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Alibaba Cloud Model Studio:Python SDK

Last Updated:Mar 24, 2026

The parameters and interfaces of the Paraformer real-time speech recognition Python SDK.

Important

This document applies only to the China (Beijing) region. To use the models, you must use an API key from the China (Beijing) region.

User guide: For model descriptions and selection guidance, see Real-time speech recognition - Fun-ASR/Gummy/Paraformer.

Prerequisites

  • You have activated Model Studio and created an API key. To prevent security risks from code leaks, do not hard-code the API key in your code. Instead, export it as an environment variable.

    Note

    Use a temporary authentication token to grant temporary access to third-party applications or users, or to strictly control high-risk operations like accessing or deleting sensitive data.

    A temporary authentication token is more secure than a long-lived API key because it expires in 60 seconds. This short lifespan makes it ideal for temporary call scenarios and significantly reduces the risk of a compromised API key.

    To use this method, replace the API key in your code with the temporary authentication token.

  • Install the latest version of the DashScope SDK.

Model list

paraformer-realtime-v2

paraformer-realtime-8k-v2

Scenarios

Live streaming and meetings

For 8 kHz Audio recognition in scenarios such as telephone customer service and voicemail.

Sample rate

Any

8 kHz

Languages

Chinese (including Mandarin and various dialects), English, Japanese, Korean, German, French, and Russian

Supported Chinese dialects: Shanghainese, Wu, Minnan, Northeastern, Gansu, Guizhou, Henan, Hubei, Hunan, Jiangxi, Ningxia, Shanxi, Shaanxi, Shandong, Sichuan, Tianjin, Yunnan, and Cantonese

Chinese

Punctuation prediction

✅ Supported by default. No configuration is required.

✅ Supported by default. No configuration is required.

Inverse Text Normalization (ITN)

✅ Supported by default. No configuration is required.

✅ Supported by default. No configuration is required.

Specify recognition language

✅ Specify the language using the language_hints parameter.

Emotion recognition

✅ (Click to view usage)

Emotion recognition has the following constraints:

  • It is available only for the paraformer-realtime-8k-v2 model.

  • You must disable semantic punctuation (controlled by the request parameter semantic_punctuation_enabled). By default, semantic punctuation is disabled.

  • The emotion recognition result is displayed only when the RecognitionResult object's is_sentence_end method returns True.

To obtain the emotion detection results, retrieve the emotion and the emotion confidence level of the current sentence from the emo_tag and emo_confidence fields of the single-sentence information (Sentence), respectively.

Getting started

The Recognition class provides methods for non-streaming and bidirectional streaming calls. You can select the appropriate method based on your requirements:

  • Non-streaming call: Recognizes a local file and returns the complete result at once. This is suitable for processing pre-recorded audio.

  • Bidirectional streaming call: Recognizes an audio stream and outputs the results in real time. The audio stream can come from an external device, such as a microphone, or be read from a local file. This is suitable for scenarios that require immediate feedback.

Non-streaming call

This method submits a real-time speech-to-text task for a local file. The process is blocked until the complete transcription result is returned.

image

Instantiate the Recognition class, set the request parameters, and call the call method to perform recognition and obtain the RecognitionResult.

Click to view the complete example

The audio file used in the example is: asr_example.wav.

from http import HTTPStatus
from dashscope.audio.asr import Recognition

# If you have not configured the API key in the environment variable, uncomment the following line of code and replace apiKey with your API key.
# import dashscope
# dashscope.api_key = "apiKey"

recognition = Recognition(model='paraformer-realtime-v2',
                          format='wav',
                          sample_rate=16000,
                          # The "language_hints" parameter is supported only by the paraformer-realtime-v2 model.
                          language_hints=['zh', 'en'],
                          callback=None)
result = recognition.call('asr_example.wav')
if result.status_code == HTTPStatus.OK:
    print('Recognition result:')
    print(result.get_sentence())
else:
    print('Error: ', result.message)
    
print(
    '[Metric] requestId: {}, first package delay ms: {}, last package delay ms: {}'
    .format(
        recognition.get_last_request_id(),
        recognition.get_first_package_delay(),
        recognition.get_last_package_delay(),
    ))

Bidirectional streaming call

This method submits a real-time speech-to-text task and returns real-time recognition results through a callback interface.

image
  1. Start streaming speech recognition

    Instantiate the Recognition class, set the request parameters and the callback interface (RecognitionCallback), and then call the start method.

