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festvox's Introduction

              "Building Voices in Festival"
 Alan W Black ([email protected]), Kevin Lenzo ([email protected])
                 and see ACKNOWLEDGEMENTS
                  http://www.festvox.org

For full details about voice building see the document itself

http://festvox.org/bsv/

The included documentation, scripts and examples should be sufficient for an interested person to build their own synthetic voices in currently supported languages or new languages in the University of Edinburgh's Festival Speech Synthesis System. The quality of the result depends much on the time and skill of the builder. For English it may be possible to build a new voice in a couple of days work, a new language may take months or years to build. It should be noted that even the best voices in Festival (or any other speech synthesis system for that matter) are still nowhere near perfect quality.

This distribution includes

Support for designing, recording and autolabelling statistical parametric
    synthesis voices
Support for designing, recording and autolabelling diphone databases
Support for designing, recording and autolabelling unit selection dbs
Building simple limited domain synthesis engines
Support for building rule driven and data driven prosody models
   (duration, intonation and phrasing)
Support for building rule driven and data driven text analysis
Lexicon and building Letter to Sound rule support
Predefined scripts for building new US (and UK) English voices
Predefined scripts for building grapheme(++) voices for any language
Scripts for designing and selecting prompts to record for
   arbitrary languages

New in 2.8

https://github.com/festvox/festival/
Grapheme built voices can be converted to .flitevox files for android
Database size reduction for random forest clustergen voices
Random Forests for F0 prediction too
18 English voices, and 13 Indic voices

New in 2.7

Random forest models building for spectrum and duration in clustergen
Grapheme based synthesizers (with specific support for large number
  of unicode writing systems)
Clustergen state and stop value optimization
Wavesurfer label support
SPAM F0 support
Phrase break support
Support for SPTK's mgc parameterization

New in 2.3

Support for cygwin tools under Windows
Substantially improved CLUSTERGEN support with mlpg and mlsb

WARNING

This is not a pointy/clicky plug and play program to build new voices. It is instructions with discussion on the problems and an attempt to document the expertise we have gained in building other voices. Although we have tried to automate the task as much as possible this is no substitute for careful correction and understanding of the processes involved. There are significant pointers into the literature throughout the document that allow for more detailed study and further reading.

REQUIREMENTS

A Unix Machine

although there is nothing inheritantly Unix about the scripts, no
attempt has yet been made about porting this to other platforms

Festival and Speech Tools

This uses speech tools programs and festival itself at various
stages in builidng voices as well as (of course) for the final
voices.  Festival and the Edinburgh Speech Tools are available from

   http://www.cstr.ed.ac.uk/projects/festival/
   
or

   http://www.festvox.org/festival

or

   https://github.com/festvox
   
It is recommended that you compile your own versions of these
as you will need the libraries and include files to build some
programs in this festvox.

Wavesurfer

To display waveforms, spectragrams and phoneme labels.

Patience and understanding

Building a new voice is a lot of work, and something will probably
go wrong which may require the repetition of some long boring and
tedious process.  Even with lots of care a new voice still might 
just not work.  In distributing this document we hope to increase the
basic knowledge of synthesis out there and hopefully find people 
who can improve on this making the processing easier and more reliable
in the future.

INSTALLATION

You must have the Edinburgh Speech Tools and Festival instllation before you can build the tools in the festvox distribution.

Unpack festvox-2.8-release.tar.gz or clone it from github

git clone https://github.com/festvox/festvox
cd festvox
./configure
make

The configuration basically tries to find your version of the Edinburgh Speech Tools and uses its configuration to set compiler type etc. So you must have that installed. If configure fails try expliciting setting your ESTDIR environment variable to point ot your compiled version of the Speech Tools.

A pre-generated version of the document in html and postscript are distributed in the html/ directory

If you need to build the document itself, you will need a working version of the docbook tools, which may (or may not) already be installed on your system

To build the documenation

cd docbook
make doc

Note that even if the documentation build fails you can still use all the scripts and programs.

