This topic describes the parameters and interfaces of the Paraformer real-time speech recognition Python SDK.
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 recommendations, see Real-time Speech Recognition.
Prerequisites
You have activated the service and obtained an API key. To prevent security risks from code leakage, configure the API key as an environment variable instead of hard-coding it in your code.
NoteWhen you need to provide temporary access permissions to third-party applications or users, or want to strictly control high-risk operations such as accessing or deleting sensitive data, we recommend that you use a temporary authentication token.
Compared to a long-term API key, a temporary authentication token is short-lived (60 seconds) and more secure. It is suitable for temporary call scenarios and effectively reduces the risk of API key leakage.
Usage: In your code, replace the API key used for authentication with the temporary authentication token.
Model list
paraformer-realtime-v2 | paraformer-realtime-8k-v2 | |
Scenarios | Live streaming, meetings, and other similar scenarios | Recognition scenarios for 8 kHz audio, 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. |
Custom vocabulary | ✅ See Custom hotwords | ✅ See Custom hotwords |
Specify recognition language | ✅ Specify the language using the | ❌ |
Emotion recognition | ❌ |
Getting started
The Recognition class provides methods for synchronous and streaming calls. You can select the appropriate method based on your requirements:
Synchronous call: Recognizes a local file and returns the complete result at once. This is suitable for processing pre-recorded audio.
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.
Synchronous 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.
Instantiate the Recognition class, set the request parameters, and call the call method to perform recognition and obtain the RecognitionResult.
Streaming call
This method submits a real-time speech-to-text task and returns real-time recognition results through a callback interface.
Start streaming speech recognition
Instantiate the Recognition class, set the request parameters and the callback interface (RecognitionCallback), and then call the
startmethod.Stream audio
Repeatedly call the `Recognition` class's
send_audio_framemethod 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_eventmethod 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.
End processing
Call the
stopmethod of the Recognition class to stop speech recognition.This method blocks the current thread until the
on_completeoron_errorcallback of the callback interface (RecognitionCallback) is triggered.
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 to be recognized, in Hz. Varies by model:
|
format | str | - | Yes | The format of the audio to be recognized. Supported audio formats: pcm, wav, mp3, opus, speex, aac, and amr. Important opus/speex: Must use Ogg encapsulation. wav: Must be PCM encoded. amr: Only the AMR-NB type is supported. |
vocabulary_id | str | - | No | The vocabulary ID. This parameter is used to set the vocabulary ID for v2 and later models. If not set, it does not take effect. In this speech recognition request, the service applies the hotword information that corresponds to the specified hotword ID. For more information, see Custom hotwords. |
disfluency_removal_enabled | bool | False | No | Specifies whether to filter disfluent words:
|
language_hints | list[str] | ["zh", "en"] | No | The language code for recognition. If you cannot determine the language in advance, you can leave this parameter unset. The model automatically detects the language. Currently supported language codes:
This parameter applies only to multilingual models. For more information, see Model list. |
semantic_punctuation_enabled | bool | False | No | Specifies whether to enable semantic punctuation. It is disabled by default.
Semantic punctuation provides higher accuracy and is suitable for meeting transcription scenarios. VAD punctuation has lower latency and is suitable for interactive scenarios. By adjusting the This parameter is effective only for v2 and later models. |
max_sentence_silence | int | 800 | No | The silence duration threshold for VAD punctuation, in ms. When the silence duration after a speech segment exceeds this threshold, the system determines that the sentence has ended. The parameter range is from 200 ms to 6000 ms. The default value is 800 ms. This parameter is effective only when the |
multi_threshold_mode_enabled | bool | False | No | When this switch is enabled (true), it prevents VAD from creating excessively long segments. It is disabled by default. This parameter is effective only when the |
punctuation_prediction_enabled | bool | True | No | Specifies whether to automatically add punctuation to the recognition results:
This parameter is effective only for v2 and later models. |
heartbeat | bool | False | No | Use this switch to maintain a persistent connection with the server:
This parameter is effective only for 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). It is enabled by default (true). When enabled, Chinese numerals are converted to Arabic numerals. This parameter is effective only for v2 and later models. |
callback | RecognitionCallback | - | No |
Key interfaces
Recognition class
The Recognition class is imported using `from dashscope.audio.asr import *`.
