Ahora que ya tenemos la voz, necesitamos un SST (Speech to
Text), o Text Recognition.De todos los que he probado, me decanto
por el de Google. El problema es que para el uso de ste, deberemos
guardar en un wav lo que digamos, convertirlo a flac, y enviarlo
por la api usando un navegador (menudo pitote).
Despus de unas recopilaciones de aqu y de all lo traigo ya
resumido:
Grabamos con el comando "rec" el wav (Aseguraos de que el
volumen de la entrada de micrfono del alsamixer est al 100%, si no
le costar la vida reconocer que estamos diciendo)rec -r 16000 -e
signed-integer -b 16 -c 1 audio.wav trim 0 4Convertimos de wav a
flac:sox audio.wav -r 16000 -b 16 -c 1 audio.flac vad reverse vad
reverse lowpass -2 2500Y ahora viene la magia... sin usar el
navegador, pero usando una cabecera (simulndolo):curl --data-binary
@audio.flac --header 'Content-type: audio/x-flac; rate=16000'
'http://www.google.com/speech-api/v1/recognize?xjerr=1&client=chromium&lang=es-ES&maxresults=1'
1>audio.txtAlmacenamos en un audio.txt lo que nos devuelve la
api de Google. Podremos cambiar el idioma en lang=es-ES (por
defecto puse Espaa-Espaol) o si queremos que nos muestre ms
resultados en vez de 1 (el ms acertado).
Para resumir lo que nos devuelve en una simple frase formateamos
el resultado:FILETOOBIG=`cat audio.txt | grep ""`TRANSCRIPT=`cat
audio.txt | cut -d"," -f3 | sed
's/^.*utterance\":\"\(.*\)\"$/\1/g'`CONFIDENCE=`cat audio.txt | cut
-d"," -f4 | sed 's/^.*confidence\":0.\([0-9][0-9]\).*$/\1/g'`Donde
el Confidence (por si queremos usarlo) ser el porcentaje de
probabilidad de acierto en el reconocimiento, y el Transcript lo
que entendi.
Ya con esto, procedemos con un simple if la prueba de que
funciona:if echo "$TRANSCRIPT" |grep -q "Hola"; then aoss espeak
-ves "$TRANSCRIPT"elif echo "$TRANSCRIPT" |grep -q "quin eres";
then aoss espeak -ves "Soy Yarvis, la mquina de Yuki Sekisan"elif
echo "$TRANSCRIPT" |grep -q "saluda a Alberto"; then aoss espeak
-ves "Hola Alberto, eres muy pesado, vete ya"elif echo
"$TRANSCRIPT" |grep -q "main craft"; then aoss espeak -ves
"Abriendo Maincraft" | java -Xmx1024M -Xms512M -cp
/home/yuki/Escritorio/Minecraft.jar
#net.minecraft.LauncherFrameelse aoss espeak -ves "No te
entiendo";aoss espeak -ves "$TRANSCRIPT"fiFunciona perfectamente
:). Ahora el siguiente paso... Migrarlo a una Base de datos! Me he
decantado por MYSQL
Basically, the library encodes the sample into FLAC using a
third party FLAC library, then issues a request to
"https://www.google.com/speech-api/v1/recognize?xjerr=1&client=chromium&lang=en-US"
with a Content-Type header specifying the format (FLAC) and bitrate
(8 kbit).
http://www.codeproject.com/Articles/338010/Fun-with-Google-Speech-Recognition-service
IntroductionI was excited to discover open web services like
Google has, and it was very amazing when I heard about Google
speech recognition.In this article, I write some tips to use Google
speech recognition API in Windows application with direct recording
voice from audio input devices. And also, like a delicious spice -
wear simple program for speech recognition into the utility for
quick issues adding in Redmine project.BackgroundThe basic idea
was: you push the button, some timer starts elapse together with
wave-in device opening, main loop starts and pcm data from buffers
with your voice records to file, timer stops and audio file is
posted to Google for recognition.First task was in understanding
flac encoding in realtime, you can tell 'In *nix,I can write couple
commands in terminal and do all: record, encode, post flac file and
receive answer from server. So, why do you not encode file with
encoder program started after recording wave file?' - because it's
boring, just imagine: your program writes already prepared flac
audio file!From the time then I wrote some application for batch
converting mp3 files to OGG/Vorbis, I have stayed library that can
encode pcm to vorbis in realtime, there also was ring buffer for
that.At that point, the appropriate handler for the flac did not
wait. You might know that Google accepts flac in 16 kHz and 16 bit
per sample with 1(mono) channel format. By using example in
libflac, I add three
functions:InitialiseEncoder,ProcessEncoder,CloseEncoderwhich are,
respectively: open file and prepare encoder, upload to encoder
16bit pcm samples, close file and destroy encoder. One thing: don't
understand why it can't add metadata to flac file?