To use the scripts and programs in the festvox distribution each user is expected to have the environment variables ESTDIR and FESTVOXDIR set for example as (if you use bash, zsh, ksh or sh)

export ESTDIR=/home/awb/projects/speech_tools
export FESTVOXDIR=/home/awb/projects/festvox
export FLITEDIR=/home/awb/projects/flite
export SPTKDIR=/home/awb/projects/SPTK

Or if you use csh or tcsh

setenv ESTDIR /home/awb/projects/speech_tools
setenv FESTVOXDIR /home/awb/projects/festvox
setenv FLITEDIR /home/awb/projects/flite
setenv SPTKDIR /home/awb/projects/SPTK

Remember to set these to where your installations are, not ours.

festvox's People

Contributors

awbcmu avatar festvox avatar lenzo-ka avatar saikrishnarallabandi avatar zeehio avatar

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festvox's Issues

What is the code to convert new sentences into speech?

I did everything you've given in this grapheme method. Also I have created the test files. They sound good. I want to provide some new sentences and get the voice for them.
May I know what is the code to provide a new sentence and get its voice output from here?

Thank you.

Cannot build clustergen voice

I tried to follow the steps to create clustergen voice.
However the sequence is failing to generate voice.

The script
./bin/build_cg_rfs_voice
invokes the script
./bin/build_cg_voice

This script invokes the following line
./bin/do_clustergen dur etc/txt.done.data.train

Everything goes fine upto that line however

I start getting errors like
Cannot open file festival/utts_hmm/arctic_a0050.utt as tokenstream
load_utt: can't open utterance input file festival/utts_hmm/arctic_a0050.utt
utt.load: loading from "festival/utts_hmm/arctic_a0050.utt" failed
Cannot open file festival/utts_hmm/arctic_a0048.utt as tokenstream
load_utt: can't open utterance input file festival/utts_hmm/arctic_a0048.utt

etc

I looked into what is being done by above line. It invokes ./bin/do_clustergen. That has following snippet

if [ $1 = "dur" ]
then
./bin/make_dur_model_mcep $PROMPTFILE
cp -p festvox/$FV_VOICENAME"_dur.scm" festvox/$FV_VOICENAME"_durdata_cg.scm"
exit 0
fi
........the above snippet invokes ./bin/make_dur_model_mcep. It has following logic.
...
awk '{printf("festival/utts_hmm/%s.utt\n",$2)}' >utthmmfile
...

however the folder festival/utts_hmm is not populated at all at the time the above code is invoked.
.... and thus the errors start.

Further down the line .........it appears to go through an infinite loop.........not sure about this.

Please let me know if I am following any wrong sequence of steps.
Please direct me to proper source where I can get any guideline, if you are busy.

Thanks in advance.
-Mahesh

tempo make_voicing

if TYPE == tempo use $FESTVOXDIR/src/vc/src/tempo/tempo

$FESTVOXDIR/src/vc/src/tempo/tempo has gone ...

src/clustergen/get_cd_dtw_mgc no "$SAMPFREQ" = "44100"

in do_clustergen
FRAMELEN=$(echo | awk "{print int(0.025*$SAMPFREQ)}")
FRAMESHIFT=$(echo | awk "{print int(0.005*$SAMPFREQ)}")

	FWARP[8000]=0.312
	FWARP[11025]=0.357
	FWARP[16000]=0.42
	FWARP[22050]=0.455
	FWARP[32000]=0.504
	FWARP[44100]=0.544
	FWARP[48000]=0.554
    FREQWARP=FREQWARP=${FWARP[$SAMPFREQ]}        

in src/clustergen/get_cd_dtw_mgc we only have the cases of
16000 and 48000

ch_wave and -scaleN Scaling factor with normalization.

for do_pm -scaleN is fixed to 0.9
but in general/simple_powernormalize
0.65 is chosen ?
Are the different options "historical" 'due to the quality of the rcording at the time ?
What are the impact of the choice made ?

