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Non-real-time speech recognition (Qwen-Audio-3.0-ASR-Flash-Filetrans/Fun-ASR)

Qwen-Audio-3.0-ASR-Flash-Filetrans/Fun-ASR non-real-time speech recognition Python SDK

This topic describes the parameters and interfaces of the Qwen-Audio-3.0-ASR-Flash-Filetrans/Fun-ASR non-real-time speech recognition Python SDK.

User guide:Non-real-time speech recognition. For input requirements such as supported audio formats, file size limits, and duration limits, see Audio specifications.

Prerequisites

You have activated the service and Obtain an API key. Please Configure API key as an environment variable instead of hardcoding it in your code to prevent security risks caused by code leakage.
When you need to provide temporary access to third-party applications or users, or when you want to strictly control high-risk operations such as accessing or deleting sensitive data, we recommend using temporary authentication tokens.Compared with long-term API Keys, temporary authentication tokens have a short validity period (60 seconds) and higher security, making them suitable for temporary call scenarios and effectively reducing the risk of API Key leakage.Usage: In your code, replace the API Key originally used for authentication with the obtained temporary authentication token.

Quick start

The Core class (Transcription) provides interfaces to submit a task asynchronously, wait synchronously for a task to finish, and query task results asynchronously. You can run non-real-time speech recognition in either of the following ways:
  • Submit the task asynchronously and wait synchronously: after submitting the task, block the current thread until the task finishes and return the recognition result.
  • Submit the task asynchronously and query the result asynchronously: after submitting the task, call the query interface to retrieve the result whenever you need it.

Submit asynchronously and wait synchronously

image
  1. Call the async_call method of the Core class (Transcription) and set the Request parameters.
    • The file transcription service processes tasks submitted through the API on a best-effort basis. After you submit a task, it enters the queued (PENDING) state. The queuing time depends on the queue length and the file duration, so it cannot be stated precisely, but it is usually within a few minutes. Once processing starts, speech recognition completes at hundreds of times real-time speed.
    • After each task finishes, the recognition result and the download URL are valid for 24 hours. After they expire, you can no longer query the task or download the result through the URL returned in a previous query.
  2. Call the wait method of the Core class (Transcription) to wait synchronously for the task to finish. A task can be in one of the following states: PENDING, RUNNING, SUCCEEDED, and FAILED. While the task is in the PENDING or RUNNING state, the wait interface blocks. When the task reaches the SUCCEEDED or FAILED state, the wait interface stops blocking and returns the task result. wait returns a TranscriptionResponse.
from http import HTTPStatus
from dashscope.audio.asr import Transcription
import dashscope
import os
import json

# The following is the configuration for the Singapore region. Replace "{WorkspaceId}" with your actual workspace ID. Configurations differ across regions.
dashscope.base_http_api_url = 'https://{WorkspaceId}.ap-southeast-1.maas.aliyuncs.com/api/v1'

# The API keys for the Singapore and Beijing regions differ. Get your API key: https://www.alibabacloud.com/help/zh/model-studio/get-api-key
# If you have not configured the environment variable, replace the following line with your Model Studio API key: dashscope.api_key = "sk-xxx"
dashscope.api_key = os.getenv("DASHSCOPE_API_KEY")

task_response = Transcription.async_call(
    model='qwen-audio-3.0-asr-flash-filetrans',
    file_urls=['{YOUR_AUDIO_URL}']
)

transcribe_response = Transcription.wait(task=task_response.output.task_id)
if transcribe_response.status_code == HTTPStatus.OK:
    print(json.dumps(transcribe_response.output, indent=4, ensure_ascii=False))
    print('transcription done!')

Submit asynchronously and query the result asynchronously

image
  1. Call the async_call method of the Core class (Transcription) and set the Request parameters.
    • The file transcription service processes tasks submitted through the API on a best-effort basis. After you submit a task, it enters the queued (PENDING) state. The queuing time depends on the queue length and the file duration, so it cannot be stated precisely, but it is usually within a few minutes. Once processing starts, speech recognition completes at hundreds of times real-time speed.
    • After each task finishes, the recognition result and the download URL are valid for 24 hours. After they expire, you can no longer query the task or download the result through the URL returned in a previous query.
  2. Call the fetch method of the Core class (Transcription) in a loop until you get the final task result. When the task status is SUCCEEDED or FAILED, stop polling and process the result. fetch returns a TranscriptionResponse.
from http import HTTPStatus
from dashscope.audio.asr import Transcription
import dashscope
import os
import json

