Speech File Transcriber Lite is an offline service that transcribes pre-recorded audio files. This is an asynchronous service. To transcribe a file, you must submit a URL that is accessible over HTTP. Local files are not supported. This service differs from the standard Speech File Recognition service in its response time. The Lite version returns transcription results within 24 hours.
Billing and concurrency limits
Speech File Transcriber Lite is available only as a commercial version. A trial version is not available. For more information, see Trial and commercial versions. To use this feature, you must activate the commercial version. For more information, see Upgrade from a trial version to a commercial version.
For more information about billing methods, see Billing methods.
For more information about concurrency limits, see Concurrency and QPS.
Limits
You must strictly follow these requirements when you write your code. Otherwise, your transcription request may fail and return an empty result.
The service supports the transcription of single-channel and dual-channel audio files in WAV, MP3, MP4, M4A, WMA, AAC, OGG, AMR, and FLAC formats.
The audio file size cannot exceed 512 MB. The video file size cannot exceed 2 GB. The total file duration cannot exceed 12 hours.
The audio file to be transcribed must be stored on a server and be accessible through a URL.
You can use Alibaba Cloud Object Storage Service (OSS). If the access permissions for the file in OSS are set to public-read, see Public-read Object to obtain the file access link. If the access permissions for the file in OSS are set to private, see Private Object to generate an access link with a validity period using the software development kit (SDK).
Alternatively, you can store the audio file on a self-built file server that allows file downloads. Ensure that the `Content-Length` value in the HTTP response header matches the actual length of the data in the body. Otherwise, the download will fail.
The access permissions for the uploaded audio file URL must be public. The URL must use a domain name, not an IP address. The URL cannot contain spaces or Chinese characters.
Valid URL
Invalid URL
https://gw.alipayobjects.com/os/bmw-prod/0574ee2e-f494-45a5-820f-63aee583045a.wav
http://127.0.XX.XX/sample.wav
D:\files\sample.wav
Speech File Transcriber Lite is an asynchronous transcription service and has no concurrency limit for transcription tasks. However, the following Queries Per Second (QPS) limits apply to API calls:
The user-level QPS limit is 200 for POST requests to the Speech File Transcriber Lite API.
The user-level QPS limit is 500 for GET requests to the Speech File Transcriber Lite API.
The QPS limit is 1 for the same TaskId when you query the result of a Speech File Transcriber Lite task.
After you submit a Speech File Transcriber Lite request, the transcription task is completed and the result is returned within 24 hours. The transcription result is stored on the server for 72 hours.
ImportantThis does not apply to large-scale data uploads, such as uploading more than 500 hours of audio within 30 minutes. To transcribe large-scale data, contact pre-sales experts.
The intelligent speaker diarization feature supports only 8 kHz single-channel and 16 kHz single-channel audio.
Supported invocation methods: polling and callback.
The service supports language model customization. For more information, see Language model customization.
The service supports hotwords. For more information, see Hotwords.
The service supports transcription models for multiple languages and dialects, such as Mandarin Chinese, Chinese dialects, and English. The following language and dialect models are currently supported:
You cannot specify the language and dialect models in the code. In the Intelligent Speech Interaction console, go to the All Projects page. For the relevant project, perform the Configure Project Features operation and select the corresponding model. For more information, see Manage projects.
Steps
Determine the format and sample rate of your audio file. Then, select a suitable scenario model in the console based on your business scenario.
Store the audio file in OSS.
If the access permissions for the file in OSS are set to public-read, see Public-read Object to obtain the file access link. If the access permissions for the file in OSS are set to private, see Private Object to generate an access link with a validity period using the SDK.
NoteAlternatively, you can store the audio file on a self-built file server that allows file downloads. Ensure that the value of
Content-Lengthin the HTTP response header matches the actual length of the data in the body. Otherwise, the download will fail.The client submits a Speech File Transcriber Lite request.
The server returns a task ID for the request. You can use this ID to query the transcription result.
The client sends a request to query the transcription result.
Use the task ID that you obtained in step 3 to query the result of the Speech File Transcriber Lite task. After you submit the request, the transcription task is completed and the result is returned within 24 hours. The transcription result is stored on the server for 72 hours.