right usage of ch_wave

https://github.com/festvox/festvox/blob/3be8dd4e3d2ac680d8d6b33fe6212322636b45c8/src/intonation/make_f0_v_sptk#L126C6-L127C63

doesn't follow the usage of ch_wave:

Usage: ch_wave [input file0] [input file1] ... -o [output file]
Summary: change/copy/combine waveform files
use "-" to make input and output files stdin/out

and lead to the error

Write access failed for temporary file
Cannot open file _tmp.3067154/est_3067160_00000 as tokenstream

Could it be:

`$ESTDIR/bin/ch_wave -otype raw  wav/$fname.wav > $TMP.raw`

(without <)

Use num2words for universal numbers to words converter in Festival

I can use num2words for automatic numbers words converter. Using (system) function, in my function (define (festpb_pt::number token name)

(define (festpb_pt::number token name)
  "(festpb_pt::number token name)
Return list of words that pronounce this number in pt."

;Set name random, using rand function
(set! ra (rand))

;Set command for system function, piper output of num2words in file with name of variable ra 
(set! command (format nil "num2words -l pt-BR %s > %s" name ra))

;Execute command using system function
(system command)

;Load file how a list with load function
(load ra t)
)

For deletion of cache file have function in Festvox Doc with this in my function (festpb_pt::number token name)

(delete-file ra)

But my function, not return when have this function.

pitch output format in festvox/src/clustergen/get_str_sptk

in the doc, there is no output format "5" : only 0 for pitch, 1 for f0, 2 for log(f0)
pitch - pitch extraction
usage:
pitch [ options ] [ infile ] > stdout
options:
-a a : algorithm used for pitch [0]
estimation
0 (RAPT)
1 (SWIPE')
-s s : sampling frequency (kHz) [16.0]
-p p : frame shift [80]
-T T : voiced/unvoiced threshold [0.0]
(used only for RAPT algorithm)
-t t : voiced/unvoiced threshold [0.3]
(used only for SWIPE' algorithm)
-L L : minimum fundamental [60]
frequency to search for (Hz)
-H H : maximum fundamental [240]
frequency to search for (Hz)
-o o : output format [0]
0 (pitch)
1 (f0)
2 (log(f0))

 for i in `seq 1 5`
    do
	$ESTDIR/bin/sigfilter -firfilter $FTMP.h$i -forder $FORDER wav/$fname.wav -otype raw | \
	    $X2X +sf | $SPTKDIR/bin/pitch -a 0 -H $F0MAX -L $F0MIN -p $FRAMESHIFT -s $SAMPKHZ -o 5 | \
	    $X2X +fa4 | awk '{print $4}' > $TMP.h$i.str
    done

luckily ifor swipe (SPTK-3.6/bin/pitch/swipe/swipe.c ) the default is : /* pitch */
but what about rapt ? (pitch -a 0 )

What are archives .frq generated in the process of voice construction?

Hello, In building the voice model, I noticed that it has files called.frq what he are? Are files related to frequency and pitch and etc? I have a database that has some files .frq and files .llsm, is it possible to utilize it these files in the database for voice construction?

problematic pitchmark's call in make_f0_pm need in voicing

$ESTDIR/bin/ch_wave -scaleN 0.9 wav/$fname.wav -F 16000 |

but 'pitchmark' has a file name as input

    F0MIN=50
    F0MAX=200
    F0MEAN=110
    # >....<
    F0_ARGS=`echo $F0MIN $F0MAX $F0MEAN | awk '{printf("-min %f -max %f -def %f",1.0/$2,1.0/$1,1.0/$3)}'`
    PM_ARGS='-wave_end -lx_lf 140 -lx_lo 111 -lx_hf 80 -lx_ho 51 -med_o 0'
    $ESTDIR/bin/ch_wave -scaleN 0.9 wav/$fname.wav -F 16000 | 
    $ESTDIR/bin/pitchmark -o pm_unfilled/$fname.pm -otype est $PM_ARGS $F0_ARGS