# The following is the configuration for the Singapore region. When calling, replace "{WorkspaceId}" with your actual workspace ID. Configurations differ across regions.
dashscope.base_http_api_url = 'https://{WorkspaceId}.ap-southeast-1.maas.aliyuncs.com/api/v1'

# The API Key differs between the Singapore and Beijing regions. Get an API Key: https://www.alibabacloud.com/help/zh/model-studio/get-api-key
# If you have not configured the environment variable, replace the following line with your Model Studio API Key: dashscope.api_key = "sk-xxx"
dashscope.api_key = os.getenv("DASHSCOPE_API_KEY")

transcribe_response = Transcription.async_call(
    model='qwen-audio-3.0-asr-flash-filetrans',
    file_urls=['{YOUR_AUDIO_URL}']
)

while True:
    if transcribe_response.output.task_status == 'SUCCEEDED' or transcribe_response.output.task_status == 'FAILED':
        break
    transcribe_response = Transcription.fetch(task=transcribe_response.output.task_id)

if transcribe_response.status_code == HTTPStatus.OK:
    print(json.dumps(transcribe_response.output, indent=4, ensure_ascii=False))
    print('transcription done!')

Service endpoints

By default, the SDK uses the service endpoint of the Beijing region. To switch to another region, modify dashscope.base_http_api_url before initialization.
  • Singapore
  • China (Beijing)
https://{WorkspaceId}.ap-southeast-1.maas.aliyuncs.com/api/v1Replace {WorkspaceId} with your actual Workspace ID.
To switch to the Singapore region:
import dashscope

# Set this at the beginning of your code
dashscope.base_http_api_url = 'https://{WorkspaceId}.ap-southeast-1.maas.aliyuncs.com/api/v1'
Note:
  • API keys differ across regions. Make sure you use the API key for the corresponding region.
  • The region configuration is a global setting that affects all API calls made through the DashScope SDK.

Request parameters

Set request parameters through the async_call method of the Core class (Transcription).
ParameterTypeRequiredDescription
modelstrYesThe model name. Supported values include the Qwen-Audio-3.0-ASR-Flash-Filetrans and Fun-ASR model families. For details, see Supported models and regions.
file_urlslist[str]YesA list of URLs of the audio or video files to transcribe. HTTP and HTTPS are supported. A single request supports only one URL. For input requirements such as supported audio formats, file size limits, and duration limits, see Audio specifications.If the recording is stored in Alibaba Cloud OSS, the RESTful API supports temporary URLs prefixed with oss://, whereas the SDK does not support oss://-prefixed temporary URLs.
  • A temporary URL is valid for 48 hours and cannot be used after it expires. Do not use it in production.
  • The upload credential interface is rate-limited to 100 QPS and cannot be scaled up. Do not use it in production, high-concurrency, or load-testing scenarios.
  • For production, use stable storage such as Alibaba Cloud OSS to keep files available long-term and avoid rate limiting.
  • If an audio file URL set to an OSS temporary public URL is unreachable, set X-DashScope-OssResourceResolve to enable in the request header (not recommended). The SDK does not support configuring request headers.
vocabulary_idstrNoThe ID of a precompiled hot word list.Generate this ID in advance by calling the create hot word list API. Pass the ID during recognition to use the hot words in the list.Suitable for scenarios where the vocabulary is known and relatively stable, and where you need to reuse the same word list across requests.For usage details, see Precompiled hotwords.
vocabularydictNoInstant hot words.Passed as key-value pairs, where the key is the hot word text (string) and the value is the hot word weight (integer). No hot word list needs to be created in advance. The weight ranges from [1, 5] or is set to 50: a value in [1, 5] makes the model more likely to output the word as the value increases; a value of 50 designates a super hot word, which greatly improves recall, but the number of super hot words cannot exceed 50.Suitable for temporary, session-level hot word optimization.When instant and precompiled hotwords are configured together, the system merges both sets. If the merged set contains more than 2000 hotwords, the system randomly selects 2000 to use. For usage details, see Instant hotwords.
Only qwen-audio-3.0-asr-flash-filetrans supports inline hotwords.
Example:
from dashscope.audio.asr import Transcription

