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Ultra Low Bitrates (REALLY!), 22khz Mono at 16kbps
Freya
post Mar 15 2012, 10:55
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I'm encoding at around 16kbps/24kbps in Mono at 22khz. I'm trying to achieve something like FM radio quality.
Sometimes I get close...

I've tried lame VBR with maxbitrate and more recently have tried ABR but I'm not sure the VBR wasn't better?

Does anyone have any tips on how I can improve my encodes in Lame? Should I add a highpass filter? What do others think of ABR vs VBR? I know ABR is recommended for low bitrates...

I know this is really far away from the usual Lame stuff but I hope people might be able to help a bit anyway!

love

Freya
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Fedot L
post Mar 15 2012, 11:36
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QUOTE (Freya @ Mar 15 2012, 10:55) *
I'm trying to achieve something like FM radio quality.
…Does anyone have any tips on how I can improve my encodes in Lame?

To achieve something like FM radio quality, you can code in “mp3 CBR-128 kbps”. Listened to and verified on spectrum analyzer display on music and white noise samples. FM radio frequency band is 30…15000 Hz, “mp3 CBR-128 kbps” frequency band is 30…16000 Hz. If you want to simulate FM radio sound/noise ratio for the best reception conditions too, add hiss at “minus 60 dB” relative to the 0 dB (peak) level of a program listened to.
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Freya
post Mar 15 2012, 11:52
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QUOTE (Fedot L @ Mar 15 2012, 10:36) *
QUOTE (Freya @ Mar 15 2012, 10:55) *
I'm trying to achieve something like FM radio quality.
…Does anyone have any tips on how I can improve my encodes in Lame?

To achieve something like FM radio quality, you can code in “mp3 CBR-128 kbps”. Listened to and verified on spectrum analyzer display on music and white noise samples. FM radio frequency band is 30…15000 Hz, “mp3 CBR-128 kbps” frequency band is 30…16000 Hz. If you want to simulate FM radio sound/noise ratio for the best reception conditions too, add hiss at “minus 60 dB” relative to the 0 dB (peak) level of a program listened to.


Heh heh! Love your reply Fedot! Really made me smile, especially the stuff about adding hiss! It's very tempting! You've given me all manner of ideas now!

Anyway I should be more careful about what I'm saying. The low bitrate stuff is the important bit more than emulating Fm radio, I was trying to get across some kind of abstract quality idea but now I'm actually starting to wonder about emulating short wave!

Anyway, I'm looking to improve my low bitrate encodings by quite a bit, so they can be less swirly!

Also would love to hear from anyone with experience with encoding speech content at low bitrates too but thats a bit of an aside as well!

Come on, there's got to be at least a few low bitraters out there!

love

Freya
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Porcus
post Mar 15 2012, 12:07
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For really low bitrates, you might want to check out this thread: http://www.hydrogenaudio.org/forums/index....showtopic=80510
(Haven't tried it.)


--------------------
One day in the Year of the Fox came a time remembered well
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KMD
post Mar 15 2012, 12:30
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If you just want to reduce the quality dont use an low bit rate mp3 encoder as that will also carry out perceptual audio compression of data , hence the swirly noise.


-
Owen.
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Freya
post Mar 15 2012, 12:47
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QUOTE (KMD @ Mar 15 2012, 11:30) *
If you just want to reduce the quality dont use an low bit rate mp3 encoder as that will also carry out perceptual audio compression of data , hence the swirly noise.


-
Owen.



No, as I say I want low bitrate mp3's.
In order to get those I have to use a mp3 encoder.
I do understand why I am getting the swirly noise, I'm just trying to minimise that.

To re-iterate, what I'm trying to achieve. Mp3, Mono, 22khz, 16kbps.
I'm just going to say all those things are neccessary to keep it simple.
In fact it also needs to be linux and gstreamer but lets not worry about that right now.