will raise

Write access failed for temporary file
Cannot open file TMP/est_868612_00000 as tokenstream
Wave load: can't open file "TMP/est_868612_00000"
Cannot recognize file format or cannot access file: "-"
Could it be something like ?
    # pitchmark has a file name as input
    "$ESTDIR"/bin/ch_wave -scaleN 0.9 -F 16000 wav/"$fname".wav -o tmp_"$fname"
    "$ESTDIR"/bin/pitchmark -o pm_unfilled/"$fname".pm -otype est  $PM_ARGS $F0_ARGS tmp_"$fname"

Building Indic Voices using Falcon

@saikrishnarallabandi I have been trying to build a synthesizer from these steps

the ./bin/do_build command is not able to find the option get_phseq. I have installed all my libraries using
fest_build.sh.txt

`
(falconenv) gnani@prajwal:~/projects/falcon/festvox/voices/indic/hindi_female/vox$ ./bin/do_build get_phseq

./bin/do_build unknown option get_phseq
`

mysp_settings

https://github.com/festvox/festvox/blob/ec3e8b4ee739447b7c2a062506759049d4ab1b5a/src/ehmm/etc/mysp_settings#L1-Ll119

leave alone

Lporder: 12
CepsNum: 16

WaveDir: wav/
HeaderBytes: 44
SamplingFreq: 16000
FrameSize: 160
FrameShift: 80
Lporder: 12
CepsNum: 16
FeatDir: ./ehmm/feat
Ext: .wav

does it mean that that we don't have to to touch it if we work with other SamplingFreq than 16000? it will help to know what exactly CepsNum is ( it's link with MCEP Order if any ..)

Unable to build voice for non-English language (Bengali)

I followed the steps mentioned in Building Indic voices and another issue's comment and tried to build voice for Bengali language, but failed to do so. What I did is given below:

    1. I downloaded the data available in Fesvox's example Indic database. I used only a small portion of it (20 files) for the process, as training took long time in the first try.
    1. I installed SPTK-3.6,festvox2.4, festival2.4, speech-tools2.4 in a directory, installed them (without error) and configured FESTVOXDIR, ESTDIR, SPTKDIR environment variables (in Ubuntu 16.04).
    1. I installed SPTK-3.6,festvox2.5, festival2.5, speech-tools2.5 in a directory, installed them (without error) and configured FESTVOXDIR, ESTDIR, SPTKDIR environment variables (in Ubuntu 16.04).
    1. I followed the instructions given in the link Building Indic voices, as well as from another issue's comment for both version-2.4 and version-2.5 (I know about the change in these version about setting (defvar lex:language 'Hindi) to Bengali and set them accordingly).
    1. When building with festvox2.4 and other 2.4-releases, I didn't face problem during training, the whole process went smoothly. But when I ran the following lines mentioned in Building Indic voices, I get error.

And generate some test examples, the first to give MCD and F0D objective measures, the second to generate standard tts output
./bin/do_clustergen cg_test resynth cgp etc/txt.done.data.test
./bin/do_clustergen cg_test tts tts etc/txt.done.data.test
Error I faced:

