Sample Rate and Bit Depth Checker – Bit Rate (Bitrate) Test
Overview
Sample Rate and Bit Depth Checker. Drop an audio file into the Sample Rate and Bit Depth Checker and it reads the file's real sample rate, bit depth, bit rate and channels, draws a spectrogram and flags upsampled or lossy-sourced files — all locally, nothing is uploaded. Below that, Check My Audio Pipeline and Check Microphone Input report the sample rate, latency and channel setup your browser is using right now. Use the tone generator online to produce a clean signal in seconds, right in your browser.
Drop a track into this sample rate and bit depth checker and, in seconds, you see the true bitrate, codec and resolution hiding behind the filename. Think of it as an audio checker that runs in your browser: it reads the file, draws a spectrogram and shows whether a "lossless" FLAC or "hi-res" 96 kHz WAV is genuine, or a lossy copy in disguise. Your music, your verdict.
Sample Rate and Bit Depth Checker for Any Audio File
Choose any audio file and this free online tool reports its sample rate, bit depth, bit rate, channels and length. You can drag and drop the file straight into the page, one file at a time. Because the analysis runs in your browser, there is no software to install and no account to create. Producers use it to confirm a delivery spec; listeners use it to confirm they got what they paid for. The same sample rate and bit depth readout also tells you how large the file should be. The online mosquito tone check takes under a minute and needs nothing but your browser.
Upload a File and Run the Check
- Upload a track by dragging it onto the drop area. WAV, MP3, AIFF, ALAC, OGG, Opus and M4A all work.
- Wait a few seconds while the tool decodes the audio; press Analyze File to run it again.
- Read the report: format, sample rate, bit depth, bitrate and a pass, warning or fail verdict for each quality check.
Reading the File Summary
The file summary is the first block of the report. It lists the data every other check builds on:
- Codec and format – the container (WAV, FLAC, M4A) and the codec inside it.
- Channels – one channel is mono, two is stereo, more means a surround mix.
- Duration and file size – used to work out the average bit rate.
- Metadata – tags such as title and encoder, which are easy to edit and not proof of quality.
What Sampling Frequency Means in Digital Audio
Digital audio turns a continuous analog sound wave into numbers, and every digital file, from a phone memo to a studio master, is built that way. Thousands of times every second, the amplitude of the waveform is measured, taking snapshots of its level. Each snapshot is quantized to the nearest available value and stored as binary bits. The sampling frequency, counted in samples per second, is written in hertz or kilohertz. It sets the highest frequency the file can hold: exactly half the rate. Run the timing discrimination test online for a few minutes a day to build real listening skill.
$$f_{max} = \frac{f_s}{2}$$
Human hearing tops out near 20 kHz, so anything above that is inaudible, but the ceiling still matters when you check a file for fakery.
Choosing a Sample Rate for Recording
For recording and music production, most sessions use 44.1 kHz or 48 kHz; the choice mostly follows the destination. Unlike a video frame rate, a higher number does not make the signal "smoother" – it only moves the ceiling up. Pick the audio sample rate your streaming or delivery target expects, and keep the same audio sample rate through mixing to avoid conversion in the digital chain.
44.1 kHz vs. 48 kHz
44.1 kHz is the CD quality standard, with a ceiling of 22.05 kHz. 48 kHz is the norm for video and broadcast. Both cover the audible range, so the audible difference is negligible.
Higher Rates and Aliasing
Hi-res formats run at 96 kHz or 192 kHz. They push aliasing – false tones created when frequencies above the ceiling fold back down – far out of the audible band, which eases filter design. Dither is a tiny noise added when reducing word length. Higher rates cost file size and CPU, which is why a high resolution claim deserves a check.
How Resolution Shapes the Noise Floor
Where sample rate sets the highest frequency, resolution in the bit-depth sense sets how finely each sample's level is measured. More bits mean smaller rounding steps, so the noise floor drops and the dynamic range grows. In digital audio, rounding error, the gap between the analog original and the stored value, shows up as distortion at low levels, and the quality of later processing depends on that headroom.
16-bit vs. 24-bit Audio Bit Depth
Each added bit gives roughly 6 dB of range. 16-bit audio offers about 96 dB, enough for finished music. 24-bit gives about 144 dB, which is why it is used for recording. A 32-bit float file adds headroom for editing, while loudness is set by the mix and not by the bit depth.
Bitrate Checker: True Bit Rate vs. Stated Rate
A bitrate checker compares what the label says with what the data supports. The number in the tag is a claim: a 128 kbps file re-saved at 320 kbps reports 320 but keeps the damage. For uncompressed audio the bit rate follows directly from the settings, so an audio bitrate checker needs only the sample rate, bit depth and channel count. For a lossy file, audio quality is judged from its frequency content instead.
CBR vs. VBR Behavior
A CBR encoder spends the same bits every second; a VBR encoder spends more on busy passages, so its variable bitrate changes through the track, and its cutoff can step up and down. AAC and MP3 both use either mode, so one number rarely tells the whole story. The tool reports the average figure for such a file, and a stepping edge in its spectrogram is normal there, not a sign of tampering.
Audio Checker Tips for Spotting Fake Lossless Files
A fake file is usually a lossy source transcoded into a lossless or hi-res container. Every lossy-to-lossless transcode keeps the old losses, and names and extensions can be changed in seconds, but lost information never comes back. Use the audio checker to compare the claimed lossless audio against the evidence, and treat lossy audio dressed up as lossless as a refund request.
