Activity for rezaee
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rezaee posted a comment on discussion Help
Great @Q3Varnam , finally I solved the problem! It was because of my missing to run this command sphinxtrain -t robot setup Before sphinxtrain run Thank you again!
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I have these sentences in my transcription file for example: <s> become blue <\s> <s> become red <\s> But I wrote them like this in my grammar .jsgf file: <colors> = (red | blue); <action> = become <colors> public <commands> = <action> Is this okay or the problem is here?
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I have these sentences in my transcription file for example: <s> become blue <\s> <s> become red <\s> But I wrote them like this in my grammar .jsgf file: <colors> = (red | blue); <action> = become <colors> public <commands> = <action> Is this okay or the problem is here?
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I checked this and seems all are same and well
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It seems all the thing are well but it gives me 100% ERROR :)
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This is robot.align file but the result is strange! SALAM (test-1) (test-1) Words: 1 Correct: 0 Errors: 1 Percent correct = 0.00% Error = 100.00% Accuracy = 0.00% Insertions: 0 Deletions: 1 Substitutions: 0 HALET CHETORE (test-2) (test-2) Words: 2 Correct: 0 Errors: 2 Percent correct = 0.00% Error = 100.00% Accuracy = 0.00% Insertions: 0 Deletions: 2 Substitutions: 0 ESMET CHIE (test-3) (test-3) Words: 2 Correct: 0 Errors: 2 Percent correct = 0.00% Error = 100.00% Accuracy = 0.00% Insertions: 0 Deletions:...
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Okay, thank you so much! Did you see my attachment? was that true?
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I attached my .jsgf file. it gives me : SENTENCE ERROR: 100.0% (10/10) WORD ERROR RATE: 100.0% (22/22) After training and ERROR: "fsg_search.c", line 940: Final result does not match the grammar in frame 97 In run-time!
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I attached my .jsgf file. it gives me : SENTENCE ERROR: 100.0% (10/10) WORD ERROR RATE: 100.0% (22/22) After training and ERROR: "fsg_search.c", line 940: Final result does not match the grammar in frame 97 In run-time!
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I attached my .jsgf file. it gives me : SENTENCE ERROR: 100.0% (10/10) WORD ERROR RATE: 100.0% (22/22) After training and ERROR: "fsg_search.c", line 940: Final result does not match the grammar in frame 97 In run-time!
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I tried to add public in front of all the rules but I get terrible results with this error messages: ERROR: "fsg_search.c", line 940: Final result does not match the grammar in frame xxx
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I tried the above .jsfg file but when I wanted to test it got this error message: ERROR: "pocketsphinx.c", line 680: No public rules found in /home/m/myrobot3/robot/etc/robot.jsgf But I don't know which rule should be publiced? I have only one .jsfg file like above with 5 times more sentences!
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Is this a right grammar file? #JSGF V1.0; grammar introduction; <hello> = hello | hi; //isn't problem same name of hello? <feeling> = how are you; <name> = what is your name; <old> = how old are you; <origin> = where are you from; <live> = where is your home; <food> = what is your favorite food; <love> = do you love me | do you like (red | blue | green); <education> = are you educated | what grade are you in; These are some of my recorded voices for my command&control system.
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Is this a right grammar file? #JSGF V1.0; grammar introduction; <hello> = hello | hi; <feeling> = how are you; <name> = what is your name; <old> = how old are you; <origin> = where are you from; <live> = where is your home; <food> = what is your favorite food; <love> = do you love me; <education> = are you educated | what grade are you in; These are some of my recorded voices for my command&control system.
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Is this a right grammar file? #JSGF V1.0; grammar introduction; <hello> = hello | hi; <feeling> = how are you; <name> = what is your name; <old> = how old are you; <origin> = where are you from; <live> = where is your home; <food> = what is your favorite food; <love> = do you love me; <education> = "are you educated" | "what grade are you in"; These are some of my recorded voices for my command&control system.
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Is this a right grammar file? #JSGF V1.0; grammar introduction; <hello> = hello | hi <feeling> = how are you <name> = what is your name <old> = how old are you <origin> = where are you from <live> = where is your home <food> = what is your favorite food <love> = do you love me <education> = "are you educated" | "what grade are you in" These are some of my recorded voices for my command&control system.