Interaction flow
The interaction flow between the client and the server is shown in the following figure.
For all server responses except audio streams, the response header contains the TaskId parameter for the transcription task. This is the unique identifier for the request.
POP call parameters for each region
Region | Call parameters |
China (Shanghai) |
|
China (Beijing) |
|
China (Shenzhen) |
|
API call method
The Speech File Transcriber Lite service provides Remote Procedure Call (RPC) style APIs that follow the POP API specification. Parameters are encapsulated in each request, and each request corresponds to a specific method. The result of the operation is included in the response. The audio file to be transcribed must be stored on a server (Alibaba Cloud OSS is recommended) and be accessible through a URL. If you use Alibaba Cloud OSS, you can access it over the internal network within the same region to avoid outbound traffic fees. For more information, see How do I set the OSS internal network address for speech transcription?.
The Speech File Transcriber Lite POP API consists of two parts: a POST API for submitting transcription requests (user-level QPS limit of 200) and a GET API for querying transcription results (user-level QPS limit of 500).
Invoke the detection API:
When you use the polling method, you can submit a Speech File Transcriber Lite task to obtain a task ID for subsequent polling.
When you use the callback method, you can submit a Speech File Transcriber Lite task and a callback URL. After the task is completed, the transcription result is sent to the webhook address in a POST request. The webhook address must be able to accept POST requests.
Input parameters and descriptions:
When you submit a Speech File Transcriber Lite request, you must set the input parameters. You must pass the parameters as a JSON string in the body of the request object. The JSON format is as follows:
{ "appkey": "your-appkey", // To get an Appkey, go to the console: https://nls-portal.console.aliyun.com/applist "file_link": "https://gw.alipayobjects.com/os/bmw-prod/0574ee2e-f494-45a5-820f-63aee583045a.wav", "auto_split":false, "enable_words": false, "enable_sample_rate_adaptive": true, // valid_times: Retrieves the recognition content for a specified time period of the audio. If not needed, you do not need to fill in this field. "valid_times": [ { "begin_time": 200, "end_time":2000, "channel_id": 0 } ] }Parameter
Value type
Required
Description
appkey
String
Yes
The Appkey of the project in the console.
file_link
String
Yes
The address where the audio file is stored. In the console, you must set the model of the corresponding project to a model that supports the audio scenario.
enable_words
Boolean
No
Specifies whether to return word information. The default is false.
enable_sample_rate_adaptive
Boolean
No
Specifies whether to automatically downsample audio with a sample rate greater than 16 kHz to 16 kHz. The default value is false.
enable_callback
Boolean
No
Specifies whether to enable the callback feature. The default value is false.
callback_url
String
No
The address of the callback service. This parameter is required if `enable_callback` is set to true. The URL supports HTTP and HTTPS. The host cannot be an IP address.
speaker_num
Integer
No
Used to help specify the number of speakers for voiceprint recognition. The value must be an integer from 2 to 100. The default value is 2 for 8k audio and 100 for 16k audio.
This parameter only helps the algorithm try to output the specified number of speakers. It does not guarantee that this number of speakers will be output. This parameter must be used with the auto_split and supervise_type parameters.
auto_split
Boolean
No
Specifies whether to enable intelligent speaker diarization. If enabled, you can determine the speaker of each sentence based on the ChannelId in the recognition result for a two-party conversation. Typically, the ChannelId of the first speaker is 0. For 8k dual-channel audio, if speaker diarization is enabled, the default number of speakers is 2, and sound channels 0 and 1 are the track numbers.
supervise_type
Integer
No
The method for determining the number of speakers for speaker separation. This parameter must be used with the auto_split and speaker_num parameters.
Default (empty): The number of speakers is specified by the user for 8k audio and determined by the algorithm for 16k audio.
1: The number of speakers is specified by the user. The specific number is confirmed by the speaker_num parameter.