  2. Stream audio

    Repeatedly call the `Recognition` class's send_audio_frame method to send the binary audio stream from a local file or a device (such as a microphone) to the server in segments.

    As audio data is sent, the server uses the RecognitionCallback callback interface's on_event method to return the recognition results to the client in real time.

    We recommend that the duration of each audio segment sent is about 100 milliseconds, and the data size is between 1 KB and 16 KB.

  3. End processing

    Call the stop method of the Recognition class to stop speech recognition.

    This method blocks the current thread until the on_complete or on_error callback of the callback interface (RecognitionCallback) is triggered.

Click to view the complete example

Recognize speech from a microphone

import os
import signal  # for keyboard events handling (press "Ctrl+C" to terminate recording)
import sys

import dashscope
import pyaudio
from dashscope.audio.asr import *

mic = None
stream = None

# Set recording parameters
sample_rate = 16000  # sampling rate (Hz)
channels = 1  # mono channel
dtype = 'int16'  # data type
format_pcm = 'pcm'  # the format of the audio data
block_size = 3200  # number of frames per buffer


def init_dashscope_api_key():
    """
        Set your DashScope API-key. More information:
        https://github.com/aliyun/alibabacloud-bailian-speech-demo/blob/master/PREREQUISITES.md
    """

    if 'DASHSCOPE_API_KEY' in os.environ:
        dashscope.api_key = os.environ[
            'DASHSCOPE_API_KEY']  # load API-key from environment variable DASHSCOPE_API_KEY
    else:
        dashscope.api_key = '<your-dashscope-api-key>'  # set API-key manually


# Real-time speech recognition callback
class Callback(RecognitionCallback):
    def on_open(self) -> None:
        global mic
        global stream
        print('RecognitionCallback open.')
        mic = pyaudio.PyAudio()
        stream = mic.open(format=pyaudio.paInt16,
                          channels=1,
                          rate=16000,
                          input=True)

    def on_close(self) -> None:
        global mic
        global stream
        print('RecognitionCallback close.')
        stream.stop_stream()
        stream.close()
        mic.terminate()
        stream = None
        mic = None

    def on_complete(self) -> None:
        print('RecognitionCallback completed.')  # recognition completed

    def on_error(self, message) -> None:
        print('RecognitionCallback task_id: ', message.request_id)
        print('RecognitionCallback error: ', message.message)
        # Stop and close the audio stream if it is running
        if 'stream' in globals() and stream.active:
            stream.stop()
            stream.close()
        # Forcefully exit the program
        sys.exit(1)

    def on_event(self, result: RecognitionResult) -> None:
        sentence = result.get_sentence()
        if 'text' in sentence:
            print('RecognitionCallback text: ', sentence['text'])
            if RecognitionResult.is_sentence_end(sentence):
                print(
                    'RecognitionCallback sentence end, request_id:%s, usage:%s'
                    % (result.get_request_id(), result.get_usage(sentence)))


def signal_handler(sig, frame):
    print('Ctrl+C pressed, stop recognition ...')
    # Stop recognition
    recognition.stop()
    print('Recognition stopped.')
    print(
        '[Metric] requestId: {}, first package delay ms: {}, last package delay ms: {}'
        .format(
            recognition.get_last_request_id(),
            recognition.get_first_package_delay(),
            recognition.get_last_package_delay(),
        ))
    # Forcefully exit the program
    sys.exit(0)