Maybecharsetproblems?The wonderful article:WaveLib, which has
wave-in API implementation included, that usesRecorderclass: starts
theWaveInRecorderand in parallel uses thread for transmitting pcm
data to encoder.File UploadingThe basic upload function usage
isbelow, change lang parameter optionally:Collapse|Copy Codestring
result = WebUpload.UploadFileEx(flacpath,
"http://www.google.com/speech-api/v1/recognize?lang=ru&client=chromium","file",
"audio/x-flac; rate=16000", parameters, null);Response from server
isreceived in JSON format.Issue CreatingIn which case can youuse
the speech recognition? Maybe for issue creating? Maybe it is not
practical, but certainly funny.The Redmine web application includes
REST web service. By it, we can create issues as much as we need
to, just specify project and tracker, by the way the list of
trackersI could only get younger version 1.3*.Collapse|Copy
CodeRedmineManager manager = new
RedmineManager(Configuration.RedmineHost,
Configuration.RedmineUser, Configuration.RedminePassword); // New
ISSUEvar newIssue = new Issue{ Subject = Title, Description =
Description, Project = new IdentifiableName() { Id = ProjectId },
Tracker = new IdentifiableName() { Id = TrackerId }};// GET ID OF
CURRENT USERUser thisuser = (from u in manager.GetObjectList(new
System.Collections.Specialized.NameValueCollection()) where u.Login
== Configuration.RedmineUser select u).FirstOrDefault();if
(thisuser != null) newIssue.AssignedTo = new IdentifiableName() {
Id = thisuser.Id }; manager.CreateObject(newIssue);Points of
InterestWhen it was over, I drew attention to record timeout, it
gives you 4 secs for your speech: not for all expressions of it may
be appropriate, form maybe needs some stop button?Ring buffer will
save you from data loss in case of such records directly to flac.
When the data comes from the wave-in, they go in the ring
buffer.
FondoLa idea bsica era: se presiona el botn, el temporizador
comienza a transcurrir algunos con olas en la apertura del
dispositivo, las principales salidas de bucle y los datos PCM de
buffers con sus registros de voz en un archivo, se detiene el
temporizador y el archivo de audio se envi a Google para su
reconocimiento.Primera tarea consista en entender la codificacin
flac en tiempo real, se puede decir 'En * nix, puedo escribir par
de comandos en el terminal y hacer todo: grabar, codificar,
publicar archivos flac y recibir respuesta del servidor.As que, por
qu no codificar archivos con el programa codificador iniciado
despus de grabar archivos de onda?- Porque es aburrido, imagnense:
el programa graba archivos de audio flac ya preparado!Desde el
momento en que escrib entonces alguna aplicacin para la conversin
por lotes archivos MP3 a OGG / Vorbis, me he alojado biblioteca que
puede codificar pcm a vorbis en tiempo real, tambin hubo memoria
cclica para eso.En ese punto, el controlador apropiado para el FLAC
no esper.Usted puede saber que Google acepta flac en 16 kHz y 16
bits por muestra con 1 (mono) Formato de canal.Por ejemplo, en el
uso de libflac, aado tres
funciones:InitialiseEncoder,ProcessEncoder,CloseEncoderque son,
respectivamente: archivos abiertos y os preparare encoder, subo a
muestras PCM encoder de 16 bits, cerrar el archivo y destruyo
encoder.Una cosa: no entiendo por qu no se puede aadir metadatos a
los archivos flac?Quizscharsetproblemas?El artculo
