Re: [FFmpeg-user] How to enforce VBR minimum bitrate

2018-06-04 Thread Zak

On 2018-06-04 at 1:40 PM EDT, Carl Eugen Hoyos wrote:
>

2018-06-04 16:59 GMT+02:00, DopeLabs :

>

you can combine steps 1 and 2 into a single command:

$ ffmpeg- i input -options -f wav - | lame -V 5 -F -b 128 - output.mp3


wav is a format that cannot easily be piped.
It may work in many cases, but generally you cannot pipe a valid file.

Please avoid top-posting here, it is rude.

Carl Eugen



Okay, I have new findings and opinions:

1. The original poster should use CBR (constant bitrate), almost 
definitely. My reasoning:


1.1. At -V 5 with normal music and no minimum frame size, the output is 
125 to 145 kbps on the small corpus of music that I tested.


1.2. With a minimum frame size of 128 kbps, the same corpus of music is 
132 to 150 kbps, and it sounds the same to me. (It doesn't sound perfect 
at all, -V 0 sounds significantly better.)


1.3. With CBR at 128 kbps, it sounds the same, it is slightly smaller, 
and it will play on very rudimentary hardware decoders. Fast scrubbing 
will also be possible because it is CBR, so quickly seeking to a 
specific time index will not require complex math using the VbrTag (also 
known as the toc, table of contents, the VBRI header).


1.4. MP3 frames do not come in arbitrary sizes, and setting a target 
bitrate of 135 kbps but only allowing the smallest frames to be 128 kbps 
means that almost all of the frames will be 128 kbps. Rather few frames 
are higher than 128 kbps (as shown by the final deduced bitrates ranging 
from 132 to 150 kbps). These files are almost entirely 128 kbps frames. 
The next frame size is 160 kbps, so there is a big jump from 128 to 160, 
and 160 is not being used often with -V 5. After 160, the next frame 
size is 192 kbps, and it shows up less than 1% of the time in all of the 
songs that I spot-checked (at -V 5). For the curious: The MP3 standard 
can only accommodate 14 sizes of frame, because the frame size is stored 
in 4 bits with 0b and 0b both being invalid.


1.5. The benefit of -V 5 is that the files are smaller because of the 
small frames. If you don't let small frames happen, then you are making 
far less compatible files for no reason.


1.6. Using -V 0 with a minimum frame size of 128 kbps is possibly 
sensible. -V 0 gives output around 270 kbps, and adding a minimum frame 
size of 128 kbps barely changes the file size (they get a little bit 
bigger). The -V 0 music sounds perfect to me, I cannot tell the 
difference between -V 0 and lossless, not by any stretch of the 
imagination. It sounds better than the V5 with or without the frame size 
limit.


1.7. The 128 kbps CBR code in LAME has actually been heavily optimized 
for sound quality. The VBR code has also been optimized, but not if you 
do something silly like forbid small frames. If you do something closer 
to what the LAME developers were expecting, you will get better-sounding 
results in a smaller file. Though to be fair, I could hear the same 
compression artifacts in both configurations.


-

The following is not good advice, I just wanted to do some experiments 
and confirm that it is bad advice.


2. Surprisingly, the following does work with 44.1 kHz 16-bit stereo:

ffmpeg -i input.flac -f wav - | lame -V 5 -F -b 128 - output.mp3

However, using more exotic inputs resulted in MP3s that sounded really 
bad, MP3s that consisted of white noise, and one MP3 file that crashed 
my entire computer when I tried to open it with VLC, which was impressive.


The WAV file contains information about the file structure at the 
beginning. LAME will make educated guesses, but if the input is exotic 
then it will guess wrong.


If LAME prints "Assuming raw pcm input file" then you are probably going 
to get garbage out. This warning is the very first thing LAME prints 
(after FFmpeg finishes printing).


Your best bet is to make a WAV file, use that, then delete it. In 
theory, you could have FFmpeg produce an actual raw PCM stream with 
known characteristics, but you would need to tell LAME what to expect.


-

3. I figured out how to get ABR working via FFmpeg:

ffmpeg -i input -codec:a libmp3lame -abr true -b:a 135k output.mp3

The key is "-abr true". I figured this out from this command:

ffmpeg -h encoder=libmp3lame

This was strikingly hard to figure out.

