Merge pull request #1945 from PaddlePaddle/asr_line

[server][asr] refactor asr streaming server and remove useless code

demos/audio_searching/src/operations/load.py

@@ -26,9 +26,8 @@ def get_audios(path):
     """
     supported_formats = [".wav", ".mp3", ".ogg", ".flac", ".m4a"]
     return [
-        item
-        for sublist in [[os.path.join(dir, file) for file in files]
-                        for dir, _, files in list(os.walk(path))]
+        item for sublist in [[os.path.join(dir, file) for file in files]
+                             for dir, _, files in list(os.walk(path))]
         for item in sublist if os.path.splitext(item)[1] in supported_formats
     ]
 

demos/streaming_asr_server/websocket_client.py

@@ -13,9 +13,7 @@
 # limitations under the License.
 #!/usr/bin/python
 # -*- coding: UTF-8 -*-
-
 # script for calc RTF: grep -rn RTF log.txt | awk '{print $NF}' | awk -F "=" '{sum += $NF} END {print "all time",sum, "audio num", NR,  "RTF", sum/NR}'
-
 import argparse
 import asyncio
 import codecs

docs/source/asr/PPASR_cn.md

@@ -92,5 +92,3 @@ server 的 demo： [streaming_asr_server](https://github.com/PaddlePaddle/Paddle
 ## 4. 快速开始
 
 关于如果使用 PP-ASR，可以看这里的 [install](https://github.com/PaddlePaddle/PaddleSpeech/blob/develop/docs/source/install_cn.md)，其中提供了 **简单**、**中等**、**困难** 三种安装方式。如果想体验 paddlespeech 的推理功能，可以用 **简单** 安装方式。
-
-

paddlespeech/cli/utils.py

@@ -24,11 +24,11 @@ from typing import Any
 from typing import Dict
 
 import paddle
+import paddleaudio
 import requests
 import yaml
 from paddle.framework import load
 
-import paddleaudio
 from . import download
 from .entry import commands
 try:

paddlespeech/s2t/io/sampler.py

@@ -51,7 +51,7 @@ def _batch_shuffle(indices, batch_size, epoch, clipped=False):
     """
     rng = np.random.RandomState(epoch)
     shift_len = rng.randint(0, batch_size - 1)
-    batch_indices = list(zip(*[iter(indices[shift_len:])] * batch_size))
+    batch_indices = list(zip(* [iter(indices[shift_len:])] * batch_size))
     rng.shuffle(batch_indices)
     batch_indices = [item for batch in batch_indices for item in batch]
     assert clipped is False

paddlespeech/server/engine/acs/__init__.py

@@ -0,0 +1,13 @@
+# Copyright (c) 2022 PaddlePaddle Authors. All Rights Reserved.
+#
+# Licensed under the Apache License, Version 2.0 (the "License");
+# you may not use this file except in compliance with the License.
+# You may obtain a copy of the License at
+#
+#     http://www.apache.org/licenses/LICENSE-2.0
+#
+# Unless required by applicable law or agreed to in writing, software
+# distributed under the License is distributed on an "AS IS" BASIS,
+# WITHOUT WARRANTIES OR CONDITIONS OF ANY KIND, either express or implied.
+# See the License for the specific language governing permissions and
+# limitations under the License.

paddlespeech/server/engine/acs/python/__init__.py

@@ -0,0 +1,13 @@
+# Copyright (c) 2022 PaddlePaddle Authors. All Rights Reserved.
+#
+# Licensed under the Apache License, Version 2.0 (the "License");
+# you may not use this file except in compliance with the License.
+# You may obtain a copy of the License at
+#
+#     http://www.apache.org/licenses/LICENSE-2.0
+#
+# Unless required by applicable law or agreed to in writing, software
+# distributed under the License is distributed on an "AS IS" BASIS,
+# WITHOUT WARRANTIES OR CONDITIONS OF ANY KIND, either express or implied.
+# See the License for the specific language governing permissions and
+# limitations under the License.

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

@@ -38,7 +38,7 @@ from paddlespeech.server.engine.base_engine import BaseEngine
 from paddlespeech.server.utils.audio_process import pcm2float
 from paddlespeech.server.utils.paddle_predictor import init_predictor
 
-__all__ = ['ASREngine']
+__all__ = ['PaddleASRConnectionHanddler', 'ASRServerExecutor', 'ASREngine']
 
 
 # ASR server connection process class
@@ -67,7 +67,7 @@ class PaddleASRConnectionHanddler:
         # tokens to text
         self.text_feature = self.asr_engine.executor.text_feature
 
-        if "deepspeech2online" in self.model_type or "deepspeech2offline" in self.model_type:
+        if "deepspeech2" in self.model_type:
             from paddlespeech.s2t.io.collator import SpeechCollator
             self.am_predictor = self.asr_engine.executor.am_predictor
 
@@ -89,8 +89,8 @@ class PaddleASRConnectionHanddler:
                 cfg.decoding_method, cfg.lang_model_path, cfg.alpha, cfg.beta,
                 cfg.beam_size, cfg.cutoff_prob, cfg.cutoff_top_n,
                 cfg.num_proc_bsearch)
-            # frame window samples length and frame shift samples length
 
+            # frame window and frame shift, in samples unit
             self.win_length = int(self.model_config.window_ms / 1000 *
                                   self.sample_rate)
             self.n_shift = int(self.model_config.stride_ms / 1000 *
@@ -109,16 +109,15 @@ class PaddleASRConnectionHanddler:
             self.preprocess_args = {"train": False}
             self.preprocessing = Transformation(self.preprocess_conf)
 
