Vaughan Johnson | 2 Feb 01:05

Re: [Audacity-quality] RM , and schedule for 2.0 release

On 1/31/2012 5:04 PM, Michael Chinen wrote:
> On Mon, Jan 30, 2012 at 3:22 PM, Martyn Shaw <martynshaw99 <at> gmail.com> wrote:
>> Looking good to go here!
>>
>> Martyn
>>
>> On 30/01/2012 06:03, Vaughan Johnson wrote:
>>> This is to confirm I am Release Manager for 2.0, per
>>> http://wiki.audacityteam.org/wiki/Release_Process#Process, step 2.
>>>
>>> Are we good for Feb 14 freeze on all fronts, and release on Feb 29?
> 
> I can do the linux tarballs again.
> I can also do mac if no one responds by the 7th.

Excellent. Thanks, Michael. Anybody up for Windows builds?

- V

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Norm C | 2 Feb 07:12
Picon
Favicon

Spectrogram problems

I've found three bugs in the spectral display, and the attached patch fixes
all of them. They are:
- if EXPERIMENTAL_USE_REALFFTF is not defined, then PowerSpectrum calculates
the DC power incorrectly, by adding together the squares of the (correct) DC
value and the Nyquist value. To verify this:
   - build without EXPERIMENTAL_USE_REALFFTF
   - generate a 5 second tone at 21900 Hz, amplitude 0.5 (with 44100 Hz
sample rate)
   - view as spectrogram (with window size at 256)
   - observe that the bottom frequencies (0 to about 200 Hz) display in a
colour that indicates substantial power
   - rebuild with EXPERIMENTAL_USE_REALFFTF and repeat
   - observe that the bottom frequencies now indicate no power there, as you
would expect
   - in addition, it's clear from Numerical Recipes in C (from which the
source code for the FFT was derived) that the DC
     bin is found in data(1) of the result (our variable rt), and the
Nyquist bin is found in data(2) of the result (our
     variable it). (See pg 513 of ver 2 of the book). "it" should not be
used for calculation of the DC value. Furthermore
     Goldwave displays zero power in the DC bin under the same conditions
(CEPro doesn't; it appears to have this same bug)
- in Spectrum.cpp a check is made for temp > 0 before taking the log10 to
turn it into dB. But if temp is 0 (or -ve) then
  the value used is 0 dB, which is clearly wrong - it should be a large
negative number (I've used -300 dB here)
- the alignment of the spectrogram's bins and the frequency axis is wrong,
by half a bin width. Here's how to verify the problem:
   - generate a sine wave, frequency 280 Hz, duration 5 seconds
   - set Spectrogram preferences to a window size of 1024, Hann window
(Continue reading)

Martyn Shaw | 3 Feb 01:48
Picon

Re: Spectrogram problems

Summary for others:
* Most of this is about code that hasn't been activated since mid-2009.
* A possible patch could be made to put the Spectrogram bins centred 
on their frequency, not offset by half a bin like they are now.
* We should really complete the transition to REALFFTF and get rid of 
the other FFT stuff we have.

Thanks Norm, great analysis work!

On 02/02/2012 06:12, Norm C wrote:
> I've found three bugs in the spectral display, and the attached patch fixes
> all of them. They are:
> - if EXPERIMENTAL_USE_REALFFTF is not defined, then PowerSpectrum calculates
> the DC power incorrectly, by adding together the squares of the (correct) DC
> value and the Nyquist value.

Interesting, but EXPERIMENTAL_USE_REALFFTF has been defined since 6th 
July 2009 and so looks like it always will be.  The remainder of the 
work need to be done to only use the newer code and remove the older 
stuff.

  To verify this:
>     - build without EXPERIMENTAL_USE_REALFFTF
>     - generate a 5 second tone at 21900 Hz, amplitude 0.5 (with 44100 Hz
> sample rate)
>     - view as spectrogram (with window size at 256)
>     - observe that the bottom frequencies (0 to about 200 Hz) display in a
> colour that indicates substantial power
>     - rebuild with EXPERIMENTAL_USE_REALFFTF and repeat
>     - observe that the bottom frequencies now indicate no power there, as you
(Continue reading)

Martyn Shaw | 3 Feb 02:39
Picon

Re: Should display track name in waveform pane

Hi

I don't think I ever got to responding to this.  My students suggested 
that the text should be in yellow, which has a much better contrast 
with all the possible track colours (blue/grey).  It does work better 
IMHO (and I tried it on Win and Mac, at least)

I have so many patches in my source right now it is hard to send a 
single patch but
    dc.SetTextForeground(wxColour(255, 255, 0));
in  TrackArtist::DrawTracks after
  dc.SetFont(labelFont);
does make a considerable improvement.

