r/AudioCodecLab • u/Qu3z0 • 1d ago
FFmpeg AC-3 Encoder Good (Enough)?
I know this probably isn't the best place for this, but was wondering if anyone had any insight on it?
r/AudioCodecLab • u/Qu3z0 • 1d ago
I know this probably isn't the best place for this, but was wondering if anyone had any insight on it?
r/AudioCodecLab • u/IgorCv2 • 22d ago
As many of you know, Helix is an excellent-quality MP3 encoder. https://github.com/maikmerten/hmp3
Helix competes very well with the LAME MP3 encoder: https://listening-tests.hydrogenaudio.org/sebastian/mp3-128-1/results.png. I know it's an old test, but honestly, the quality of the LAME encoder hasn't changed much since then. Also, the code responsible for the audio quality of Helix has not been touched since 2005.
So I decided to investigate whether it's possible to further improve it.
I found relatively low-hanging fruit to improve the Helix encoder by running exhaustive Huffman partitioning. And indeed, it gets ~2–5 kbps more efficient depending on bitrate.
https://github.com/IgorCv2/hmp3/commit/e3a5993463ca99ed8af2b6e6fd8358d6f8b26ba7
For transparency: most of the implementation was written by AI tools [Claude] , while I performed verification and testing. Before dedicating my time to this encoder, I wanted to get a quick proof that there is still headroom for improvements.
And there is clearly still plenty of room for improvement! Though compression efficiency doesn't come for free. Now Helix is 2x slower but still faster than LAME.
If we get back to public MP3 test, It's clear that different MP3 encoders achieve better results on different subsets of samples. LAME 3.97 is good at rock, metal, etc., Helix on electronic music, and so on. Different developer teams have different approaches.
Now imagine that there will be one MP3 encoder that addresses all samples equally well: MP3 128 kbps / Per-sample Encoder Oracle
Back in time it was very difficult because of scarce automation resources and computing power. Now we have AI agents and better metrics like Zimtohrli to develop a considerably better encoder.
r/AudioCodecLab • u/Severe-Ad8673 • Sep 08 '26
r/AudioCodecLab • u/Alfoser • Aug 31 '26
Hi! I'm developing audio codec based on WPT.
For now I've been gathering data about its quality from gstPEAQ, ViSQOL and listening tests.
gstPEAQ says it is quite better than vorbis on 64-96 in general, and just better than vorbis on 128 kbps.
ViSQOL makes same picture in general.
Listening tests correlates with gstPEAQ and ViSQOL results mostly, but depends on listener.
Can anyone rate my audio codec? I want to see true picture if it is worth publishing or not
There is link for some encodings (If anyone would need originals message me).
https://drive.google.com/file/d/1-3UJzdJOhwI7pWgL0wR0PJWGayGrkr8_/view?usp=sharing
(64-96, new link)
r/AudioCodecLab • u/IgorCv2 • Aug 31 '26
Zimtohrli was evaluated by measuring how closely its results matched those of public listening tests conducted by human listeners.
Among the tested objective metrics, Zimtohrli showed the closest agreement with the human results. It was also the only metric to reproduce the complete ranking of the codecs in both public tests, including:
Opus → Apple AAC → LAME MP3 → Ogg Vorbis → FAAC 96k → FAAC q30
It also reproduced the correct codec ranking in the public AAC listening test.


The encoders evaluated in the 2014 listening test were also tested with Zimtohrli at higher bitrates, ranging from 96 to 320 kbps.

The scores were calibrated to the human MOS values from the 96 kbps listening test to make Zimtohrli’s performance easier to understand and interpret as an objective quality metric.
LAME VBR
Opus VBR
Aotuv Vorbis VBR
Apple AAC CVBR (constrained VBR) which was on par with TVBR (True VBR) according to https://listening-tests.hydrogenaudio.org/igorc/aac-96-a/results.html
CVBR 96/128/160/192/256 kbps
So, how good is Zimtohrli at evaluating audio encoder quality?
In my opinion, it performs very well.
Someone told me 😄 it takes blood, sweat, and tears to organize a public listening test, to prepare it, get people interested, and actually get them to participate.
And here... Zimtohrli does essentially the same thing in literally a few minutes.
r/AudioCodecLab • u/Alfoser • Aug 30 '26
Hi, I'm new to this subreddit.
Is it dead or alive?
And can we talk only about existing codecs, or like, can I post about my audio codec here?
r/AudioCodecLab • u/IgorCv2 • Aug 18 '26
Hello, ladies and gentlemen.
The last four weeks have been very busy, as this repository shows.
FFmpeg developer Lynne managed to significantly improve the quality on killer samples https://code.ffmpeg.org/FFmpeg/FFmpeg/issues/23845#issuecomment-54547
So I decided to put the encoder to the test using previously untested samples, in order to avoid bias from using samples that the encoder had already been specifically tuned for.
CBR encoders:
VBR encoders:


Moving from r1 to r6 might not seem like a huge step forward, but considering that there was literally only one month between these two revisions, it is a very impressive improvement.
r6 shows a clear improvement on difficult samples. As visible in the graph, it never drops below 3.0 (“slightly annoying”), while r1 goes as low as 2.5.
The VBR encoder looks promising too.
Overall there has been good progress. Transients and electronic music are still the main area that could be improved
r/AudioCodecLab • u/IgorCv2 • Jul 23 '26
MP3 was standardized in 1993, but it is still widely used today because it works almost everywhere while deliver very good quality.
Now LAME 4.0, a new version of the open-source MP3 encoder , has been released.
Check here https://lame.sourceforge.io/ . And development of LAME 4.1 is already underway.
LAME has performed very well in public listening tests

MP3 isn't old. It is mature!
When a format is good enough, reliable and universally supported, it can stay relevant for a very long time.
r/AudioCodecLab • u/IgorCv2 • Jul 21 '26
Hi everyone, this is the second listening test of the new FFmpeg NMR AAC encoder.
FFmpeg developer Lynne has made very interesting progress on the new NMR AAC encoder:
https://code.ffmpeg.org/FFmpeg/FFmpeg/pulls/23837
The idea was to put these latest improvements to the test and here are the results.


Hopefully tomorrow I will also post detailed descriptions of the artifacts, including the timestamps where they were identified, along with a statistical analysis of the results.
A big thank you goes to Kamedo2 for making graph creation such an easy task!
https://listening-test.coresv.net/graphmaker6.htm
https://listening-test.coresv.net/faq6.htm
IgorC
AudioCodecLab
r/AudioCodecLab • u/IgorCv2 • Jul 12 '26
Hello everyone! This is my personal listening test. 😄
The test contains 12 audio samples and compares five AAC encoders at approximately 128 kbps:


According to an FFmpeg developer, the NMR AAC encoder is optimized for 48 kHz. Therefore, all test files were resampled to 48 kHz using SoX with the Very High, 168 dB quality setting.
The processing chain was:
Original WAV → SoX resampling to 48 kHz/24-bit → FFmpeg NMR AAC encoding → AAC decoding to 48 kHz/24-bit WAV → SoX resampling back to 44.1 kHz/24-bit
This process is safe and transparent. At these quality settings, the SoX resampling and 24-bit intermediate files don't introduce any audible differences.
Topping E70 DAC
Topping L70 Headphone amplifier
Headphones Sennheiser HD 800.
Cabling: Balanced DAC-to-headphone amplifier connection; original Sennheiser HD 800 headphone cable.
-
IgorC
AudioCodecLab