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Is this a right grammar file? #JSGF V1.0; grammar introduction; <hello> = hello | hi <feeling> = how are you <name> = what is your name <old> = how old are you <origin> = where are you from <live> = where is your home <food> = what is your favorite food <love> = do you love me <education> = "are you educated" | :what grade are you in" These are some of my recorded voices for my command&control system.
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OK! but I don't know how to use import<> command. In the .jsgf documentation it uses java like addressing <com.example.x> but I don't know how should I do that in Linux with C++? I think: import </home/m/robot/grammer/x> ?
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Sorry for that! I edited my question and tried to write the code more simple. May you test this code and try to solve the problem with ad_stop_rec(ad); or any other solutions you may know? I tried a lot but couldn't al all!
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I am trying to pass pocketsphinx's recognized hyp to espeak TTS engine by this simple code: #include <stdio.h> #include <string.h> #include <assert.h> #include <sphinxbase/err.h> #include <sphinxbase/ad.h> #include "pocketsphinx.h" #include <espeak/speak_lib.h> #include <string> #include <iostream> static ps_decoder_t *ps; static cmd_ln_t *config; static FILE *rawfd; ad_rec_t *ad; static void recognize_from_microphone() { int16 adbuf[2048]; uint8 utt_started, in_speech; int32 k; char *hyp; if ((ad...
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May you upload a grammar folder if you have! I am confused about the sructure. I don't know should I write all my possible sentences that I have in a single .jsgf file? Or I must create many files for each sentence and import all in one? How should I address them in < > to import?
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Thanks again, but how to change this command in my C++ code: config = cmd_ln_init(NULL, ps_args(), TRUE, "-hmm", "/home/m/robot/model_parameters /robot.cd_cont_2000", "-lm","/home/m/robot/etc/robot.lm", "-dict", "/home/m/robot/etc/robot.dic", NULL);
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Thank you! But may you explain more about how can I build and use a grammer with pocketsphinx? Can I use with C++ code? How?
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Thank you! But may you explain more about how can I build and use a grammer with pocketsphinx?
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I am trying to create a single speaker command control acoustic model.. I have about 100 little sentences like what is your name , how are you, that I repeated each sentence 30 times, and put them in a structure like this: ==sentence_one 1.wav 2.wav . 30.wav ==sentence_two 1.wav 2.wav . 30.wav I think I have about 30-45mins of .wav voices totally. I used SRILM to creat language model, then trained acoustic model using sphinx train, but my accuracy is low! I tried to change senonses from 200 to 2000...
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Hi I have build and tested language and acoustic models on my laptop successfully(amd64 Ubuntu 16.04 Desktop). And tried to just copy them and use them on my mini-pc(arm7-sunxi Ubuntu server 16.04). But it seems it doesn't work! Should I create them again on my ARM based pc?
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Hi I have build and tested language and acoustic models on my laptop successfully(amd64 Ubuntu 16.04 Desktop). And tried to just copy them and use them on my mini-pc(arm7-sunxi Ubuntu server 16.04). But it seems it doesn't work! Should I create them for my ARM based pc again?
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Let's consider I have a for(;;) loop, it gets a new string by every iteration, and passes it to system("espeak") like this: for(;;) { string str = getNewString(); string str2 = "espeak '" +str+ "'"; system(str2.c_str()); } But when the first str passes to espeak it starts to say it, but before it ends the speaking, for loop runs again and passes a new string and it crashes. I am looking a way, my program pauses the for loop untill the espeak done! Is it possible? How?
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I tried a lof ot things but couldn't reach success till now!
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What is the reason if i get the error message in this part of the code(in continuous.c default code): if ((k = ad_read(ad, adbuf, 2048)) < 0) E_FATAL("Failed to read audio\n");
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I am trying to recognize voices from microphone by pocketsphinx and pass it through espeak to say what it recognized in real-time/continuous speech(I mean user says a word/sentence, pocketsphinx will recognize it and collects it inside char* hyp, then passes the hyp to speech() function that uses espeak to say what is inside the hyp. But I don't know why do I get Failed to read audio In this part of the code: if ((k = ad_read(ad, adbuf, 2048)) < 0) E_FATAL("Failed to read audio\n"); This is the complete...