If I have no negative comments, I will commit that before 2.0.

Thanks
Martyn

On 18/01/2012 20:45, Martyn Shaw wrote:
> Thanks Gale and Edgar!
> Martyn
>
> On 18/01/2012 18:36, Gale (Audacity Team) wrote:
>> Martyn wrote:
>>> Somebody might like to update the documentation, but that's not a
>>> release blocker.
>>
>> Actually, it would be a blocker in terms of Manual "bugs":
>> http://manual.audacityteam.org/man/ToDo-1
(Continue reading)

Picon

Re: Converting video from English to another Language

Thank you Reid.
About the billing thing, now that a novice at this, how much time would it take a professional like you to do(converting from English to another language like French) one video? The video is as long as 10mins. I'll base my charge on the time it would take a professional to deliver the job. lol

Am preparing to send the client a copy of my costing. Hehehehe.

Thanking you again,
Simon.

On Mon, Jan 30, 2012 at 1:17 AM, Reid Zak <reidzak <at> gmail.com> wrote:
If you do want to split and then rejoin the audio from the video, you'll need some video software to do that. The reason is because Audacity doesn't edit video, only audio. So you can use Audacity with ffmpeg to import the audio from the video, you can export as a wav file, but then you will need to use another program to combine it back with the video.
One program you could use would be SUPER, which will let you mux the video and audio streams back into one finished file.

--Reid


On Sun, Jan 29, 2012 at 8:05 AM, Simon Peter Muwanga <smuwanga <at> gmail.com> wrote:
Thanks Josiah,
These guys(my clients) gave me a website done in English. The website has video clips that are tutorials to their users. The users are not able to understand English. They use the local language to communicate.

Have tried doing some dev't with audacity, 2yrs ago, I felt my problem would solved. :)
The client would want a new video, have the same content as the original but every audio in the local language.

I see am likely to use the voiceover effect. I hope audacity will save me, I am not sure though.

Regards,
Simon.

On Sun, Jan 29, 2012 at 4:38 PM, Reid Zak <reidzak <at> gmail.com> wrote:
This video?  http://www.youtube.com/watch?feature=player_embedded&v=I3tTUmeWztU 
Maybe you could do a voiceover effect, where you leave their voices in the background, but you overdub your translation a couple seconds later, like they do on national television. Did they send you any specific instructions on what to do?

Are you wanting to split the audio from the video, put your audio in the same file as the video, then send them back a file with audio and video in it? Or do you just need to send them an audio file, and they can match it up with the video at their leisure? (that would be easier)

--Josiah

On Sun, Jan 29, 2012 at 6:47 AM, Simon Peter Muwanga <smuwanga <at> gmail.com> wrote:
 Hi Reid,
Thanks man. I like the tips.

About the video format, meant the video. I will confirming the type of video format it is. Thanks for the correction. :)
 Its accessible at http://learn.go-on.co.uk/ . One of the video clips is on that page, just check at the top-right side of the website. Others are viewable when you have an account. Its free subscription for an account.

The format being a video. How is that meant to be put in the local language?
Here is my work-plan.
-Listen as I watch the video
-Write down the text in English as is.
-Record an audio file to match the video clip as advised by Reid.
-Play the video, with the audio at the same time.
 ->Suppress the video's audio in favour of the recorded/translated audio
-Ship product to the client.

I would appreciate help on informing the work-plan. lol

Thanking you,
Simon.

On Sun, Jan 29, 2012 at 8:18 AM, Reid Zak <reidzak <at> gmail.com> wrote:
Mp3 is an audio format, not a video format. Do you mean audio? If so, just try writing out what the text is in English, then translate it into your language and write it down, then use Audacity to record yourself.
Some tips:
 - Make sure you have a good quality mic
 - Check your recording to make sure you aren't clipping or distorting
 - Speak at a specific (constant) distance away from the microphone.
 - If you make a mistake, you can re-record the part, or you can start over from the beginning.
 - You'll need the lame libraries if you want to export back as MP3:  http://lame1.buanzo.com.ar/#lamewindl 

I don't know how much you should charge... maybe $25/hour? It all depends on the difficulty of the work, how much experience you've had, and how good you are at doing it.

--Reid


On Sat, Jan 28, 2012 at 10:44 PM, Simon Peter Muwanga <smuwanga <at> gmail.com> wrote:
Dear All,

I am given a task to convert six video .mp3 clips into my local language. Each clip is in English. Of course the actions will remain.

