This topic describes the parameters and API details of the Qwen-Audio-3.0-ASR-Flash-Filetrans/Fun-ASR non-real-time speech recognition Java 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
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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.
NoteWhen 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
Core class (Transcription) provides interfaces to submit a task asynchronously, wait synchronously for the task to finish, and query the task result asynchronously. You can run non-real-time speech recognition in either of the following two ways:
Submit a task asynchronously and wait synchronously for it to finish: after you submit a task, the current thread blocks until the task finishes and the recognition result is returned.
Submit a task asynchronously and query the task result asynchronously: after you submit a task, call the query interface to get the task result whenever you need it.
Submit a task asynchronously and wait synchronously for it to finish
Configure the Request parameters.
Instantiate a Core class (Transcription).
Call the
asyncCallmethod of the Core class (Transcription) to submit the task asynchronously.NoteThe 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.
Call the
waitmethod of the Core class (Transcription) to wait synchronously for the task to finish.A task can be in the
PENDING,RUNNING,SUCCEEDED, orFAILEDstate. While the task is in thePENDINGorRUNNINGstate, thewaitinterface blocks. When the task reaches theSUCCEEDEDorFAILEDstate, thewaitinterface stops blocking and returns the task result.waitreturns a Task result (TranscriptionResult).
Submit a task asynchronously and query the task result asynchronously
Configure the Request parameters.
Instantiate a Core class (Transcription).
Call the
asyncCallmethod of the Core class (Transcription) to submit the task asynchronously.NoteThe 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.
Call the
fetchmethod of the Core class (Transcription) in a loop until you get the final task result.When the task status is
SUCCEEDEDorFAILED, stop polling and process the result.fetchreturns a Task result (TranscriptionResult).
Endpoints
By default, the SDK uses the endpoint of the China (Beijing) region. To switch to another region, modify Constants.baseHttpApiUrl before initialization.
China (Beijing)
https://{WorkspaceId}.cn-beijing.maas.aliyuncs.com/api/v1
When you make a call, replace {WorkspaceId} with your actual Workspace ID.
Singapore
https://{WorkspaceId}.ap-southeast-1.maas.aliyuncs.com/api/v1
When you make a call, replace {WorkspaceId} with your actual Workspace ID.
Alibaba Cloud Model Studio has released workspace-specific domains for the China (Beijing) and Singapore regions. The new dedicated domains deliver superior performance and higher stability for inference requests. We recommend migrating to the new domains:
China (Beijing): from
dashscope.aliyuncs.comto{WorkspaceId}.cn-beijing.maas.aliyuncs.comSingapore: from
dashscope-intl.aliyuncs.comto{WorkspaceId}.ap-southeast-1.maas.aliyuncs.com
Replace {WorkspaceId} with your actual Workspace ID. The existing domains remain fully functional.
Switch to the Singapore region:
import com.alibaba.dashscope.utils.Constants;
// Set this at the beginning of your code
Constants.baseHttpApiUrl = "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 target region.
The region setting is global and affects the API calls of all DashScope SDKs.
Request parameters
Configure request parameters using the chained methods of TranscriptionParam.
Parameter | Type | Required | Description |
model | String | Yes | The 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. |
fileUrls | List<String> | Yes | A 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 Important
|
vocabularyId | String | No | The 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. |
vocabulary | Map<String, Integer> | No | Instant hot words. Passed as key-value pairs, where the key is the hot word text ( Suitable for temporary, session-level hot word optimization. When configured together with precompiled hot words, only the instant hot words take effect. For usage details, see Instant hotwords. Important Only Note Set Set through parameterSet through parameters |
channelId | List<Integer> | No | The 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. Important Each specified track is billed independently. For example, requesting [0, 1] for a single file incurs two separate charges. Default value: [0]. |
specialWordFilter | String | No | The sensitive words to process during speech recognition. You can set a different handling method for each sensitive word. For details, see Sensitive word filtering. |
diarizationEnabled | Boolean | No | Whether 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 Note 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 |
speakerCount | Integer | No | Important Takes effect only when speaker diarization is enabled ( 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_hints | String[] | No | The 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. Note Set Set through parameterSet through parameters |
apiKey | String | No | Your API key. If you have configured the API key as an environment variable, you do not need to set it in your code. Otherwise, you must set it in your code. |
Response
Task result (TranscriptionResult)
TranscriptionResult encapsulates the result of the current task.
Interface/Method | Parameter | Return value | Description |
| None | requestId | Gets the requestId. |
| None | taskId | Gets the taskId. |
| None |
| Gets the task status.
Note When a task contains multiple subtasks, the overall task status is marked as |
| None | Gets the Subtask result (TranscriptionTaskResult). Each task recognizes one or more audio files. Different audio files are processed in separate subtasks, so each task corresponds to one or more subtasks. | |
| None | The task result, in JSON format | Gets the task result. The result is data in JSON format. If you want to get the task result through the |
Subtask result (TranscriptionTaskResult)
TranscriptionTaskResult encapsulates the result of a subtask. A subtask recognizes a single audio file.
Interface/Method | Parameter | Return value | Description |
| None | The URL of the recognized audio file | Gets the URL of the recognized audio file. |
| None | The URL of the recognition result | Gets the URL of the recognition result. This URL is valid for 24 hours. After it expires, you can no longer query the task or download the result through the URL returned in a previous query. The recognition result is saved as a JSON file. You can download the file through the URL or read its content directly through an HTTP request. For the meaning of each field in the JSON data, see Recognition result description. |
| None |
| Gets the subtask status.
|
| None | Key information generated during task execution, which may be empty | Gets the key information generated during task execution. When a task fails, check this content to analyze the cause. |
Recognition result description
The recognition result is saved as a JSON file.
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. Important 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
Task query parameter class (TranscriptionQueryParam)
TranscriptionQueryParam is used when waiting for a task to finish (calling the wait method of Transcription) or querying the task result (calling the fetch method of Transcription).
Create a TranscriptionQueryParam instance through the static method FromTranscriptionParam.
Interface/method | Parameters | Return value | Description |
|
| a | Creates a |
Core class (Transcription)
Import Transcription with "import com.alibaba.dashscope.audio.asr.transcription.*;". Its key interfaces are as follows:
Interface/method | Parameters | Return value | Description |
|
| Submits a speech recognition task asynchronously. | |
|
| Blocks the current thread until the asynchronous task ends (the task status is | |
|
| Queries the current task result asynchronously. |
Additional interfaces: batch query task status / cancel tasks
For details, see Manage asynchronous tasks. It supports batch querying non-real-time speech recognition tasks submitted within the past 24 hours, and canceling tasks in the PENDING (queued) state.
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:
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:
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The temporary URL is valid for 48 hours and cannot be used after it expires. Do not use it in a production environment.
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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.
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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 your code returns an error, troubleshoot it 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