fix ds2 online edge bug, test=doc

paddlespeech/cli/asr/pretrained_models.py

@@ -88,6 +88,8 @@ model_alias = {
     "paddlespeech.s2t.models.ds2_online:DeepSpeech2ModelOnline",
     "conformer":
     "paddlespeech.s2t.models.u2:U2Model",
+    "conformer_online":
+    "paddlespeech.s2t.models.u2:U2Model",
     "transformer":
     "paddlespeech.s2t.models.u2:U2Model",
     "wenetspeech":

paddlespeech/server/engine/asr/online/asr_engine.py

@@ -130,9 +130,10 @@ class PaddleASRConnectionHanddler:
                 cfg.num_proc_bsearch)
             # frame window samples length and frame shift samples length
 
-            self.win_length = int(self.model_config.window_ms *
+            self.win_length = int(self.model_config.window_ms / 1000 *
                                   self.sample_rate)
-            self.n_shift = int(self.model_config.stride_ms * self.sample_rate)
+            self.n_shift = int(self.model_config.stride_ms / 1000 *
+                               self.sample_rate)
 
         elif "conformer" in self.model_type or "transformer" in self.model_type:
             # acoustic model
@@ -158,6 +159,11 @@ class PaddleASRConnectionHanddler:
             samples = np.frombuffer(samples, dtype=np.int16)
             assert samples.ndim == 1
 
+            # pcm16 -> pcm 32
+            # pcm2float will change the orignal samples, 
+            # so we shoule do pcm2float before concatenate
+            samples = pcm2float(samples)
+
             if self.remained_wav is None:
                 self.remained_wav = samples
             else:
@@ -167,11 +173,9 @@ class PaddleASRConnectionHanddler:
                 f"The connection remain the audio samples: {self.remained_wav.shape}"
             )
 
-            # pcm16 -> pcm 32
-            samples = pcm2float(self.remained_wav)
             # read audio
             speech_segment = SpeechSegment.from_pcm(
-                samples, self.sample_rate, transcript=" ")
+                self.remained_wav, self.sample_rate, transcript=" ")
             # audio augment
             self.collate_fn_test.augmentation.transform_audio(speech_segment)
 
@@ -474,6 +478,7 @@ class PaddleASRConnectionHanddler:
         self.hyps = self.searcher.get_one_best_hyps()
         assert self.cached_feat.shape[0] == 1
         assert end >= cached_feature_num
+
         self.cached_feat = self.cached_feat[0, end -
                                             cached_feature_num:, :].unsqueeze(0)
         assert len(
@@ -515,7 +520,6 @@ class PaddleASRConnectionHanddler:
             return
 
         # assert len(hyps) == beam_size
-        paddle.save(self.encoder_out, "encoder.out")
         hyp_list = []
         for hyp in hyps:
             hyp_content = hyp[0]
@@ -815,7 +819,7 @@ class ASRServerExecutor(ASRExecutor):
 
     def advanced_decoding(self, xs: paddle.Tensor, x_chunk_lens):
         logger.info("start to decode with advanced_decoding method")
-        encoder_out, encoder_mask = self.decode_forward(xs)
+        encoder_out, encoder_mask = self.encoder_forward(xs)
         ctc_probs = self.model.ctc.log_softmax(
             encoder_out)  # (1, maxlen, vocab_size)
         ctc_probs = ctc_probs.squeeze(0)
@@ -827,7 +831,7 @@ class ASRServerExecutor(ASRExecutor):
         if "attention_rescoring" in self.config.decode.decoding_method:
             self.rescoring(encoder_out, xs.place)
 
-    def decode_forward(self, xs):
+    def encoder_forward(self, xs):
         logger.info("get the model out from the feat")
         cfg = self.config.decode
         decoding_chunk_size = cfg.decoding_chunk_size