-

4. Passing in the -F option via FFmpeg will be impossible. I am 
convinced of this after reading this file:


ffmpeg/libavcodec/libmp3lame.c

Here it is in the Git repo:

https://git.ffmpeg.org/gitweb/ffmpeg.git/blob/refs/heads/release/4.0:/libavcodec/libmp3lame.c

Only a few of the libmp3lame options can be customized in FFmpeg, and 
you cannot pass in arbitrary options.


Side note: I'm going to try changing this line and recompiling FFmpeg, 
because I like the LAME Info Tag (Xing header):


At line 132 in libmp3lame.c:

   /* do not get a Xing VBR header frame from LAME */
-- lame_set_bWriteVbrTag(s->gfp,0);
++ lame_set_bWriteVbrTag(s->gfp,1);

Will I crash my computer for the 

Re: [FFmpeg-user] How to enforce VBR minimum bitrate

2018-06-04 Thread Carl Eugen Hoyos
2018-06-04 16:59 GMT+02:00, DopeLabs :
> you can combine steps 1 and 2 into a single command:
>
> $ ffmpeg- i input -options -f wav - | lame -V 5 -F -b 128 - output.mp3

wav is a format that cannot easily be piped.
It may work in many cases, but generally you cannot pipe a valid file.

Please avoid top-posting here, it is rude.

Carl Eugen
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Re: [FFmpeg-user] How to enforce VBR minimum bitrate

2018-06-04 Thread DopeLabs
you can combine steps 1 and 2 into a single command:

$ ffmpeg- i input -options -f wav - | lame -V 5 -F -b 128 - output.mp3


> On Jun 4, 2018, at 7:52 42AM, Zak  wrote:
> 
> On 2018-06-04 at 3:03 AM EDT, sook...@expressinmusic.com wrote:
> 
>> Hi experts!
>> 
>> I am currently using ffmpeg as a solution to audio file conversion and
>> normalization.
>> 
>> I understand that my setting (-q:a 5) below should produce output with
>> bitrate range in between 120-150kbps, or sometimes even lower than that if
>> the input already in being a lower bitrate than requested.
>> 
>> By using VBR Encoding, is there an option that is able to constrain the
>> bitrate minimum at 128kbps?
>> 
>> ffmpeg -i "input" -af "silenceremove=0:0:0:-1:1:-80dB,
>> dynaudnorm=p=1:m=3:r=1" -vn -acodec libmp3lame -q:a 5 -ar 44100 "output.mp3"
>> 
>> Please kindly advise. Thanks!
>> 
>> Regards,
>> 
>> Sook Sin
> 
> Hello Sook Sin,
> 
> Disclamer: It might depend on your version of FFmpeg and libmp3lame, but I 
> doubt it (assuming versions from 2012 or more recent). The libmp3lame 
> behavior in this respect has been the same since at least 2012, and the 
> FFmpeg behavior has probably been the same in this respect. (If it changed, 
> it would indicate that FFmpeg removed a feature, and I don't think they did 
> this. There is no reason to remove features.)
> 
> Short version: You can set a minimum bitrate in VBR mode with the LAME 
> command line interface, but I think you CANNOT do this with the version of 
> libmp3lame that is linked by FFmpeg during compilation of FFmpeg. I just 
> tried, the minimum bitrate was not respected, every frame of silence was 
> encoded at 32 kbps. A work-around is to use your FFmpeg command to make an 
> uncompressed PCM file, such as a WAV file, and then convert the WAV to MP3 
> using the LAME command line interface (going forward, I will call it the LAME 
> CLI). I give an example of how to do this below.
> 
> Question for the whole list:
> Is it possible to pass arbitrary options from FFmpeg to libmp3lame as if they 
> came from argc and argv? I am guessing no. libx264 seems to take arbitrary 
> options via FFmpeg, but I am guessing libmp3lame does not. Sook Sin wants to 
> pass in the option "-F" as if it came from the command line. (Or rather, if 
> this were possible it would solve the problem at hand.)
> 
> Why:
> 
> In order to tell the LAME CLI to do VBR but also enforce a minimum bitrate 
> for all frames of MP3 audio data, you need to give the LAME CLI a command 
> like this (this is valid Bash syntax for GNU/Linux, MacOS, or Cygwin on 
> Windows):
> 
> bash$ lame -V 5 -F -b 128 input.wav output.mp3
> 
> The option "-V 5" for the LAME CLI is the same as "-q:a 5" in FFmpeg, and in 
> fact FFmpeg passes the number 5 directly to libmp3lame to be used by 
> libmp3lame as it sees fit. The FFmpeg option "-q:a 5" should not be confused 
> with the LAME CLI option "-q 5", which is a different encoder parameter.
> 
> The option -V (LAME) and its sister -q:a (FFmpeg) control the file size and 
> target bitrate.
> 
> The option -q (LAME) and its sister -compression_level:a (FFmpeg) control how 
> fast or slow the encoder is. "-q 0" (LAME CLI) is equal to 
> "-compression_level:a 0" (FFmpeg), and it will be the slowest during 
> compression, equally fast during playback, and give the best-sounding 
> results. "-q 3" is the default in VBR mode for LAME (both CLI and 
> FFmpeg-linked). In CBR mode, -q 3 is often the same as -q 0, but it depends 
> on the sample rate of the input and the output bitrate of the MP3. For CD 
> audio (44.1 kHz stereo) at CBR at 320 kbps, -q 3 is actually a tiny bit 
> faster than -q 0. At lower bitrates they are often identical algorithms, 
> speed, and output. -q 0 is the highest quality and slowest compression, -q 9 
> is the lowest quality and fastest, the output file sizes are generally about 
> the same with no systematic trend larger or smaller.
> 
> The option -F (LAME) is the key option that you want. It means FORCE the 
> encoder to use frames of a certain minimum size, even if it can achieve the 
> target quality (implied by -V 5) using a small frame. You say -F and then you 
> say -b N with the frame size in kbps. So for a minimum frame size of 128 
> kbps, it is the command above with "-F -b 128". This does indeed work in LAME 
> v3.100, I just tested it. Even silence will be 128 kbps. In fact, if I set -V 
> 9 (which should be about 65 kbps and should sound terrible), the whole file 
> is created using 128 kbps frames and the quality is much better, because 128 
> kbps sounds markedly better than ~65 kbps VBR. (These examples and numbers 
> are all for CD audio, in which silence would normally be 32 kbps in -V N for 
> any value of N. Frames smaller than 32 kbps are not valid at CD sample rates 
> in the MP3 standard - VLC will play such files, iTunes will not. LAME CLI 
> will make such a file if you compile in debug mode. FFmpeg will not make such 
> files, as 