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I edited the code like below but the problem didn't solve! #include <stdio.h> #include <string.h> #include <assert.h> #include <sphinxbase/err.h> #include <sphinxbase/ad.h> #include "pocketsphinx.h" #include <malloc.h> #include <espeak/speak_lib.h> #include <string> using namespace std; static ps_decoder_t *ps; static cmd_ln_t *config; static FILE *rawfd; espeak_POSITION_TYPE position_type; espeak_AUDIO_OUTPUT output; char *path=NULL; int Buflength = 1000, Options=0; void* user_data; char Voice[]...
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I edited the code like below but the problem didn't solve! #include <stdio.h> #include <string.h> #include <assert.h> #include <sphinxbase/err.h> #include <sphinxbase/ad.h> #include "pocketsphinx.h" #include <malloc.h> #include <espeak/speak_lib.h> #include <string> using namespace std; static ps_decoder_t *ps; static cmd_ln_t *config; static FILE *rawfd; espeak_POSITION_TYPE position_type; espeak_AUDIO_OUTPUT output; char *path=NULL; int Buflength = 1000, Options=0; void* user_data; char Voice[]...
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I tried to change my command to : ngram-count -text train-text.txt -lm your.lm And got this result: warning: discount coeff 1 is out of range: 0 warning: discount coeff 4 is out of range: 2.60909 warning: discount coeff 5 is out of range: 1.21455 warning: discount coeff 6 is out of range: 1.17879 warning: discount coeff 7 is out of range: 1.15325 warning: count of count 8 is zero -- lowering maxcount warning: discount coeff 3 is out of range: -0.0630631 warning: count of count 4 is zero warning:...
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I have downloaded SRILM and did make world in it's main directory successfully(without errors),Also i tried make test without errors. I fount there is a file named ngram-count in this path../srilm-1.7.2/bin/i686-m64 then I copied it to inside my etc file inside my database, and tried to run this command as cmusphinx tutorials say here https://cmusphinx.github.io/wiki/tutoriallm (I copied all my words and sentences inside train-text.txt file) : ./ngram-count -kndiscount -interpolate -text train-text.txt...
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I have downloaded SRILM and did make world in it's main directory successfully(without errors),Also i tried make test without errors. but when I go to my database(like an4) inside etc folder and run ngram-count -kndiscount -interpolate -text train-text.txt -lm your.lm it gives me this result: ngram-count: command not found . How can I use SRILM?
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For example, when I tried to write the Persian words with English letters, and tried to make a language model, the dictionary is look like this: A AH A(2) EY AB AE B ABI AE B IY AGHAB AE AH B AHLE AE L AIR EH R AM AE M AM(2) EY EH M ANGUR AE N G AH R ANJIR AE N JH AH R APPLE AE P AH L ARE AA R ARE(2) ER AROUND ER AW N D AROUND(2) ER AW N BABAT B AE B AH T BACK B AE K BALA B AA L AH BEBAR B IH B AA R BECHARKH B AH CH AA R K BECOME B IH K AH M BEGU B IH G Y UW BENEVIS B IY AH N IY V IY Z BIRD B ER...
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Thank you for the paper Nikolay! But it seems as I said above, I must use the standard phoneset that pocketsphinx use for English language, but don't know where can I find a full list of them? I mean a complete phoneset file of what pocketsphinx use
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I found that the problem is because of my phoneset file. I used my own phoneset for both English and Persian languages as you can see in my question(first post) but it seems language model only accepts Persian words and don't wotk for English words with this phoneset! It seems I must use the standard English phoneset(I mean what the pocketsphinx use for English) for my Persian words as well to have a bi-langual model....But I don't know where can I find a complete set of them and an instruction to...
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This is the etc file(I have problem with editing my posts)
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Also the command I use to test is: pocketsphinx_continuous -inmic yes -hmm /home/m/myrobot2/robot/model_parameters/robot.cd_cont_200 -lm /home/m/myrobot2/robot/etc/robot.lm -dict /home/m/myrobot2/robot/etc/robot.di And it works. But only recognizes Persian sentences and words.