2: The algorithm counts the number of people.
enable_inverse_text_normalization
Boolean
No
Inverse text normalization (ITN) converts Chinese numerals to Arabic numerals. If set to True, Chinese numerals are converted to Arabic numerals in the output. The default value is False.
enable_disfluency
Boolean
No
Filters filler words, also known as disfluency detection. The default value is false (disabled). To enable this feature, set version to 4.0.
enable_punctuation_prediction
Boolean
No
Specifies whether to add punctuation to sentences. The default value is true (add punctuation).
valid_times
List< ValidTime >
No
Information about valid time periods, used to exclude unwanted time periods.
max_end_silence
Integer
No
The maximum allowed ending silence. The value must be in the range of 200 to 6000. The default value is 800. Unit: milliseconds.
This parameter is invalid if semantic sentence detection (enable_semantic_sentence_detection) is enabled.
max_single_segment_time
Integer
No
The maximum allowed end time for a single sentence. The minimum value is 5000. The default value is 60000. Unit: milliseconds.
This parameter is invalid if semantic sentence detection (enable_semantic_sentence_detection) is enabled.
customization_id
String
No
The ID of the custom model created through the POP API. By default, no custom model is added.
class_vocabulary_id
String
No
The ID of the created class-based hotword vocabulary. By default, no vocabulary is added.
vocabulary_id
String
No
The ID of the created general hotword vocabulary. By default, no vocabulary is added.
enable_semantic_sentence_detection
Boolean
No
Specifies whether to enable semantic sentence detection. Valid values: true/false. The default value is false.
enable_timestamp_alignment
Boolean
No
Specifies whether to enable the timestamp calibration feature. Valid values: true/false. The default value is false.
first_channel_only
Boolean
No
Specifies whether to recognize only the first sound channel. Valid values: true/false. (If the speech recognition result is duplicated, you can enable this parameter.)
Default (empty): Processes dual-channel for 8k audio and single-channel for 16k audio.
false: Processes dual-channel for 8k audio and dual-channel for 16k audio.
true: Processes single-channel for 8k audio and single-channel for 16k audio.
ImportantBilling model:
For 8k dual-channel processing, billing is based on a single channel, which is the audio duration.
For 16k dual-channel processing, billing is based on dual channels, which is the number of sound channels × audio duration.
special_word_filter
String (in JSON format)
No
The sensitive word filtering feature. You can enable or disable it and customize sensitive words. This parameter lets you:
Do not process (default, displays the original text), Filter, or Replace with *.
For a specific call example, see the custom word filtering example below.
NoteIf this feature is enabled but no sensitive words are configured, the default word list is filtered: Sensitive word list.
punctuation_mark
String
No
Custom punctuation for sentence segmentation.
If not specified, periods, question marks, and exclamation marks are used for sentence segmentation by default. If you specify a value for this parameter, the specified punctuation marks will also be used for sentence segmentation.
Example:
To segment by English commas, enter ","
To segment by Chinese and English commas, enter "，,"
NoteIf you enter non-punctuation characters, they will not take effect.
This field can contain multiple punctuation marks, and it distinguishes between Chinese and English punctuation. There are no spaces between the punctuation marks.
sentence_max_length
Integer
No
The maximum number of characters to display per sentence. The value must be in the range of [4, 50]. By default, this feature is disabled. If enabled without specifying the number of characters, long sentences will be segmented. This parameter can be used in caption generation scenarios to control the maximum number of characters per line of captions.
The following is an example of a custom word filtering call:
// Take real-time transcription as an example, JSONObject root = new JSONObject(); root.put("system_reserved_filter", true); // Replace the following words with an empty string JSONObject root1 = new JSONObject(); JSONArray array1 = new JSONArray(); array1.add("start"); array1.add("happen"); root1.put("word_list", array1); // Replace the following words with * JSONObject root2 = new JSONObject(); JSONArray array2 = new JSONArray(); array2.add("test"); root2.put("word_list", array2); // You can set all or some of them root.put("filter_with_empty", root1); root.put("filter_with_signed", root2); transcriber.addCustomedParam("special_word_filter", root);The parameters of the ValidTime object are described in the following table.