-            # frame window samples length and frame shift samples length
+            # frame window and frame shift, in samples unit
             self.win_length = self.preprocess_conf.process[0]['win_length']
             self.n_shift = self.preprocess_conf.process[0]['n_shift']
+        else:
+            raise ValueError(f"Not supported: {self.model_type}")
 
     def extract_feat(self, samples):
-
         # we compute the elapsed time of first char occuring 
         # and we record the start time at the first pcm sample arraving
-        # if self.first_char_occur_elapsed is not None:
-        #     self.first_char_occur_elapsed = time.time()
 
         if "deepspeech2online" in self.model_type:
             # self.reamined_wav stores all the samples, 
@@ -154,28 +153,27 @@ class PaddleASRConnectionHanddler:
                 spectrum = self.collate_fn_test._normalizer.apply(spectrum)
 
             # spectrum augment
-            audio = self.collate_fn_test.augmentation.transform_feature(
-                spectrum)
+            feat = self.collate_fn_test.augmentation.transform_feature(spectrum)
 
-            audio_len = audio.shape[0]
-            audio = paddle.to_tensor(audio, dtype='float32')
-            # audio_len = paddle.to_tensor(audio_len)
-            audio = paddle.unsqueeze(audio, axis=0)
+            # audio_len is frame num
+            frame_num = feat.shape[0]
+            feat = paddle.to_tensor(feat, dtype='float32')
+            feat = paddle.unsqueeze(feat, axis=0)
 
             if self.cached_feat is None:
-                self.cached_feat = audio
+                self.cached_feat = feat
             else:
-                assert (len(audio.shape) == 3)
+                assert (len(feat.shape) == 3)
                 assert (len(self.cached_feat.shape) == 3)
                 self.cached_feat = paddle.concat(
-                    [self.cached_feat, audio], axis=1)
+                    [self.cached_feat, feat], axis=1)
 
                 # set the feat device
             if self.device is None:
                 self.device = self.cached_feat.place
 
-            self.num_frames += audio_len
-            self.remained_wav = self.remained_wav[self.n_shift * audio_len:]
+            self.num_frames += frame_num
+            self.remained_wav = self.remained_wav[self.n_shift * frame_num:]
 
             logger.info(
                 f"process the audio feature success, the connection feat shape: {self.cached_feat.shape}"
@@ -183,25 +181,30 @@ class PaddleASRConnectionHanddler:
             logger.info(
                 f"After extract feat, the connection remain the audio samples: {self.remained_wav.shape}"
             )
+
         elif "conformer_online" in self.model_type:
             logger.info("Online ASR extract the feat")
             samples = np.frombuffer(samples, dtype=np.int16)
             assert samples.ndim == 1
 
-            logger.info(f"This package receive {samples.shape[0]} pcm data")
             self.num_samples += samples.shape[0]
+            logger.info(
+                f"This package receive {samples.shape[0]} pcm data. Global samples:{self.num_samples}"
+            )
 
             # self.reamined_wav stores all the samples, 
             # include the original remained_wav and this package samples
             if self.remained_wav is None:
                 self.remained_wav = samples
             else:
-                assert self.remained_wav.ndim == 1
+                assert self.remained_wav.ndim == 1  # (T,)
                 self.remained_wav = np.concatenate([self.remained_wav, samples])
             logger.info(
-                f"The connection remain the audio samples: {self.remained_wav.shape}"
+                f"The concatenation of remain and now audio samples length is: {self.remained_wav.shape}"
             )
+
             if len(self.remained_wav) < self.win_length:
+                # samples not enough for feature window
                 return 0
 
             # fbank
@@ -209,11 +212,13 @@ class PaddleASRConnectionHanddler:
                                          **self.preprocess_args)
             x_chunk = paddle.to_tensor(
                 x_chunk, dtype="float32").unsqueeze(axis=0)
+
+            # feature cache
             if self.cached_feat is None:
                 self.cached_feat = x_chunk
             else:
-                assert (len(x_chunk.shape) == 3)
-                assert (len(self.cached_feat.shape) == 3)
+                assert (len(x_chunk.shape) == 3)  # (B,T,D)
+                assert (len(self.cached_feat.shape) == 3)  # (B,T,D)
                 self.cached_feat = paddle.concat(
                     [self.cached_feat, x_chunk], axis=1)
 
@@ -221,20 +226,28 @@ class PaddleASRConnectionHanddler:
             if self.device is None:
                 self.device = self.cached_feat.place
 
+            # cur frame step
             num_frames = x_chunk.shape[1]
+
+            # global frame step
             self.num_frames += num_frames
+
+            # update remained wav
             self.remained_wav = self.remained_wav[self.n_shift * num_frames:]
 
             logger.info(
-                f"process the audio feature success, the connection feat shape: {self.cached_feat.shape}"
+                f"process the audio feature success, the cached feat shape: {self.cached_feat.shape}"
             )
             logger.info(
-                f"After extract feat, the connection remain the audio samples: {self.remained_wav.shape}"
+                f"After extract feat, the cached remain the audio samples: {self.remained_wav.shape}"
             )
-            # logger.info(f"accumulate samples: {self.num_samples}")       
+            logger.info(f"global samples: {self.num_samples}")
+            logger.info(f"global frames: {self.num_frames}")
+        else:
+            raise ValueError(f"not supported: {self.model_type}")
 
     def reset(self):
-        if "deepspeech2online" in self.model_type or "deepspeech2offline" in self.model_type:
+        if "deepspeech2" in self.model_type:
             # for deepspeech2 
             self.chunk_state_h_box = copy.deepcopy(
                 self.asr_engine.executor.chunk_state_h_box)
@@ -242,35 +255,61 @@ class PaddleASRConnectionHanddler:
                 self.asr_engine.executor.chunk_state_c_box)
             self.decoder.reset_decoder(batch_size=1)
 