What settings will best achieve this? Has anyone tried this? ABR vs VBR, is ABR really better in this context? Are certain versions of lame better at this as some people seem to be suggesting?

love

Freya
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forart.eu
post Mar 15 2012, 14:18
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You have to consider Stereo Tool too for mono downmixing. wink.gif
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onkl
post Mar 15 2012, 15:16
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If it doesn't need to be MP3, you should give OGG Vorbis and AAC a try. These usually work much more efficient at very low bitrates.

This post has been edited by onkl: Mar 15 2012, 15:17
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DonP
post Mar 15 2012, 15:36
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QUOTE (Freya @ Mar 15 2012, 05:55) *
I'm encoding at around 16kbps/24kbps in Mono at 22khz. I'm trying to achieve something like FM radio quality.
Sometimes I get close...


The lame web site recommends 16 khz and LP filter for 24 kbps. Best way to beat the swirlies is narrowing the passband that lame has to encode.
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LithosZA
post Mar 15 2012, 15:43
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QUOTE
In fact it also needs to be linux and gstreamer but lets not worry about that right now.


HE-AAC might be your best bet.

To use AAC on gstreamer you will need to use gst-aacplus:

QUOTE
gst-aacplus is GStreamer plug-in, that able to encode audio to
modern mp4 audiocodec: AAC+ and eAAC+. The specific mode
(SBR or SBR+PS) - is selected automatically, depending on channels
number, samplerate and bitrate of input stream.

The plug-in based on libaacplus library, that also depends on proprietary
3GPP's code, so you have to be very careful with this plug-ion usage,
because the encoding logic part is not free.


FAAC won't work, because it is LC-AAC and It barely works at 128Kbit/s anyway.

Opus/CELT should also work if it is available for gstreamer yet.

Maybe try to use GSM codec?

If you can't get any of those working then Vorbis might be your last resort.

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Freya
post Mar 15 2012, 15:47
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QUOTE (forart.eu @ Mar 15 2012, 13:18) *
You have to consider Stereo Tool too for mono downmixing. wink.gif


While most of the stuff I'm working with will be mono all the way, that is still an awesome suggestion that will come in very handy!
THANKYOU! Very much appreciated!

love

Freya
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Freya
post Mar 15 2012, 16:00
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QUOTE (DonP @ Mar 15 2012, 14:36) *
QUOTE (Freya @ Mar 15 2012, 05:55) *
I'm encoding at around 16kbps/24kbps in Mono at 22khz. I'm trying to achieve something like FM radio quality.
Sometimes I get close...


The lame web site recommends 16 khz and LP filter for 24 kbps. Best way to beat the swirlies is narrowing the passband that lame has to encode.


Thanks I was thinking it might be worth adding a low pass filter.

Believe it or not, a lot of the 16kbps stuff is actually really good, just not quite right.
I've started trying again at 24 to see how it compares and was checking it out, thinking "this is waaaay better, only a tiny tinge of high frequency swirl" and then I realised I was listening to the wrong file. It was the 16kbps version!

At 24kbps things seem really good.

I'm just going to run a bunch of tests now but I definitely want to try a low pass filter at 16 too.

Do you have a link to where on the lame website you saw this info?
I'm going to go there now and look for it!

Thankyou.

love

Freya
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LithosZA
post Mar 15 2012, 16:01
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A few speech codecs that you could try:
speex
amrnb
gsm
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Remedial Sound
post Mar 15 2012, 16:01
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Have you tried "--abr 16 -mm." with the latest version of lame? It does indeed downsample to 16 khz.

As onkl noted HE-AAC is really better suited for what you're trying to achieve, as it can make use of efficiency schemes (SBR & PS) that mp3 cannot.
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DonP
post Mar 15 2012, 16:07
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QUOTE (Freya @ Mar 15 2012, 11:00) *
[
I'm just going to run a bunch of tests now but I definitely want to try a low pass filter at 16 too.