Parameter
Value type
Required
Description
begin_time
Int
Yes
The start time offset of the valid time period. Unit: milliseconds.
end_time
Int
Yes
The end time offset of the valid time period. Unit: milliseconds.
channel_id
Int
Yes
The sound channel number (starting from 0) to which the valid time period applies.
Output parameters and descriptions:
The server returns a response to the Speech File Transcriber Lite request. The output parameters of the response are in a JSON string:
{ "TaskId": "4b56f0c4b7e611e88f34c33c2a60****", "RequestId": "E4B183CC-6CFE-411E-A547-D877F7BD****", "StatusText": "SUCCESS", "StatusCode": 21050000 }An HTTP status code of 200 indicates a successful operation. For more information about status codes, see HTTP status codes.
Property
Value type
Required
Description
TaskId
String
Yes
The recognition task ID.
RequestId
String
Yes
The request ID, used only for joint debugging.
StatusCode
Int
Yes
The status code.
StatusText
String
Yes
The status description.
Transcription result query API:
After you submit a Speech File Transcriber Lite request, you can poll for the transcription result by setting the following parameters.
Input parameters:
Use the task ID that you obtained from the Speech File Transcriber Lite request as a parameter for the transcription result query API to retrieve the transcription result. You must set an appropriate polling interval when you call the API.
ImportantThe query API has a QPS limit of 100. If you exceed this limit, the error
Throttling.User : Request was denied due to user flow control.may be returned. To avoid this error, set a longer polling interval for the query API.Property
Value type
Required
Description
TaskId
String
Yes
The recognition task ID.
Output parameters and descriptions:
The server returns a response to the transcription result query request. The output parameters of the response are in a JSON string.
Normal response: The following example shows the transcription result for the audio file nls-sample-16k.wav (a single-channel file).
{ "TaskId": "d429dd7dd75711e89305ab6170fe****", "RequestId": "9240D669-6485-4DCC-896A-F8B31F94****", "StatusText": "SUCCESS", "BizDuration": 2956, "SolveTime": 1540363288472, "StatusCode": 21050000, "Result": { "Sentences": [{ "EndTime": 2365, "SilenceDuration": 0, "BeginTime": 340, "Text": "The weather in Beijing.", "ChannelId": 0, "SpeechRate": 177, "EmotionValue": 5.0 }] } }If `enable_callback` and `callback_url` are enabled, the callback transcription result is as follows:
{ "Result": { "Sentences": [{ "EndTime": 2365, "SilenceDuration": 0, "BeginTime": 340, "Text": "The weather in Beijing.", "ChannelId": 0, "SpeechRate": 177, "EmotionValue": 5.0 }] }, "TaskId": "36d01b244ad811e9952db7bb7ed2****", "StatusCode": 21050000, "StatusText": "SUCCESS", "RequestTime": 1553062810452, "SolveTime": 1553062810831, "BizDuration": 2956 }NoteRequestTime is a timestamp in milliseconds (for example, 1553062810452 corresponds to 2019/3/20 14:20:10 UTC+8). It indicates the time when the Speech File Transcriber Lite request was submitted.
SolveTime is a timestamp in milliseconds. It indicates the time when the Speech File Transcriber Lite task was completed.
Queuing:
{ "TaskId": "c7274235b7e611e88f34c33c2a60****", "RequestId": "981AD922-0655-46B0-8C6A-5C836822****", "StatusText": "QUEUEING", "StatusCode": 21050002 }Recognizing:
{ "TaskId": "c7274235b7e611e88f34c33c2a60****", "RequestId": "8E908ED2-867F-457E-82BF-4756194A****", "StatusText": "RUNNING", "BizDuration": 0, "StatusCode": 21050001 }Abnormal response: The following example shows a file download failure.
{ "TaskId": "4cf25b7eb7e711e88f34c33c2a60****", "RequestId": "098BF27C-4CBA-45FF-BD11-3F532F26****", "StatusText": "FILE_DOWNLOAD_FAILED", "BizDuration": 0, "SolveTime": 1536906469146, "StatusCode": 41050002 }NoteFor more information about exceptions, see the error codes and solutions in the Service status codes section below.
An HTTP status code of 200 indicates a successful operation. For more information about status codes, see HTTP status codes.