maravilloso:WaveLib, que tiene olas de aplicacin API incluida, que
utiliza lagrabadora declase: se inicia laWaveInRecordery
paralelamente utiliza hilos para transmitir datos PCM para
encoder.Carga de archivosEl uso bsico funcin de carga est por
debajo, cambie el parmetro lang opcionalmente:Collapse|Copiar
cdigostring resultado = rate = 16000 " , los parmetros, nulo
);Respuesta del servidor se recibe en formato JSON.Emitir CreacinEn
este caso se puede utilizar el reconocimiento de voz?Tal vez para
la creacin de tema?Tal vez no es prctico, pero sin duda
divertido.La aplicacin web Redmine incluye servicio web REST.Por
ello, podemos crear problemas tanto como necesitamos, basta con
especificar los proyectos y seguimiento, por la forma en la lista
de seguidores que slo poda obtener la versin ms joven 1.3
*.Collapse|Copiar cdigoRedmineManager gerente = nueva
RedmineManager (Configuration.RedmineHost,
Configuration.RedmineUser, Configuration.RedminePassword); / /
NUEVO NMERO var = newIssue nueva emisin{ Subject = Ttulo,
Description = Descripcin, = Proyecto de nueva IdentifiableName ()
{id = Projectid} Rastreador = nueva IdentifiableName () {Id =
TrackerId}};/ / Obtener ID DE USUARIO ACTUAL usuario thisuser = (de
u en manager.GetObjectList ( nueva
System.Collections.Specialized.NameValueCollection ()) donde
u.Login == Configuration.RedmineUser . seleccione u) FirstOrDefault
();si (thisuser! = NULL ) newIssue.AssignedTo = nueva
IdentifiableName () {Id = thisuser.Id}; manager.CreateObject
(newIssue);Puntos de intersCuando todo termin, me llam la atencin
para registrar tiempo de espera, que te da 4 segundos para su
discurso: no para todas las expresiones de la misma puede ser
apropiado, la forma tal vez necesita un poco de botn de
parada?Memoria circular te salvar de la prdida de datos en caso de
tales registros directamente a flac.Cuando los datos proceden de la
onda-in, se van en el bfer de anillo.Historia 28 de febrero 2012:
Primera versinLicencia
IntroductionAs I already mentioned in my articleA low-level
audio player in C#, there are no built-in classes in the .NET
framework for dealing with sound. This holds true not only for
audio playback, but also for audio capture.It should be noted,
though, that the Managed DirectX 9 SDK does include classes for
high-level and low-level audio manipulation. However, sometimes you
dont want your application to depend on the full DX 9 runtime, just
to do basic sound playback and capture, and there are also some
areas where Managed DirectSound doesnt help at all (for example,
multi-channel sound playback and capture).Nevertheless, I strongly
recommend you to use Managed DirectSound for sound playback and
capture unless you have a good reason for not doing so.This article
describes a sample application that uses thewaveInandwaveOutAPIs in
C# through P/Invoke to capture an audio signal from the sound cards
input, and play it back (almost) at the same time.Using the codeThe
sample code reuses theWaveOutPlayerclass from my articleA low-level
audio player in C#. The new classes in this sample
areWaveInRecorderandFifoStream.TheFifoStreamclass
extendsSystem.IO.Streamto implement a FIFO (first-in first-out) of
bytes. The overriddenWritemethod adds data to the FIFOs tail, and
theReadmethod peeks and removes data from the FIFOs head.
TheLengthproperty returns the amount of buffered data at any time.
