fix the ws send bug, cache buffer, text=doc

demos/speech_recognition/run.sh

 #!/bin/bash
 
-wget -c https://paddlespeech.bj.bcebos.com/PaddleAudio/zh.wav https://paddlespeech.bj.bcebos.com/PaddleAudio/en.wav
+# wget -c https://paddlespeech.bj.bcebos.com/PaddleAudio/zh.wav https://paddlespeech.bj.bcebos.com/PaddleAudio/en.wav
 
 # asr
-paddlespeech asr --input ./zh.wav
+export CUDA_VISIBLE_DEVICES=0
+paddlespeech asr --input audio/119994.wav -v
 
 
 # asr + punc
-paddlespeech asr --input ./zh.wav | paddlespeech text --task punc
+# paddlespeech asr --input ./zh.wav | paddlespeech text --task punc
\ No newline at end of file

examples/aishell/asr1/conf/preprocess.yaml

@@ -5,7 +5,7 @@ process:
     n_mels: 80
     n_shift: 160
     win_length: 400
-    dither: 0.1
+    dither: 0.0
   - type: cmvn_json
     cmvn_path: data/mean_std.json
   # these three processes are a.k.a. SpecAugument

examples/aishell/asr1/conf/tuning/decode.yaml

@@ -3,9 +3,9 @@ decode_batch_size: 128
 error_rate_type: cer 
 decoding_method: attention # 'attention', 'ctc_greedy_search', 'ctc_prefix_beam_search', 'attention_rescoring'
 ctc_weight: 0.5 # ctc weight for attention rescoring decode mode.
-decoding_chunk_size: -1 # decoding chunk size. Defaults to -1.
+decoding_chunk_size: 1 # decoding chunk size. Defaults to -1.
     # <0: for decoding, use full chunk.
     # >0: for decoding, use fixed chunk size as set.
     # 0: used for training, it's prohibited here. 
 num_decoding_left_chunks: -1  # number of left chunks for decoding. Defaults to -1.
-simulate_streaming: False  # simulate streaming inference. Defaults to False.
+simulate_streaming: True  # simulate streaming inference. Defaults to False.

examples/aishell/asr1/run.sh

@@ -3,12 +3,12 @@ source path.sh
 set -e
 
 gpus=0,1,2,3
-stage=0
-stop_stage=50
-conf_path=conf/conformer.yaml
+stage=5
+stop_stage=5
+conf_path=conf/chunk_conformer.yaml
 decode_conf_path=conf/tuning/decode.yaml
 avg_num=20
-audio_file=data/demo_01_03.wav
+audio_file=audio/zh.wav
 
 source ${MAIN_ROOT}/utils/parse_options.sh || exit 1;
 
@@ -44,7 +44,7 @@ fi
 # Optionally, you can add LM and test it with runtime.
 if [ ${stage} -le 5 ] && [ ${stop_stage} -ge 5 ]; then
     # test a single .wav file
-    CUDA_VISIBLE_DEVICES=0 ./local/test_wav.sh ${conf_path} ${decode_conf_path} exp/${ckpt}/checkpoints/${avg_ckpt} ${audio_file} || exit -1
+    CUDA_VISIBLE_DEVICES=0 ./local/test_wav.sh ${conf_path} ${decode_conf_path} exp/chunk_conformer/checkpoints/multi_cn ${audio_file} || exit -1
 fi
 
 # Not supported at now!!!

paddlespeech/__init__.py

@@ -14,3 +14,6 @@
 import _locale
 
 _locale._getdefaultlocale = (lambda *args: ['en_US', 'utf8'])
+
+
+

paddlespeech/s2t/exps/u2/bin/test_wav.py

@@ -79,7 +79,6 @@ class U2Infer():
 
             ilen = paddle.to_tensor(feat.shape[0])
             xs = paddle.to_tensor(feat, dtype='float32').unsqueeze(axis=0)
-
             decode_config = self.config.decode
             result_transcripts = self.model.decode(
                 xs,
@@ -129,9 +128,12 @@ if __name__ == "__main__":
     args = parser.parse_args()
 
     config = CfgNode(new_allowed=True)
+    
     if args.config:
+        print(f"load config: {args.config}")
         config.merge_from_file(args.config)
     if args.decode_cfg:
+        print(f"load decode cfg: {args.decode_cfg}")
         decode_confs = CfgNode(new_allowed=True)
         decode_confs.merge_from_file(args.decode_cfg)
         config.decode = decode_confs

paddlespeech/server/conf/ws_application.yaml

@@ -4,7 +4,7 @@
 #                             SERVER SETTING                                    #
 #################################################################################
 host: 0.0.0.0
-port: 8091
+port: 8096
 