vocab = {"Zhang San": 5, "Li Si": 5}
result = Transcription.async_call(
    model="qwen-audio-3.0-asr-flash-filetrans",
    vocabulary=vocab,
    file_urls=['{YOUR_AUDIO_URL}']
)
channel_idlist[int]NoThe index of the audio tracks to recognize in a multi-track audio file. The index starts at 0. For example, [0] recognizes the first track, and [0, 1] recognizes the first and second tracks at the same time. If you omit this parameter, only the first track is processed.
Each specified track is billed independently. For example, requesting [0, 1] for a single file incurs two separate charges.
Default value: [0].
special_word_filterstrNoThe sensitive words to process during speech recognition. You can set a different handling method for each sensitive word. For details, see Sensitive word filtering.
diarization_enabledboolNoWhether to enable speaker diarization. Disabled by default.Applies only to mono audio. Multi-channel audio does not support speaker diarization.When enabled, the recognition result includes a speaker_id field that distinguishes different speakers.
When speaker diarization is enabled, keep the audio duration within 2 hours. Otherwise, recognition may fail or time out.
Default value: False.For an example of speaker_id, see Recognition result description.
speaker_countintNo
Takes effect only when speaker diarization is enabled (diarization_enabled is set to True).
A reference value for the number of speakers. The valid range is an integer from 2 to 100 (inclusive).By default, the number of speakers is detected automatically. If you set this value, it only guides the algorithm to output the specified count when possible and does not guarantee that exact count.No default value.
language_hintslist[str]NoThe language codes to recognize. If you can't determine the language in advance, leave it unset and the model detects the language automatically.For Qwen-Audio-3.0-ASR-Flash-Filetrans models, you can set up to 4 values; any values beyond the first 4 are ignored. For Fun-ASR models, you can set only 1 value; if you set multiple, only the first takes effect.
  • qwen-audio-3.0-asr-flash-filetrans, fun-asr, fun-asr-2025-11-07, fun-asr-mtl, fun-asr-mtl-2025-08-25:
    • zh: Chinese
    • en: English
    • ja: Japanese
    • ko: Korean
    • vi: Vietnamese
    • th: Thai
    • id: Indonesian
    • ms: Malay
    • tl: Filipino
    • hi: Hindi
    • ar: Arabic
    • fr: French
    • de: German
    • es: Spanish
    • pt: Portuguese
    • ru: Russian
    • it: Italian
    • nl: Dutch
    • sv: Swedish
    • da: Danish
    • fi: Finnish
    • no: Norwegian
    • el: Greek
    • pl: Polish
    • cs: Czech
    • hu: Hungarian
    • ro: Romanian
    • bg: Bulgarian
    • hr: Croatian
    • sk: Slovak
  • fun-asr-2025-08-25:
    • zh: Chinese
    • en: English

Response

TranscriptionResponse

TranscriptionResponse wraps the basic task information (task_id and task_status) and the task result (the content of the output attribute, see TranscriptionOutput).
{
    "status_code":200,
    "request_id":"251aceab-a6aa-9fc4-b7f7-0cc6d3e2a9f3",
    "code":null,
    "message":"",
    "output":{
        "task_id":"7d0a58a3-1dbe-4de9-8cff-5f48213128b0",
        "task_status":"PENDING",
        "submit_time":"2025-02-13 16:55:08.573",
        "scheduled_time":"2025-02-13 16:55:08.592",
        "task_metrics":{
            "TOTAL":1,
            "SUCCEEDED":0,
            "FAILED":0
        }
    },
    "usage":null
}
Parameters to note:

Parameter

Description

status_code

HTTP status code of the request.

code

  • The outermost code can be ignored.

  • The code under output.results is the error code. Combine it with the message field and refer to Error codes to troubleshoot the issue.

message

  • The outermost message can be ignored.

  • The message under output.results is the error message. Combine it with the code field and refer to Error codes to troubleshoot the issue.

task_id

Task ID.

task_status

Task status.

One of four states: PENDING, RUNNING, SUCCEEDED, and FAILED.

When a task contains multiple subtasks, the overall task status is marked as SUCCEEDED as long as any subtask succeeds. Check the subtask_status field to determine the result of each subtask.

results

Subtask recognition results.

subtask_status

Subtask status.

One of four states: PENDING, RUNNING, SUCCEEDED, and FAILED.

file_url

URL of the recognized audio.

transcription_url

URL of the audio recognition result.

The recognition result is saved as a JSON file. You can download the file from the link associated with transcription_url or read its content directly through an HTTP request. For the contents of the JSON file, see Recognition result description.