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Hi dear Nikolay! I attached my etc folder to my post. I tried to create language model with the online tools the tutorial suggested, but it only works for Persian languages and it can not recognize English words at the same time.
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I am trying : sudo svn co https://cmusphinx.svn.sourceforge.net/svnroot/cmusphinx/trunk/cmuclmtk But I get: svn: E170013: Unable to connect to a repository at URL 'https://cmusphinx.svn.sourceforge.net/svnroot/cmusphinx/trunk/cmuclmtk' svn: E160013: '/svnroot/cmusphinx/trunk/cmuclmtk' path not found How can I fix it? I wanna use cmuclmtk to create a language model.
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I did follow this instructions(https://cmusphinx.github.io/wiki/tutorialadapt/) till creating a transformation with MLLR, but when I try : ./mllr_solve \ -meanfn en-us/means \ -varfn en-us/variances \ -outmllrfn mllr_matrix -accumdir . I get bash: ./mllr_solve: No such file or directory .
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I am trying to write a code that use spocketsphinx to recognize what the user says, then pass it to espeak to say what the pocketsphinx recognized! I wrote this code: #include <stdio.h> #include <string.h> #include <assert.h> #include <sphinxbase/err.h> #include <sphinxbase/ad.h> #include "pocketsphinx.h" #include <malloc.h> #include <espeak/speak_lib.h> #include <string> using namespace std; static ps_decoder_t *ps; static cmd_ln_t *config; static FILE *rawfd; espeak_POSITION_TYPE position_type;...
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I got my model from here: https://sourceforge.net/projects/cmusphinx/files/Acoustic%20and%20Language%20Models/US%20English/cmusphinx-en-us-ptm-5.2.tar.gz/download And tried to configure the above command like this: ./bw -hmmdir en-us -moddeffn en-us/mdef.txt -ts2cbfn .cont. -feat 1s_c_d_dd -cmn current -agc none -dictfn cmudict-en-us.dict -ctlfn robot_train.fileids -lsnfn robot_train.transcription -accumdir -lda feature_transform . But I get these error messages: INFO: main.c(229): Compiled on Mar...
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I got my model from here: https://sourceforge.net/projects/cmusphinx/files/Acoustic%20and%20Language%20Models/US%20English/cmusphinx-en-us-ptm-5.2.tar.gz/download And tried to configure the above command like this: ./bw -hmmdir en-us -moddeffn en-us/mdef.txt -ts2cbfn .cont. -feat 1s_c_d_dd -cmn current -agc none -dictfn cmudict-en-us.dict -ctlfn robot_train.fileids -lsnfn robot_train.transcription -accumdir -lda feature_transform . But I get these error messages: INFO: main.c(229): Compiled on Mar...
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I got my model from here: https://sourceforge.net/projects/cmusphinx/files/Acoustic%20and%20Language%20Models/US%20English/cmusphinx-en-us-ptm-5.2.tar.gz/download And tried to configure the above command like this: ./bw -hmmdir en-us -moddeffn en-us/mdef.txt -ts2cbfn .cont. -feat 1s_c_d_dd -cmn current -agc none -dictfn cmudict-en-us.dict -ctlfn robot_train.fileids -lsnfn robot_train.transcription -accumdir -lda feature_transform . But I get these error messages: INFO: main.c(229): Compiled on Mar...
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I have tried to change and execute this command: ./bw -hmmdir en-us -moddeffn en-us/mdef.txt -ts2cbfn .cont -feat 1s_c_d_dd -cmn current -agc none -dictfn cmudict-en-us.dict -ctlfn robot_train.fileids -lsnfn robot_train.transcription -accumdir . But got this error message:~~~ INFO: feat.c(715): Initializing feature stream to type: '1s_c_d_dd', ceplen=13, CMN='batch', VARNORM='no', AGC='none' INFO: main.c(318): Reading en-us/mdef.txt INFO: model_def_io.c(573): Model definition info: INFO: model_def_io.c(574):...