 # The task format in the engin_list is: <speech task>_<engine type>
 # task choices = ['asr_online', 'tts_online']

paddlespeech/server/tests/asr/online/websocket_client.py

@@ -17,7 +17,8 @@ import argparse
 import asyncio
 import json
 import logging
-
+import os
+import codecs
 import numpy as np
 import soundfile
 import websockets
@@ -32,22 +33,23 @@ class ASRAudioHandler:
     def read_wave(self, wavfile_path: str):
         samples, sample_rate = soundfile.read(wavfile_path, dtype='int16')
         x_len = len(samples)
-        chunk_stride = 40 * 16  #40ms, sample_rate = 16kHz
+        # chunk_stride = 40 * 16  #40ms, sample_rate = 16kHz
         chunk_size = 80 * 16  #80ms, sample_rate = 16kHz
-
-        if (x_len - chunk_size) % chunk_stride != 0:
-            padding_len_x = chunk_stride - (x_len - chunk_size) % chunk_stride
+        
+        if x_len % chunk_size != 0:
+            padding_len_x =  chunk_size - x_len % chunk_size
         else:
             padding_len_x = 0
 
         padding = np.zeros((padding_len_x), dtype=samples.dtype)
         padded_x = np.concatenate([samples, padding], axis=0)
 
-        num_chunk = (x_len + padding_len_x - chunk_size) / chunk_stride + 1
+        assert ( x_len + padding_len_x ) % chunk_size == 0
+        num_chunk = (x_len + padding_len_x ) / chunk_size
         num_chunk = int(num_chunk)
 
         for i in range(0, num_chunk):
-            start = i * chunk_stride
+            start = i * chunk_size
             end = start + chunk_size
             x_chunk = padded_x[start:end]
             yield x_chunk
@@ -77,8 +79,10 @@ class ASRAudioHandler:
             for chunk_data in self.read_wave(wavfile_path):
                 await ws.send(chunk_data.tobytes())
                 msg = await ws.recv()
+                msg = json.loads(msg)
                 logging.info("receive msg={}".format(msg))
 
+            result = msg
             # finished 
             audio_info = json.dumps(
                 {
@@ -91,16 +95,36 @@ class ASRAudioHandler:
                 separators=(',', ': '))
             await ws.send(audio_info)
             msg = await ws.recv()
+            msg = json.loads(msg)
             logging.info("receive msg={}".format(msg))
+        
+
+        return result
 
 
 def main(args):
     logging.basicConfig(level=logging.INFO)
     logging.info("asr websocket client start")
-    handler = ASRAudioHandler("127.0.0.1", 8091)
+    handler = ASRAudioHandler("127.0.0.1", 8096)
     loop = asyncio.get_event_loop()
-    loop.run_until_complete(handler.run(args.wavfile))
-    logging.info("asr websocket client finished")
+
+    # support to process single audio file
+    if args.wavfile and os.path.exists(args.wavfile):
+        logging.info(f"start to process the wavscp: {args.wavfile}")
+        result = loop.run_until_complete(handler.run(args.wavfile))
+        result = result["asr_results"]
+        logging.info(f"asr websocket client finished : {result}")
+
+    # support to process batch audios from wav.scp 
+    if args.wavscp and os.path.exists(args.wavscp):
+        logging.info(f"start to process the wavscp: {args.wavscp}")
+        with codecs.open(args.wavscp, 'r', encoding='utf-8') as f,\
+             codecs.open("result.txt", 'w', encoding='utf-8') as w:
+            for line in f:
+                utt_name, utt_path = line.strip().split()
+                result = loop.run_until_complete(handler.run(utt_path))
+                result = result["asr_results"]
+                w.write(f"{utt_name} {result}\n")
 