Property
Value type
Required
Description
TaskId
String
Yes
The recognition task ID.
StatusCode
Int
Yes
The status code.
StatusText
String
Yes
The status description.
RequestId
String
Yes
The request ID, used for debugging.
Result
Object
Yes
The recognition result object.
Sentences
List< SentenceResult >
Yes
The recognition result data. This exists when StatusText is SUCCEED.
Words
List< WordResult >
No
Word-level timestamps. To get this information, set enable_words to true.
BizDuration
Long
Yes
The total duration of the recognized audio file. Unit: milliseconds.
SolveTime
Long
Yes
A timestamp that indicates when the Speech File Transcriber Lite task was completed. Unit: milliseconds.
The parameters of the single-sentence result SentenceResult are as follows.
Property
Value type
Required
Description
ChannelId
Int
Yes
The sound channel ID to which the sentence belongs.
BeginTime
Int
Yes
The start time offset of the sentence. Unit: milliseconds.
EndTime
Int
Yes
The end time offset of the sentence. Unit: milliseconds.
Text
String
Yes
The recognized text of the sentence.
EmotionValue
Float
Yes
The emotional energy value, which is the volume in decibels divided by 10. The value is in the range of [1, 10]. A higher value indicates a stronger emotion.
SilenceDuration
Int
Yes
The duration of silence between this sentence and the previous one. Unit: seconds.
SpeechRate
Int
Yes
The average speech rate of this sentence.
If the recognized language is Chinese, the unit is characters per minute.
If the recognized language is English, the unit is words per minute.
Enabling word-level information:
If `enable_words` is set to true, the transcription result from the server includes word-level timestamps. The word information that is retrieved through polling and callback methods is the same. The following example shows a transcription result from the polling method:
{ "StatusCode": 21050000, "Result": { "Sentences": [{ "SilenceDuration": 0, "EmotionValue": 5.0, "ChannelId": 0, "Text": "The weather in Beijing.", "BeginTime": 340, "EndTime": 2365, "SpeechRate": 177 }], "Words": [{ "ChannelId": 0, "Word": "Beijing", "BeginTime": 640, "EndTime": 940 }, { "ChannelId": 0, "Word": "'s", "BeginTime": 940, "EndTime": 1120 }, { "ChannelId": 0, "Word": "weather", "BeginTime": 1120, "EndTime": 2020 }] }, "SolveTime": 1553236968873, "StatusText": "SUCCESS", "RequestId": "027B126B-4AC8-4C98-9FEC-A031158F****", "TaskId": "b505e78c4c6d11e9a213e11db149****", "BizDuration": 2956 }Words object description:
Property
Value type
Required
Description
BeginTime
Int
Yes
The start time of the word. Unit: milliseconds.
EndTime
Int
Yes
The end time of the word. Unit: milliseconds.
ChannelId
Int
Yes
The sound channel ID to which the word belongs.
Word
String
Yes
The word information.