# main function
if __name__ == '__main__':
    init_dashscope_api_key()
    print('Initializing ...')

    # Create the recognition callback
    callback = Callback()

    # Call recognition service by async mode, you can customize the recognition parameters, like model, format,
    # sample_rate
    recognition = Recognition(
        model='paraformer-realtime-v2',
        format=format_pcm,
        # 'pcm', 'wav', 'opus', 'speex', 'aac', or 'amr'. You can check the supported formats in the document.
        sample_rate=sample_rate,
        # 8000 or 16000 is supported.
        semantic_punctuation_enabled=False,
        callback=callback)

    # Start recognition
    recognition.start()

    signal.signal(signal.SIGINT, signal_handler)
    print("Press 'Ctrl+C' to stop recording and recognition...")
    # Create a keyboard listener until "Ctrl+C" is pressed

    while True:
        if stream:
            data = stream.read(3200, exception_on_overflow=False)
            recognition.send_audio_frame(data)
        else:
            break

    recognition.stop()

Recognize a local audio file

The audio file used in the example is: asr_example.wav.

import os
import time
from dashscope.audio.asr import *

# If you have not configured the API key in the environment variable, uncomment the following line of code and replace apiKey with your API key.
# import dashscope
# dashscope.api_key = "apiKey"

from datetime import datetime

def get_timestamp():
    now = datetime.now()
    formatted_timestamp = now.strftime("[%Y-%m-%d %H:%M:%S.%f]")
    return formatted_timestamp

class Callback(RecognitionCallback):
    def on_complete(self) -> None:
        print(get_timestamp() + ' Recognition completed')  # recognition complete

    def on_error(self, result: RecognitionResult) -> None:
        print('Recognition task_id: ', result.request_id)
        print('Recognition error: ', result.message)
        exit(0)

    def on_event(self, result: RecognitionResult) -> None:
        sentence = result.get_sentence()
        if 'text' in sentence:
            print(get_timestamp() + ' RecognitionCallback text: ', sentence['text'])
            if RecognitionResult.is_sentence_end(sentence):
                print(get_timestamp() + 
                    'RecognitionCallback sentence end, request_id:%s, usage:%s'
                    % (result.get_request_id(), result.get_usage(sentence)))


callback = Callback()

recognition = Recognition(model='paraformer-realtime-v2',
                          format='wav',
                          sample_rate=16000,
                          # The "language_hints" parameter is supported only by the paraformer-realtime-v2 model.
                          language_hints=['zh', 'en'],
                          callback=callback)

recognition.start()

try:
    audio_data: bytes = None
    f = open("asr_example.wav", 'rb')
    if os.path.getsize("asr_example.wav"):
        while True:
            audio_data = f.read(3200)
            if not audio_data:
                break
            else:
                recognition.send_audio_frame(audio_data)
            time.sleep(0.1)
    else:
        raise Exception(
            'The supplied file was empty (zero bytes long)')
    f.close()
except Exception as e:
    raise e

recognition.stop()

print(
    '[Metric] requestId: {}, first package delay ms: {}, last package delay ms: {}'
    .format(
        recognition.get_last_request_id(),
        recognition.get_first_package_delay(),
        recognition.get_last_package_delay(),
    ))

Concurrent calls

In Python, because of the Global Interpreter Lock, only one thread can execute Python code at a time, although some performance-oriented libraries may remove this limitation. To better utilize the computing resources of a multi-core computer, we recommend that you use multiprocessing or concurrent.futures.ProcessPoolExecutor. Multi-threading can significantly increase SDK call latency under high concurrency.

Request parameters

Request parameters are set in the constructor (__init__) of the Recognition class.

Parameter

Type

Default

Required

Description

model

str

-

Yes

The model used for real-time speech recognition. For more information, see Model list.

sample_rate

int

-

Yes

The sample rate of the audio, in Hz.