-        # for conformer online
+        self.device = None
+
+        ## common
+
+        # global sample and frame step
+        self.num_samples = 0
+        self.num_frames = 0
+
+        # cache for audio and feat
+        self.remained_wav = None
+        self.cached_feat = None
+
+        # partial/ending decoding results
+        self.result_transcripts = ['']
+
+        ## conformer
+
+        # cache for conformer online 
         self.subsampling_cache = None
         self.elayers_output_cache = None
         self.conformer_cnn_cache = None
         self.encoder_out = None
-        self.cached_feat = None
-        self.remained_wav = None
-        self.offset = 0
-        self.num_samples = 0
-        self.device = None
+        # conformer decoding state
+        self.chunk_num = 0  # globa decoding chunk num
+        self.offset = 0  # global offset in decoding frame unit
         self.hyps = []
-        self.num_frames = 0
-        self.chunk_num = 0
-        self.global_frame_offset = 0
-        self.result_transcripts = ['']
+
+        # token timestamp result
         self.word_time_stamp = []
+
+        # one best timestamp viterbi prob is large.
         self.time_stamp = []
-        self.first_char_occur_elapsed = None
 
     def decode(self, is_finished=False):
+        """advance decoding
+
+        Args:
+            is_finished (bool, optional): Is last frame or not. Defaults to False.
+
+        Raises:
+            Exception: when not support model.
+
+        Returns:
+            None: nothing
+        """
         if "deepspeech2online" in self.model_type:
-            # x_chunk 是特征数据
-            decoding_chunk_size = 1  # decoding_chunk_size=1 in deepspeech2 model
-            context = 7  # context=7 in deepspeech2 model
-            subsampling = 4  # subsampling=4 in deepspeech2 model
-            stride = subsampling * decoding_chunk_size
+            decoding_chunk_size = 1  # decoding chunk size = 1. int decoding frame unit
+            context = 7  # context=7, in audio frame unit
+            subsampling = 4  # subsampling=4, in audio frame unit
+
             cached_feature_num = context - subsampling
-            # decoding window for model
+            # decoding window for model, in audio frame unit
             decoding_window = (decoding_chunk_size - 1) * subsampling + context
+            # decoding stride for model, in audio frame unit
+            stride = subsampling * decoding_chunk_size
 
             if self.cached_feat is None:
                 logger.info("no audio feat, please input more pcm data")
@@ -280,6 +319,7 @@ class PaddleASRConnectionHanddler:
             logger.info(
                 f"Required decoding window {decoding_window} frames, and the connection has {num_frames} frames"
             )
+
             # the cached feat must be larger decoding_window
             if num_frames < decoding_window and not is_finished:
                 logger.info(
@@ -293,6 +333,7 @@ class PaddleASRConnectionHanddler:
                     "flast {num_frames} is less than context {context} frames, and we cannot do model forward"
                 )
                 return None, None
+
             logger.info("start to do model forward")
             # num_frames - context + 1 ensure that current frame can get context window
             if is_finished:
@@ -302,6 +343,7 @@ class PaddleASRConnectionHanddler:
                 # we only process decoding_window frames for one chunk
                 left_frames = decoding_window
 
+            end = None
             for cur in range(0, num_frames - left_frames + 1, stride):
                 end = min(cur + decoding_window, num_frames)
                 # extract the audio
@@ -311,7 +353,9 @@ class PaddleASRConnectionHanddler:
 
             self.result_transcripts = [trans_best]
 
+            # update feat cache
             self.cached_feat = self.cached_feat[:, end - cached_feature_num:, :]
+
             # return trans_best[0]            
         elif "conformer" in self.model_type or "transformer" in self.model_type:
             try:
@@ -328,7 +372,16 @@ class PaddleASRConnectionHanddler:
 
     @paddle.no_grad()
     def decode_one_chunk(self, x_chunk, x_chunk_lens):
-        logger.info("start to decoce one chunk with deepspeech2 model")
+        """forward one chunk frames
+
+        Args:
+            x_chunk (np.ndarray): (B,T,D), audio frames.
+            x_chunk_lens ([type]): (B,), audio frame lens
+
+        Returns:
+            logprob: poster probability.
+        """
+        logger.info("start to decoce one chunk for deepspeech2")
         input_names = self.am_predictor.get_input_names()
         audio_handle = self.am_predictor.get_input_handle(input_names[0])
         audio_len_handle = self.am_predictor.get_input_handle(input_names[1])
@@ -365,24 +418,32 @@ class PaddleASRConnectionHanddler:
 
         self.decoder.next(output_chunk_probs, output_chunk_lens)
         trans_best, trans_beam = self.decoder.decode()
-        logger.info(f"decode one best result: {trans_best[0]}")
+        logger.info(f"decode one best result for deepspeech2: {trans_best[0]}")
         return trans_best[0]
 
     @paddle.no_grad()
     def advance_decoding(self, is_finished=False):
-        logger.info("start to decode with advanced_decoding method")
+        logger.info(
+            "Conformer/Transformer: start to decode with advanced_decoding method"
+        )
         cfg = self.ctc_decode_config
+
+        # cur chunk size, in decoding frame unit
         decoding_chunk_size = cfg.decoding_chunk_size
+        # using num of history chunks
         num_decoding_left_chunks = cfg.num_decoding_left_chunks
-
         assert decoding_chunk_size > 0
+
         subsampling = self.model.encoder.embed.subsampling_rate
         context = self.model.encoder.embed.right_context + 1
-        stride = subsampling * decoding_chunk_size
-        cached_feature_num = context - subsampling  # processed chunk feature cached for next chunk
 
-        # decoding window for model
+        # processed chunk feature cached for next chunk
+        cached_feature_num = context - subsampling
+        # decoding stride, in audio frame unit
+        stride = subsampling * decoding_chunk_size
+        # decoding window, in audio frame unit
         decoding_window = (decoding_chunk_size - 1) * subsampling + context
+
         if self.cached_feat is None:
             logger.info("no audio feat, please input more pcm data")
             return
@@ -407,6 +468,7 @@ class PaddleASRConnectionHanddler:
             return None, None
 
         logger.info("start to do model forward")
+        # hist of chunks, in deocding frame unit
         required_cache_size = decoding_chunk_size * num_decoding_left_chunks
         outputs = []
 
@@ -423,8 +485,11 @@ class PaddleASRConnectionHanddler:
         for cur in range(0, num_frames - left_frames + 1, stride):
             end = min(cur + decoding_window, num_frames)
 
