Audio Sample Rate Converter

Supports batch audio sample rate conversion, allowing you to adjust the sample rate of MP3, WAV, AAC, and other audio formats to improve sound quality.

Bug report

Browser execution mode: Your data is processed in your browser and is not uploaded to the server.

Speed and Stability: Processing speed depends on your device and browser. For large batch work, the desktop version may be more stable.

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If the online tool fails to load or run, try the desktop tool.https://tools.yikeaigc.com/

Tool Usage

Return to old version

A desktop version is available with more powerful, stable features. Open desktop version

Import audio and set the sample rate before processing. All batch files will be converted. Only the first 20 are shown for review. Output files use original names by default; duplicates get numbers.

Waiting for local transcoding core

Choose Audio File

Supports common audio formats such as MP3, WAV, FLAC, AAC, M4A, OGG, OPUS, WMA, AIFF, AMR, and CAF.

Click to choose audio, or drag files here Files are converted on this device. No server upload needed.

Review the file list before setting parameters.

File Review

Only the first 20 files are shown. All selected files will be converted.

File Count

0 KB

Total Size

Shown

Not Shown

Choose audio files or load a sample to view the conversion flow.

Sample Rate & Output

44.1 kHz suits music, 48 kHz suits video audio, and 16 kHz is common for speech.

Target Sample Rate

Custom sample rate Hz

Output Format

Compressed Bitrate

MP3 Encoding Mode

MP3 VBR Quality

WAV Bit Depth

FLAC Compression Level

Channels

Processing Options

Keep tags, normalize loudness, adjust gain, and generate batch reports as needed.

Keep Audio TagsKeep title, artist, album, and other info when possible Loudness NormalizationGreat for unifying volume from different sources in batches Generate Processing ReportLog parameters, successful files, and failure reasons ZIP compressionEnable compression when packaging downloads

Volume gain

100%

Resampling strategy

High-pass filter

Low-pass filter

ZIP file name

Set parameters before converting and downloading.

Convert & download

Confirm parameters to start converting. The first 20 results can be previewed. The ZIP includes all successful results.

Pending0

Target Sample Rate44.1 kHz

Output FormatMP3

Result count0

Set parameters first, then start processing.

Instructions

Software Instructions

Select audio files: Click the selection area or drag audio files into the tool. Common formats such as MP3, WAV, FLAC, AAC, M4A, OGG, OPUS, WMA, AIFF, AMR, and CAF are supported, and you can select multiple files at once.
Check the file list: The tool will display the names, formats, and sizes of the first 20 files. If there are more than 20 files, a text prompt will be shown, and all selected files will be processed during the actual conversion.
Set the target sample rate:
- 44.1 kHz: Suitable for music playback and organizing CD-quality audio.
- 48 kHz: Suitable for video dubbing, editing projects, and common digital audio projects.
- 16 kHz: Suitable for speech materials, meeting recordings, and transcription content.
- 96 kHz: Suitable for WAV or FLAC files that need to retain a higher sampling specification.
- Custom: You can enter a sample rate from 8000 to 192000 Hz.
Select the output format:
- MP3: Highly compatible, suitable for everyday playback and sharing.
- WAV: PCM audio, suitable for post-production editing and audio engineering.
- FLAC: Lossless compression, suitable for audio archiving.
- M4A/AAC/OGG: Suitable for mobile devices, web audio, or compressed output.
Configure audio parameters: You can set the bitrate, MP3 encoding mode, WAV bit depth, FLAC compression level, channels, volume gain, loudness normalization, high-pass filter, low-pass filter, and ZIP file name.
Save common settings: Click “Save Settings” to keep the current parameters and continue using them the next time you open the tool; click “Restore Defaults” to return to the initial configuration.
Start conversion: Go to the “Convert & Download” step, confirm the number of files to process, the target sample rate, and the output format, then click “Start Conversion”.
Preview and download: After conversion is complete, you can preview the first 20 results, download an individual file, or click “Download All Results ZIP” to get all successfully converted audio files.

FAQ

A: 44.1 kHz is commonly used for music playback and CD-quality audio; 48 kHz is commonly used for video dubbing, editing projects, and digital audio projects. If the audio will be used with video, 48 kHz is usually recommended.

A: It will not create sound details that do not exist in the original file. Upsampling is mainly used to standardize project specifications or meet device requirements. If the source file itself is low quality, it is recommended to keep a reasonable sample rate first and choose an appropriate output format.

A: Result files use the original file name by default, with only the extension replaced. If multiple files have the same name after conversion, the tool automatically adds a number to the file name to prevent overwriting.

A: Different encoding formats support different sample rate ranges. MP3 commonly uses standard sample rates from 8 kHz to 48 kHz, while OGG Opus commonly uses 8, 12, 16, 24, and 48 kHz. If you need 88.2 kHz, 96 kHz, or higher sample rates, WAV or FLAC is recommended.

A: 16-bit PCM is suitable for everyday playback and general delivery; 24-bit PCM is suitable for post-production editing and material that requires a wider dynamic range; 32-bit Float is suitable for professional editing workflows, but the file size will be larger.

A: Loudness normalization makes audio from different sources play at more consistent volumes, making it suitable for batch organizing course audio, podcasts, meeting recordings, and voice-over materials. If the original file has already been mixed, you can also leave it turned off.

A: Common causes include corrupted files, unsupported encoding, insufficient memory due to overly large files, or incompatibility between the target format and sample rate combination. You can try switching to WAV or FLAC output, lowering the sample rate, processing files in batches, or confirming that the source file plays normally.