CallingFlushwill clear all pending data.TheWaveInRecorderclass is
analogous to theWaveOutPlayerclass. In fact, if you look at the
source files, youll notice that the implementations of these
classes are very similar. As withWaveOutPlayer, the interface of
this class has been reduced to the strict minimum.Creating an
instance ofWaveInRecorderwill cause the system to start recording
immediately. Heres the code that creates
theWaveOutPlayerandWaveInRecorderinstances.Collapse|Copy
Codeprivate void Start(){ Stop(); try { WaveLib.WaveFormat fmt =
new WaveLib.WaveFormat(44100, 16, 2); m_Player = new
WaveLib.WaveOutPlayer(-1, fmt, 16384, 3, new
WaveLib.BufferFillEventHandler(Filler)); m_Recorder = new
WaveLib.WaveInRecorder(-1, fmt, 16384, 3, new
WaveLib.BufferDoneEventHandler(DataArrived)); } catch { Stop();
throw; }}TheWaveInRecorderconstructor takes five parameters. Except
for the last parameter, their meaning is the same as
inWaveOutPlayer.The first parameter is the ID of the wave input
device that you want to use. The value-1represents the default
system device, but if your system has more than one sound card,
then you can pass any number from0to the number of installed sound
cards minus one, to select a particular device.The second parameter
is the format of the audio samples.The third and forth parameters
are the size of the internal wave buffers and the number of buffers
to allocate. You should set these to reasonable values. Smaller
buffers will give you less latency, but the captured audio may have
gaps on it if your computer is not fast enough.The fifth and last
parameter is a delegate that will be called periodically as
internal audio buffers are full of captured data. In the sample
application we just write the captured data to the FIFO, like
this:Collapse|Copy Codeprivate void DataArrived(IntPtr data, int
size){ if (m_RecBuffer == null || m_RecBuffer.Length < size)
m_RecBuffer = new byte[size];
System.Runtime.InteropServices.Marshal.Copy(data, m_RecBuffer, 0,
size); m_Fifo.Write(m_RecBuffer, 0, m_RecBuffer.Length);}Similarly,
theFillermethod is called every time the player needs more data.
Our implementation just reads the data from the FIFO, as shown
below:Collapse|Copy Codeprivate void Filler(IntPtr data, int size){
if (m_PlayBuffer == null || m_PlayBuffer.Length < size)
m_PlayBuffer = new byte[size]; if (m_Fifo.Length >= size)
m_Fifo.Read(m_PlayBuffer, 0, size); else for (int i = 0; i <
m_PlayBuffer.Length; i++) m_PlayBuffer[i] = 0;
System.Runtime.InteropServices.Marshal.Copy(m_PlayBuffer, 0, data,
size);}Note that we declared the temporary
buffersm_RecBufferandm_PlayBufferas member fields in order to
improve performance by saving some garbage collections.To stop
streaming, just callDisposeon the player and capture objects. We
also need to flush the FIFO so that the next timeStartis called
there is no residual data to play.Collapse|Copy Codeprivate void
Stop(){ if (m_Player != null) try { m_Player.Dispose(); } finally {
m_Player = null; } if (m_Recorder != null) try {
m_Recorder.Dispose(); } finally { m_Recorder = null; }
m_Fifo.Flush(); // clear all pending data}Conclusion
# Curl-H "Content-Type: audio / x-flac, tasa = 16000"
"https://www.google.com/speech-api/v1/recognize?xjerr=1&client=chromium&lang=en-US"-F
miarchivo = "@ C: \ input.flac"-k-o "C: \ output.txt"Funciona
excelente!Slo algunas notas:1) copiar y pegar: utilizar diferentes
seales cotizacin2) hacer que tipo de seguro = 16000 corresponde a
la tasa de bits (audacity: antes de grabar)!3) que tena un poco
mejor los resultados de grabacin monoAlguien tiene algo sobre:*
Hecho espera de 100-continueMe falta esta milisegundos para esperar
a que ...
1. Path path=Paths.get("out.flac");2.
byte[]data=Files.readAllBytes(path);3. 4.
Stringrequest="https://www.google.com/"+5.
"speech-api/v1/recognize?"+6.
"xjerr=1&client=speech2text&lang=en-US&maxresults=10";7.
URLurl=newURL(request);8.
HttpURLConnectionconnection=(HttpURLConnection)url.openConnection();
9. connection.setDoOutput(true);10. connection.setDoInput(true);11.
connection.setInstanceFollowRedirects(false);12.
connection.setRequestMethod("POST");13.
connection.setRequestProperty("Content-Type","audio/x-flac;
rate=16000");14.
connection.setRequestProperty("User-Agent","speech2text");15.
connection.setConnectTimeout(60000);16.
connection.setUseCaches(false);17. 18.