Re: [FFmpeg-user] How to enforce VBR minimum bitrate

2018-06-04 Thread Zak

On 2018-06-04 at 3:03 AM EDT, sook...@expressinmusic.com wrote:


Hi experts!

I am currently using ffmpeg as a solution to audio file conversion and
normalization.

I understand that my setting (-q:a 5) below should produce output with
bitrate range in between 120-150kbps, or sometimes even lower than that if
the input already in being a lower bitrate than requested.

By using VBR Encoding, is there an option that is able to constrain the
bitrate minimum at 128kbps?

ffmpeg -i "input" -af "silenceremove=0:0:0:-1:1:-80dB,
dynaudnorm=p=1:m=3:r=1" -vn -acodec libmp3lame -q:a 5 -ar 44100 "output.mp3"

Please kindly advise. Thanks!

Regards,

Sook Sin


Hello Sook Sin,

Disclamer: It might depend on your version of FFmpeg and libmp3lame, but I 
doubt it (assuming versions from 2012 or more recent). The libmp3lame behavior 
in this respect has been the same since at least 2012, and the FFmpeg behavior 
has probably been the same in this respect. (If it changed, it would indicate 
that FFmpeg removed a feature, and I don't think they did this. There is no 
reason to remove features.)

Short version: You can set a minimum bitrate in VBR mode with the LAME command 
line interface, but I think you CANNOT do this with the version of libmp3lame 
that is linked by FFmpeg during compilation of FFmpeg. I just tried, the 
minimum bitrate was not respected, every frame of silence was encoded at 32 
kbps. A work-around is to use your FFmpeg command to make an uncompressed PCM 
file, such as a WAV file, and then convert the WAV to MP3 using the LAME 
command line interface (going forward, I will call it the LAME CLI). I give an 
example of how to do this below.

