This tool allows you to compare the audio quality of different codecs used in VoIP telecommunications. Each codec compresses audio differently, balancing file size against audio quality. The files are produced by transcoding the source file using ffmpeg and some tools for specific codecs.
Use the codec buttons to switch between different codecs while listening to a sample from wikipedia.
Tip: Try switching between high-bitrate codecs (like g722 at 64 Kbps) and low-bitrate codecs (like codec2 at 3.2 Kbps) to hear the difference compression makes.
The visualization above shows a real-time frequency spectrum of the audio. Each vertical red bar represents the intensity of different frequencies in the sound:
Lower-quality codecs often show reduced detail in the high frequencies, which affects clarity. Watch how the spectrum changes as you switch between codecs—you may notice some codecs cut off higher frequencies to save bandwidth.
| Codec | Year | Sample rate | Filesize | Average bitrate | Play |
|---|
A codec (coder-decoder) compresses audio to reduce file size and bandwidth usage. In VoIP calls, lower bandwidth means calls work better on slow connections, but may sacrifice audio quality.
To learn more about each codec: Click the ▸ button next to any codec name in the table above to see its description, history, and characteristics. Hover over the column headers (Year, Sample rate, Filesize, Average bitrate) for explanations of each metric.
Pro tip: Try comparing codecs with similar bitrates (e.g., GSM vs. iLBC vs. opus-nb around 13-15 Kbps) to see how modern codecs (Opus) outperform older designs.
Use filters to isolate different parts of the audio spectrum and hear how each codec handles them:
Current filter: allpass
Note: The filter is applied during playback and doesn't affect the codec comparison itself. It's a tool to help you focus on specific aspects of audio quality.