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| author | schneefux <schneefux+commit@schneefux.xyz> | 2014-09-27 13:40:11 +0200 |
|---|---|---|
| committer | schneefux <schneefux+commit@schneefux.xyz> | 2014-09-27 13:40:11 +0200 |
| commit | 5d6d5177f9bf4947d861815db837dfb813a4964c (patch) | |
| tree | bbbc5b3e1b9125e2f7ff137d8ad3fbd6828c8368 | |
| parent | 16905e89b0e0d15f86df4cdf6536a396e06c49e6 (diff) | |
| parent | 8d2e10462a79fc34c446d837cb9af6a5d5171efd (diff) | |
| download | jasper-client-5d6d5177f9bf4947d861815db837dfb813a4964c.tar.gz jasper-client-5d6d5177f9bf4947d861815db837dfb813a4964c.zip | |
Merge pull request #176 from Holzhaus/improved-stt-module
Improved stt module
| -rw-r--r-- | client/mic.py | 51 | ||||
| -rw-r--r-- | client/stt.py | 195 |
2 files changed, 156 insertions, 90 deletions
diff --git a/client/mic.py b/client/mic.py index 0351592..2e8d603 100644 --- a/client/mic.py +++ b/client/mic.py @@ -4,11 +4,13 @@ """ import os -from wave import open as open_audio +import tempfile +import wave import audioop import pyaudio import alteration import jasperpath +from stt import TranscriptionMode class Mic: @@ -80,7 +82,6 @@ class Mic: """ THRESHOLD_MULTIPLIER = 1.8 - AUDIO_FILE = "passive.wav" RATE = 16000 CHUNK = 1024 @@ -154,15 +155,17 @@ class Mic: stream.stop_stream() stream.close() audio.terminate() - write_frames = open_audio(AUDIO_FILE, 'wb') - write_frames.setnchannels(1) - write_frames.setsampwidth(audio.get_sample_size(pyaudio.paInt16)) - write_frames.setframerate(RATE) - write_frames.writeframes(''.join(frames)) - write_frames.close() - - # check if PERSONA was said - transcribed = self.passive_stt_engine.transcribe(AUDIO_FILE, PERSONA_ONLY=True) + + with tempfile.NamedTemporaryFile(mode='w+b') as f: + wav_fp = wave.open(f, 'wb') + wav_fp.setnchannels(1) + wav_fp.setsampwidth(audio.get_sample_size(pyaudio.paInt16)) + wav_fp.setframerate(RATE) + wav_fp.writeframes(''.join(frames)) + wav_fp.close() + f.seek(0) + # check if PERSONA was said + transcribed = self.passive_stt_engine.transcribe(f, mode=TranscriptionMode.KEYWORD) if PERSONA in transcribed: return (THRESHOLD, PERSONA) @@ -187,18 +190,10 @@ class Mic: Returns a list of the matching options or None """ - AUDIO_FILE = "active.wav" RATE = 16000 CHUNK = 1024 LISTEN_TIME = 12 - # user can request pre-recorded sound - if not LISTEN: - if not os.path.exists(AUDIO_FILE): - return None - - return self.active_stt_engine.transcribe(AUDIO_FILE) - # check if no threshold provided if THRESHOLD == None: THRESHOLD = self.fetchThreshold() @@ -238,14 +233,18 @@ class Mic: stream.stop_stream() stream.close() audio.terminate() - write_frames = open_audio(AUDIO_FILE, 'wb') - write_frames.setnchannels(1) - write_frames.setsampwidth(audio.get_sample_size(pyaudio.paInt16)) - write_frames.setframerate(RATE) - write_frames.writeframes(''.join(frames)) - write_frames.close() - return self.active_stt_engine.transcribe(AUDIO_FILE, MUSIC=MUSIC) + with tempfile.SpooledTemporaryFile(mode='w+b') as f: + wav_fp = wave.open(f, 'wb') + wav_fp.setnchannels(1) + wav_fp.setsampwidth(audio.get_sample_size(pyaudio.paInt16)) + wav_fp.setframerate(RATE) + wav_fp.writeframes(''.join(frames)) + wav_fp.close() + f.seek(0) + mode = TranscriptionMode.MUSIC if MUSIC else TranscriptionMode.NORMAL + transcribed = self.active_stt_engine.transcribe(f, mode=mode) + return transcribed def say(self, phrase, OPTIONS=" -vdefault+m3 -p 40 -s 160 --stdout > say.wav"): # alter phrase before speaking diff --git a/client/stt.py b/client/stt.py index fc1bd1f..899775b 100644 --- a/client/stt.py +++ b/client/stt.py @@ -2,9 +2,11 @@ # -*- coding: utf-8-*- import os import traceback +import wave import json import tempfile import logging +from abc import ABCMeta, abstractmethod import requests import yaml @@ -12,12 +14,37 @@ import yaml The default Speech-to-Text implementation which relies on PocketSphinx. """ +class TranscriptionMode: + NORMAL, KEYWORD, MUSIC = range(3) -class PocketSphinxSTT(object): +class AbstractSTTEngine(object): + """ + Generic parent class for all STT engines + """ + + __metaclass__ = ABCMeta + + @classmethod + def get_config(cls): + return {} + + @classmethod + @abstractmethod + def is_available(cls): + return True + + @abstractmethod + def transcribe(self, fp, mode=TranscriptionMode.NORMAL): + pass + +class PocketSphinxSTT(AbstractSTTEngine): + + SLUG = 'sphinx' def __init__(self, lmd="languagemodel.lm", dictd="dictionary.dic", lmd_persona="languagemodel_persona.lm", dictd_persona="dictionary_persona.dic", - lmd_music=None, dictd_music=None, **kwargs): + lmd_music=None, dictd_music=None, + hmm_dir="/usr/local/share/pocketsphinx/model/hmm/en_US/hub4wsj_sc_8k"): """ Initiates the pocketsphinx instance. @@ -37,38 +64,51 @@ class PocketSphinxSTT(object): except: import pocketsphinx as ps - hmm_dir = None + self._logfiles = {} + with tempfile.NamedTemporaryFile(prefix='psdecoder_music_', suffix='.log', delete=False) as f: + self._logfiles[TranscriptionMode.MUSIC] = f.name + with tempfile.NamedTemporaryFile(prefix='psdecoder_keyword_', suffix='.log', delete=False) as f: + self._logfiles[TranscriptionMode.KEYWORD] = f.name + with tempfile.NamedTemporaryFile(prefix='psdecoder_normal_', suffix='.log', delete=False) as f: + self._logfiles[TranscriptionMode.NORMAL] = f.name + + self._decoders = {} + if lmd_music and dictd_music: + self._decoders[TranscriptionMode.MUSIC] = ps.Decoder(hmm=hmm_dir, lm=lmd_music, dict=dictd_music, logfn=self._logfiles[TranscriptionMode.MUSIC]) + self._decoders[TranscriptionMode.KEYWORD] = ps.Decoder(hmm=hmm_dir, lm=lmd_persona, dict=dictd_persona, logfn=self._logfiles[TranscriptionMode.KEYWORD]) + self._decoders[TranscriptionMode.NORMAL] = ps.Decoder(hmm=hmm_dir, lm=lmd, dict=dictd, logfn=self._logfiles[TranscriptionMode.NORMAL]) + def __del__(self): + for filename in self._logfiles.values(): + os.remove(filename) + + @classmethod + def get_config(cls): #FIXME: Replace this as soon as we have a config module + config = {} + # HMM dir # Try to get hmm_dir from config profile_path = os.path.join(os.path.dirname(__file__), 'profile.yml') if os.path.exists(profile_path): with open(profile_path, 'r') as f: profile = yaml.safe_load(f) - if 'pocketsphinx' in profile and 'hmm_dir' in profile['pocketsphinx']: - hmm_dir = profile['pocketsphinx']['hmm_dir'] - - if not hmm_dir: - hmm_dir = "/usr/local/share/pocketsphinx/model/hmm/en_US/hub4wsj_sc_8k" - - with tempfile.NamedTemporaryFile(prefix='psdecoder_music_', suffix='.log', delete=False) as f: - self.logfile_music = f.name - with tempfile.NamedTemporaryFile(prefix='psdecoder_persona_', suffix='.log', delete=False) as