Service status codes
General-purpose error codes
Status code | Status message | Cause | Solution |
40000000 | The default client error code. This code corresponds to multiple error messages. | Invalid parameters or call logic was used. | Compare your code with the sample code in the official documentation to test and verify it. |
40000001 | The token 'xxx' has expired. The token 'xxx' is invalid | Invalid parameters or call logic was used. This is a general-purpose client error code that usually indicates an incorrect token, such as an expired or invalid token. | Compare your code with the sample code in the official documentation to test and verify it. |
40000002 | Gateway:MESSAGE_INVALID:Can't process message in state'FAILED'! | The message is invalid or incorrect. | Compare your code with the sample code in the official documentation to test and verify it. |
40000003 | PARAMETER_INVALID Failed to decode url params | The parameters passed by the user are incorrect. This error is common for RESTful API calls. | Compare your code with the sample code in the official documentation to test and verify it. |
40000005 | Gateway:TOO_MANY_REQUESTS:Too many requests! | Too many concurrent requests. | If you are using the Free Edition, you can upgrade to a commercial version to increase the concurrency. If you are already using a commercial version, you can purchase a concurrency resource plan to increase your concurrency quota. |
40000009 | Invalid wav header! | The message header is invalid. | If you send a WAV audio file and set the |
40000009 | Too large wav header! | The WAV header of the transmitted audio is invalid. | You can send the audio stream in a format such as PCM or OPUS. If you use the WAV format, make sure that the WAV header of the audio file contains the correct data length. |
40000010 | Gateway:FREE_TRIAL_EXPIRED:The free trial has expired! | The trial period has ended, and the commercial version is not activated or your account has an overdue payment. | You can log on to the console to check the service activation status and your account balance. |
40010001 | Gateway:NAMESPACE_NOT_FOUND:RESTful url path illegal | The operation or parameter is not supported. | Check whether the parameters passed in the call are consistent with the requirements in the official documentation. You can compare them with the error message to identify and set the correct parameters. For example, if you are using a curl command to make a RESTful API request, check whether the URL you constructed is valid. |
40010003 | Gateway:DIRECTIVE_INVALID:[xxx] | A general-purpose client-side error code. | This error indicates that the client passed an incorrect parameter or instruction. Detailed error messages are available for different operations. You can refer to the corresponding documentation to set the parameters correctly. |
40010004 | Gateway:CLIENT_DISCONNECT:Client disconnected before task finished! | The client actively terminated the connection before the request was processed. | None. Alternatively, you can close the connection after the server responds. |
40010005 | Gateway:TASK_STATE_ERROR:Got stop directive while task is stopping! | The client sent a message instruction that is not currently supported. | Compare your code with the sample code in the official documentation to test and verify it. |
40020105 | Meta:APPKEY_NOT_EXIST:Appkey not exist! | A non-existent Appkey was used. | Confirm whether a non-existent Appkey was used. You can log on to the console and view the project configuration to find the Appkey. |
40020106 | Meta:APPKEY_UID_MISMATCH:Appkey and user mismatch! | The Appkey and token passed in the call were not created by the same Alibaba Cloud account UID. This causes a mismatch. | Check whether you are using resources from two different accounts. Do not use an Appkey from Account A with a token generated from Account B. |
403 | Forbidden | The token is invalid. For example, the token does not exist or has expired. | Set a valid token. Tokens have an expiration period. You must obtain a new token before the current one expires. |
41000003 | MetaInfo doesn't have end point info | Failed to retrieve the routing information for this Appkey. | Check whether you are using resources from two different accounts. Do not use an Appkey from Account A with a token generated from Account B. |
41010101 | UNSUPPORTED_SAMPLE_RATE | The sample rate is not supported. | Real-time speech recognition currently supports only audio with a sample rate of 8000 Hz or 16000 Hz. |
41040201 | Realtime:GET_CLIENT_DATA_TIMEOUT:Client data does not send continuously! | Failed to retrieve data from the client due to a timeout. | When you call real-time speech recognition, the client must send data at a real-time rate and close the connection promptly after the data is sent. |
50000000 | GRPC_ERROR:Grpc error! | An exception caused by factors such as machine load or network issues. This error usually occurs randomly. | You can retry the call to resolve the issue. |
50000001 | GRPC_ERROR:Grpc error! | An exception caused by factors such as machine load or network issues. This error usually occurs randomly. | You can retry the call to resolve the issue. |
52010001 | GRPC_ERROR:Grpc error! | An exception caused by factors such as machine load or network issues. This error usually occurs randomly. | You can retry the call to resolve the issue. |
