shixian.shi <shixian.shi@alibaba-inc.com>
Date: 星期五, 12 一月 2024 18:04:42 +0800
Subject: [PATCH] update device

---
 funasr/models/paraformer_streaming/model.py |   97 ++++++++++++++++++++++++++++--------------------
 1 files changed, 57 insertions(+), 40 deletions(-)

diff --git a/funasr/models/paraformer_streaming/model.py b/funasr/models/paraformer_streaming/model.py
index 304c0f7..b736aa9 100644
--- a/funasr/models/paraformer_streaming/model.py
+++ b/funasr/models/paraformer_streaming/model.py
@@ -64,8 +64,8 @@
 		
 		super().__init__(*args, **kwargs)
 		
-		import pdb;
-		pdb.set_trace()
+		# import pdb;
+		# pdb.set_trace()
 		self.sampling_ratio = kwargs.get("sampling_ratio", 0.2)
 
 
@@ -375,11 +375,10 @@
 		
 		return pre_acoustic_embeds, pre_token_length, pre_alphas, pre_peak_index
 	
-	def calc_predictor_chunk(self, encoder_out, encoder_out_lens, cache=None):
-		
-		pre_acoustic_embeds, pre_token_length = \
-			self.predictor.forward_chunk(encoder_out, cache["encoder"])
-		return pre_acoustic_embeds, pre_token_length
+	def calc_predictor_chunk(self, encoder_out, encoder_out_lens, cache=None, **kwargs):
+		is_final = kwargs.get("is_final", False)
+
+		return self.predictor.forward_chunk(encoder_out, cache["encoder"], is_final=is_final)
 	
 	def cal_decoder_with_predictor(self, encoder_out, encoder_out_lens, sematic_embeds, ys_pad_lens):
 		decoder_outs = self.decoder(
@@ -416,7 +415,7 @@
 		            "chunk_size": chunk_size}
 		cache["decoder"] = cache_decoder
 		cache["frontend"] = {}
-		cache["prev_samples"] = []
+		cache["prev_samples"] = torch.empty(0)
 		
 		return cache
 	
@@ -432,12 +431,12 @@
 		speech.to(device=kwargs["device"]), speech_lengths.to(device=kwargs["device"])
 		
 		# Encoder
-		encoder_out, encoder_out_lens = self.encode_chunk(speech, speech_lengths, cache=cache)
+		encoder_out, encoder_out_lens = self.encode_chunk(speech, speech_lengths, cache=cache, is_final=kwargs.get("is_final", False))
 		if isinstance(encoder_out, tuple):
 			encoder_out = encoder_out[0]
 		
 		# predictor
-		predictor_outs = self.calc_predictor_chunk(encoder_out, encoder_out_lens, cache=cache)
+		predictor_outs = self.calc_predictor_chunk(encoder_out, encoder_out_lens, cache=cache, is_final=kwargs.get("is_final", False))
 		pre_acoustic_embeds, pre_token_length, alphas, pre_peak_index = predictor_outs[0], predictor_outs[1], \
 		                                                                predictor_outs[2], predictor_outs[3]
 		pre_token_length = pre_token_length.round().long()
@@ -476,10 +475,7 @@
 				)
 				nbest_hyps = [Hypothesis(yseq=yseq, score=score)]
 			for nbest_idx, hyp in enumerate(nbest_hyps):
-				ibest_writer = None
-				if ibest_writer is None and kwargs.get("output_dir") is not None:
-					writer = DatadirWriter(kwargs.get("output_dir"))
-					ibest_writer = writer[f"{nbest_idx + 1}best_recog"]
+				
 				# remove sos/eos and get results
 				last_pos = -1
 				if isinstance(hyp.yseq, list):
@@ -490,22 +486,15 @@
 				# remove blank symbol id, which is assumed to be 0
 				token_int = list(filter(lambda x: x != self.eos and x != self.sos and x != self.blank_id, token_int))
 				