f: - self.logfile_persona = f.name - with tempfile.NamedTemporaryFile(prefix='psdecoder_default_', suffix='.log', delete=False) as f: - self.logfile_default = f.name - - if lmd_music and dictd_music: - self.speechRec_music = ps.Decoder(hmm=hmm_dir, lm=lmd_music, dict=dictd_music, logfn=self.logfile_music) - self.speechRec_persona = ps.Decoder( - hmm=hmm_dir, lm=lmd_persona, dict=dictd_persona, logfn=self.logfile_persona) - self.speechRec = ps.Decoder(hmm=hmm_dir, lm=lmd, dict=dictd, logfn=self.logfile_default) - - def __del__(self): - os.remove(self.logfile_music) - os.remove(self.logfile_persona) - os.remove(self.logfile_default) + if 'pocketsphinx' in profile: + if 'hmm_dir' in profile['pocketsphinx']: + config['hmm_dir'] = profile['pocketsphinx']['hmm_dir'] + if 'lmd' in profile['pocketsphinx']: + config['lmd'] = profile['pocketsphinx']['lmd'] + if 'dictd' in profile['pocketsphinx']: + config['dictd'] = profile['pocketsphinx']['dictd'] + if 'lmd_persona' in profile['pocketsphinx']: + config['lmd_persona'] = profile['pocketsphinx']['lmd_persona'] + if 'dictd_persona' in profile['pocketsphinx']: + config['dictd_persona'] = profile['pocketsphinx']['dictd_persona'] + if 'lmd_music' in profile['pocketsphinx']: + config['lmd'] = profile['pocketsphinx']['lmd_music'] + if 'dictd_music' in profile['pocketsphinx']: + config['dictd_music'] = profile['pocketsphinx']['dictd_music'] + return config - def transcribe(self, audio_file_path, PERSONA_ONLY=False, MUSIC=False): + def transcribe(self, fp, mode=TranscriptionMode.NORMAL): """ Performs STT, transcribing an audio file and returning the result. @@ -77,30 +117,27 @@ class PocketSphinxSTT(object): PERSONA_ONLY -- if True, uses the 'Persona' language model and dictionary MUSIC -- if True, uses the 'Music' language model and dictionary """ + decoder = self._decoders[mode] - wavFile = file(audio_file_path, 'rb') - wavFile.seek(44) + fp.seek(44) - if MUSIC: - self.speechRec_music.decode_raw(wavFile) - result = self.speechRec_music.get_hyp() - with open(self.logfile_music, 'r+') as f: + # FIXME: Can't use the Decoder.decode_raw() here, because + # pocketsphinx segfaults with tempfile.SpooledTemporaryFile() + data = fp.read() + decoder.start_utt() + decoder.process_raw(data, False, True) + decoder.end_utt() + + result = decoder.get_hyp() + with open(self._logfiles[mode], 'r+') as f: + if mode == TranscriptionMode.KEYWORD: + modename = "[KEYWORD]" + elif mode == TranscriptionMode.MUSIC: + modename = "[MUSIC]" + else: + modename = "[NORMAL]" for line in f: - self._logger.debug("speechRec_music %s", line.strip()) - f.truncate() - elif PERSONA_ONLY: - self.speechRec_persona.decode_raw(wavFile) - result = self.speechRec_persona.get_hyp() - with open(self.logfile_persona, 'r+') as f: - for line in f: - self._logger.debug("speechRec_persona %s", line.strip()) - f.truncate() - else: - self.speechRec.decode_raw(wavFile) - result = self.speechRec.get_hyp() - with open(self.logfile_default, 'r+') as f: - for line in f: - self._logger.debug("speechRec_default %s", line.strip()) + self._logger.debug("%s %s", modename, line.strip()) f.truncate() print "===================" @@ -109,6 +146,10 @@ class PocketSphinxSTT(object): return [result[0]] + @classmethod + def is_available(cls): + return (pkgutil.get_loader('pocketsphinx') is not None