Question for the whole list:
Is it possible to pass arbitrary options from FFmpeg to libmp3lame as if they came from 
argc and argv? I am guessing no. libx264 seems to take arbitrary options via FFmpeg, but 
I am guessing libmp3lame does not. Sook Sin wants to pass in the option "-F" as 
if it came from the command line. (Or rather, if this were possible it would solve the 
problem at hand.)

Why:

In order to tell the LAME CLI to do VBR but also enforce a minimum bitrate for 
all frames of MP3 audio data, you need to give the LAME CLI a command like this 
(this is valid Bash syntax for GNU/Linux, MacOS, or Cygwin on Windows):

bash$ lame -V 5 -F -b 128 input.wav output.mp3

The option "-V 5" for the LAME CLI is the same as "-q:a 5" in FFmpeg, and in fact FFmpeg passes the 
number 5 directly to libmp3lame to be used by libmp3lame as it sees fit. The FFmpeg option "-q:a 5" should 
not be confused with the LAME CLI option "-q 5", which is a different encoder parameter.

The option -V (LAME) and its sister -q:a (FFmpeg) control the file size and 
target bitrate.

The option -q (LAME) and its sister -compression_level:a (FFmpeg) control how fast or slow the encoder is. 
"-q 0" (LAME CLI) is equal to "-compression_level:a 0" (FFmpeg), and it will be the 
slowest during compression, equally fast during playback, and give the best-sounding results. "-q 
3" is the default in VBR mode for LAME (both CLI and FFmpeg-linked). In CBR mode, -q 3 is often the same 
as -q 0, but it depends on the sample rate of the input and the output bitrate of the MP3. For CD audio (44.1 
kHz stereo) at CBR at 320 kbps, -q 3 is actually a tiny bit faster than -q 0. At lower bitrates they are 
often identical algorithms, speed, and output. -q 0 is the highest quality and slowest compression, -q 9 is 
the lowest quality and fastest, the output file sizes are generally about the same with no systematic trend 
larger or smaller.

The option -F (LAME) is the key option that you want. It means FORCE the encoder to use 
frames of a certain minimum size, even if it can achieve the target quality (implied by 
-V 5) using a small frame. You say -F and then you say -b N with the frame size in kbps. 
So for a minimum frame size of 128 kbps, it is the command above with "-F -b 
128". This does indeed work in LAME v3.100, I just tested it. Even silence will be 
128 kbps. In fact, if I set -V 9 (which should be about 65 kbps and should sound 
terrible), the whole file is created using 128 kbps frames and the quality is much 
better, because 128 kbps sounds markedly better than ~65 kbps VBR. (These examples and 
numbers are all for CD audio, in which silence would normally be 32 kbps in -V N for any 
value of N. Frames smaller than 32 kbps are not valid at CD sample rates in the MP3 
standard - VLC will play such files, iTunes will not. LAME CLI will make such a file if 
you compile in debug mode. FFmpeg will not make such files, as far as I know. This seems 
like good behavior on all accounts.)

Sadly, you need the -F option, and FFmpeg doesn't seem to know how to accept 
that option and pass it to libmp3lame. Using a realistic music file with some 
silence, I was not able to get FFmpeg to respect the minimum frame size. In VBR 
mode, FFmpeg wants to make 32 kbps frames if it feels like they are high-enough 

[FFmpeg-user] How to enforce VBR minimum bitrate

2018-06-04 Thread sooksin
Hi experts!

 

I am currently using ffmpeg as a solution to audio file conversion and
normalization.

 

I understand that my setting (-q:a 5) below should produce output with
bitrate range in between 120-150kbps, or sometimes even lower than that if
the input already in being a lower bitrate than requested.

 

By using VBR Encoding, is there an option that is able to constrain the
bitrate minimum at 128kbps?

 

ffmpeg -i "input" -af "silenceremove=0:0:0:-1:1:-80dB,
dynaudnorm=p=1:m=3:r=1" -vn -acodec libmp3lame -q:a 5 -ar 44100 "output.mp3"

 

Please kindly advise. Thanks!

 

Regards,

Sook Sin

 



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