Could someone tell me what I need to get the job done?
Probably how long it will take to finish the work?
How much money should I charge ?

Regards,
Simon.


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Steve the Fiddle | 3 Feb 14:09
Picon

Potential P2 in Audacity 2.0.0 alpha

This relates to bug 152 http://bugzilla.audacityteam.org/show_bug.cgi?id=152
and this forum topic: http://forum.audacityteam.org/viewtopic.php?f=16&t=63406

Summary:

Very high sample values in audio data can cause Exported audio to be
corrupted from the position of very high sample value for the
remaining duration of the file.
The corruption in Audacity is persistent until Audacity is restarted.

Steps to reproduce:

Dither set to "shaped".
32 bit float format audio track containing 1 or more samples with
values of 100 dB or more (NAN values may also cause this problem, but
I've not fully tested that).
Export track in any integer format.

Test the Imported track and it will be found that it flat-lines at -1
dB from the position of where the very high sample value was.

Create a new Audacity project without restarting Audacity.
Record or generate some audio.
Export in any integer format.
The Exported file is corrupted (flat-line at -1 dB).

Gale wrote here:
http://forum.audacityteam.org/viewtopic.php?f=16&t=63406&start=20#p169975
"1.3.3 processes the project correctly, but I haven't tried later
versions. I'm not surprised, because I've seen several other reports
of "flatlining" similar to this on the Forum recently and I didn't
think they were all inability to open the source data (though one
was)."

So it appears that this is not an isolated incident, and it does not
matter if the "invalid" samples are caused by an improper Nyquist
command (as in bug 152) or by a corrupt sample value occurring during
recording (as in the forum topic).

For testing purposes, simple way to create an "illegal" sample is to
run the following code in the Nyquist Prompt on one or more samples of
an audio track:
(snd-const (db-to-linear 100) 0 *sound-srate* (get-duration 1))

Is there any way that audio track samples can be limited to sensible
values, for example, less than +50 dB?
Can this be fixed before 2.0?

I think that this bug must be at least P3, but really it seems more
like a P2. Can anything be done before 2.0?

(this e-mail was written in a bit of a hurry as I think you all need
to be aware of this asap, so apologies for any errors)

Steve

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Steve the Fiddle | 3 Feb 14:12
Picon

Re: Potential P2 in Audacity 2.0.0 alpha

On 3 February 2012 13:09, Steve the Fiddle <stevethefiddle <at> gmail.com> wrote:
> This relates to bug 152 http://bugzilla.audacityteam.org/show_bug.cgi?id=152
> and this forum topic: http://forum.audacityteam.org/viewtopic.php?f=16&t=63406
>
> Summary:
>
> Very high sample values in audio data can cause Exported audio to be
> corrupted from the position of very high sample value for the
> remaining duration of the file.
> The corruption in Audacity is persistent until Audacity is restarted.
>
> Steps to reproduce:
>
> Dither set to "shaped".
> 32 bit float format audio track containing 1 or more samples with
> values of 100 dB or more (NAN values may also cause this problem, but
> I've not fully tested that).
> Export track in any integer format.
>
> Test the Imported track and it will be found that it flat-lines at -1
> dB from the position of where the very high sample value was.

Sorry, that should be:
"it flat-lines at -1.0 (linear)"

>
> Create a new Audacity project without restarting Audacity.
> Record or generate some audio.
> Export in any integer format.
> The Exported file is corrupted (flat-line at -1 dB).

and again, "flat-line at -1.0 linear"

>
>
> Gale wrote here:
> http://forum.audacityteam.org/viewtopic.php?f=16&t=63406&start=20#p169975
> "1.3.3 processes the project correctly, but I haven't tried later
> versions. I'm not surprised, because I've seen several other reports
> of "flatlining" similar to this on the Forum recently and I didn't
> think they were all inability to open the source data (though one
> was)."
>
> So it appears that this is not an isolated incident, and it does not
> matter if the "invalid" samples are caused by an improper Nyquist
> command (as in bug 152) or by a corrupt sample value occurring during
> recording (as in the forum topic).
>
> For testing purposes, simple way to create an "illegal" sample is to
> run the following code in the Nyquist Prompt on one or more samples of
> an audio track:
> (snd-const (db-to-linear 100) 0 *sound-srate* (get-duration 1))
>
> Is there any way that audio track samples can be limited to sensible
> values, for example, less than +50 dB?
> Can this be fixed before 2.0?
>
> I think that this bug must be at least P3, but really it seems more
> like a P2. Can anything be done before 2.0?
>
> (this e-mail was written in a bit of a hurry as I think you all need
> to be aware of this asap, so apologies for any errors)
>
> Steve

