Re: [Flightgear-devel] ATC voice howto

2004-02-16 Thread Erik Hofman
Martin Spott wrote:

When you take the 'philosophical' route, I agree - in almost _every_
situation it's a big fault to delete detail/resolution from your raw
data.
On the other hand: 8 kHz, 8 bit is not that bad. German ISDN telephony
has this resolution and to my impression the audible quality is far
better than usual radio in an aircraft  ;-)
ISDN uses the uLaw compression which means, compress 16-bit audio into 
8-bit by removing less relevant bits as the audio gets louder.

Erik

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Re: [Flightgear-devel] ATC voice howto

2004-02-16 Thread Martin Spott
Erik Hofman [EMAIL PROTECTED] wrote:
 Martin Spott wrote:

 On the other hand: 8 kHz, 8 bit is not that bad. German ISDN telephony
 has this resolution and to my impression the audible quality is far
 better than usual radio in an aircraft  ;-)

 ISDN uses the uLaw compression which means, compress 16-bit audio into 
 8-bit by removing less relevant bits as the audio gets louder.

Oh, sorry, I didn't expect the system to be _that_ sophisticated  ;-)

Martin.
-- 
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Re: [Flightgear-devel] ATC voice howto

2004-02-15 Thread Matevz Jekovec




Jonathan Richards wrote:

  On Thursday 12 Feb 2004 5:31 pm, David Luff wrote:
  
  
OK, here's some instructions on how to generate new ATC voices for
FlightGear.  Hopefully this will make some sense to somebody, ask if it's
unclear.

  
  snip
  
  
Two files are required for each voice - a wave file containing the actual
sounds, and an index file that basically describes where in the wave file
buffer to find each word or phrase.  The current voices for the ATIS can be
found in $FG_ROOT/data/ATC and are called default.wav and default.vce for
the wave and index file respectively.  Note that one important change will
be made in default.vce - currently it is indexed by byte position into the
sound buffer, but I've decided it would be better to index by time into the
buffer, since that is more robust to changing the recording quality, and in
the future possiby using encoding such as Ogg Vorbis.  Also, the first line
currently contains the number of subsequent lines, but I think that can be
ditched!

  
  massive snip
David
In what units shall the time index be specified?  The sampling rate sets a 
resolution limit on the timing, so for 8kHz we only need 1/8000 sec = 125 
microseconds precision, but if we have an ambition for higher rates, we need 
more.  [1]
In reality, radio comms are not HiFi standard.  Does anyone know what the 
typical bandwidth is?  Or should we simulate by taking a beautiful 22kHz 
recording and filtering it to sound like the real thing?  Perhaps as an 
option, so one can do radio practice with bell-like clarity at first, and 
graduate to crackly reception of foreign languages and accents later!

Regards
	Jonathan

[1] It occurs to me that for chunked formats like WAV, there is a mathematical 
relationship between the byte position and the time offset which could be 
used for conversion, no?
  

Does anyone know, does Plib include realtime sound mixing, downsampling
and volume control (the sound should be downsampled to match the real
aircraft comms, it should be more soft when you're further away from
the transmitter and we should mix some noise sounds with a speech to
sound everything more like in real life).


- Matevz


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RE: [Flightgear-devel] ATC voice howto

2004-02-13 Thread Richard Bytheway
snip 
 Anyone got any wavefile editor recommendations BTW?  I used 
 CoolEdit (Windows) for the ATIS, but the trial period is now 
 long gone, and when I went to buy it I found the guy had sold 
 it to Adobe and the price had tripled.  No thanks!  I'm using 
 Audacity now, but it's not entirely polished, and the sound 
 is not in sync with cursor movement when playing a small 
 selection in recent versions.  Crucial factors are the 
 ability to extend or contract a selection by either edge, and 
 easy copy and pasting into a new buffer, plus display of time 
 or byte position.  There's loads for Linux, but I haven't 
 found a really likeable one yet, Audacity being the nearest so far.
 
 Cheers - Dave
 
 
I use WaveFlow (http://www.waveflow.com). Good interface, copes with massive (30 
minute+) recordings reasonably fast, and is under active development.