+            # global chunk_num
             self.chunk_num += 1
+            # cur chunk
             chunk_xs = self.cached_feat[:, cur:end, :]
+            # forward chunk
             (y, self.subsampling_cache, self.elayers_output_cache,
              self.conformer_cnn_cache) = self.model.encoder.forward_chunk(
                  chunk_xs, self.offset, required_cache_size,
@@ -432,7 +497,7 @@ class PaddleASRConnectionHanddler:
                  self.conformer_cnn_cache)
             outputs.append(y)
 
-            # update the offset
+            # update the global offset, in decoding frame unit
             self.offset += y.shape[1]
 
         ys = paddle.cat(outputs, 1)
@@ -445,12 +510,15 @@ class PaddleASRConnectionHanddler:
         ctc_probs = self.model.ctc.log_softmax(ys)  # (1, maxlen, vocab_size)
         ctc_probs = ctc_probs.squeeze(0)
 
+        # advance decoding
         self.searcher.search(ctc_probs, self.cached_feat.place)
-
+        # get one best hyps
         self.hyps = self.searcher.get_one_best_hyps()
+
         assert self.cached_feat.shape[0] == 1
         assert end >= cached_feature_num
 
+        # advance cache of feat
         self.cached_feat = self.cached_feat[0, end -
                                             cached_feature_num:, :].unsqueeze(0)
         assert len(
@@ -462,50 +530,81 @@ class PaddleASRConnectionHanddler:
         )
 
     def update_result(self):
+        """Conformer/Transformer hyps to result.
+        """
         logger.info("update the final result")
         hyps = self.hyps
+
+        # output results and tokenids
         self.result_transcripts = [
             self.text_feature.defeaturize(hyp) for hyp in hyps
         ]
         self.result_tokenids = [hyp for hyp in hyps]
 
     def get_result(self):
+        """return partial/ending asr result.
+
+        Returns:
+            str: one best result of partial/ending.
+        """
         if len(self.result_transcripts) > 0:
             return self.result_transcripts[0]
         else:
             return ''
 
     def get_word_time_stamp(self):
+        """return token timestamp result.
+
+        Returns:
+            list: List of ('w':token, 'bg':time, 'ed':time)
+        """
         return self.word_time_stamp
 
     @paddle.no_grad()
     def rescoring(self):
-        if "deepspeech2online" in self.model_type or "deepspeech2offline" in self.model_type:
+        """Second-Pass Decoding, 
+        only for conformer and transformer model.
+        """
+        if "deepspeech2" in self.model_type:
+            logger.info("deepspeech2 not support rescoring decoding.")
             return
 
-        logger.info("rescoring the final result")
         if "attention_rescoring" != self.ctc_decode_config.decoding_method:
+            logger.info(
+                f"decoding method not match: {self.ctc_decode_config.decoding_method}, need attention_rescoring"
+            )
             return
 
+        logger.info("rescoring the final result")
+
+        # last decoding for last audio
         self.searcher.finalize_search()
+        # update beam search results
         self.update_result()
 
         beam_size = self.ctc_decode_config.beam_size
         hyps = self.searcher.get_hyps()
         if hyps is None or len(hyps) == 0:
+            logger.info("No Hyps!")
             return
 
+        # rescore by decoder post probability
+
         # assert len(hyps) == beam_size
+        # list of Tensor
         hyp_list = []
         for hyp in hyps:
             hyp_content = hyp[0]
             # Prevent the hyp is empty
             if len(hyp_content) == 0:
                 hyp_content = (self.model.ctc.blank_id, )
+
             hyp_content = paddle.to_tensor(
                 hyp_content, place=self.device, dtype=paddle.long)
             hyp_list.append(hyp_content)
-        hyps_pad = pad_sequence(hyp_list, True, self.model.ignore_id)
+
+        hyps_pad = pad_sequence(
+            hyp_list, batch_first=True, padding_value=self.model.ignore_id)
         hyps_lens = paddle.to_tensor(
             [len(hyp[0]) for hyp in hyps], place=self.device,
             dtype=paddle.long)  # (beam_size,)
@@ -531,10 +630,12 @@ class PaddleASRConnectionHanddler:
             score = 0.0
             for j, w in enumerate(hyp[0]):
                 score += decoder_out[i][j][w]
+
             # last decoder output token is `eos`, for laste decoder input token.
             score += decoder_out[i][len(hyp[0])][self.model.eos]
             # add ctc score (which in ln domain)
             score += hyp[1] * self.ctc_decode_config.ctc_weight
+
             if score > best_score:
                 best_score = score
                 best_index = i
@@ -542,43 +643,52 @@ class PaddleASRConnectionHanddler:
         # update the one best result
         # hyps stored the beam results and each fields is:
 
-        logger.info(f"best index: {best_index}")
+        logger.info(f"best hyp index: {best_index}")
         # logger.info(f'best result: {hyps[best_index]}')
         # the field of the hyps is:
+        ## asr results
         # hyps[0][0]: the sentence word-id in the vocab with a tuple
         # hyps[0][1]: the sentence decoding probability with all paths
+        ## timestamp 
         # hyps[0][2]: viterbi_blank ending probability
-        # hyps[0][3]: viterbi_non_blank probability
+        # hyps[0][3]: viterbi_non_blank dending probability
         # hyps[0][4]: current_token_prob,
-        # hyps[0][5]: times_viterbi_blank, 
-        # hyps[0][6]: times_titerbi_non_blank 
+        # hyps[0][5]: times_viterbi_blank ending timestamp, 
+        # hyps[0][6]: times_titerbi_non_blank encding timestamp.
         self.hyps = [hyps[best_index][0]]
+        logger.info(f"best hyp ids: {self.hyps}")
 
         # update the hyps time stamp
         self.time_stamp = hyps[best_index][5] if hyps[best_index][2] > hyps[
             best_index][3] else hyps[best_index][6]
         logger.info(f"time stamp: {self.time_stamp}")
 
