libebur128 - (yet another) EBU R 128 implementation Options

[Raiden]

Hi,
inspired by the wonderful work in this thread, I wrote my own implementation of the EBU R 128 standard.
It is written in plain ANSI C and designed as a library, so you can use it in your own code. It's licensed under the MIT license.

I've also implemented a simple scanning tool, which outputs something like this:

CODE

$ ./r128-sndfile -l -p both ~/music/bad\ loop\ -\ Luo/*.flac

-12.81 LUFS, LRA: 14.16 LU, sample peak: 0.89151001, true peak: 0.99826229, /home/jan/music/bad loop - Luo/bad loop - Luo - 01 Nio.flac
-11.15 LUFS, LRA: 8.26 LU, sample peak: 0.89163208, true peak: 0.99095666, /home/jan/music/bad loop - Luo/bad loop - Luo - 02 Eri Valeire.flac
-10.14 LUFS, LRA: 11.79 LU, sample peak: 0.89154053, true peak: 0.99171823, /home/jan/music/bad loop - Luo/bad loop - Luo - 03 Kauniit Ihmiset.flac
-11.31 LUFS, LRA: 11.75 LU, sample peak: 0.89157104, true peak: 0.92898595, /home/jan/music/bad loop - Luo/bad loop - Luo - 04 Mmin.flac
-26.13 LUFS, LRA: 14.87 LU, sample peak: 0.25204468, true peak: 0.25203928, /home/jan/music/bad loop - Luo/bad loop - Luo - 05 3b Or T.flac
-14.10 LUFS, LRA: 11.40 LU, sample peak: 0.89151001, true peak: 1.02603507, /home/jan/music/bad loop - Luo/bad loop - Luo - 06 Kannas Nsp.flac
--------------------------------------------------------------------------------
-11.75 LUFS, LRA: 13.34 LU, sample peak: 0.89163208, true peak: 1.02603507

There is also ReplayGain tagging, using a reference level of -18 LUFS to match RG's loudness:

CODE

r128-mpg123 -t album [FILE|DIRECTORY]...
r128-mpg123 -t track [FILE|DIRECTORY]...

Download current version here.

This post has been edited by Raiden: Feb 27 2011, 22:29

[Raiden]

And in the example of a 600ms input, it should be calculating the loudness for 0-400 ms and then 200-600 ms. Because in the code it looks like it might be doing 0-400ms and then 400-600ms. But again I might just be following along wrong - the #define macros made it hard to debug and follow along

ebur128_calc_gating_block in line 390 is always called with samplerate*2/5 (400ms) as a parameter. So it will always analyse the last 400ms for each new block. A bit difficult are lines 256-275: I must check if the last 400ms "wrap around" the circular buffer, and handle that case separately.

Another test:

CODE

$ sox -n sine.wav synth 0.2 sine
$ sox -n noise.wav synth 0.2 noise
$ sox sine.wav noise.wav sine.wav noise.wav sine.wav noise.wav sine.wav noise.wav test.wav
$ ./r128-sndfile test.wav
0.654417
0.654412
0.654412
0.654412
0.654412
0.654412
0.654412
global loudness: -2.53 LUFS

The first block differs from the others. This is probably because the filter states must get filled.

[benski]

And in the example of a 600ms input, it should be calculating the loudness for 0-400 ms and then 200-600 ms. Because in the code it looks like it might be doing 0-400ms and then 400-600ms. But again I might just be following along wrong - the #define macros made it hard to debug and follow along

ebur128_calc_gating_block in line 390 is always called with samplerate*2/5 (400ms) as a parameter. So it will always analyse the last 400ms for each new block. A bit difficult are lines 256-275: I must check if the last 400ms "wrap around" the circular buffer, and handle that case separately.
The first block differs from the others. This is probably because the filter states must get filled.

I looked at the code again and it's definitely not doing any sort of window overlap. It's treating each 200ms block of audio separately, except for the first block which is done as 400ms. We should get this fixed. Unfortunately, the IIR filter code is writing the filtered data back into the audio_data buffer. In order to do overlapped windows, you'd need to keep the audio intact as each sample will be used twice. One simple solution would be to do the sum-of-squares inside the filter function, so that you don't need to store the filtered audio.

Just to be clear, this is essentially what needs to be done for overlap. For the sake of clarity, let's use a 1000Hz sampling rate.

CODE

ebur128_filter_double(st, &audio_data[0], 400);
ebur128_filter_double(st, &audio_data[200], 400);
ebur128_filter_double(st, &audio_data[400], 400);
ebur128_filter_double(st, &audio_data[600], 400);
/* etc. */