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I have tried to change and execute this command: ./bw -hmmdir en-us -moddeffn en-us/mdef.txt -ts2cbfn .conf. -feat 1s_c_d_dd -ts2cbfn .cont -cmn current -agc none -dictfn cmudict-en-us.dict -ctlfn robot_train.fileids -lsnfn robot_train.transcription -accumdir . But got this error message: INFO: main.c(229): Compiled on Mar 22 2018 at 12:54:02 ERROR: "cmd_ln.c", line 641: Duplicate argument name in arguments: -ts2cbfn ERROR: "cmd_ln.c", line 704: Failed to parse arguments list ERROR: "cmd_ln.c", line...
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In tutorial it says: Make sure the arguments in the bw command match the parameters in the feat.params file inside the acoustic model folder. And: For example, for a continuous model you don’t need to include the svspec option. Instead, you need to use just -ts2cbfn .cont. But as you can see above, I have no-ts2cbfn .continside myfeat.params` file! Also, the tutorial says: If you are missing the noisedict file, you also need an extra step. Copy the fillerdict file into the directory that you choose...
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In tutorial it says: Make sure the arguments in the bw command match the parameters in the feat.params file inside the acoustic model folder. And: For example, for a continuous model you don’t need to include the svspec option. Instead, you need to use just -ts2cbfn .cont. But as you can see above, I have no-ts2cbfn .continside myfeat.params` file! Also, the tutorial says: If you are missing the noisedict file, you also need an extra step. Copy the fillerdict file into the directory that you choose...
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In tutorials it has written this command: ~~~ bw \ -hmmdir en-us \ -moddeffn en-us/mdef.txt \ -ts2cbfn .ptm. \ -feat 1s_c_d_dd \ -svspec 0-12/13-25/26-38 \ -cmn current \ -agc none \ -dictfn cmudict-en-us.dict \ -ctlfn arctic20.fileids \ -lsnfn arctic20.transcription \ -accumdir . But I checked my feat.params and it has this content: -lowerf 130 -upperf 6800 -nfilt 25 -transform dct -lifter 22 -feat 1s_c_d_dd -svspec 0-12/13-25/26-38 -agc none -cmn current -varnorm no -model ptm -cmninit 40,3,-1...
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In tutorials it has written this command: ~~~ ./~~~ bw \ -hmmdir en-us \ -moddeffn en-us/mdef.txt \ -ts2cbfn .ptm. \ -feat 1s_c_d_dd \ -svspec 0-12/13-25/26-38 \ -cmn current \ -agc none \ -dictfn cmudict-en-us.dict \ -ctlfn arctic20.fileids \ -lsnfn arctic20.transcription \ -accumdir . But I checked my feat.params and it has this content: -lowerf 130 -upperf 6800 -nfilt 25 -transform dct -lifter 22 -feat 1s_c_d_dd -svspec 0-12/13-25/26-38 -agc none -cmn current -varnorm no -model ptm -cmninit 40,3,-1...
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In tutorials it has written this command: ~~~ ./~~~ bw \ -hmmdir en-us \ -moddeffn en-us/mdef.txt \ -ts2cbfn .ptm. \ -feat 1s_c_d_dd \ -svspec 0-12/13-25/26-38 \ -cmn current \ -agc none \ -dictfn cmudict-en-us.dict \ -ctlfn arctic20.fileids \ -lsnfn arctic20.transcription \ -accumdir . But I checked my feat.params and it has this content: -lowerf 130 -upperf 6800 -nfilt 25 -transform dct -lifter 22 -feat 1s_c_d_dd -svspec 0-12/13-25/26-38 -agc none -cmn current -varnorm no -model ptm -cmninit 40,3,-1...
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In tutorials it has written this command: ~~~ ./bw \ -hmmdir en-us \ -moddeffn en-us/mdef.txt \ -ts2cbfn .ptm. \ -feat 1s_c_d_dd \ -svspec 0-12/13-25/26-38 \ -cmn current \ -agc none \ -dictfn cmudict-en-us.dict \ -ctlfn arctic20.fileids \ -lsnfn arctic20.transcription \ -accumdir . But I checked my feat.params and it has this content: -lowerf 130 -upperf 6800 -nfilt 25 -transform dct -lifter 22 -feat 1s_c_d_dd -svspec 0-12/13-25/26-38 -agc none -cmn current -varnorm no -model ptm -cmninit 40,3,-1...