CG test_resynth bn_00010
SIOD ERROR: unbound variable : rfs_info
CG test_resynth bn_00010 Failed
CG test_resynth bn_00021 
Segmentation fault (core dumped)
awk: cmd. line:2: fatal: division by zero attempted
awk: cmd. line:2: fatal: division by zero attempted
awk: cmd. line:2: fatal: division by zero attempted
awk: cmd. line:2: fatal: division by zero attempted
    1. When tried to build with festvox-2.5 and other 2.5-releases from this github account, I faced error, after the following command mentioned in both of the sources (another issue's comment and Building Indic voices ). The command is:
      ./bin/do_clustergen parallel cluster etc/txt.done.data.train
      and the error I found:
Setting clustergen params
SIOD ERROR: unbound variable : cg:f0_clustersize
BACKTRACE:
   0: cg:f0_clustersize
   1: (list (quote wagon_cluster_size_f0) cg:f0_clustersize)
   2: (list
    (quote (clunit_relation mcep))
    (list (quote wagon_cluster_size_mcep) cg:mcep_clustersize)
    ...)
   3: (append
    (list
     (quote (clunit_relation mcep))
     (list (quote wagon_cluster_size_mcep) cg:mcep_clustersize)
     ...)
    clunits_params)
   4: (set!
    clustergen_params
    (append
     (...)
     clunits_params))
   5: (begin
    (set! cg:parallel_tree_build t)
    (build_clustergen "etc/txt.done.data.train"))

Please let me know, if I am doing something wrong, and if possible, direct me to the correct steps. If possible, direct me to some proper sources that I can use to build Indic voices in Festival/Festvox.

Thanks,
~Numan

$PDA_PARAMS, PDAPARAMS=

in src/general/make_f0 we have

$ESTDIR/bin/pda -L -fmin 70 -fmax 240 -o f0/$fname.f0 -otype ssff $PDA_PARAMS wav/$fname.wav

and we have in src/general/prune_middle_silences

PDAPARAMS='-fmin 70 -fmax 240'
followed by
$ESTDIR/bin/pda -L $PDAPARAMS -otype est_ascii wav/$fname.wav | ...

so PDA_PARAMS and PDAPARAMS are not the same .. or ... should be the same

  • in make_f0, unless we knew about it the variable $PDA_PARAMS is unset.
  • in src/intonation/find_f0_stats or hts_build/build_utts
    are used fmin=$FMIN and fmax=$FMAX as provided by etc/f0.params

Festvox emulabel: not found

Hello,
I built out a new system on Ubunto 20.x. I am trying to do the 'Telling the time' exercise from the documentation. I have run into a problem with the display_sg utility. It returns the following message when I execute the command:

/home/greg/projects/build/festvox/src/general/display_sg: 68: emulabel: not found

I reinstalled the festvox tools and did not see any errors. Does anyone have any suggestions?

help needed for 44100 Hz : mcep

SIG2FV=$ESTDIR/bin/sig2fv
#SIG2FVPARAMS='-coefs melcep -delta melcep -melcep_order 12 -fbank_order 24 -shift 0.01 -factor 2.5 -preemph 0.97'
SIG2FVPARAMS='-coefs melcep -melcep_order 12 -fbank_order 24 -shift 0.01 -factor 2.5 -preemph 0.97'

# These parameters are chosen by becuase they seems vaguely right
# and do produce reasonable results, but if you think you know better
# you probably do, feel free to experiment
SIG2FVPARAMS='-coefs melcep -delta melcep -melcep_order 12 -fbank_order 24 -shift 0.005 -factor 5.0 -preemph 0.97'

Could it be that these parameters are not "reasonable"
for a voice in 44100 Hz?
I certainly not know better, and I desperately need help.
Thanks for any light

How to integrate my build voice with festival

I have already build some indic tts models using festvox, I want to integrate them into festival so that i can play the utterance from command line. Like:
festival '(SayText "नमस्कार आपका स्वागत है.")'.

MGCEP versus MGEP : typo ?

in /festvox/src/clustergen/get_str_sptk (for example)

MGCEP=$SPTKDIR/bin/mcep
doesn't follow what it seems a pattern
   X2X=$SPTKDIR/bin/x2x
   ..
   SOPR=$SPTKDIR/bin/sopr
but both $SPTKDIR/bin/mcep and $SPTKDIR/bin/mgcep exists

could it be a typo ?