+        # update one best result
         self.update_result()
 
         # update each word start and end time stamp
-        frame_shift_in_ms = self.model.encoder.embed.subsampling_rate * self.n_shift / self.sample_rate
-        logger.info(f"frame shift ms: {frame_shift_in_ms}")
+        # decoding frame to audio frame
+        frame_shift = self.model.encoder.embed.subsampling_rate
+        frame_shift_in_sec = frame_shift * (self.n_shift / self.sample_rate)
+        logger.info(f"frame shift sec: {frame_shift_in_sec}")
+
         word_time_stamp = []
         for idx, _ in enumerate(self.time_stamp):
             start = (self.time_stamp[idx - 1] + self.time_stamp[idx]
                      ) / 2.0 if idx > 0 else 0
-            start = start * frame_shift_in_ms
+            start = start * frame_shift_in_sec
 
             end = (self.time_stamp[idx] + self.time_stamp[idx + 1]
                    ) / 2.0 if idx < len(self.time_stamp) - 1 else self.offset
-            end = end * frame_shift_in_ms
+
+            end = end * frame_shift_in_sec
             word_time_stamp.append({
                 "w": self.result_transcripts[0][idx],
                 "bg": start,
                 "ed": end
             })
-            # logger.info(f"{self.result_transcripts[0][idx]}, start: {start}, end: {end}")
+            # logger.info(f"{word_time_stamp[-1]}")
+
         self.word_time_stamp = word_time_stamp
         logger.info(f"word time stamp: {self.word_time_stamp}")
 
@@ -610,6 +720,7 @@ class ASRServerExecutor(ASRExecutor):
         self.sample_rate = sample_rate
         sample_rate_str = '16k' if sample_rate == 16000 else '8k'
         tag = model_type + '-' + lang + '-' + sample_rate_str
+
         if cfg_path is None or am_model is None or am_params is None:
             logger.info(f"Load the pretrained model, tag = {tag}")
             res_path = self._get_pretrained_path(tag)  # wenetspeech_zh
@@ -639,7 +750,7 @@ class ASRServerExecutor(ASRExecutor):
         self.config.merge_from_file(self.cfg_path)
 
         with UpdateConfig(self.config):
-            if "deepspeech2online" in model_type or "deepspeech2offline" in model_type:
+            if "deepspeech2" in model_type:
                 from paddlespeech.s2t.io.collator import SpeechCollator
                 self.vocab = self.config.vocab_filepath
                 self.config.decode.lang_model_path = os.path.join(
@@ -655,6 +766,7 @@ class ASRServerExecutor(ASRExecutor):
                 self.download_lm(
                     lm_url,
                     os.path.dirname(self.config.decode.lang_model_path), lm_md5)
+
             elif "conformer" in model_type or "transformer" in model_type:
                 logger.info("start to create the stream conformer asr engine")
                 if self.config.spm_model_prefix:
@@ -682,7 +794,8 @@ class ASRServerExecutor(ASRExecutor):
                 ], f"we only support ctc_prefix_beam_search and attention_rescoring dedoding method, current decoding method is {self.config.decode.decoding_method}"
             else:
                 raise Exception("wrong type")
-        if "deepspeech2online" in model_type or "deepspeech2offline" in model_type:
+
+        if "deepspeech2" in model_type:
             # AM predictor
             logger.info("ASR engine start to init the am predictor")
             self.am_predictor_conf = am_predictor_conf
@@ -719,6 +832,7 @@ class ASRServerExecutor(ASRExecutor):
             self.chunk_state_c_box = np.zeros(
                 (self.config.num_rnn_layers, 1, self.config.rnn_layer_size),
                 dtype=float32)
+
         elif "conformer" in model_type or "transformer" in model_type:
             model_name = model_type[:model_type.rindex(
                 '_')]  # model_type: {model_name}_{dataset}
@@ -737,277 +851,14 @@ class ASRServerExecutor(ASRExecutor):
             # update the ctc decoding
             self.searcher = CTCPrefixBeamSearch(self.config.decode)
             self.transformer_decode_reset()
-            
-        return True
-
-    def reset_decoder_and_chunk(self):
-        """reset decoder and chunk state for an new audio
-        """
-        if "deepspeech2online" in self.model_type or "deepspeech2offline" in self.model_type:
-            self.decoder.reset_decoder(batch_size=1)
-            # init state box, for new audio request
-            self.chunk_state_h_box = np.zeros(
-                (self.config.num_rnn_layers, 1, self.config.rnn_layer_size),
-                dtype=float32)
-            self.chunk_state_c_box = np.zeros(
-                (self.config.num_rnn_layers, 1, self.config.rnn_layer_size),
-                dtype=float32)
-        elif "conformer" in self.model_type or "transformer" in self.model_type:
-            self.transformer_decode_reset()
-
-    def decode_one_chunk(self, x_chunk, x_chunk_lens, model_type: str):
-        """decode one chunk
-
-        Args:
-            x_chunk (numpy.array): shape[B, T, D]
-            x_chunk_lens (numpy.array): shape[B]
-            model_type (str): online model type
-
-        Returns:
-            str: one best result
-        """
-        logger.info("start to decoce chunk by chunk")
-        if "deepspeech2online" in model_type:
-            input_names = self.am_predictor.get_input_names()
-            audio_handle = self.am_predictor.get_input_handle(input_names[0])
-            audio_len_handle = self.am_predictor.get_input_handle(
-                input_names[1])
-            h_box_handle = self.am_predictor.get_input_handle(input_names[2])
-            c_box_handle = self.am_predictor.get_input_handle(input_names[3])
-
-            audio_handle.reshape(x_chunk.shape)
-            audio_handle.copy_from_cpu(x_chunk)
-
-            audio_len_handle.reshape(x_chunk_lens.shape)
-            audio_len_handle.copy_from_cpu(x_chunk_lens)
-
-            h_box_handle.reshape(self.chunk_state_h_box.shape)
-            h_box_handle.copy_from_cpu(self.chunk_state_h_box)
-
-            c_box_handle.reshape(self.chunk_state_c_box.shape)
-            c_box_handle.copy_from_cpu(self.chunk_state_c_box)
-
-            output_names = self.am_predictor.get_output_names()
-            output_handle = self.am_predictor.get_output_handle(output_names[0])
-            output_lens_handle = self.am_predictor.get_output_handle(
-                output_names[1])
-            output_state_h_handle = self.am_predictor.get_output_handle(
-                output_names[2])
-            output_state_c_handle = self.am_predictor.get_output_handle(
-                output_names[3])
-
-            self.am_predictor.run()
-
-            output_chunk_probs = output_handle.copy_to_cpu()
-            output_chunk_lens = output_lens_handle.copy_to_cpu()
-            self.chunk_state_h_box = output_state_h_handle.copy_to_cpu()
-            self.chunk_state_c_box = output_state_c_handle.copy_to_cpu()
-
-            self.decoder.next(output_chunk_probs, output_chunk_lens)
-            trans_best, trans_beam = self.decoder.decode()
-            logger.info(f"decode one best result: {trans_best[0]}")
-            return trans_best[0]
-
-        elif "conformer" in model_type or "transformer" in model_type:
-            try:
-                logger.info(
-                    f"we will use the transformer like model : {self.model_type}"
-                )
-                self.advanced_decoding(x_chunk, x_chunk_lens)
-                self.update_result()
-
-                return self.result_transcripts[0]
-            except Exception as e:
-                logger.exception(e)
         else:
-            raise Exception("invalid model name")
+            raise ValueError(f"Not support: {model_type}")
 