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I am trying to do adaption by following the tutorials: https://cmusphinx.github.io/wiki/tutorialadapt/ But when I want to download this : cmusphinx-en-us-ptm-5.2.tar.gz , it gives me this error message: The "/Acoustic and Langua..en-us-ptm-5.2.tar.gz" file could not be found or is not available. Please select another file.
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Hi I am trying to use espeak within my Android app. I use Android Strudio 2.3 and Win10 OS. I am looking for a simple sample code or a step-by-step guide to learn how can I use this greate library inside my Android app? Help me please!
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no idea? Is it mean no one have such problem? all of you get better result than google or microsoft voice recognizers?
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when I compare pocketsphinx results to google or microsoft cortana or siri results, pocketsphinx can understand 50% of my words correct, but others recognize them about 100% !! What is the problem? Is this mean CMU toolkit is weaker than obove online tools?
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I am trying to connect pocketsphinx to espeak, so I send the hyp result to espeak, but when espeak says the first hyp, pocketsphinx listening to espeak sound and repeat it for ever! So I am loiking for a way to disable listening until espeak silents!
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This is my code. I want to get 5 strings from the user and espeak reads each of them when user interred it. But I get segmentation fault(core dumped) message. #include <string.h> #include <malloc.h> #include <espeak/speak_lib.h> int test() { espeak_POSITION_TYPE position_type; espeak_AUDIO_OUTPUT output; char *path=NULL; int Buflength = 500, Options=0; void* user_data; t_espeak_callback *SynthCallback; espeak_PARAMETER Parm; char Voice[] = {"English"}; int i=0; char text[1000]; unsigned int Size,position=0,...
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This is my code. I want to get 5 strings from the user and espeak reads each of them when user interred it. But I get segmentation fault(core dumped) message. #include <string.h> #include <malloc.h> #include <espeak/speak_lib.h> int test() { espeak_POSITION_TYPE position_type; espeak_AUDIO_OUTPUT output; char *path=NULL; int Buflength = 500, Options=0; void* user_data; t_espeak_callback *SynthCallback; espeak_PARAMETER Parm; char Voice[] = {"English"}; int i=0; char text[1000]; unsigned int Size,position=0,...
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I found this code on the internet: #include <string.h> #include <malloc.h> #include <espeak/speak_lib.h> espeak_POSITION_TYPE position_type; espeak_AUDIO_OUTPUT output; char *path=NULL; int Buflength = 1000, Options=0; void* user_data; t_espeak_callback *SynthCallback; espeak_PARAMETER Parm; char Voice[] = {"English"}; char text[30] = {"this is a english test"}; unsigned int Size,position=0, end_position=0, flags=espeakCHARS_AUTO, *unique_identifier; int main(int argc, char* argv[] ) { output = AUDIO_OUTPUT_PLAYBACK;...
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Hi there I tried the default continous.c program, but it seems it's so far from our accent?! or maybe it has very bad language model! When I test this words I get this results: me:hi pocketsphinx: i m: how old are you p: hold on you m:where do you live p: good you leave/ where the you/ where do you/ when you leave So, if I want to get better results, which way is better? creating a ristricted dictionary and buld a new language model for that(although I don' know if I have not the acoustig model,...
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Hi, I have installed pocketsphinx on an ARM-based board that has not very much proccessing power. When I try to test the pocketsphinx_continuous program, this happens: Board: ready ...listening Me: Hi, how are you? Board: hi ... on who Me: how are you? Borad: on ...(some processing) Me: what's your name? Board: who are you Me:what's your name? Board: (does some proceesing) Me: how old are you? Board: wha'ts your name Me:how old are you? Board: (some processing) Me: what's your name Me:silent Borad:...
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It's interesting for me to know how many people for how many years worked on this toolkit?
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I know there is a dcument here: http://www.speech.cs.cmu.edu/sphinx/doc/ and another one here: https://cmusphinx.github.io/doc and a little guid on sourceforge site, but if someone wants to know how this toolkit works step by step, he should start from where? how to learn and follow the files,structures,variables,functions,etc? I mean a road map about how to read and learn it?