DataOutputStreamwr=newDataOutputStream(connection.getOutputStream());19.
wr.writeBytes(newString(data));20. wr.flush();21. wr.close();22.
connection.disconnect();23. 24. System.out.println("Done");25. 26.
BufferedReaderin=newBufferedReader(27. newInputStreamReader(28.
connection.getInputStream()));29. StringdecodedString;30.
while((decodedString=in.readLine())!=null){31.
System.out.println(decodedString);32. }
Path path = Paths.get("out.flac"); byte[] data =
Files.readAllBytes(path); String request =
"https://www.google.com/"+ "speech-api/v1/recognize?"+
"xjerr=1&client=speech2text&lang=en-US&maxresults=10";
URL url = new URL(request); HttpURLConnection connection =
(HttpURLConnection) url.openConnection();
connection.setDoOutput(true); connection.setDoInput(true);
connection.setInstanceFollowRedirects(false);
connection.setRequestMethod("POST");
connection.setRequestProperty("Content-Type", "audio/x-flac;
rate=16000"); connection.setRequestProperty("User-Agent",
"speech2text"); connection.setConnectTimeout(60000);
connection.setUseCaches (false); DataOutputStream wr = new
DataOutputStream(connection.getOutputStream ()); wr.writeBytes(new
String(data)); wr.flush(); wr.close(); connection.disconnect();
System.out.println("Done"); BufferedReader in = new BufferedReader(
new InputStreamReader( connection.getInputStream()));String
decodedString;while ((decodedString = in.readLine()) != null)
{System.out.println(decodedString);}
Debe utilizarwr.write (datos);lugar dewr.writeBytes (new String
(datos));Google la respuesta:{ "status" : 0 , "id" :
"e0f4ced346ad18bbb81756ed4d639164-1" , "hiptesis" : [{ "hablasen" :
"hola cmo ests" , "confianza" : 0,94028234 }, { "hablasen" : "hello
how r que " }, { "hablasen" : "Cmo ests hoy u" }, { "hablasen" :
"hola cmo ests en" }]}
1. package test;2. 3. import java.io.BufferedReader;4. import
java.io.DataOutputStream;5. import java.io.InputStreamReader;6.
import java.net.HttpURLConnection;7. import
java.net.MalformedURLException;8. import java.net.URL;9. import
java.nio.file.Files;10. import java.nio.file.Path;11. import
java.nio.file.Paths;12. 13. public class
TestGoogleApiForSpeechRecognition {14. 15. public static void
main(String[] args) throws Exception {16. 17. Path path =
Paths.get("C:\\Users\\CDAC\\Downloads\\priyanka.flac");18. byte[]
data = Files.readAllBytes(path);19. 20. String request =
"https://www.google.com/"+21. "speech-api/v1/recognize?"+22.
"xjerr=0&client=speech2text&lang=en-US&maxresults=20";23.
URL url = new URL(request);24. HttpURLConnection connection =
(HttpURLConnection) url.openConnection(); 25.
connection.setDoOutput(true);26. connection.setDoInput(true);27.
connection.setInstanceFollowRedirects(false);28.
connection.setRequestMethod("POST");29.
connection.setRequestProperty("Content-Type", "audio/x-flac;
rate=16000");30. connection.setRequestProperty("User-Agent",
"speech2text");31. connection.setConnectTimeout(60000);32.
connection.setUseCaches (false);33. 34. DataOutputStream wr = new
DataOutputStream(connection.getOutputStream ());35.
wr.write(data);36. wr.flush();37. wr.close();38.
connection.disconnect();39. 40. System.out.println("Done");41. 42.
