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Dial-up bitrate listening test

Reply #75
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Fair enough would be to use recommended default settings for each bit-rate, as this is the typical usage scenario - which reflects the biggest number of available streams/content available around.

I agree completely

Well this would mean also using the internal Vorbis resampler. If it's not as good as external SSRC, then it's a problem in the encoder, and too bad if it's not fixed.
However, I think that Vorbis supporters wouldn't like that, if an external component like SSRC can make Vorbis sound better in this test. 
Another question is, is it fair to use external components here.
Juha Laaksonheimo

Dial-up bitrate listening test

Reply #76
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Did a quick test using SSRC and Vorbis:

(...)

Seems like the samplerate used is a lot less important than the SRC tool used.

That's very interesting. Thank-you for testing it.

Dial-up bitrate listening test

Reply #77
OK, so the codecs list would now be:

- Ahead HE AAC + PS at 32kbps, 44.1kHz
- Ogg Vorbis 1.0.1CVS at managed bitrate 32kbps, resampled with SSRC to 22050Hz
- MP3pro 32kbps, resampled to 32kHz
- QDesign 32kbps at either 44.1 or 32kHz
- WMA Std 32kbps at either 44.1, 32 or 22.5kHz
- Real Audio Cook 32kbps at either 44.1, 32 or 22.5kHz
- Low anchor: MP3 at 32kbps, 12kHz
- High anchor: lowpass


What do you guys think?

Regards;

Roberto.

Dial-up bitrate listening test

Reply #78
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Another question is, is it fair to use external components here.

I believe it is. This test's scope is to compare streaming audio. Streaming stations use external components for several purposes. So I think it's fair if I use these components, if that leads to improved quality - what the stations are ultimately after.

Dial-up bitrate listening test

Reply #79
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Fair enough would be to use recommended default settings for each bit-rate, as this is the typical usage scenario - which reflects the biggest number of available streams/content available around.

I agree completely

Well this would mean also using the internal Vorbis resampler. If it's not as good as external SSRC, then it's a problem in the encoder, and too bad if it's not fixed.
However, I think that Vorbis supporters wouldn't like that, if an external component like SSRC can make Vorbis sound better in this test. 
Another question is, is it fair to use external components here.

You have a point here, but I don't think is fair to use external tools in this test mainly because I don't imagine the "average joe" using them to get the best quality.
[edit] Ok rjamorim cleared this in the previous post [/edit]

I'd like to test both version of the sample (the original one, eventually resampled by internal resampler versus the one SSRCed): it would double the amount of samples in the test, but is should still remain "doable" since the low bitrate makes abx easier (?).
Vital papers will demonstrate their vitality by spontaneously moving from where you left them to where you can't find them.

Dial-up bitrate listening test

Reply #80
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You have a point here, but I don't think is fair to use external tools in this test mainly because I don't imagine the "average joe" using them to get the best quality.

Well, I personally don't think the average joe would use 32kbps for his daily encodings :B

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I'd like to test both version of the sample (the original one, eventually resampled by internal resampler versus the one SSRCed): it would double the amount of samples in the test, but is should still remain "doable" since the low bitrate makes abx easier (?).


That would be difficult. We already have 8 samples. Even though testing is easy at this bitrate, so many samples require several iterations so that you can decide what sample sounds better than the other.

Dial-up bitrate listening test

Reply #81
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I'd like to test both version of the sample (the original one, eventually resampled by internal resampler versus the one SSRCed): it would double the amount of samples in the test, but is should still remain "doable" since the low bitrate makes abx easier (?).

I'd rather do more tests beforehand. I only quickly tested two (very similiar) samples, the results could be less dramatic for other types of music.
With more SSRC vs. <encoder's resampler> test results it should be easier to weight 'ease of use' against 'best quality'.

Real word question: How many broadcasting tools (Oddcast?) use their own resampler? How many don't?
"To understand me, you'll have to swallow a world." Or maybe your words.

Dial-up bitrate listening test

Reply #82
Is it possible to do live streaming with SSRC+Vorbis encoding option with currently available tools?
I think this is a pretty essential question regarding whether external SSRC could be fairly used with Vorbis here.
Juha Laaksonheimo

Dial-up bitrate listening test

Reply #83
Can't streaming servers resample the WAVs with SSRC beforehand and then feed them to oddcast?

Dial-up bitrate listening test

Reply #84
Couldn't one of the excellent Vorbis fork authors simply make a niche branch of Vorbis which replaces the internal resampler with SSRC.  This would remove the need for this particular argument
< w o g o n e . c o m / l o l >

Dial-up bitrate listening test

Reply #85
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Can't streaming servers resample the WAVs with SSRC beforehand and then feed them to oddcast?

With live content, I don't know.
Juha Laaksonheimo

Dial-up bitrate listening test

Reply #86
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With live content, I don't know.

Oh, but I'm not testing live content. It's all prerecorded stuff

Dial-up bitrate listening test

Reply #87
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With live content, I don't know.

Oh, but I'm not testing live content. It's all prerecorded stuff

Ok, it's just that you said: "Streaming stations use external components for several purposes. So I think it's fair if I use these components, if that leads to improved quality - what the stations are ultimately after."