Richard

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Re: [Flightgear-devel] ATC voice howto

2004-02-12 Thread Jonathan Richards
On Thursday 12 Feb 2004 5:31 pm, David Luff wrote:
 OK, here's some instructions on how to generate new ATC voices for
 FlightGear.  Hopefully this will make some sense to somebody, ask if it's
 unclear.
snip
 Two files are required for each voice - a wave file containing the actual
 sounds, and an index file that basically describes where in the wave file
 buffer to find each word or phrase.  The current voices for the ATIS can be
 found in $FG_ROOT/data/ATC and are called default.wav and default.vce for
 the wave and index file respectively.  Note that one important change will
 be made in default.vce - currently it is indexed by byte position into the
 sound buffer, but I've decided it would be better to index by time into the
 buffer, since that is more robust to changing the recording quality, and in
 the future possiby using encoding such as Ogg Vorbis.  Also, the first line
 currently contains the number of subsequent lines, but I think that can be
 ditched!
massive snip
David
In what units shall the time index be specified?  The sampling rate sets a 
resolution limit on the timing, so for 8kHz we only need 1/8000 sec = 125 
microseconds precision, but if we have an ambition for higher rates, we need 
more.  [1]
In reality, radio comms are not HiFi standard.  Does anyone know what the 
typical bandwidth is?  Or should we simulate by taking a beautiful 22kHz 
recording and filtering it to sound like the real thing?  Perhaps as an 
option, so one can do radio practice with bell-like clarity at first, and 
graduate to crackly reception of foreign languages and accents later!

Regards
Jonathan

[1] It occurs to me that for chunked formats like WAV, there is a mathematical 
relationship between the byte position and the time offset which could be 
used for conversion, no?

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Re: [Flightgear-devel] ATC voice howto

2004-02-12 Thread David Luff
Jonathan Richards writes:

 In what units shall the time index be specified?  The sampling rate sets a 
 resolution limit on the timing, so for 8kHz we only need 1/8000 sec = 125 
 microseconds precision, but if we have an ambition for higher rates, we need 
 more.  [1]
 In reality, radio comms are not HiFi standard.  Does anyone know what the 
 typical bandwidth is?  Or should we simulate by taking a beautiful 22kHz 
 recording and filtering it to sound like the real thing?  Perhaps as an 
 option, so one can do radio practice with bell-like clarity at first, and 
 graduate to crackly reception of foreign languages and accents later!
 
 Regards
   Jonathan
 
 [1] It occurs to me that for chunked formats like WAV, there is a mathematical 
 relationship between the byte position and the time offset which could be 
 used for conversion, no?
 

Fantastic, a post not talking about rotating planes on their noses ;-))  Time index 
should be specified in seconds I think.  Yes there'll be a lot of decimal points, but 
it's SI, and it's typically what wavefile editors natively display.

In practice it probably doesn't matter if the time resolution spans a few bytes, 
especially at higher rates, since there's generally a slight gap between words, and if 
there isn't it's a good candidate for a run-together phrase.

I guess comms aren't always that clear in real life, but at least if recordings are 
made at good quality to start with its optional whether to degrade them or not.  At 
present the 8,8 limitation forces some degradation, and I'm not sure if the sort of 
'muffled' quality that it imparts is what we're after.

Yes, there is a relationship between byte position and time for wave files, which is 
how I'm going to convert the current default.vce.  I think time is the best way 
forward though ever since someone mentioned Ogg Vorbis at one point.  Plus I think 
Audacity only displays time, not byte position.

As I say, I'm not an audio guy, so I'm open to being persuaded that another way is 
better...

Another thing, some folk running Linux (David Megginson and at least one other) have 
reported only being able to hear the ATIS extremely faintly in the past.  I'm not sure 
why, but it might be best *not* to copy the volume characteristics of default.wav, but 
to do what you think is best and see how it goes.

Anyone got any wavefile editor recommendations BTW?  I used CoolEdit (Windows) for the 
ATIS, but the trial period is now long gone, and when I went to buy it I found the guy 
had sold it to Adobe and the price had tripled.  No thanks!  I'm using Audacity now, 
but it's not entirely polished, and the sound is not in sync with cursor movement when 
playing a small selection in recent versions.  Crucial factors are the ability to 
extend or contract a selection by either edge, and easy copy and pasting into a new 
buffer, plus display of time or byte position.  There's loads for Linux, but I haven't 
found a really likeable one yet, Audacity being the nearest so far.

Cheers - Dave


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Re: [Flightgear-devel] ATC voice howto

2004-02-12 Thread Roy Vegard Ovesen
On Thu, 12 Feb 2004 21:05:25 +, David Luff [EMAIL PROTECTED] 
wrote:

Anyone got any wavefile editor recommendations BTW?
I like GoldWave. www.goldwave.com

I used CoolEdit (Windows) for the ATIS, but the trial period is now long 
gone, and when I went to buy it I found the guy had sold it to Adobe and 
the price had tripled.  No thanks!  I'm using Audacity now, but it's not 
entirely polished, and the sound is not in sync with cursor movement 
when playing a small selection in recent versions.  Crucial factors are 
the ability to extend or contract a selection by either edge, and easy 
copy and pasting into a new buffer, plus display of time or byte 
position.
You can chose to display time or sample position, but the time display has 
only 3 decimal numbers.

--
Roy Vegard Ovesen
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