silences

      the option -silences  of  $FESTVOXDIR/src/general/smooth_f0  is a comma separated list of silence names
so I expected  the LIST of phones I declared as silences in my phoneset.scm will be taken into account, after all I'm not te only one to want a list not reduced to a single element  for radio-phones (PhoneSet.silences '(pau h# brth)) 

but indeed we have in make_f0_v_sptk
    $FESTVOXDIR/src/general/smooth_f0 -o f0/$fname.f0 sptk_f0/$fname.f0 -otype ssff -lab lab/$fname.lab -silences $SILENCE  ....
where  SILENCE=awk '{print $1}' etc/silence
It looks fine but if you did not have a etc/silence` to start with, find_silence_name is invoked

(define (find_silence_name)
  (set! INST_LANG_VOX::clunits_prompting_stage t)
  (voice_INST_LANG_VOX_clunits)
  (set! silence (car (cadr (car (PhoneSet.description '(silences))))))
  (set! sfd (fopen "etc/silence" "w"))
  (format sfd "%s\n" silence)
  (fclose sfd)
)
and $SILENCE will be reduced to one silence name ..

By the way, despite its reassuring name "find_silence_name" is able to insidiously "destroy" a "etc/silence" if you are inside the directory of your voice and spits an error if you are not

Problem in building clustergen indic voice

Following the link "http://festvox.org/bsv/x3528.html", tried to built hindi tts from scratch on sample to 100 'hindi' wav files ['hindi_0001' - 'hindi_0102'] obtained from 'cmu_indic_hin_ab.tar.bz2'.

Every script works fine upto the following command--
./bin/do_clustergen parallel cluster etc/txt.done.data.train

getting lots of file not found & Segmentation fault (core dumped) Errors

example------------ final lines of above command:

Segmentation fault (core dumped)
Dataset of 0 vectors of 64 parameters from: festival/feats/9r_1.feats
Attempt to access channel 1 of 0 channel track
Dataset of 0 vectors of 64 parameters from: festival/feats/9r_1.feats
Segmentation fault (core dumped)
Segmentation fault (core dumped)
Dataset of 0 vectors of 64 parameters from: festival/feats/9r=_1.feats
Attempt to access channel 1 of 0 channel track
Dataset of 0 vectors of 64 parameters from: festival/feats/9r=_1.feats
Dataset of 0 vectors of 64 parameters from: festival/feats/9r_1.feats
Attempt to access channel 1 of 0 channel track
Dataset of 0 vectors of 64 parameters from: festival/feats/9r_1.feats
Segmentation fault (core dumped)
Dataset of 0 vectors of 64 parameters from: festival/feats/9r=_1.feats
Attempt to access channel 1 of 0 channel track
Dataset of 0 vectors of 64 parameters from: festival/feats/9r=_1.feats
Segmentation fault (core dumped)
Dataset of 0 vectors of 64 parameters from: festival/feats/9r_1.feats
Attempt to access channel 1 of 0 channel track
Dataset of 0 vectors of 64 parameters from: festival/feats/9r_1.feats
Segmentation fault (core dumped)
Segmentation fault (core dumped)
Dataset of 0 vectors of 64 parameters from: festival/feats/9r=_1.feats
Attempt to access channel 1 of 0 channel track
Dataset of 0 vectors of 64 parameters from: festival/feats/9r=_1.feats
Dataset of 0 vectors of 64 parameters from: festival/feats/9r_1.feats
Attempt to access channel 1 of 0 channel track
Dataset of 0 vectors of 64 parameters from: festival/feats/9r_1.feats
Segmentation fault (core dumped)
Dataset of 0 vectors of 64 parameters from: festival/feats/9r=_1.feats
Attempt to access channel 1 of 0 channel track
Dataset of 0 vectors of 64 parameters from: festival/feats/9r=_1.feats
Segmentation fault (core dumped)
Dataset of 0 vectors of 64 parameters from: festival/feats/9r_1.feats
Attempt to access channel 1 of 0 channel track
Dataset of 0 vectors of 64 parameters from: festival/feats/9r_1.feats
Segmentation fault (core dumped)
Dataset of 0 vectors of 64 parameters from: festival/feats/9r=_1.feats
Attempt to access channel 1 of 0 channel track
Dataset of 0 vectors of 64 parameters from: festival/feats/9r=_1.feats
Segmentation fault (core dumped)
Segmentation fault (core dumped)
Dataset of 0 vectors of 64 parameters from: festival/feats/9r_1.feats
Attempt to access channel 1 of 0 channel track
Dataset of 0 vectors of 64 parameters from: festival/feats/9r_1.feats
Segmentation fault (core dumped)
Segmentation fault (core dumped)
Segmentation fault (core dumped)
Segmentation fault (core dumped)
Segmentation fault (core dumped)
Segmentation fault (core dumped)
Segmentation fault (core dumped)
Collect trees
SIOD ERROR: wrong type of argument to setcar
BACKTRACE:
0: (set-car! (car tree) vector_num)
1: (clustergen::dump_tree_vectors tree rawtrackfd)
2: (set! tree (clustergen::dump_tree_vectors tree rawtrackfd))
3: (f (car l2))
4: (cons (f (car l2)) r)
5: (set! r (cons (f (car l2)) r))
6: (while l2 (set! r (cons (f (car l2)) r)) (set! l2 (cdr l2)))
7: (mapcar
(lambda
(unit)
(...)
...)
unittypes)
8: (if
(consp cg:multimodel)
(mapcar
(...)
cg:multimodel)
...)
9: (begin
(set! cg:parallel_tree_build t)
(build_clustergen "etc/txt.done.data.train"))
closing a file left open: festival/trees/cmu_indic_ss_mcep.rawparams
closing a file left open: festival/trees/cmu_indic_ss_mcep.tree