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Reid Zak | 3 Feb 16:15
Picon

Re: Converting video from English to another Language

I replied off list, as even though these are good questions, they are becoming increasingly non-relevant to an Audacity Dev list :-)


--Reid

On Fri, Feb 3, 2012 at 3:38 AM, Simon Peter Muwanga <smuwanga <at> gmail.com> wrote:
Thank you Reid.
About the billing thing, now that a novice at this, how much time would it take a professional like you to do(converting from English to another language like French) one video? The video is as long as 10mins. I'll base my charge on the time it would take a professional to deliver the job. lol

Am preparing to send the client a copy of my costing. Hehehehe.

Thanking you again,
Simon.


On Mon, Jan 30, 2012 at 1:17 AM, Reid Zak <reidzak <at> gmail.com> wrote:
If you do want to split and then rejoin the audio from the video, you'll need some video software to do that. The reason is because Audacity doesn't edit video, only audio. So you can use Audacity with ffmpeg to import the audio from the video, you can export as a wav file, but then you will need to use another program to combine it back with the video.
One program you could use would be SUPER, which will let you mux the video and audio streams back into one finished file.

--Reid


On Sun, Jan 29, 2012 at 8:05 AM, Simon Peter Muwanga <smuwanga <at> gmail.com> wrote:
Thanks Josiah,
These guys(my clients) gave me a website done in English. The website has video clips that are tutorials to their users. The users are not able to understand English. They use the local language to communicate.

Have tried doing some dev't with audacity, 2yrs ago, I felt my problem would solved. :)
The client would want a new video, have the same content as the original but every audio in the local language.

I see am likely to use the voiceover effect. I hope audacity will save me, I am not sure though.

Regards,
Simon.

On Sun, Jan 29, 2012 at 4:38 PM, Reid Zak <reidzak <at> gmail.com> wrote:
This video?  http://www.youtube.com/watch?feature=player_embedded&v=I3tTUmeWztU 
Maybe you could do a voiceover effect, where you leave their voices in the background, but you overdub your translation a couple seconds later, like they do on national television. Did they send you any specific instructions on what to do?

Are you wanting to split the audio from the video, put your audio in the same file as the video, then send them back a file with audio and video in it? Or do you just need to send them an audio file, and they can match it up with the video at their leisure? (that would be easier)

--Josiah

On Sun, Jan 29, 2012 at 6:47 AM, Simon Peter Muwanga <smuwanga <at> gmail.com> wrote:
 Hi Reid,
Thanks man. I like the tips.

About the video format, meant the video. I will confirming the type of video format it is. Thanks for the correction. :)
 Its accessible at http://learn.go-on.co.uk/ . One of the video clips is on that page, just check at the top-right side of the website. Others are viewable when you have an account. Its free subscription for an account.

The format being a video. How is that meant to be put in the local language?
Here is my work-plan.
-Listen as I watch the video
-Write down the text in English as is.
-Record an audio file to match the video clip as advised by Reid.
-Play the video, with the audio at the same time.
 ->Suppress the video's audio in favour of the recorded/translated audio
-Ship product to the client.

I would appreciate help on informing the work-plan. lol

Thanking you,
Simon.

On Sun, Jan 29, 2012 at 8:18 AM, Reid Zak <reidzak <at> gmail.com> wrote:
Mp3 is an audio format, not a video format. Do you mean audio? If so, just try writing out what the text is in English, then translate it into your language and write it down, then use Audacity to record yourself.
Some tips:
 - Make sure you have a good quality mic
 - Check your recording to make sure you aren't clipping or distorting
 - Speak at a specific (constant) distance away from the microphone.
 - If you make a mistake, you can re-record the part, or you can start over from the beginning.
 - You'll need the lame libraries if you want to export back as MP3:  http://lame1.buanzo.com.ar/#lamewindl 

I don't know how much you should charge... maybe $25/hour? It all depends on the difficulty of the work, how much experience you've had, and how good you are at doing it.

--Reid


On Sat, Jan 28, 2012 at 10:44 PM, Simon Peter Muwanga <smuwanga <at> gmail.com> wrote:
Dear All,

I am given a task to convert six video .mp3 clips into my local language. Each clip is in English. Of course the actions will remain.

Could someone tell me what I need to get the job done?
Probably how long it will take to finish the work?
How much money should I charge ?