Audio file recognition/Audio file recognition (off-peak) error codes
Status code | Status message | Cause | Solution |
21050000 | SUCCESS | Success. | None. |
21050001 | RUNNING | The audio file recognition task is running. | Send a GET request to query the recognition result later. |
21050002 | QUEUEING | The audio file recognition task is in the queue. | Send a GET request to query the recognition result later. |
21050003 | SUCCESS_WITH_NO_VALID_FRAGMENT | The operation to query the recognition result was successful, but the voice activity detection (VAD) module did not detect any valid speech. | In this case, check the following: Check whether the audio file contains valid speech. If it contains only invalid speech, such as pure silence, it is normal for no recognition result to be returned. |
ASR_RESPONSE_HAVE_NO_WORDS | The operation to query the recognition result was successful, but the final recognition result is empty. | In this case, check the following: Check whether the audio file contains valid speech, or whether the valid speech consists only of filler words and the enable_disfluency parameter is enabled, causing the filler words to be filtered out. In these cases, it is normal for no recognition result to be returned. | |
41050001 | USER_BIZDURATION_QUOTA_EXCEED | The daily duration limit is exceeded. Free Edition users can recognize audio files with a total duration of no more than 2 hours per day. | Upgrade from the Free Edition to the commercial version. If you have a high volume of business, contact our business team by email at nls_support@service.aliyun.com. |
41050002 | FILE_DOWNLOAD_FAILED | File download failed. | Check whether the audio file path is correct and whether the file can be accessed and downloaded from the internet. |
41050003 | FILE_CHECK_FAILED | The file format is incorrect. | Check whether the audio file is in single-channel or dual-channel WAV or MP3 format. |
41050004 | FILE_TOO_LARGE | The file is too large. | Check if the audio file size exceeds 512 MB. If it does, segment the audio file. |
41050005 | FILE_NORMALIZE_FAILED | File normalization failed. | Check whether the audio file is damaged and whether it can be played normally. |
41050006 | FILE_PARSE_FAILED | File parsing failed. | Check whether the audio file is damaged and whether it can be played normally. |
41050007 | MKV_PARSE_FAILED | MKV parsing failed. | Check whether the audio file is damaged and whether it can be played normally. |
41050008 | UNSUPPORTED_SAMPLE_RATE | The sample rate does not match. | Check whether the actual audio sample rate is consistent with the sample rate of the ASR model bound to the Appkey in the console, or set the automatic downsampling parameter enable_sample_rate_adaptive in this document to true. |
41050010 | FILE_TRANS_TASK_EXPIRED | The audio file recognition task has expired. | The TaskId does not exist or has expired. |
41050011 | REQUEST_INVALID_FILE_URL_VALUE | The file_link parameter in the request is invalid. | Confirm whether the format of the file_link parameter is correct. |
41050012 | REQUEST_INVALID_CALLBACK_VALUE | The callback_url parameter in the request is invalid. | Confirm whether the format of the callback_url parameter is correct and whether it is empty. |
41050013 | REQUEST_PARAMETER_INVALID | The request parameter is invalid. | Confirm that the task value in the request is a valid JSON format string. |
41050014 | REQUEST_EMPTY_APPKEY_VALUE | The appkey parameter value in the request is empty. | Confirm whether the appkey parameter value is set. |
41050015 | REQUEST_APPKEY_UNREGISTERED | The appkey in the request parameter is not registered. | Confirm whether the appkey parameter value in the request is set correctly, or whether it belongs to the same account as the AccessKey ID of the Alibaba Cloud account. |
41050021 | RAM_CHECK_FAILED | RAM check failed. | Check whether your RAM user is authorized to call the API operations of Voice Service. For more information, see Configure RAM user permissions. |
41050023 | CONTENT_LENGTH_CHECK_FAILED | content-length check failed. | When downloading a file, check whether the content-length in the HTTP response is consistent with the actual file size. |
41050024 | FILE_404_NOT_FOUND | The file to be downloaded does not exist. | Check whether the file to be downloaded exists. |
41050025 | FILE_403_FORBIDDEN | You do not have permission to download the required file. | Check whether you have permission to download the audio file. |
41050026 | FILE_SERVER_ERROR | The service where the requested file is located is unavailable. | Check whether the service where the requested file is located is available. |
41050103 | AUDIO_DURATION_TOO_LONG | The duration of the requested file exceeds 12 hours. | Segment the audio and submit multiple recognition tasks. For more information, see Segmentation command reference. |
40270003 | DECODER_ERROR | Failed to detect audio file information. | Confirm that the file at the download link is in a supported audio format. |
51050000 | INTERNAL_ERROR | An exception caused by factors such as machine load or network issues. This error usually occurs randomly. | Retry the call to resolve the issue. If the issue persists, contact technical support. |