and os.path.exits(cls._get_hmm_dir())) + """ Speech-To-Text implementation which relies on the Google Speech API. @@ -134,19 +175,34 @@ Excerpt from sample profile.yml: """ -class GoogleSTT(object): +class GoogleSTT(AbstractSTTEngine): - RATE = 16000 + SLUG = 'google' - def __init__(self, api_key, **kwargs): + def __init__(self, api_key=None): #FIXME: get init args from config """ Arguments: api_key - the public api key which allows access to Google APIs """ + if not api_key: + raise ValueError("No Google API Key given") self.api_key = api_key self.http = requests.Session() - def transcribe(self, audio_file_path, PERSONA_ONLY=False, MUSIC=False): + @classmethod + def get_config(cls): #FIXME: Replace this as soon as we have a config module + config = {} + # HMM dir + # Try to get hmm_dir from config + profile_path = os.path.join(os.path.dirname(__file__), 'profile.yml') + if os.path.exists(profile_path): + with open(profile_path, 'r') as f: + profile = yaml.safe_load(f) + if 'keys' in profile and 'GOOGLE_SPEECH' in profile['keys']: + config['api_key'] = profile['keys']['GOOGLE_SPEECH'] + return config + + def transcribe(self, fp, mode=TranscriptionMode.NORMAL): """ Performs STT via the Google Speech API, transcribing an audio file and returning an English string. @@ -154,15 +210,18 @@ class GoogleSTT(object): Arguments: audio_file_path -- the path to the .wav file to be transcribed """ + + wav = wave.open(fp, 'rb') + frame_rate = wav.getframerate() + wav.close() + url = "https://www.google.com/speech-api/v2/recognize?output=json&client=chromium&key=%s&lang=%s&maxresults=6&pfilter=2" % ( self.api_key, "en-us") - wav = open(audio_file_path, 'rb') - data = wav.read() - wav.close() + data = fp.read() try: - headers = {'Content-type': 'audio/l16; rate=%s' % GoogleSTT.RATE} + headers = {'Content-type': 'audio/l16; rate=%s' % frame_rate} response = self.http.post(url, data=data, headers=headers) response.encoding = 'utf-8' response_read = response.text @@ -181,6 +240,10 @@ class GoogleSTT(object): except Exception: traceback.print_exc() + @classmethod + def is_available(cls): + return True + """ Returns a Speech-To-Text engine. @@ -191,13 +254,17 @@ Arguments: engine_type - one of "sphinx" or "google" kwargs - keyword arguments passed to the constructor of the STT engine """ +def get_engines(): + return [stt_engine for stt_engine in AbstractSTTEngine.__subclasses__() if hasattr(stt_engine, 'SLUG') and stt_engine.SLUG] - -def newSTTEngine(engine_type, **kwargs): - t = engine_type.lower() - if t == "sphinx": - return PocketSphinxSTT(**kwargs) - elif t == "google": - return GoogleSTT(**kwargs) +def newSTTEngine(stt_engine, **kwargs): + selected_engines = filter(lambda engine: hasattr(engine, "SLUG") and engine.SLUG == stt_engine, get_engines()) + if len(selected_engines) == 0: + raise ValueError("No STT engine found for slug '%s'" % stt_engine) else: - raise ValueError("Unsupported STT engine type: " + engine_type) + if len(selected_engines) > 1: + print("WARNING: Multiple STT engines found for slug '%s'. This is most certainly a bug." % stt_engine) + engine = selected_engines[0] + if not engine.is_available(): + raise ValueError("STT engine '%s' is not available (due to missing dependencies, missing dependencies, etc.)" % stt_engine) + return engine(**engine.get_config()) |