 
 if __name__ == "__main__":
@@ -110,6 +134,8 @@ if __name__ == "__main__":
         action="store",
         help="wav file path ",
         default="./16_audio.wav")
+    parser.add_argument(
+        "--wavscp", type=str, default=None, help="The batch audios dict text")
     args = parser.parse_args()
 
     main(args)

paddlespeech/server/utils/buffer.py

@@ -24,15 +24,25 @@ class Frame(object):
 
 class ChunkBuffer(object):
     def __init__(self,
-                 frame_duration_ms=80,
-                 shift_ms=40,
+                 window_n=7,    # frame
+                 shift_n=4,     # frame
+                 window_ms=20,  # ms
+                 shift_ms=10,   # ms
                  sample_rate=16000,
                  sample_width=2):
-        self.sample_rate = sample_rate
-        self.frame_duration_ms = frame_duration_ms
+        self.window_n = window_n
+        self.shift_n = shift_n
+        self.window_ms = window_ms
         self.shift_ms = shift_ms
-        self.remained_audio = b''
+        self.sample_rate = sample_rate
         self.sample_width = sample_width  # int16 = 2; float32 = 4
+        self.remained_audio = b''
+
+        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)
+
+        self.window_bytes = int(self.window_sec * self.sample_rate * self.sample_width)
+        self.shift_bytes = int(self.shift_sec * self.sample_rate * self.sample_width)
 
     def frame_generator(self, audio):
         """Generates audio frames from PCM audio data.
@@ -43,17 +53,12 @@ class ChunkBuffer(object):
         audio = self.remained_audio + audio
         self.remained_audio = b''
 
-        n = int(self.sample_rate * (self.frame_duration_ms / 1000.0) *
-                self.sample_width)
-        shift_n = int(self.sample_rate * (self.shift_ms / 1000.0) *
-                      self.sample_width)
         offset = 0
         timestamp = 0.0
-        duration = (float(n) / self.sample_rate) / self.sample_width
-        shift_duration = (float(shift_n) / self.sample_rate) / self.sample_width
-        while offset + n <= len(audio):
-            yield Frame(audio[offset:offset + n], timestamp, duration)
-            timestamp += shift_duration
-            offset += shift_n
+
+        while offset + self.window_bytes <= len(audio):
+            yield Frame(audio[offset:offset + self.window_bytes], timestamp, self.window_sec)
+            timestamp += self.shift_sec
+            offset += self.shift_bytes
 
         self.remained_audio += audio[offset:]

paddlespeech/server/ws/asr_socket.py

@@ -36,6 +36,10 @@ async def websocket_endpoint(websocket: WebSocket):
     # init buffer
     chunk_buffer_conf = asr_engine.config.chunk_buffer_conf
     chunk_buffer = ChunkBuffer(
+        window_n=7,
+        shift_n=4,
+        window_ms=20,
+        shift_ms=10,
         sample_rate=chunk_buffer_conf['sample_rate'],
         sample_width=chunk_buffer_conf['sample_width'])
     # init vad
@@ -76,9 +80,9 @@ async def websocket_endpoint(websocket: WebSocket):
                 message = message["bytes"]
 
                 # vad for input bytes audio
-                vad.add_audio(message)
-                message = b''.join(f for f in vad.vad_collector()
-                                   if f is not None)
+                # vad.add_audio(message)
+                # message = b''.join(f for f in vad.vad_collector()
+                #                    if f is not None)
 
                 engine_pool = get_engine_pool()
                 asr_engine = engine_pool['asr']
@@ -89,9 +93,9 @@ async def websocket_endpoint(websocket: WebSocket):
                     sample_rate = asr_engine.config.sample_rate
                     x_chunk, x_chunk_lens = asr_engine.preprocess(samples,
                                                                   sample_rate)
+                    print(x_chunk_lens)
                     asr_engine.run(x_chunk, x_chunk_lens)
                     asr_results = asr_engine.postprocess()
-
                 asr_results = asr_engine.postprocess()
                 resp = {'asr_results': asr_results}
 

setup.py

@@ -168,7 +168,7 @@ class DevelopCommand(develop):
     def run(self):
         develop.run(self)
         # must after develop.run, or pkg install by shell will not see
-        self.execute(_post_install, (self.install_lib, ), msg="Post Install...")
+    #    self.execute(_post_install, (self.install_lib, ), msg="Post Install...")
 
 
 class InstallCommand(install):