This parameter varies by model:

  • paraformer-realtime-v2 supports any sample rate.

  • paraformer-realtime-8k-v2 supports only an 8000 Hz sample rate.

format

str

-

Yes

The format of the audio.

Supported audio formats: pcm, wav, mp3, opus, speex, aac, and amr.

Important

opus/speex requires Ogg encapsulation.

wav requires PCM encoding.

amr: Only the AMR-NB type is supported.

disfluency_removal_enabled

bool

False

No

Specifies whether to filter out disfluent words:

  • true: Filters out disfluent words.

  • false (default): Does not filter out disfluent words.

language_hints

list[str]

["zh", "en"]

No

Specifies the language codes for recognition. If this parameter is unset, the model automatically detects the language.

Supported language codes:

  • zh: Chinese

  • en: English

  • ja: Japanese

  • yue: Cantonese

  • ko: Korean

  • de: German

  • fr: French

  • ru: Russian

This parameter applies only to multilingual models. For more information, see Model list.

semantic_punctuation_enabled

bool

False

No

Specifies whether to enable semantic sentence segmentation.

  • true: Enables semantic sentence segmentation and disables Voice Activity Detection (VAD) based segmentation.

  • false (default): Enables VAD-based segmentation and disables semantic sentence segmentation.

Semantic sentence segmentation offers higher accuracy and is ideal for meeting transcription scenarios. VAD-based segmentation has lower latency and is better suited for interactive scenarios.

By adjusting the semantic_punctuation_enabled parameter, you can flexibly change the punctuation method for speech recognition to suit different scenarios.

This parameter applies only to v2 and later models.

max_sentence_silence

int

800

No

Specifies the silence duration threshold for VAD-based segmentation, in milliseconds (ms).

If a pause in speech exceeds this threshold, the system considers the sentence complete.

Valid range: 200 ms to 6000 ms. Default: 800 ms.

This parameter applies only to v2 and later models when semantic_punctuation_enabled is set to false.

multi_threshold_mode_enabled

bool

False

No

If true, this parameter prevents VAD-based segmentation from creating excessively long sentences.

This parameter applies only to v2 and later models when semantic_punctuation_enabled is set to false.

punctuation_prediction_enabled

bool

True

No

Specifies whether to automatically add punctuation to the recognition results:

  • true (default): Adds punctuation.

  • false: Does not add punctuation.

This parameter applies only to v2 and later models.

heartbeat

bool

False

No

Specifies whether to maintain a persistent connection with the server:

  • true: Maintains the connection even when continuously sending silent audio.

  • false (default): The connection terminates after a 60-second timeout, even if silent audio is continuously sent.

    Silent audio is an audio file or data stream with no sound. You can generate silent audio using audio editing software like Audacity or Adobe Audition, or with command-line tools like FFmpeg.

This parameter applies only to v2 and later models.

When using this field, the SDK version must be 1.23.1 or later.

inverse_text_normalization_enabled

bool

True

No

Specifies whether to enable Inverse Text Normalization (ITN).

When enabled, Chinese numerals are converted to Arabic numerals.

This parameter applies only to v2 and later models.

callback

RecognitionCallback

-

No

RecognitionCallback interface.

Key interfaces

Recognition class

The Recognition class is imported using `from dashscope.audio.asr import *`.

Member method

Method signature

Description

call

def call(self, file: str, phrase_id: str = None, **kwargs) -> RecognitionResult

A non-streaming call that uses a local file. This method blocks the current thread until the entire audio file is read. The file must have read permissions.

The recognition result is returned as a RecognitionResult type.

start

def start(self, phrase_id: str = None, **kwargs)

Starts speech recognition.

This is a callback-based streaming real-time recognition method that does not block the current thread. It must be used with send_audio_frame and stop.

send_audio_frame

def send_audio_frame(self, buffer: bytes)

Pushes an audio stream. The audio stream pushed each time should not be too large or too small. We recommend that each audio packet has a duration of about 100 ms and a size between 1 KB and 16 KB.

