The WebSocket Endpoints supports all languages as listed on here and the following audio formats
| File Format | Audio Bit Depth | Max File Size | Languages |
|---|---|---|---|
| RAW | PCM 16-bit | 10MB | Singapore/Malaysia English, Bahasa Melayu |
You can try with this Sample audio !
Whenever an API request is made to Ins8.ai STT, API Token must be included as a HTTP Parameter.
To enable a specific feature e.g. Punctuation, Timestamp, the corresponding must be added as a HTTP Parameter with the value True in the URL.
wss://stt.ins8.ai/api/v1/stt/websocket/recognize?api_token={TOKEN}×tamp=True`api_token` - Your generated API Token.
`language` - Refer to Language Model. (en-sg, bh-sg, bh-id, ch-sg, ch-ml)
`punctuation` - True/False
`timestamp` - True/False
`confidence` - True/False
`enable_speaker_diarization` - True/False
`diarization_speaker_count` - Int (Between 2 to 5)
`pii` - True/False (Redacts Name, IC, Location, Email and Date)
`itn` - True/False (Inverse normalizes text e.g. there are twenty eight birds -> there are 28 birds)When creating a WebSocket Connection, the base url should look like
wss://stt.ins8.ai/api/v1/stt/websocket/recognize?api_token={TOKEN}&language=en-sgAfter a connection is initialized, a client will have to first send a configuration packet in json detailling the configuration and metadata details of the audio to be sent.
`encoding` - Specify the type of encoding the audio file. Please use `raw`.
`sample_rate_hertz` - Sample rate in kilohertz e.g. 16000.
`stop_string` - The stop string that marks the end of an audio transmission.
`custom_vocab` - Enable greater emphasis on key words
`boost` - Float value between -20.0 to 20.0After sending the initial start message, the client can begin transmitting audio packets to the websocket in the format as defined during the initial transmission.
{'encoding': 'raw', 'sample_rate_hertz': 16000, 'stop_string': 'EOS', 'custom_vocab':'metaverse,bitcoin', 'boost':8}From the websocket server, the websocket client can close the websocket connection. Upon receiving:
!!END_OF_TRANSCRIPTION!!# Install required packages using
# pip install pywav
# pip install websockets
import json
from pywav import WavRead
import asyncio
import ssl
import websockets
path = "Path_to_example_audio.wav"
async def receiver(websocket):
""" (Asynchronous) function to recieve transcript for chunks asynchronously
Args:
websocket ([websockets.legacy.client.WebSocketClientProtocol])
"""
while True:
data = await websocket.recv()
print(f"Hypothesis of recognised speech: {data}")
if data == "!!END_OF_TRANSCRIPTION!!":
break
async def sender(websocket, data: list, stop_str:str ):
"""
(Asynchronous) function to send audio chunks asynchronously
Args:
websocket ([websockets.legacy.client.WebSocketClientProtocol])
data (list): [ List of audio chunks in bytes ]
stop_str (str): [ the string that is sent to the server to mark the end of audio streaming]
"""
for chunk in data:
await asyncio.sleep(0.05)
await websocket.send(chunk)
# ! Required: A stop string has to be sent
#await websocket.send(stop_str)
await websocket.send(stop_str.encode())
def get_chunks(filepath, filetype=None):
if not filetype:
filetype = filepath.split('.')[-1]
# Loads the audio into memory
assert filetype.lower() == 'wav', f"WSS endpoint only support wave file."
audio = WavRead(filepath)
sample_rate = audio.getsamplerate()
assert (audio.getnumofchannels() == 1)
content = audio.getdata()
chunk_size = int(sample_rate * audio.getbytespersample() / 10) # 100ms
assert (audio.getaudioformat() in [1, 7])
assert audio.getaudioformat() == 1, "WSS endpoint only support PCM encoded wave files"
encoding = 'raw'
config = {'encoding': encoding,
'sample_rate_hertz': sample_rate,
# stop string could string of any choice
'stop_string': 'EOS'}
stream = [content[i:i+chunk_size] for i in range(0, len(content), chunk_size)]
return config, stream
async def wss_test(hostname="wss://stt.ins8.ai/api/v1/stt/websocket/recognize", token=''):
audio_config, data_stream_list = get_chunks(filepath=path)
url = hostname + "?api_token=" + token + "&language=en-sg" + "&itn=True" + "&punctuation=True" + "&pii=True"
ssl_context = None
ssl_context = ssl.SSLContext(ssl.PROTOCOL_TLS_CLIENT)
ssl_context.check_hostname = False
ssl_context.verify_mode = ssl.CERT_NONE
if ssl_context is not None:
async with websockets.connect(url, ssl=ssl_context) as websocket:
#, ssl=True
stop_string = audio_config['stop_string']
await websocket.send(json.dumps(audio_config))
print(audio_config)
while True:
await asyncio.gather(sender(websocket, data_stream_list,
stop_string
),
receiver(websocket))
break
print(f"Client closing connection")
else:
async with websockets.connect(url) as websocket:
#, ssl=True
stop_string = audio_config['stop_string']
await websocket.send(json.dumps(audio_config))
print(audio_config)
while True:
await asyncio.gather(sender(websocket, data_stream_list,
stop_string
),
receiver(websocket))
break
print(f"Client closing connection")
api_key = "<YOUR SERVICE TOKEN>"
asyncio.get_event_loop().run_until_complete(wss_test(token=api_key))Below is the sample JSON output obtained from Ins8.ai STT Websocket streaming.
Testing testing, one, two three.{
"transcript":"Testing testing, one, two three.",
"confidence":0.982926,
"words":[
{
"startTime":"3.424s",
"endTime":"3.668s",
"word":"testing",
"confidence":0.92237765,
"start_time":3.424,
"end_time":3.668
},
{
"startTime":"3.871s",
"endTime":"4.114s",
"word":"testing",
"confidence":0.919446,
"start_time":3.871,
"end_time":4.114
},
{
"startTime":"4.357s",
"endTime":"4.398s",
"word":"one",
"confidence":0.99807906,
"start_time":4.357,
"end_time":4.398
},
{
"startTime":"4.560s",
"endTime":"4.601s",
"word":"two",
"confidence":0.9857209,
"start_time":4.56,
"end_time":4.601
},
{
"startTime":"4.763s",
"endTime":"4.804s",
"word":"three",
"confidence":0.9959221,
"start_time":4.763,
"end_time":4.804
}
]
}