TranscriptionOutput

TranscriptionOutput corresponds to the output attribute of the TranscriptionResponse and represents the result of the current task.
  • PENDING state
  • RUNNING state
  • SUCCEEDED state
  • FAILED state
{
    "task_id":"f2f7c2fa-0cd9-4bb2-a283-27b26ee4bb67",
    "task_status":"PENDING",
    "submit_time":"2025-02-13 17:59:27.754",
    "scheduled_time":"2025-02-13 17:59:27.789",
    "task_metrics":{
        "TOTAL":1,
        "SUCCEEDED":0,
        "FAILED":0
    }
}
Parameters to note:

Parameter

Description

code

The error code. Combine it with the message field and refer to Error codes to troubleshoot the issue.

message

The error message. Combine it with the code field and refer to Error codes to troubleshoot the issue.

task_id

Task ID.

task_status

Task status.

One of four states: PENDING, RUNNING, SUCCEEDED, and FAILED.

When a task contains multiple subtasks, the overall task status is marked as SUCCEEDED as long as any subtask succeeds. Check the subtask_status field to determine the result of each subtask.

results

Subtask recognition results.

subtask_status

Subtask status.

One of four states: PENDING, RUNNING, SUCCEEDED, and FAILED.

file_url

URL of the recognized audio.

transcription_url

URL of the audio recognition result.

The recognition result is saved as a JSON file. You can download the file from the link associated with transcription_url or read its content directly through an HTTP request. For the contents of the JSON file, see Recognition result description.

Recognition result description

The recognition result is saved as a JSON file.
{
    "file_url":"{YOUR_AUDIO_URL}",
    "properties":{
        "audio_format":"pcm_s16le",
        "channels":[
            0
        ],
        "original_sampling_rate":16000,
        "original_duration_in_milliseconds":3834
    },
    "transcripts":[
        {
            "channel_id":0,
            "content_duration_in_milliseconds":3720,
            "text":"Hello world, this is Alibaba Speech Lab.",
            "sentences":[
                {
                    "begin_time":100,
                    "end_time":3820,
                    "text":"Hello world, this is Alibaba Speech Lab.",
                    "sentence_id":1,
                    "speaker_id":0, //This field is displayed only when automatic speaker diarization is enabled
                    "words":[
                        {
                            "begin_time":100,
                            "end_time":596,
                            "text":"Hello ",
                            "punctuation":""
                        },
                        {
                            "begin_time":596,
                            "end_time":844,
                            "text":"world",
                            "punctuation":", "
                        }
                        // Other content is omitted here
                    ]
                }
            ]
        }
    ]
}
The following parameters are worth noting:

Parameter

Type

Description

audio_format

string

The audio format of the source file.

channels

array[integer]

The track index of the audio in the source file. For single-track audio, [0] is returned; for dual-track audio, [0, 1] is returned; and so on.

original_sampling_rate

integer

The sampling rate (Hz) of the audio in the source file.

original_duration_in_milliseconds

integer

The original audio duration (ms) in the source file.

channel_id

integer

The track index of the transcription result, starting from 0.

content_duration

integer

The duration (ms) of content in the track that is identified as speech.

The speech recognition model service transcribes only the content in a track that is identified as speech, and meters and bills based on that duration. Non-speech content is not metered or billed. Typically, the speech content duration is shorter than the original audio duration. Because whether speech content exists is determined by an AI model, the result may differ slightly from the actual situation.

transcript

string

The paragraph-level transcription result.

sentences

array

The sentence-level transcription result.

words

array

The word-level transcription result.

begin_time

integer

The start timestamp (ms).

end_time

integer

The end timestamp (ms).

text

string

The transcription result.

speaker_id

integer

The index of the current speaker, starting from 0, used to distinguish between different speakers.

This field appears in the recognition result only when speaker diarization is enabled.

punctuation

string

The punctuation predicted after the word, if any.

Key interfaces

Core class (Transcription)

Import Transcription with "from dashscope.audio.asr import Transcription".
MethodSignatureDescription
async_call
@classmethod
def async_call(cls,
               model: str,
               file_urls: List[str],
               phrase_id: str = None,
               api_key: str = None,
               workspace: str = None,
               **kwargs) -> TranscriptionResponse
Submits a speech recognition task asynchronously.
wait
@classmethod
def wait(cls,
         task: Union[str, TranscriptionResponse],
         api_key: str = None,
         workspace: str = None,
         **kwargs) -> TranscriptionResponse
Blocks the current thread until the asynchronous task finishes (the task status is SUCCEEDED or FAILED).This method returns a TranscriptionResponse.
fetch
@classmethod
def fetch(cls,
          task: Union[str, TranscriptionResponse],
          api_key: str = None,
          workspace: str = None,
          **kwargs) -> TranscriptionResponse
Queries the result of the current task asynchronously.This method returns a TranscriptionResponse.