I think many radio stations are/will be streaming live content, be it Vorbis, wma, AAC-PS (or Digital Radio Mondiale) etc.
Juha Laaksonheimo

Dial-up bitrate listening test

Reply #88
I'm all for a 32kbps test, and not a 48kbps one. 

We would see how some of these codecs fare at or near the bottom of their range.  Some sites are streaming at even lower rates of 16kbps, admittedly usually mono.

Streamed video will usually include audio at this lower rate as well.

I expect that most of the codecs will lowpass automatically, so there should be no decision making about whether to lowpass or not.  However, should we lowpass beyond the encoder's defaults (Ivan is suggesting not to)?  Where there is choice offered by an encoder (and not just one default), I say we make the choice ourselves.  We could run a few pre-tests to see how some of the (more flexible) encoders fare at various lowpass frequencies.  There should be an obvious point where we select for frequency content versus artifact content.

Just something I would like confirmed at this point: this is a stereo test, isn't it?

edit: looks like my reply took some while and several replies got in before me.

Dial-up bitrate listening test

Reply #89
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I think many radio stations are/will be streaming live content, be it Vorbis, wma, AAC-PS (or Digital Radio Mondiale) etc.

Well, if we are to emulate live stations, we can sack HE AAC+PS right away, because so far there's no way to stream it ATM, live or not (DRM equipment still only broadcasts HE AAC).

Dial-up bitrate listening test

Reply #90
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I think many radio stations are/will be streaming live content, be it Vorbis, wma, AAC-PS (or Digital Radio Mondiale) etc.

Well, if we are to emulate live stations, we can sack HE AAC+PS right away, because so far there's no way to stream it ATM, live or not (DRM equipment still only broadcasts HE AAC).

Well, ok. If the live streaming concept is ditched, then of course anything is possible I suppose. DRM with PS should be ready anytime, but IIRC the PS-part is a bit different than in the MPEG standard. I'm not sure but FhG&CT might have a streaming solution in principle ready which does AAC-PS.
Juha Laaksonheimo

Dial-up bitrate listening test

Reply #91
I'm not trying to ditch encoders that have no streaming support.

My point here is that I'm trying to cut some slack to the encoders. I'm not interested in conducing a test as anal as it has been suggested here, because my test isn't even a formal one for starters.

If we are to start wildly speculating that "FhG and CT might have a streaming solution ready", we can also wildly speculate that Monty will have a fixed version of oggenc soon.

Dial-up bitrate listening test

Reply #92
Heh, you are right. To me it just sounds questional in a principle level to use external component to compensate for a non-optimized feature in an encoder.
This problem would be solved if somebody would actually make a Vorbis binary with better resampling.
But fixing this using an external component instead of the encoder's own component in order to gain better quality for the test, it is in the limit imo..

What do other people think?
Juha Laaksonheimo

Dial-up bitrate listening test

Reply #93
So you're now setting limits 

Dial-up bitrate listening test

Reply #94
I'm not setting any limits. I said in my opinion it is in the limit (of what should be allowed when testing encoders).
I personally would draw the line to the point where totally separate external components are used to replace the encoder's internal component in order to compensate for an encoder software's deficiency.
Juha Laaksonheimo

Dial-up bitrate listening test

Reply #95
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Well, if we are to emulate live stations, we can sack HE AAC+PS right away, because so far there's no way to stream it ATM, live or not (DRM equipment still only broadcasts HE AAC).

DRM equipment broadcasts HE-AAC streams with Parametric Stereo since December 2003.


Dial-up bitrate listening test

Reply #96
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Well, if we are to emulate live stations, we can sack HE AAC+PS right away, because so far there's no way to stream it ATM, live or not (DRM equipment still only broadcasts HE AAC).

DRM equipment broadcasts HE-AAC streams with Parametric Stereo since December 2003.


Since before it was even standardized? 

Dial-up bitrate listening test

Reply #97
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Well, if we are to emulate live stations, we can sack HE AAC+PS right away, because so far there's no way to stream it ATM, live or not (DRM equipment still only broadcasts HE AAC).

DRM equipment broadcasts HE-AAC streams with Parametric Stereo since December 2003.


Since before it was even standardized? 

New DRM standard for audio coding was finalized December 15 2003 and this already included PS AAC.

All transmitters switched to PS AAC shortly afterwards. (Sometimes mono is still used when the propagation is very bad to save a few more bits).

Dial-up bitrate listening test

Reply #98
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Is it possible to do live streaming with SSRC+Vorbis encoding option with currently available tools?
I think this is a pretty essential question regarding whether external SSRC could be fairly used with Vorbis here.

I just checked: It's possible using (who would've guseed) using foobar2000 and foo_oddcast.
It should also be possible using ssrc and ices on a *nix system
"To understand me, you'll have to swallow a world." Or maybe your words.

Dial-up bitrate listening test

Reply #99
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I just checked: It's possible using (who would've guseed) using foobar2000 and foo_oddcast.
It should also be possible using ssrc and ices on a *nix system

oh, well, cheers. It's settled then. I'll use SSRC with Vorbis.

Thanks for checking it out, dev0.

Now, it only remains to be tested what sampling rates will be used with WMA and Real. Maybe I should set up a really simple and quick pre-test?

 
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