BufferedReader in = new BufferedReader(new
InputStreamReader(connection.getInputStream()));43. 44. String
decodedString;45. while ((decodedString = in.readLine()) != null)
{46. System.out.println(decodedString);47. }48. }49. }
package test;
import java.io.BufferedReader;import
java.io.DataOutputStream;import java.io.InputStreamReader;import
java.net.HttpURLConnection;import
java.net.MalformedURLException;import java.net.URL;import
java.nio.file.Files;import java.nio.file.Path;import
java.nio.file.Paths;
public class TestGoogleApiForSpeechRecognition {
public static void main(String[] args) throws Exception {Path
path = Paths.get("C:\\Users\\CDAC\\Downloads\\priyanka.flac");
byte[] data = Files.readAllBytes(path); String request =
"https://www.google.com/"+ "speech-api/v1/recognize?"+
"xjerr=0&client=speech2text&lang=en-US&maxresults=20";
URL url = new URL(request); HttpURLConnection connection =
(HttpURLConnection) url.openConnection();
connection.setDoOutput(true); connection.setDoInput(true);
connection.setInstanceFollowRedirects(false);
connection.setRequestMethod("POST");
connection.setRequestProperty("Content-Type", "audio/x-flac;
rate=16000"); connection.setRequestProperty("User-Agent",
"speech2text"); connection.setConnectTimeout(60000);
connection.setUseCaches (false); DataOutputStream wr = new
DataOutputStream(connection.getOutputStream ()); wr.write(data);
wr.flush(); wr.close(); connection.disconnect();
System.out.println("Done"); BufferedReader in = new
BufferedReader(new InputStreamReader(connection.getInputStream()));
String decodedString; while ((decodedString = in.readLine()) !=
null) { System.out.println(decodedString); }}}
1. CAMINO=Caminos.conseguir("out.flac");2.
byte[]datos=Archivos.ReadAllBytes(ruta);3. 4.
Cadenapeticin="https://www.google.com/"+5.
"Speech-api/v1/recognize?"+6. "Xjerr = 1 & client = speech2text
& lang = es-US & maxResults = 10";7.
URLurl=nuevaURL(peticin);8.
HttpURLConnectionconexin=(HttpURLConnection)url.openConnection();
9. . conexinsetDoOutput(verdadero);10. .
conexinsetDoInput(verdadero);11.
conexin.setInstanceFollowRedirects(false);12.
conexin.setRequestMethod("POST");13.
conexin.setRequestProperty("Content-Type","audio / x-flac, tasa =
16000");14.
conexin.setRequestProperty("User-Agent","speech2text");15. .
conexinsetConnectTimeout(60000);16. conexin.setUseCaches(false);17.
18.
DataOutputStreamwr=nuevaDataOutputStream(conexin.getOutputStream());19.
. wrwriteBytes(nuevacadena(datos));20. wr.ras();21. . wrclose();22.
. conexindesconexin();23. 24. Sistema.cabo.println("Done");25. 26.
BufferedReaderen=nuevoBufferedReader(27. nuevaInputStreamReader(28.
conexin.getInputStream()));29. CadenaDecodedString;30. mientras
que((DecodedString=pulgreadLine())! =NULL){31.
Sistema.cabo.println(DecodedString);32. }
Path path = Paths.get("out.flac"); byte[] data =
Files.readAllBytes(path); String request =
"https://www.google.com/"+ "speech-api/v1/recognize?"+
"xjerr=1&client=speech2text&lang=en-US&maxresults=10";
URL url = new URL(request); HttpURLConnection connection =
(HttpURLConnection) url.openConnection();
connection.setDoOutput(true); connection.setDoInput(true);
connection.setInstanceFollowRedirects(false);
connection.setRequestMethod("POST");
connection.setRequestProperty("Content-Type", "audio/x-flac;
rate=16000"); connection.setRequestProperty("User-Agent",
"speech2text"); connection.setConnectTimeout(60000);
connection.setUseCaches (false); DataOutputStream wr = new
DataOutputStream(connection.getOutputStream ()); wr.writeBytes(new
String(data)); wr.flush(); wr.close(); connection.disconnect();
System.out.println("Done"); BufferedReader in = new BufferedReader(
new InputStreamReader( connection.getInputStream()));String
decodedString;while ((decodedString = in.readLine()) != null)
{System.out.println(decodedString);}
