Can I Convert 128 Kbps To 320 Kbps?

Not all audio files are created equal. The same song can be encoded in many ways, particularly at different bit rates. In digital audio, the bit rate is one of the most significant determiners of audio quality. In light of this, is it possible to increase a track’s bit rate?

Converting audio with a bit rate of 128 Kbps to 320 Kbps is possible but never advantageous. To recover more information about the waveform, additional information is required. Audio file converters can only generate redundant data, maintaining or reducing audio fidelity.

This might seem odd: bit rate almost always means better fidelity. Let’s examine the crucial factors, unpack what’s happening, where we might want to change the bit rate, and look behind the scenes.

Can The Bit Rate Of Audio Be Increased?

You can usually increase the bit rate of audio files up to the format’s maximum bit rate, for instance:

  • MP3: 320 kbps
  • AAC: 320 kbps
  • Opus: 510 kbps

Many online audio converters, such as ACONVERT and Online Audio Converter, allow you to specify the output bit rate within a range. However, picking a higher bit rate than your source audio may not give the expected result.

Re-encoding an audio file at a higher bit rate is rarely desirable. It increases the size of your file without any gain in quality.

The Data: Increasing The Bit Rate Of Audio

We’ve conducted a small experiment using FFmpeg to demonstrate the effect of increasing the bit rate of an audio file. The Opus format is used as it supports a wide range of bit rates from 6 kbps to 510 kbps.

  • Start with an audio file. Let’s call it “Song.opus”
  • Reduce its bit rate to 8 kbps, save as “Song_8k.opus” (this audibly worsens audio quality)
  • Finally, re-encode “Song_8k.opus” at 510 kbps, and save as “Song_510k.opus”

The final step performs an extreme version of increasing the bitrate from 128 kbps to 320 kbps, going from 8 kbps to 510 kbps. While the original file is not relevant to our conclusion, it’s included for reference. All in all, the results are clear-cut:

File Name bit rate (kbps) File Size (MiB) Fidelity
Song.opus 129 3.4 Good
Song_8k.opus 7 0.21 Terrible
Song_510k.opus 480 12.8 Terrible

The Opus codec has saved some extra data compared to what we specified. Nonetheless, there was no audible difference between the 7 kbps and 480 kbps audio, yet the 480 kbps track takes up much more space. This is what we expected, but what exactly is going on here?

Are Higher Bit Rates Always Better Quality?

A higher audio bit rate does not necessarily mean better quality. However, good quality requires a high bit rate. In other words, the number of bits per second is a soft limit of information capacity. Increasing the capacity of a container does not mean the contents of the container will automatically be greater.

Imagine you are given a large cup full of water. For the sake of analogy, water is information, and the cup is water. To save space, you might slice the cup such that it’s shorter and easier to store. However, the extra water above the slice spills out. Taping the cup back together again to its original capacity does not change the new level of the water.

After some unique audio has been synthesized or recorded, the resulting file contains the most accurate information about the original waveform that will ever exist. If the audio in that file is re-encoded, it decreases or remains equally representative.

The original recording is as high-fidelity as the audio will ever get, so you should always convert your audio as few steps removed from the source as possible. For instance, if you need an MP3 version for an old device or an Opus version for a small drive, convert from the original, which might be a downloaded file from a store.

What Data Is Generated By Increasing Bitrate?

The data generated by increasing the bitrate is redundant. For example, one could write the number 1 as 000001.00000, which is mathematically equivalent. The new representation doesn’t add any information, but it presents more data.

In the case of audio tracks, let’s imagine you have an audio file with a sequence of audio samples at 10kHz. Your device might default to playing 40kHz, so the software must generate extras to match. The software will connect the samples with a smooth curve and sample that curve in between the originals to match the sample rate.

Thus, if we increase the file’s sample rate fourfold from 10kHz to 40kHz, the encoder will use the same process, modeling a smooth curve between the samples and then sampling the curve between the original samples. This way, the audio stream does not sound different, but more data is occupied. No information has been added, but more data is used.

Where bit rate is of primary concern (rather than sample frequency and bit depth), lossy compression is usually the subject. Modern lossy codecs don’t encode the audio as samples but instead usually banks of frequencies. The result is similar when increasing the bit rate, though. For instance, many extra near-zero-amplitude frequencies may be added.

How To Modify The Bit Rate Of An Audio File

Modifying the bit rate of a file isn’t too tricky, as there are multiple suitable tools, both online and offline, that will do the job:

FFmpeg may appear daunting to use, being a command-line program. However, it’s the fastest option, especially for many or large files. It’s free, open source, and entirely local (no need to upload and download your audio). Once you have it installed, edit and run the following command:

ffmpeg -i ./path/to/file.ext -b:a 320k -minrate 256k -maxrate 400k ./path/to/newfile.ext

  • Replace the values, filenames, and extensions (e.g., “.mp3”) according to your needs
  • FFmpeg will convert the format automatically if the extensions do not match
  • The “minrate” and “maxrate” parameters are not necessary but can help further specify the bounds used for variable-rate encoding, which almost all modern codecs use

You can inspect the metadata of audio files using “ffprobe ./path/to/file.ext” which provides lots of information, including:

  • Bit rate
  • Format
  • Length
  • Encoding
  • Sample rate
  • Et cetera

Can Audio Fidelity Be Improved?

The audio fidelity of a track cannot be improved. Additional information could help, but this is more of a theoretical technicality. Data encodes information, and the more important information you have about an audio track, the better the original waveform can be reproduced and played back.

Lossy formats โ€“ like MP3, AAC, and Opus โ€“ discard information to achieve low bit rates. A given bit rate is precisely the number of bits per second of audio. Thus, lower bit rates save storage space and network bandwidth.

The encoding algorithm cleverly discards the least essential information about the source waveform. Although, below 96 kbps, compression artifacts and muddied sounds often become more easily perceptible. Above 256 kbps, the loss is almost entirely transparent across all modern lossy codecs, compared to lossless audio.

Streaming and distribution services for video, music, and podcasts will compress audio data efficiently to save bandwidth and maximize quality. As a result, inefficiently compressed audio is uncommon. However, jacking up the bit rate while performing audio conversions is the easiest way to end up with redundant bits.

Conclusion

While increasing the bit rate is possible, it will never improve the quality of the audio, and it will take up more space. When converting your audio files, always set the output bit rate to approximately equal to or less than the source. The best you can do to preserve audio quality is to keep the original or non-lossy transcoding.

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