-    def advanced_decoding(self, xs: paddle.Tensor, x_chunk_lens):
-        logger.info("start to decode with advanced_decoding method")
-        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)
-        self.searcher.search(ctc_probs, xs.place)
-        # update the one best result
-        self.hyps = self.searcher.get_one_best_hyps()
-
-        # now we supprot ctc_prefix_beam_search and attention_rescoring
-        if "attention_rescoring" in self.config.decode.decoding_method:
-            self.rescoring(encoder_out, xs.place)
-
-    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
-        num_decoding_left_chunks = cfg.num_decoding_left_chunks
-
-        assert decoding_chunk_size > 0
-        subsampling = self.model.encoder.embed.subsampling_rate
-        context = self.model.encoder.embed.right_context + 1
-        stride = subsampling * decoding_chunk_size
-
-        # decoding window for model
-        decoding_window = (decoding_chunk_size - 1) * subsampling + context
-        num_frames = xs.shape[1]
-        required_cache_size = decoding_chunk_size * num_decoding_left_chunks
-
-        logger.info("start to do model forward")
-        outputs = []
-
-        # num_frames - context + 1 ensure that current frame can get context window
-        for cur in range(0, num_frames - context + 1, stride):
-            end = min(cur + decoding_window, num_frames)
-            chunk_xs = xs[:, cur:end, :]
-            (y, self.subsampling_cache, self.elayers_output_cache,
-             self.conformer_cnn_cache) = self.model.encoder.forward_chunk(
-                 chunk_xs, self.offset, required_cache_size,
-                 self.subsampling_cache, self.elayers_output_cache,
-                 self.conformer_cnn_cache)
-            outputs.append(y)
-            self.offset += y.shape[1]
-
-        ys = paddle.cat(outputs, 1)
-        masks = paddle.ones([1, ys.shape[1]], dtype=paddle.bool)
-        masks = masks.unsqueeze(1)
-        return ys, masks
-
-    def rescoring(self, encoder_out, device):
-        logger.info("start to rescoring the hyps")
-        beam_size = self.config.decode.beam_size
-        hyps = self.searcher.get_hyps()
-        assert len(hyps) == beam_size
-
-        hyp_list = []
-        for hyp in hyps:
-            hyp_content = hyp[0]
-            # Prevent the hyp is empty
-            if len(hyp_content) == 0:
-                hyp_content = (self.model.ctc.blank_id, )
-            hyp_content = paddle.to_tensor(
-                hyp_content, place=device, dtype=paddle.long)
-            hyp_list.append(hyp_content)
-        hyps_pad = pad_sequence(hyp_list, True, self.model.ignore_id)
-        hyps_lens = paddle.to_tensor(
-            [len(hyp[0]) for hyp in hyps], place=device,
-            dtype=paddle.long)  # (beam_size,)
-        hyps_pad, _ = add_sos_eos(hyps_pad, self.model.sos, self.model.eos,
-                                  self.model.ignore_id)
-        hyps_lens = hyps_lens + 1  # Add <sos> at begining
-
-        encoder_out = encoder_out.repeat(beam_size, 1, 1)
-        encoder_mask = paddle.ones(
-            (beam_size, 1, encoder_out.shape[1]), dtype=paddle.bool)
-        decoder_out, _ = self.model.decoder(
-            encoder_out, encoder_mask, hyps_pad,
-            hyps_lens)  # (beam_size, max_hyps_len, vocab_size)
-        # ctc score in ln domain
-        decoder_out = paddle.nn.functional.log_softmax(decoder_out, axis=-1)
-        decoder_out = decoder_out.numpy()
-
-        # Only use decoder score for rescoring
-        best_score = -float('inf')
-        best_index = 0
-        # hyps is List[(Text=List[int], Score=float)], len(hyps)=beam_size
-        for i, hyp in enumerate(hyps):
-            score = 0.0
-            for j, w in enumerate(hyp[0]):
-                score += decoder_out[i][j][w]
-            # last decoder output token is `eos`, for laste decoder input token.
-            score += decoder_out[i][len(hyp[0])][self.model.eos]
-            # add ctc score (which in ln domain)
-            score += hyp[1] * self.config.decode.ctc_weight
-            if score > best_score:
-                best_score = score
-                best_index = i
-
-        # update the one best result
-        self.hyps = [hyps[best_index][0]]
-        return hyps[best_index][0]
-
-    def transformer_decode_reset(self):
-        self.subsampling_cache = None
-        self.elayers_output_cache = None
-        self.conformer_cnn_cache = None
-        self.offset = 0
-        # decoding reset
-        self.searcher.reset()
-
-    def update_result(self):
-        logger.info("update the final result")
-        hyps = self.hyps
-        self.result_transcripts = [
-            self.text_feature.defeaturize(hyp) for hyp in hyps
-        ]
-        self.result_tokenids = [hyp for hyp in hyps]
-
-    def extract_feat(self, samples, sample_rate):
-        """extract feat
-
-        Args:
-            samples (numpy.array): numpy.float32
-            sample_rate (int): sample rate
-
-        Returns:
-            x_chunk (numpy.array): shape[B, T, D]
-            x_chunk_lens (numpy.array): shape[B]
-        """
-
-        if "deepspeech2online" in self.model_type:
-            # pcm16 -> pcm 32
-            samples = pcm2float(samples)
-            # read audio
-            speech_segment = SpeechSegment.from_pcm(
-                samples, sample_rate, transcript=" ")
-            # audio augment
-            self.collate_fn_test.augmentation.transform_audio(speech_segment)
-
-            # extract speech feature
-            spectrum, transcript_part = self.collate_fn_test._speech_featurizer.featurize(
-                speech_segment, self.collate_fn_test.keep_transcription_text)
-            # CMVN spectrum
-            if self.collate_fn_test._normalizer:
-                spectrum = self.collate_fn_test._normalizer.apply(spectrum)
-
-            # spectrum augment
-            audio = self.collate_fn_test.augmentation.transform_feature(
-                spectrum)
-
-            audio_len = audio.shape[0]
-            audio = paddle.to_tensor(audio, dtype='float32')
-            # audio_len = paddle.to_tensor(audio_len)
-            audio = paddle.unsqueeze(audio, axis=0)
-
-            x_chunk = audio.numpy()
-            x_chunk_lens = np.array([audio_len])
-
-            return x_chunk, x_chunk_lens
-        elif "conformer_online" in self.model_type:
-
-            if sample_rate != self.sample_rate:
-                logger.info(f"audio sample rate {sample_rate} is not match,"
-                            "the model sample_rate is {self.sample_rate}")
-            logger.info(f"ASR Engine use the {self.model_type} to process")
-            logger.info("Create the preprocess instance")
-            preprocess_conf = self.config.preprocess_config
-            preprocess_args = {"train": False}
-            preprocessing = Transformation(preprocess_conf)
-
-            logger.info("Read the audio file")
-            logger.info(f"audio shape: {samples.shape}")
-            # fbank
-            x_chunk = preprocessing(samples, **preprocess_args)
-            x_chunk_lens = paddle.to_tensor(x_chunk.shape[0])
-            x_chunk = paddle.to_tensor(
-                x_chunk, dtype="float32").unsqueeze(axis=0)
-            logger.info(
-                f"process the audio feature success, feat shape: {x_chunk.shape}"
-            )
-            return x_chunk, x_chunk_lens
+        return True
 