Member method | Method signature | Description |
call | | A synchronous 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 |
start | | 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 | | 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 | | Stops speech recognition. This method blocks until the service has recognized all received audio and the task is complete. |
get_last_request_id | | Gets the request_id. This can be used after the constructor is called (the object is created). |
get_first_package_delay | | 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 | | Gets the last packet delay, which is the time taken from sending the |
get_response | | Gets the last message. This can be used to get a task-failed error. |
Callback interface (RecognitionCallback)
During a 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.
Method | Parameter | Return value | Description |
| None | None | This method is called immediately after a connection is established with the server. |
|
| None | This method is called when the service sends a response. |
| None | None | This method is called after all recognition results have been returned. |
|
| None | This method is called when an exception occurs. |
| 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 streaming call or a synchronous call.
Member method | Method signature | Description |
get_sentence | | 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 | | Gets the request_id of the request. |
is_sentence_end | | 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:
Emotion recognition has the following constraints:
|
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:
|
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 you encounter an error, see Error messages for troubleshooting.
If the issue persists, join the developer group to provide feedback. Include the Request ID to help us investigate the problem.
More examples
For more examples, see GitHub.
FAQ
Features
Q: How can I maintain a persistent connection with the server during long periods of silence?
Set the heartbeat request parameter to true and continuously send silent audio to the server.
Silent audio refers to content in an audio file or data stream that has no sound signal. You can generate silent audio using various methods, such as audio editing software (Audacity and Adobe Audition) or command-line tools (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, function: input file path, example value: audio.wav
# -c:a, function: audio encoder, example values: aac, libmp3lame, pcm_s16le
# -b:a, function: bit rate (controls audio quality), example values: 192k, 320k
# -ar, function: sample rate, example values: 44100 (CD), 48000, 16000
# -ac, function: number of sound channels, example values: 1 (mono), 2 (stereo)
# -y, function: overwrite existing file (no value needed)
ffmpeg -i input_audio.ext -c:a encoder_name -b:a bit_rate -ar sample_rate -ac num_channels output.ext
# Example: WAV → MP3 (maintain original quality)
ffmpeg -i input.wav -c:a libmp3lame -q:a 0 output.mp3
# Example: MP3 → WAV (16-bit PCM standard format)
ffmpeg -i input.mp3 -c:a pcm_s16le -ar 44100 -ac 2 output.wav
# Example: M4A → 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 improve quality
# Example: FLAC lossless → Opus (high compression)
ffmpeg -i input.flac -c:a libopus -b:a 128k -vbr on output.opusQ: 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.
For more information, see synchronous call. Pass the file path to the
callmethod of the Recognition class to directly recognize the audio file.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.
For more information, see Streaming call. You can use the
send_audio_framemethod of the Recognition class to send a binary stream to the server for recognition.
Troubleshooting
Q: Why is the speech not recognized (no recognition result)?
Check whether the audio format (
format) and sample rate (sampleRateorsample_rate) in the request parameters are set correctly and meet the parameter constraints. The following are common examples of errors:The audio file has a .wav file name extension but is actually in MP3 format, and the
formatrequest parameter is set to mp3 (incorrect parameter setting).The audio sample rate is 3600 Hz, but the
sampleRateorsample_raterequest parameter is set to 48000 (incorrect parameter setting).
You can use the ffprobe tool to retrieve information about the audio 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.xxxWhen you use the
paraformer-realtime-v2model, check whether the language set inlanguage_hintsmatches the actual language of the audio.For example, the audio is in Chinese, but
language_hintsis set toen(English).If all the preceding checks pass, you can use custom vocabulary to improve the recognition of specific words.