Error codes

If you encounter an error, see Error codes to troubleshoot. When a task contains multiple subtasks, the overall task status is marked as SUCCEEDED as long as at least one subtask succeeds. Check the subtask_status field to determine the result of each subtask. Error response example:
{
    "task_id": "7bac899c-06ec-4a79-8875-xxxxxxxxxxxx",
    "task_status": "SUCCEEDED",
    "submit_time": "2024-12-16 16:30:59.170",
    "scheduled_time": "2024-12-16 16:30:59.204",
    "end_time": "2024-12-16 16:31:02.375",
    "results": [
        {
            "file_url": "{YOUR_AUDIO_URL}",
            "code": "InvalidFile.DownloadFailed",
            "message": "The audio file cannot be downloaded.",
            "subtask_status": "FAILED"
        }
    ],
    "task_metrics": {
        "TOTAL": 1,
        "SUCCEEDED": 0,
        "FAILED": 1
    }
}

FAQ

Features

Q: Is Base64-encoded audio supported?

Base64-encoded audio is not supported. Only audio at a publicly accessible URL can be recognized. Binary streams and local files cannot be recognized directly.

Q: How do I make an audio file available at a publicly accessible URL?

The typical steps are as follows. This is one approach; the exact process varies by storage product. We recommend that you upload the audio to Alibaba Cloud OSS:
For example:
  • Object storage service (recommended):
    • Use a cloud provider's object storage service (such as Alibaba Cloud OSS) to upload the audio file to a bucket and set it to public access.
    • Advantages: high availability, CDN acceleration support, and easy management.
  • Web server:
    • Place the audio file on a web server that supports HTTP/HTTPS access (such as Nginx or Apache).
    • Advantages: suitable for small projects or local testing.
  • Content delivery network (CDN):
    • Host the audio file on a CDN and access it through the URL that the CDN provides.
    • Advantages: accelerates file delivery and suits high-concurrency scenarios.
Upload the audio according to the storage or hosting method you chose. For example:
  • Object storage service:
    • Log in to the cloud provider's console and create a bucket.
    • Upload the audio file, and set its permission to public read or generate a temporary access link.
  • Web server:
    • Place the audio file in a designated directory on the server (such as /var/www/html/audio/).
    • Make sure the file is accessible over HTTP/HTTPS.
For example:
  • Object storage service:
    • After the file is uploaded, the system automatically generates a public access URL (typically in the format https://<bucket-name>.<region>.aliyuncs.com/<file-name>).
    • For a friendlier domain name, bind a custom domain and enable HTTPS.
  • Web server:
    • The access URL is usually the server address plus the file path (such as https://your-domain.com/audio/file.mp3).
  • CDN:
    • After you configure CDN acceleration, use the URL that the CDN provides (such as https://cdn.your-domain.com/audio/file.mp3).
Make sure the generated URL is accessible over the public network. For example:
  • Open the URL in a browser and check whether the audio file plays.
  • Use a tool (such as curl or Postman) to verify that the URL returns the correct HTTP response (status code 200).
When using the SDK, if audio files are stored in Alibaba Cloud OSS, temporary URLs with the oss:// prefix are not supported. When using the RESTful API, if audio files are stored in Alibaba Cloud OSS, temporary URLs with the oss:// prefix are supported:
  • The temporary URL is valid for 48 hours and cannot be used after it expires. Do not use it in a production environment.
  • The API for obtaining an upload credential is limited to 100 QPS and does not support scaling out. Do not use it in production environments, high-concurrency scenarios, or stress testing scenarios.
  • For production environments, use a stable storage service such as OSS to ensure long-term file availability and avoid rate limiting issues.

Q: How long does it take to get the recognition result?

After a task is submitted, it enters the queued (PENDING) state. The queuing time depends on the queue length and the audio duration, so it cannot be stated exactly, but it is usually within a few minutes. In general, the longer the audio, the longer it takes.

Troubleshooting

If you encounter a code error, troubleshoot the issue based on the information in Error codes.

Q: Polling never returns a result?

This may be caused by throttling. Wait a moment and try again.

Q: Why can't the speech be recognized (no recognition result)?

Check that the audio format and sample rate are correct and meet the parameter constraints. Use the ffprobe tool to get the audio container, codec, sample rate, channels, and other details:
ffprobe -v error -show_entries format=format_name -show_entries stream=codec_name,sample_rate,channels -of default=noprint_wrappers=1 input.xxx
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