 
 class ASREngine(BaseEngine):
-    """ASR server engine
+    """ASR server resource
 
     Args:
         metaclass: Defaults to Singleton.
@@ -1015,7 +866,7 @@ class ASREngine(BaseEngine):
 
     def __init__(self):
         super(ASREngine, self).__init__()
-        logger.info("create the online asr engine instance")
+        logger.info("create the online asr engine resource instance")
 
     def init(self, config: dict) -> bool:
         """init engine resource
@@ -1026,17 +877,12 @@ class ASREngine(BaseEngine):
         Returns:
             bool: init failed or success
         """
-        self.input = None
-        self.output = ""
-        self.executor = ASRServerExecutor()
         self.config = config
+        self.executor = ASRServerExecutor()
+
         try:
-            if self.config.get("device", None):
-                self.device = self.config.device
-            else:
-                self.device = paddle.get_device()
-            logger.info(f"paddlespeech_server set the device: {self.device}")
-            paddle.set_device(self.device)
+            default_dev = paddle.get_device()
+            paddle.set_device(self.config.get("device", default_dev))
         except BaseException as e:
             logger.error(
                 f"Set device failed, please check if device '{self.device}' is already used and the parameter 'device' in the yaml file"
@@ -1045,6 +891,8 @@ class ASREngine(BaseEngine):
                 "If all GPU or XPU is used, you can set the server to 'cpu'")
             sys.exit(-1)
 
+        logger.info(f"paddlespeech_server set the device: {self.device}")
+
         if not self.executor._init_from_path(
                 model_type=self.config.model_type,
                 am_model=self.config.am_model,
@@ -1062,42 +910,11 @@ class ASREngine(BaseEngine):
         logger.info("Initialize ASR server engine successfully.")
         return True
 
-    def preprocess(self,
-                   samples,
-                   sample_rate,
-                   model_type="deepspeech2online_aishell-zh-16k"):
-        """preprocess
-
-        Args:
-            samples (numpy.array): numpy.float32
-            sample_rate (int): sample rate
-
-        Returns:
-            x_chunk (numpy.array): shape[B, T, D]
-            x_chunk_lens (numpy.array): shape[B]
-        """
-        # if "deepspeech" in model_type:
-        x_chunk, x_chunk_lens = self.executor.extract_feat(samples, sample_rate)
-        return x_chunk, x_chunk_lens
+    def preprocess(self, *args, **kwargs):
+        raise NotImplementedError("Online not using this.")
 