-				if tokenizer is not None:
-					# Change integer-ids to tokens
-					token = tokenizer.ids2tokens(token_int)
-					text = tokenizer.tokens2text(token)
-					
-					text_postprocessed, _ = postprocess_utils.sentence_postprocess(token)
-					
-					result_i = {"key": key[i], "text": text_postprocessed}
-					
-					if ibest_writer is not None:
-						ibest_writer["token"][key[i]] = " ".join(token)
-						# ibest_writer["text"][key[i]] = text
-						ibest_writer["text"][key[i]] = text_postprocessed
-				else:
-					result_i = {"key": key[i], "token_int": token_int}
-				results.append(result_i)
+
+				# Change integer-ids to tokens
+				token = tokenizer.ids2tokens(token_int)
+				# text = tokenizer.tokens2text(token)
+				
+				result_i = token
+
+
+				results.extend(result_i)
 		
 		return results
 	
@@ -515,6 +504,7 @@
 	             key: list = None,
 	             tokenizer=None,
 	             frontend=None,
+	             cache: dict={},
 	             **kwargs,
 	             ):
 
@@ -526,38 +516,65 @@
 			self.init_beam_search(**kwargs)
 			self.nbest = kwargs.get("nbest", 1)
 		
-		cache = kwargs.get("cache", {})
+
 		if len(cache) == 0:
 			self.init_cache(cache, **kwargs)
 		
+		
 		meta_data = {}
 		chunk_size = kwargs.get("chunk_size", [0, 10, 5])
-		chunk_stride_samples = chunk_size[1] * 960  # 600ms
+		chunk_stride_samples = int(chunk_size[1] * 960)  # 600ms
 		
 		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)
+		cfg = {"is_final": kwargs.get("is_final", False)}
+		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,
+														cache=cfg,
+														)
+		_is_final = cfg["is_final"] # if data_in is a file or url, set is_final=True
+		
 		time2 = time.perf_counter()
 		meta_data["load_data"] = f"{time2 - time1:0.3f}"
 		assert len(audio_sample_list) == 1, "batch_size must be set 1"
 		
-		audio_sample = cache["prev_samples"] + audio_sample_list[0]
+		audio_sample = torch.cat((cache["prev_samples"], audio_sample_list[0]))
 		
-		n = len(audio_sample) // chunk_stride_samples
-		m = len(audio_sample) % chunk_stride_samples
+		n = int(len(audio_sample) // chunk_stride_samples + int(_is_final))
+		m = int(len(audio_sample) % chunk_stride_samples * (1-int(_is_final)))
+		tokens = []
 		for i in range(n):
+			kwargs["is_final"] = _is_final and i == n -1
 			audio_sample_i = audio_sample[i*chunk_stride_samples:(i+1)*chunk_stride_samples]
 
 			# extract fbank feats
 			speech, speech_lengths = extract_fbank([audio_sample_i], data_type=kwargs.get("data_type", "sound"),
-			                                       frontend=frontend, cache=cache["frontend"])
+			                                       frontend=frontend, cache=cache["frontend"], is_final=kwargs["is_final"])
 			time3 = time.perf_counter()
 			meta_data["extract_feat"] = f"{time3 - time2:0.3f}"
 			meta_data["batch_data_time"] = speech_lengths.sum().item() * frontend.frame_shift * frontend.lfr_n / 1000
 			
-			result_i = self.generate_chunk(speech, speech_lengths, **kwargs)
+			tokens_i = self.generate_chunk(speech, speech_lengths, key=key, tokenizer=tokenizer, cache=cache, frontend=frontend, **kwargs)
+			tokens.extend(tokens_i)
+			
+		text_postprocessed, _ = postprocess_utils.sentence_postprocess(tokens)
+		
+		result_i = {"key": key[0], "text": text_postprocessed}
+		result = [result_i]
+		
 		
 		cache["prev_samples"] = audio_sample[:-m]
+		if _is_final:
+			self.init_cache(cache, **kwargs)
+		
+		if kwargs.get("output_dir"):
+			writer = DatadirWriter(kwargs.get("output_dir"))
+			ibest_writer = writer[f"{1}best_recog"]
+			ibest_writer["token"][key[0]] = " ".join(tokens)
+			ibest_writer["text"][key[0]] = text_postprocessed
+		
+		return result, meta_data
 
 

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