Do you have a link to where on the lame website you saw this info?
I'm going to go there now and look for it!


http://lame.cvs.sourceforge.net/viewvc/lame/lame/USAGE

Then search for "LOW BITRATES"
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IgorC
post Mar 15 2012, 16:18
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QUOTE (Remedial Sound @ Mar 15 2012, 12:01) *
As onkl noted HE-AAC is really better suited for what you're trying to achieve, as it can make use of efficiency schemes (SBR & PS) that mp3 cannot.

he-aac is a general audio format, hence more suitable for music. AMR/AMR-WB(+), G.7xx (I would say the last G.718SWB or G.719), Opus (speech mode:SILK) will be better for speech content as they are speech codecs.

Probably it's worth to read

http://research.nokia.com/files/FullMOSTes...ICASSP-2010.pdf

http://research.nokia.com/files/public/%5B..._Opus_Codec.pdf

This post has been edited by IgorC: Mar 15 2012, 16:40
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AndyH-ha
post Mar 15 2012, 20:54
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I've done many audiobooks with this. It isn't a standard LAME setting but my results sound quite good to me and a few other people who use them. Occasionally the sound is a little different than the original source but not enough to bother me. Music isn't the same, however, but there is little music in most audio books and I would not care if what is there wasn't.

I haven't updated anything is quite some time, so the vbr-new might not be appropriate with a newer version of LAME.
-V 8 --vbr-new -m m --resample 22 --lowpass 11
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birdie
post Mar 15 2012, 23:26
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I'd also suggest HE-AAC or even Skype's SILK, but I haven't seen its useful implementations.
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NullC
post Mar 16 2012, 04:04
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QUOTE (birdie @ Mar 15 2012, 15:26) *
I'd also suggest HE-AAC or even Skype's SILK, but I haven't seen its useful implementations.


There is no SILK anymore, SILK and CELT were merged and the result is Opus. Igor already recommended opus above.
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Fedot L
post Mar 16 2012, 14:28
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QUOTE (Freya @ Mar 15 2012, 11:52) *
QUOTE (Fedot L @ Mar 15 2012, 10:36) *
QUOTE (Freya @ Mar 15 2012, 10:55) *
I'm trying to achieve something like FM radio quality.
…Does anyone have any tips on how I can improve my encodes in Lame?

To achieve something like FM radio quality, you can code in “mp3 CBR-128 kbps”. Listened to and verified on spectrum analyzer display on music and white noise samples. FM radio frequency band is 30…15000 Hz, “mp3 CBR-128 kbps” frequency band is 30…16000 Hz. If you want to simulate FM radio sound/noise ratio for the best reception conditions too, add hiss at “minus 60 dB” relative to the 0 dB (peak) level of a program listened to.

----
…your reply Fedot! Really made me smile…

Really? And your “trying to achieve something like FM radio quality” (and only mono, I don’t know why, as since 1963, FM is stereo…) by coding in “mp3 16 kbps/24 kbps at 22 khz”, when even in “mp3 24 kbps at 22 khz” the frequency band is 8 kHz, and in “mp3 16 kbps at 22 khz” it’s 5,5 kHz (tested and watched on spectrum analyzer display). Against 15 kHz in FM…
QUOTE (Freya @ Mar 15 2012, 11:52) *
…especially the stuff about adding hiss!

You never heard hiss in FM reception?
QUOTE (onkl @ Mar 15 2012, 15:16) *
If it doesn't need to be MP3, you should give OGG Vorbis and AAC a try. These usually work much more efficient at very low bitrates.

You are quite right. “M4a VBR-16 kbps” stereo gave me 10,5 kHz on HQ music/vocal file, with a good stereo effect and rather good general quality. But far from “simulating FM radio quality”.
QUOTE (DonP @ Mar 15 2012, 15:36) *
Best way to beat the swirlies is narrowing the passband that lame has to encode.

I clearly heard that in HQ files coded in “OGG Vorbis” there was no swirling at much lower bitrates than it was present in “mp3”. On some low frequency music passages I heard swirling in “mp3 CBR-128 kbps”, while it was not in “OGG Vorbis CBR-48 kbps”.
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