Errors when using 44100Hz .wav files

I completed festival part successfully with 16kHz .wav files. But when I'm trying to use 44100Hz .wav files I'm getting errors. CAtually building prompts was okay. But the problem started in the labeling part. Only .sl files are created. .lab and .slehmm files are not created and it shows errors and labeling part stopped at the middle.
I need to know do we need to change or edit some files (codes) when we are using a sample rate other than 16kHz.
Please someone help me to overcome this issue.

Thanks in advance.

ch_wave and f=16

do_pm call ch_wave with the parameter F=16
but ch_wave -h says
-F Output sample rate in Hz. If this is different from the input sample rate, resampling will occur

and errors do occur (rateconv)

Error in building prompts

I was building indic voice using the steps given in the website and when I typed in
./bin/do_build build_prompts etc/txt.done.data It gave me error as below

SIOD ERROR: could not open file ./festvox/language_variant.scm
I got to know that it is because of the language selector in the festvox/indic_lexicon.scm

Am I wrong somewhere? or Is it still under development? Please let me know

I am currently running Festvox 2.8 with Festival 2.5

SIOD ERROR: null-arg-to-last in Brazilian Portuguese voice construction used Grapheme-based Synthesizer.

Hello, I'm building a voice in Brazilian Portuguese used "Grapheme-based in Grapheme-based Synthesizer, but in generating some test examples occurred:

./bin/do_clustergen cg_test tts tts etc/txt.done.data.test Resulting in SIOD ERROR: null-arg-to-last.

The database has 600 audios on average from 3 to 8 seconds long. All training commands and generation of label the data and prompts. and feature extraction are successful.

do_cga back to CLUSTERGENDIR

in /src/clustergen/fr/do_cga, you are supposed to

Build voice conversion model using CART prediction (and initially)

using a clustergen voice as a source

but you don't build with it your own files :
you use for example "$CLUSTERGENDIR/clustergen_build.scm"
that could not be the same as yours, particularly if work with a new language

Error in building prompts

Hi, I'm getting the following error when trying to build prompts.

free(): invalid next size (fast)
Aborted (core dumped)

Can someone tell me the reason and solution for this please.

Thank you

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