From 6e26ad0e149ae51e3fc8b89b3178684979e6bbd1 Mon Sep 17 00:00:00 2001
From: 雾聪 <wucong.lyb@alibaba-inc.com>
Date: 星期四, 09 十一月 2023 11:03:45 +0800
Subject: [PATCH] Merge branch 'main' of https://github.com/alibaba-damo-academy/FunASR into main
---
runtime/python/websocket/funasr_wss_client.py | 110 ++++++++++++++++++++++++++++++++++++++-----------------
1 files changed, 76 insertions(+), 34 deletions(-)
diff --git a/runtime/python/websocket/funasr_wss_client.py b/runtime/python/websocket/funasr_wss_client.py
index 7c96553..66b3ce0 100644
--- a/runtime/python/websocket/funasr_wss_client.py
+++ b/runtime/python/websocket/funasr_wss_client.py
@@ -27,20 +27,16 @@
help="grpc server port")
parser.add_argument("--chunk_size",
type=str,
- default="0, 10, 5",
+ default="5, 10, 5",
help="chunk")
-parser.add_argument("--encoder_chunk_look_back",
- type=int,
- default=4,
- help="number of chunks to lookback for encoder self-attention")
-parser.add_argument("--decoder_chunk_look_back",
- type=int,
- default=1,
- help="number of encoder chunks to lookback for decoder cross-attention")
parser.add_argument("--chunk_interval",
type=int,
default=10,
help="chunk")
+parser.add_argument("--hotword",
+ type=str,
+ default="",
+ help="hotword file path, one hotword perline (e.g.:闃块噷宸村反 20)")
parser.add_argument("--audio_in",
type=str,
default=None,
@@ -61,11 +57,14 @@
type=str,
default=None,
help="output_dir")
-
parser.add_argument("--ssl",
type=int,
default=1,
help="1 for ssl connect, 0 for no ssl")
+parser.add_argument("--use_itn",
+ type=int,
+ default=1,
+ help="1 for using itn, 0 for not itn")
parser.add_argument("--mode",
type=str,
default="2pass",
@@ -106,10 +105,29 @@
rate=RATE,
input=True,
frames_per_buffer=CHUNK)
+ # hotwords
+ fst_dict = {}
+ hotword_msg = ""
+ if args.hotword.strip() != "":
+ f_scp = open(args.hotword)
+ hot_lines = f_scp.readlines()
+ for line in hot_lines:
+ words = line.strip().split(" ")
+ if len(words) < 2:
+ print("Please checkout format of hotwords")
+ continue
+ try:
+ fst_dict[" ".join(words[:-1])] = int(words[-1])
+ except ValueError:
+ print("Please checkout format of hotwords")
+ hotword_msg=json.dumps(fst_dict)
- message = json.dumps({"mode": args.mode, "chunk_size": args.chunk_size, "encoder_chunk_look_back": args.encoder_chunk_look_back,
- "decoder_chunk_look_back": args.decoder_chunk_look_back, "chunk_interval": args.chunk_interval,
- "wav_name": "microphone", "is_speaking": True})
+ use_itn=True
+ if args.use_itn == 0:
+ use_itn=False
+
+ message = json.dumps({"mode": args.mode, "chunk_size": args.chunk_size, "chunk_interval": args.chunk_interval,
+ "wav_name": "microphone", "is_speaking": True, "hotwords":hotword_msg, "itn": use_itn})
#voices.put(message)
await websocket.send(message)
while True:
@@ -127,6 +145,31 @@
wavs = f_scp.readlines()
else:
wavs = [args.audio_in]
+
+ # hotwords
+ fst_dict = {}
+ hotword_msg = ""
+ if args.hotword.strip() != "":
+ f_scp = open(args.hotword)
+ hot_lines = f_scp.readlines()
+ for line in hot_lines:
+ words = line.strip().split(" ")
+ if len(words) < 2:
+ print("Please checkout format of hotwords")
+ continue
+ try:
+ fst_dict[" ".join(words[:-1])] = int(words[-1])
+ except ValueError:
+ print("Please checkout format of hotwords")
+ hotword_msg=json.dumps(fst_dict)
+ print (hotword_msg)
+
+ sample_rate = 16000
+ wav_format = "pcm"
+ use_itn=True
+ if args.use_itn == 0:
+ use_itn=False
+
if chunk_size > 0:
wavs = wavs[chunk_begin:chunk_begin + chunk_size]
for wav in wavs:
@@ -143,20 +186,13 @@
import wave
with wave.open(wav_path, "rb") as wav_file:
params = wav_file.getparams()
+ sample_rate = wav_file.getframerate()
frames = wav_file.readframes(wav_file.getnframes())
audio_bytes = bytes(frames)
else:
- import ffmpeg
- try:
- # This launches a subprocess to decode audio while down-mixing and resampling as necessary.
- # Requires the ffmpeg CLI and `ffmpeg-python` package to be installed.
- audio_bytes, _ = (
- ffmpeg.input(wav_path, threads=0)
- .output("-", format="s16le", acodec="pcm_s16le", ac=1, ar=16000)
- .run(cmd=["ffmpeg", "-nostdin"], capture_stdout=True, capture_stderr=True)
- )
- except ffmpeg.Error as e:
- raise RuntimeError(f"Failed to load audio: {e.stderr.decode()}") from e
+ wav_format = "others"
+ with open(wav_path, "rb") as f:
+ audio_bytes = f.read()
# stride = int(args.chunk_size/1000*16000*2)
stride = int(60 * args.chunk_size[1] / args.chunk_interval / 1000 * 16000 * 2)
@@ -164,8 +200,9 @@
# print(stride)
# send first time
- message = json.dumps({"mode": args.mode, "chunk_size": args.chunk_size, "chunk_interval": args.chunk_interval,
- "wav_name": wav_name, "is_speaking": True})
+ message = json.dumps({"mode": args.mode, "chunk_size": args.chunk_size, "chunk_interval": args.chunk_interval, "audio_fs":sample_rate,
+ "wav_name": wav_name, "wav_format": wav_format, "is_speaking": True, "hotwords":hotword_msg, "itn": use_itn})
+
#voices.put(message)
await websocket.send(message)
is_speaking = True
@@ -213,12 +250,17 @@
meg = await websocket.recv()
meg = json.loads(meg)
- # print(meg)
wav_name = meg.get("wav_name", "demo")
text = meg["text"]
+ timestamp=""
+ if "timestamp" in meg:
+ timestamp = meg["timestamp"]
if ibest_writer is not None:
- text_write_line = "{}\t{}\n".format(wav_name, text)
+ if timestamp !="":
+ text_write_line = "{}\t{}\t{}\n".format(wav_name, text, timestamp)
+ else:
+ text_write_line = "{}\t{}\n".format(wav_name, text)
ibest_writer.write(text_write_line)
if meg["mode"] == "online":
@@ -227,15 +269,15 @@
os.system('clear')
print("\rpid" + str(id) + ": " + text_print)
elif meg["mode"] == "offline":
- text_print += "{}".format(text)
+ if timestamp !="":
+ text_print += "{} timestamp: {}".format(text, timestamp)
+ else:
+ text_print += "{}".format(text)
+
# text_print = text_print[-args.words_max_print:]
# os.system('clear')
print("\rpid" + str(id) + ": " + wav_name + ": " + text_print)
- if ("is_final" in meg and meg["is_final"]==False):
- offline_msg_done = True
-
- if not "is_final" in meg:
- offline_msg_done = True
+ offline_msg_done = True
else:
if meg["mode"] == "2pass-online":
text_print_2pass_online += "{}".format(text)
--
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