You can obtain the recognition results through the on_event method of the callback interface (RecognitionCallback).

stop

def stop(self)

Stops speech recognition. This method blocks until the service has recognized all received audio and the task is complete.

get_last_request_id

def get_last_request_id(self)

Gets the request_id. This can be used after the constructor is called (the object is created).

get_first_package_delay

def get_first_package_delay(self)

Gets the first packet delay, which is the latency from sending the first audio packet to receiving the first recognition result packet. Use this after the task is completed.

get_last_package_delay

def get_last_package_delay(self)

Gets the last packet delay, which is the time taken from sending the stop instruction to receiving the last recognition result packet. Use this after the task is completed.

Callback interface (RecognitionCallback)

During a bidirectional streaming call, the server uses callbacks to return key process information and data to the client. You must implement a callback method to process the returned information and data.

Click to view the example

class Callback(RecognitionCallback):
    def on_open(self) -> None:
        print('Connection successful')

    def on_event(self, result: RecognitionResult) -> None:
        # Implement the logic for receiving recognition results

    def on_complete(self) -> None:
        print('Task completed')

    def on_error(self, result: RecognitionResult) -> None:
        print('An exception occurred: ', result)

    def on_close(self) -> None:
        print('Connection closed')


callback = Callback()

Method

Parameter

Return value

Description

def on_open(self) -> None

None

None

This method is called immediately after a connection is established with the server.

def on_event(self, result: RecognitionResult) -> None

result: RecognitionResult

None

This method is called when the service sends a response.

def on_complete(self) -> None

None

None

This method is called after all recognition results have been returned.

def on_error(self, result: RecognitionResult) -> None

result: Recognition result

None

This method is called when an exception occurs.

def on_close(self) -> None

None

None

This method is called after the service has closed the connection.

Response results

Recognition result (RecognitionResult)

RecognitionResult represents the recognition result of either a single real-time recognition in a bidirectional streaming call or a non-streaming call.

Member method

Method signature

Description

get_sentence

def get_sentence(self) -> Union[Dict[str, Any], List[Any]]

Gets the currently recognized sentence and timestamp information. In a callback, a single sentence is returned, so this method returns a Dict[str, Any] type.

For more information, see Sentence.

get_request_id

def get_request_id(self) -> str

Gets the request_id of the request.

is_sentence_end

@staticmethod
def is_sentence_end(sentence: Dict[str, Any]) -> bool

Determines whether the given sentence has ended.

Sentence (Sentence)

The members of the Sentence class are as follows:

Parameter

Type

Description

begin_time

int

The start time of the sentence, in ms.

end_time

int

The end time of the sentence, in ms.

text

str

The recognized text.

words

A list of Word timestamp information (Word)

Word timestamp information.

emo_tag

str

The emotion of the current sentence:

  • positive: Positive emotion, such as happy or satisfied

  • negative: Negative emotion, such as angry or sad

  • neutral: No obvious emotion

Emotion recognition has the following constraints:

  • It is available only for the paraformer-realtime-8k-v2 model.

  • You must disable semantic punctuation (controlled by the request parameter semantic_punctuation_enabled). By default, semantic punctuation is disabled.

  • The emotion recognition result is displayed only when the RecognitionResult object's is_sentence_end method returns True.

emo_confidence

float

The confidence level of the recognized emotion for the current sentence. The value ranges from 0.0 to 1.0. A larger value indicates a higher confidence level.

Emotion recognition has the following constraints:

  • It is available only for the paraformer-realtime-8k-v2 model.

  • You must disable semantic punctuation (controlled by the request parameter semantic_punctuation_enabled). By default, semantic punctuation is disabled.

  • The emotion recognition result is displayed only when the RecognitionResult object's is_sentence_end method returns True.