How to Tell if a FLAC Is Lossless
A genuine lossless FLAC holds frequency content close to its ceiling and a ragged, shifting upper edge. A suspicious one has a hard, flat shelf instead – the signature of a lossy source and upsampling. Check the audio quality against the file size too: a "lossless" file much smaller than the maths below predicts was probably compressed from something else.
Reading the Spectrogram and Nyquist Cutoff
The spectrogram plots frequency against time, with brighter colour for stronger signal. The Nyquist limit is the top edge of the chart. A natural recording rolls off gradually, while a frequency cutoff drawn as a ruler-straight line, held for the whole track, means the encoder removed everything above it. The frequency spectrum, bandwidth and frequency range together show how much of the available range holds real content.
Worked Example: Checking a 96 kHz Track With a File Analyzer
Say a download is labelled 96 kHz, 24-bit stereo and runs 4 minutes 12 seconds (252 seconds). An audio file analyzer reports a claimed bit rate and then, through sound analysis of the spectrum, the real content. The table lays out both readings.
| Measure | As labelled | If it were a true 44.1 kHz, 16-bit master |
|---|---|---|
| Sample rate | 96 kHz | 44.1 kHz |
| Bit depth | 24-bit | 16-bit |
| Bitrate | 4,608 kbps | 1,411.2 kbps |
| File size (252 s) | 145.15 MB | 44.45 MB |
| Nyquist ceiling | 48 kHz | 22.05 kHz |
Calculating Bitrate and File Size
Multiply the sample rate, the bit depth and the channel count to get the uncompressed bit rate:
$$\text{bit rate} = 96{,}000 \times 24 \times 2 = 4{,}608{,}000\ \text{bps}$$
Then multiply by the duration and divide by 8 for bytes:
$$\text{size} = \frac{4{,}608{,}000 \times 252}{8} = 145{,}152{,}000\ \text{bytes}$$
The labelled file is therefore about 3.3 times larger than the 44.1 kHz, 16-bit version of the same recording.
Spotting an Upsampled Cutoff
Now the spectrum: the analyzer finds content ending at 21.9 kHz against a 48 kHz Nyquist ceiling, so only 21.9 ÷ 48 = 45.6% of the bandwidth holds real audio. Everything above sits near silence, a clear sign the track was upsampled from a 44.1 kHz source. The extra size is padding, not detail.
Bit Depth Checker Walkthrough: A Podcast Interview That Came Back at 16-Bit
I'm editing a podcast episode for a network whose delivery spec asks for 48 kHz, 24-bit mono WAVs. The guest sent a 38 minute 20 second interview (2,300 seconds) and swears it was recorded to spec. Before I spend an evening cutting it, I want to know whether that is true.
I drop the WAV into the checker and read the file summary: 48 kHz, 16-bit, mono, 220.8 MB. The sample rate matches, but the bit depth does not. The sums agree with the tool: 48,000 × 16 × 1 = 768,000 bps, and 768,000 × 2,300 ÷ 8 = 220,800,000 bytes. A true 24-bit file would be 1,152 kbps and 331.2 MB, so the guest's recorder was set to 16-bit.
| Setting (mono, 2,300 s) | Bit rate | File size |
|---|---|---|
| 48 kHz, 24-bit (spec) | 1,152 kbps | 331.2 MB |
| 48 kHz, 16-bit (received) | 768 kbps | 220.8 MB |
| 44.1 kHz, 16-bit (CD standard) | 705.6 kbps | 202.86 MB |
Here is the catch that makes a file-size check alone unreliable: 32 kHz at 24-bit also lands on 768 kbps, so a 220.8 MB WAV could be either. Only the sample rate and bit depth fields tell them apart, and they confirm the rate is right. If the 16-bit recording were a 24-bit file padded with zeros, the size would be 331.2 MB, so it is not.
The judgement call: the number to watch is the noise floor. Sixteen bits give about 96 dB of range against 144 dB for 24-bit, and a quiet room voice with gentle gain sits comfortably inside 96 dB, so I don't ask for a re-record. I convert nothing, since stretching a 16-bit file to 24-bit adds no detail, and I note in the session that the source is 16-bit. If the guest were recorded at low level and I planned heavy noise reduction, I would ask for the original at 24-bit first, because that extra headroom is what the processing leans on.
Audio Analysis Features: Dynamic Range, Clipping and Levels
Beyond the format fields, audio analysis adds level checks that explain how the track sounds. The dynamic range check uses the crest factor, the gap between the peak and the average (RMS) level, to show how hard the mix was compressed; a crest factor above about 14 dB breathes naturally. Clipping is flagged when samples hit the 0 dBFS ceiling, adding audible distortion. The peak shown is the sample peak; a true peak meter, which oversamples, can read slightly higher. Each of these checks gets its own pass, warning or fail next to the format fields, since word length sets how much range the track can hold.
Peak, RMS and Stereo Balance
Peak level is the loudest single sample and RMS tracks perceived average loudness. The channel detail in the report shows these values, and a sound analyzer also compares left and right: music with a stereo correlation near 1.0 is close to mono, while negative values mean out-of-phase audio that can vanish on a mono speaker.
Who Uses a Sound Analyzer: Producers and Podcasters
Each group reads a different field. Producers and engineers confirm a delivery spec on stems and masters before mastering or loading a DAW session. Podcasters read the format fields to confirm every podcast episode meets a host's spec. Archivists check the sample rate of digitised recordings, and collectors use the spectrogram readout to catch inflated hi-res purchases.
Is a Free Audio Analyzer Private?
Privacy depends on the software. This checker decodes your file locally and reads every format field on your own device, so nothing is uploaded or stored and an unreleased master stays private. If you use another report service, check whether the file leaves your machine first.