Regards,
Simon.

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Gale Andrews | 3 Feb 18:01

Re: Potential P2 in Audacity 2.0.0 alpha


| From Steve the Fiddle <stevethefiddle <at> gmail.com> 
| Fri, 3 Feb 2012 13:09:51 +0000
| Subject: Potential P2 in Audacity 2.0.0 alpha
> This relates to bug 152 http://bugzilla.audacityteam.org/show_bug.cgi?id=152
> and this forum topic: http://forum.audacityteam.org/viewtopic.php?f=16&t=63406
> 
> Summary:
> 
> Very high sample values in audio data can cause Exported audio to be
> corrupted from the position of very high sample value for the
> remaining duration of the file.
> The corruption in Audacity is persistent until Audacity is restarted.
> 
> Steps to reproduce:
> 
> Dither set to "shaped".
> 32 bit float format audio track containing 1 or more samples with
> values of 100 dB or more (NAN values may also cause this problem, but
> I've not fully tested that).
> Export track in any integer format.
> 
> Test the Imported track and it will be found that it flat-lines at -1
> dB from the position of where the very high sample value was.
> 
> Create a new Audacity project without restarting Audacity.
> Record or generate some audio.
> Export in any integer format.
> The Exported file is corrupted (flat-line at -1 dB).
> 
> 
> Gale wrote here:
> http://forum.audacityteam.org/viewtopic.php?f=16&t=63406&start=20#p169975
> "1.3.3 processes the project correctly, but I haven't tried later
> versions. I'm not surprised, because I've seen several other reports
> of "flatlining" similar to this on the Forum recently and I didn't
> think they were all inability to open the source data (though one
> was)."
> 
> So it appears that this is not an isolated incident, and it does not
> matter if the "invalid" samples are caused by an improper Nyquist
> command (as in bug 152) or by a corrupt sample value occurring during
> recording (as in the forum topic).
> 
> For testing purposes, simple way to create an "illegal" sample is to
> run the following code in the Nyquist Prompt on one or more samples of
> an audio track:
> (snd-const (db-to-linear 100) 0 *sound-srate* (get-duration 1))
> 
> Is there any way that audio track samples can be limited to sensible
> values, for example, less than +50 dB?
> Can this be fixed before 2.0?
> 
> I think that this bug must be at least P3, but really it seems more
> like a P2. Can anything be done before 2.0?
> 
> (this e-mail was written in a bit of a hurry as I think you all need
> to be aware of this asap, so apologies for any errors)

Of course I'd like it fixed but I was envisaging entering it as P3 then 
raising it and some others to P2 after 2.0 so as to prioritise them
(or some other way to do so).  

The problem has been around since at least 1.3.8 (just tested). 

I can only recall one other confirmed Forum report like this before 
(though I cannot find it right now). In that case the recording had a 
very obvious visual and audible glitch where there were very high 
sample values. Exports were then corrupted at -1. But in the current 
case the fault is not obvious, so cases may be more common than we 
thought.   

I'm not sure why there have increasing numbers of reports of "flatlined" 
exports recently. It could merely be that I've been actively looking for 
them, thinking they could be project reopening bugs. These sort of 
reports look to be fairly common on the Forum, but given the reports 
are usually vague they are usually dismissed as user running out of 
disk space and not noticing the holes in the project, or having a 
fragmented drive. 

Gale  

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Gale Andrews | 3 Feb 19:02

Borderline P2/P3: Repeatable MP3 import crash if tried by libsndfile


If you want to consider other borderline P2/P3's, MP3 files that contain 
duplicated metadata tags will crash Audacity if libsndfile tries them,
which seems to be a fault in libsndfile:   
http://forum.audacityteam.org/viewtopic.php?f=16&t=62791&start=20#p166637 .

Link to the file in the Forum is dead but the file is  here: 
http://www.gaclrecords.org.uk/bugs/02_Flawless_%28Go_To_The_City%29.zip .

To reproduce, quit Audacity, set audacity.cfg to "NewPrefsInitialized=1", 
launch Audacity, File > Import > Audio, leave filter on "All Files", open the
MP3.

I think this started with the libsndfile upgrade in 1.3.13, however the 
above steps won't reproduce in 1.3.13 because in that case libmad tries
the MP3 first, whereas in 1.3.14 libsndfile now tries the file first. 

Report frequency in 1.3.14 about four per week at the moment. 

Workaround: Set an extended import rule so libmad tries the MP3 first. 

I can live with a P3. 

Gale  

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