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OK! I fixed my code by changing config line to this: config = cmd_ln_init(NULL, ps_args(), TRUE, "-hmm", MODELDIR "/en-us/en-us", "-lm", MODELDIR "/en-us/en-us.lm.bin", "-dict", MODELDIR "/en-us/cmudict-en-us.dict", NULL); ps = ps_init(config);
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Why does it say there is no config?!
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I added this line in main() function: ps = ps_init(config); But it says:ERROR: "pocketsphinx.c", line 393: No configuration specifiedSegmentation fault (core dumped)
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I know you say I must change this line: static ps_decoder_t *ps; But changing to what?
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What's the problem Nikolay?
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But the above code gives me the same result as I put in previous post (segmentation fault). I didn't remove any part of the code (except the recognition from file part). and I didn't modify any part except "main()" function and "cont_args_def" variable. So, where should I intialize the decoder? how?
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This is my full code: #include <stdio.h> #include <string.h> #include <string> #include <assert.h> #if defined(_WIN32) && !defined(__CYGWIN__) #include <windows.h> #else #include <sys/select.h> #endif #include <sphinxbase/err.h> #include <sphinxbase/ad.h> #include "pocketsphinx.h" using namespace std; static const arg_t cont_args_def[] = { POCKETSPHINX_OPTIONS, {"-inmic", ARG_BOOLEAN, "no", "Transcribe audio from microphone."}, CMDLN_EMPTY_OPTION }; static ps_decoder_t *ps; static cmd_ln_t *config;...
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I changed the code like this: static const arg_t cont_args_def[] = { POCKETSPHINX_OPTIONS, {"-inmic", ARG_BOOLEAN, "no", "Transcribe audio from microphone."}, CMDLN_EMPTY_OPTION }; int main(int argc, char *argv[]) { config = cmd_ln_parse_r(NULL, cont_args_def, argc, argv, TRUE); // if (cmd_ln_boolean_r(config, "-inmic")) { recognize_from_microphone(); // } // recognize_from_microphone(); ps_free(ps); cmd_ln_free_r(config); return 0; } But the result is: ~~~ Arguments list definition: [NAME] [DEFLT]...
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Where should I change the continuous.c's dictionary and acousticmodel?
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Help please! I want to disable needing to get -inmic yes and program starts to work with microphon automatically after starting it!
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I changed this line to : config = cmd_ln_parse_r(NULL, &cont_args_def, argc, argv, TRUE); But it gives this error when I want ru the executable: Segmentation fault (core dumped)
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May you explain more? How should I change the code?
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Also this doesn't work: static const arg_t cont_args_def= {"-inmic", ARG_BOOLEAN, "no", "Transcribe audio from microphone."}; int main(int argc, char *argv[]) { config = cmd_ln_parse_r(NULL, cont_args_def, argc, argv, TRUE); recognize_from_microphone(); ps_free(ps); cmd_ln_free_r(config); return 0; }
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I changed it like this but gives me error: int main(int argc, char *argv[]) { config = cmd_ln_parse_r(NULL, {"-inmic",ARG_BOOLEAN, "no","Transcribe audio from microphone."}, argc, argv, TRUE); recognize_from_microphone(); ps_free(ps); cmd_ln_free_r(config); return 0; } m2.cpp:174:117: error: cannot convert ‘<brace-enclosed initializer="" list="">’ to ‘const arg_t {aka const arg_s}’ for argument ‘2’ to ‘cmd_ln_t cmd_ln_parse_r(cmd_ln_t, const arg_t, int32, char*, int32)’ mic",ARG_BOOLEAN, "no","T...
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I changed it like this but gives me error: int main(int argc, char *argv[]) { config = cmd_ln_parse_r(NULL, {"-inmic",ARG_BOOLEAN, "no","Transcribe audio from microphone."}, argc, argv, TRUE); recognize_from_microphone(); ps_free(ps); cmd_ln_free_r(config); return 0; } m2.cpp:174:117: error: cannot convert ‘<brace-enclosed initializer="" list="">’ to ‘const arg_t {aka const arg_s}’ for argument ‘2’ to ‘cmd_ln_t cmd_ln_parse_r(cmd_ln_t, const arg_t, int32, char*, int32)’ mic",ARG_BOOLEAN, "no","T...