-    def run(self, x_chunk, x_chunk_lens, decoder_chunk_size=1):
-        """run online engine
-
-        Args:
-            x_chunk (numpy.array): shape[B, T, D]
-            x_chunk_lens (numpy.array): shape[B]
-            decoder_chunk_size(int)
-        """
-        self.output = self.executor.decode_one_chunk(x_chunk, x_chunk_lens,
-                                                     self.config.model_type)
+    def run(self, *args, **kwargs):
+        raise NotImplementedError("Online not using this.")
 
     def postprocess(self):
-        """postprocess
-        """
-        return self.output
-
-    def reset(self):
-        """reset engine decoder and inference state
-        """
-        self.executor.reset_decoder_and_chunk()
-        self.output = ""
+        raise NotImplementedError("Online not using this.")

paddlespeech/server/restful/api.py

@@ -17,12 +17,12 @@ from typing import List
 from fastapi import APIRouter
 
 from paddlespeech.cli.log import logger
+from paddlespeech.server.restful.acs_api import router as acs_router
 from paddlespeech.server.restful.asr_api import router as asr_router
 from paddlespeech.server.restful.cls_api import router as cls_router
 from paddlespeech.server.restful.text_api import router as text_router
 from paddlespeech.server.restful.tts_api import router as tts_router
 from paddlespeech.server.restful.vector_api import router as vec_router
-from paddlespeech.server.restful.acs_api import router as acs_router
 _router = APIRouter()
 
 

paddlespeech/server/utils/audio_handler.py

@@ -248,7 +248,7 @@ class ASRHttpHandler:
         }
 
         res = requests.post(url=self.url, data=json.dumps(data))
-        
+
         return res.json()
 
 

paddlespeech/server/utils/buffer.py

@@ -12,6 +12,7 @@
 # See the License for the specific language governing permissions and
 # limitations under the License.
 
+
 class Frame(object):
     """Represents a "frame" of audio data."""
 
@@ -45,7 +46,7 @@ class ChunkBuffer(object):
         self.shift_ms = shift_ms
         self.sample_rate = sample_rate
         self.sample_width = sample_width  # int16 = 2; float32 = 4
-        
+
         self.window_sec = float((self.window_n - 1) * self.shift_ms +
                                 self.window_ms) / 1000.0
         self.shift_sec = float(self.shift_n * self.shift_ms / 1000.0)
@@ -77,8 +78,8 @@ class ChunkBuffer(object):
 
         offset = 0
         while offset + self.window_bytes <= len(audio):
-            yield Frame(audio[offset:offset + self.window_bytes], self.timestamp,
-                        self.window_sec)
+            yield Frame(audio[offset:offset + self.window_bytes],
+                        self.timestamp, self.window_sec)
             self.timestamp += self.shift_sec
             offset += self.shift_bytes
 

paddlespeech/t2s/exps/speedyspeech/synthesize_e2e.py

@@ -176,7 +176,10 @@ def main():
     parser.add_argument(
         "--ngpu", type=int, default=1, help="if ngpu == 0, use cpu or xpu.")
     parser.add_argument(
-        "--nxpu", type=int, default=0, help="if nxpu == 0 and ngpu == 0, use cpu.")
+        "--nxpu",
+        type=int,
+        default=0,
+        help="if nxpu == 0 and ngpu == 0, use cpu.")
 
     args, _ = parser.parse_known_args()
 

paddlespeech/t2s/exps/speedyspeech/train.py

@@ -188,7 +188,10 @@ def main():
     parser.add_argument("--dev-metadata", type=str, help="dev data.")
     parser.add_argument("--output-dir", type=str, help="output dir.")
     parser.add_argument(
-        "--nxpu", type=int, default=0, help="if nxpu == 0 and ngpu == 0, use cpu.")
+        "--nxpu",
+        type=int,
+        default=0,
+        help="if nxpu == 0 and ngpu == 0, use cpu.")
     parser.add_argument(
         "--ngpu", type=int, default=1, help="if ngpu == 0, use cpu or xpu")
 

paddlespeech/t2s/modules/transformer/repeat.py

@@ -36,4 +36,4 @@ def repeat(N, fn):
     Returns:
         MultiSequential: Repeated model instance.
     """
-    return MultiSequential(*[fn(n) for n in range(N)])
+    return MultiSequential(* [fn(n) for n in range(N)])

setup.py

@@ -98,7 +98,6 @@ requirements = {
 }
 
 
-
 def check_call(cmd: str, shell=False, executable=None):
     try:
         sp.check_call(
@@ -112,12 +111,13 @@ def check_call(cmd: str, shell=False, executable=None):
             file=sys.stderr)
         raise e
 
+
 def check_output(cmd: str, shell=False):
     try:
         out_bytes = sp.check_output(cmd.split())
     except sp.CalledProcessError as e:
-        out_bytes = e.output       # Output generated before error
-        code      = e.returncode   # Return code
+        out_bytes = e.output  # Output generated before error
+        code = e.returncode  # Return code
         print(
             f"{__file__}:{inspect.currentframe().f_lineno}: CMD: {cmd}, Error:",
             out_bytes,
@@ -146,6 +146,7 @@ def _remove(files: str):
     for f in files:
         f.unlink()
 
+
 ################################# Install ##################################
 
 
@@ -308,6 +309,5 @@ setup_info = dict(
         ]
     })
 
-
 with version_info():
     setup(**setup_info)

tests/unit/cli/aishell_test_prepare.py

@@ -20,7 +20,6 @@ of each audio file in the data set.
 """
 import argparse
 import codecs
-import json
 import os
 from pathlib import Path
 
@@ -89,7 +88,7 @@ def create_manifest(data_dir, manifest_path_prefix):
                 duration = float(len(audio_data) / samplerate)
                 text = transcript_dict[audio_id]
                 json_lines.append(audio_path)
-                reference_lines.append(str(total_num+1) + "\t" + text)
+                reference_lines.append(str(total_num + 1) + "\t" + text)
 
                 total_sec += duration
                 total_text += len(text)
@@ -106,6 +105,7 @@ def create_manifest(data_dir, manifest_path_prefix):
 
         manifest_dir = os.path.dirname(manifest_path_prefix)
 
+
 def prepare_dataset(url, md5sum, target_dir, manifest_path=None):
     """Download, unpack and create manifest file."""
     data_dir = os.path.join(target_dir, 'data_aishell')