Word timestamp information (Word)

The members of the Word class are as follows:

Parameter

Type

Description

begin_time

int

The start time of the word, in ms.

end_time

int

The end time of the word, in ms.

text

str

The word.

punctuation

str

The punctuation.

Error codes

If an error occurs, see Error Messages for troubleshooting.

If the issue persists, join the Developer Group and provide the Request ID for further investigation.

More examples

For more examples, see GitHub.

FAQ

Features

Q: How do I maintain a persistent connection with the server during long silences?

Set the request parameter heartbeat to true, and continuously send silent audio to the server.

Silent audio is an audio file or data stream with no sound. You can generate silent audio using audio editing software like Audacity or Adobe Audition, or with command-line tools like FFmpeg.

Q: How do I convert an audio file to a supported format?

You can use the FFmpeg tool. For more information, see the official FFmpeg website.

# Basic conversion command (universal template)
# -i: Input file path. Example: audio.wav
# -c:a: Audio encoder. Examples: aac, libmp3lame, pcm_s16le
# -b:a: Bit rate (controls audio quality). Examples: 192k, 320k
# -ar: Sample rate. Examples: 44100 (CD), 48000, 16000
# -ac: Audio channels. Examples: 1 (mono), 2 (stereo)
# -y: Overwrite existing file (no value required).
ffmpeg -i input_audio.ext -c:a encoder_name -b:a bit_rate -ar sample_rate -ac num_channels output.ext

# Example: Convert WAV to MP3 (maintaining original quality)
ffmpeg -i input.wav -c:a libmp3lame -q:a 0 output.mp3
# Example: Convert MP3 to WAV (16-bit PCM standard format)
ffmpeg -i input.mp3 -c:a pcm_s16le -ar 44100 -ac 2 output.wav
# Example: Convert M4A to AAC (extract/convert Apple audio)
ffmpeg -i input.m4a -c:a copy output.aac  # Directly extract without re-encoding
ffmpeg -i input.m4a -c:a aac -b:a 256k output.aac  # Re-encode to a 256k bit rate
# Example: Convert FLAC lossless to Opus (high compression)
ffmpeg -i input.flac -c:a libopus -b:a 128k -vbr on output.opus
Q: Can I view the time range for each sentence?

Yes. The speech recognition result includes the start and end timestamp for each sentence, which defines its time range.

Q: How do I recognize a local file (recorded audio file)?

There are two ways to recognize a local file:

  • Directly pass the local file path: This method returns the complete recognition result after the file is fully processed. It is not suitable for scenarios that require immediate feedback.

    Pass the file path to the call method of the Recognition class to directly recognize the audio file. For more information, see non-streaming call.

  • Convert the local file into a binary stream for recognition: This method returns recognition results in a stream while the file is being processed. It is suitable for scenarios that require immediate feedback.

    You can use the send_audio_frame method of the Recognition class to send a binary stream to the server for recognition. For more information, see bidirectional streaming call.

Troubleshooting

Q: Why am I not getting any recognition results?

  1. Check if the settings for the audio format (format) and sample rate (sampleRate/sample_rate) are correct and meet the parameter constraints. The following are common error examples:

    • The audio file has a .wav extension but is actually in MP3 format, and the request parameter format is set to mp3 (an incorrect parameter setting).

    • The audio sample rate is 3600 Hz, but the request parameter sampleRate/sample_rate is set to 48000 (incorrect parameter setting).

    You can use the ffprobe tool to get information about an audio file's properties, such as its container, encoding, sample rate, and channels:

    ffprobe -v error -show_entries format=format_name -show_entries stream=codec_name,sample_rate,channels -of default=noprint_wrappers=1 input.xxx
  2. When you use the paraformer-realtime-v2 model, check that the language specified in language_hints matches the actual language of the audio.

    For example: the audio is actually in Chinese, but language_hints is set to en (English).