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In continuous.c file, how can I increase waiting time after it recognized a word? I ran pocketsphinx in an ARM-based board, so I have not much proccessing power, then after saying some words system steel trying to recognize previous words. I mean you say the sentence number 5 and system tries to recognize what you said in sentence 2! How can I fix this? is it possible?
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If I want to change the continuous.c code, as it only uses microphon automatically after compile, how should I cnahge these lines? char const *cfg; config = cmd_ln_parse_r(NULL, cont_args_def, argc, argv, TRUE); /* Handle argument file as -argfile. */ if (config && (cfg = cmd_ln_str_r(config, "-argfile")) != NULL) { config = cmd_ln_parse_file_r(config, cont_args_def, cfg, FALSE); } if (config == NULL || (cmd_ln_str_r(config, "-infile") == NULL && cmd_ln_boolean_r(config, "-inmic") == FALSE)) { E_INFO("Specify...
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If I want to change the continuous.c code, as it only uses microphon automatically after compile, how should I cnahge these lines? char const *cfg; config = cmd_ln_parse_r(NULL, cont_args_def, argc, argv, TRUE); / Handle argument file as -argfile. / if (config && (cfg = cmd_ln_str_r(config, "-argfile")) != NULL) { config = cmd_ln_parse_file_r(config, cont_args_def, cfg, FALSE); } if (config == NULL || (cmd_ln_str_r(config, "-infile") == NULL && cmd_ln_boolean_r(config, "-inmic") == FALSE)) { E_INFO("Specify...
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In the tutorial it uses "cmd_ln_init" to feed the config variab, but in continuous.c it uses "cmd_ln_parse_r" function to fill it. Why?
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In the tutorial it uses "cmd_ln_init" to fill the config variab;e, but in continuous.c it uses "cmd_ln_parse_r" function to fill it. Why?
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Thanks a lot Bilal, I have another question: I think pocketsphinx_continuous is the executable made by compiling continuous.c. Am I true? If yes, where is continuous.c ?
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I did it, but couldn't find my answer. I can find that file inside the pocketsphinx/src/programs , but I think it's the raw folder! I mean in Windows OS, when you install a program within the raw folder(I mean the folder contains the setup file), it will installs the program in other location and you can remove the raw folder. But in Linux it seems the story is difference and there is no installation folder outside the raw folder because I couldn't find any file with the name continuous.c outside...
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Is this a very bad question?
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I am new in Linux and don't know much about instalation. It's interesting for me to know where does it install the pockesphinx program? where is continuous.c? How Linux can find and execute this command: pocketsphinx_continuous -inmic yes ? I know it seems some of these questions are related to Linux but I like to know the pocketsphinx related questions answer.
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There is no help?
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I installed pocketsphinx without errors step-by-step (./autogen.sh , ./configure, make, make install for both sphinx-base and pocketsphinx), but when I did try this command to test: pocketsphinx_continuous -inmic yes I got this error message: calloc(-1090616432,1) failed from ngram_model_trie.c(313) This is the full result: INFO: pocketsphinx.c(152): Parsed model-specific feature parameters from /usr/local/share/pocketsphinx/model/en-us/en-us/feat.params Current configuration: [NAME] [DEFLT] [VALUE]...
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Hi I am looking for a function that I can include in my program and call it by passing my desire string to it and it converts my string to voice by espeak engine! How can I do that?
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If I want to use espeak in my program, Which header files should be included and by which function can I call espeak and pass a string to it?
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When I trying to install pocketsphinx by make install command(after ./autogen.sh , ./configure, make commands) , I get this error message: libtool: error: relink 'libpocketsphinx.la' with the above command before installing it
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Thank you Nickolay! But, if we want to use movie's sound for acoustic model, there is a lot of silence or environments sounds between their speech, should we conside this space between their speech? For example: Hello (2s sil) Hi (1s sil) (walking sound for 5s) What did you do? Nothing(10s silence and road sound) OK, let's talk about that night ... I hope I could convey my mean clear!
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