我正在从事一个涉及使用SpeechRecognition模块进行语音识别的项目。我想提高语音识别能力的一件事就是能够尽快输出已被识别的单词。我希望它与您讲Google Translate时一样，只要您说出一个单词，它就会在屏幕上输出它，让您知道您已经说过。
我尝试过的一些事情是拥有一个存储独立音频记录的数组，并通过该数组迭代语音识别，以识别每个音频记录，然后将其输出。这是行不通的，因为不同的单词需要花费不同的时间来说。
我进一步研究了SpeechRecognition模块提供给我的Google语音识别API，并希望了解如何通过在某些地方添加打印语句来实现目标来调整实际库。我不知道该放在哪里，因为我是语音识别的初学者，而且我对Google语音识别API不太了解。
这是Google api代码，它访问云以执行sr。
def recognize_google(self, audio_data, key=None, language="en-US", show_all=False):
"""
Performs speech recognition on ``audio_data`` (an ``AudioData`` instance), using the Google Speech Recognition API.
The Google Speech Recognition API key is specified by ``key``. If not specified, it uses a generic key that works out of the box. This should generally be used for personal or testing purposes only, as it **may be revoked by Google at any time**.
To obtain your own API key, simply following the steps on the `API Keys <http://www.chromium.org/developers/how-tos/api-keys>`__ page at the Chromium Developers site. In the Google Developers Console, Google Speech Recognition is listed as "Speech API".
The recognition language is determined by ``language``, an RFC5646 language tag like ``"en-US"`` (US English) or ``"fr-FR"`` (International French), defaulting to US English. A list of supported language tags can be found in this `StackOverflow answer <http://stackoverflow.com/a/14302134>`__.
Returns the most likely transcription if ``show_all`` is false (the default). Otherwise, returns the raw API response as a JSON dictionary.
Raises a ``speech_recognition.UnknownValueError`` exception if the speech is unintelligible. Raises a ``speech_recognition.RequestError`` exception if the speech recognition operation failed, if the key isn't valid, or if there is no internet connection.
"""
assert isinstance(audio_data, AudioData), "``audio_data`` must be audio data"
assert key is None or isinstance(key, str), "``key`` must be ``None`` or a string"
assert isinstance(language, str), "``language`` must be a string"
flac_data = audio_data.get_flac_data(
convert_rate=None if audio_data.sample_rate >= 8000 else 8000, # audio samples must be at least 8 kHz
convert_width=2 # audio samples must be 16-bit
)
if key is None: key = "AIzaSyBOti4mM-6x9WDnZIjIeyEU21OpBXqWBgw"
url = "http://www.google.com/speech-api/v2/recognize?{}".format(urlencode({
"client": "chromium",
"lang": language,
"key": key,
}))
request = Request(url, data=flac_data, headers={"Content-Type": "audio/x-flac; rate={}".format(audio_data.sample_rate)})
# obtain audio transcription results
try:
response = urlopen(request, timeout=self.operation_timeout)
except HTTPError as e:
raise RequestError("recognition request failed: {}".format(e.reason))
except URLError as e:
raise RequestError("recognition connection failed: {}".format(e.reason))
response_text = response.read().decode("utf-8")
# ignore any blank blocks
actual_result = []
for line in response_text.split("\n"):
if not line: continue
result = json.loads(line)["result"]
if len(result) != 0:
actual_result = result[0]
print(actual_result)
sleep(1000)
break
# return results
if show_all: return actual_result
if not isinstance(actual_result, dict) or len(actual_result.get("alternative", [])) == 0: raise UnknownValueError()
if "confidence" in actual_result["alternative"]:
# return alternative with highest confidence score
best_hypothesis = max(actual_result["alternative"], key=lambda alternative: alternative["confidence"])
else:
# when there is no confidence available, we arbitrarily choose the first hypothesis.
best_hypothesis = actual_result["alternative"][0]
if "transcript" not in best_hypothesis: raise UnknownValueError()
return best_hypothesis["transcript"]
这是我的基本代码（此处没有显示我之前尝试过的内容）：它能够成功进行语音识别。
r = sr.Recognizer()
m = sr.Microphone();
r = sr.Recognizer()
on = True
while on :
with sr.Microphone() as source:
audio = r.listen(source)
try:
text = r.recognize_google(audio)
print("You said: {}".format(text))
except:
print("Sorry, we did not recognize your voice")
您应该知道的最后一种方法是制作音频文件或对象的录音功能：
def listen(self, source, timeout=None, phrase_time_limit=None, snowboy_configuration=None):
"""
Records a single phrase from ``source`` (an ``AudioSource`` instance) into an ``AudioData`` instance, which it returns.
This is done by waiting until the audio has an energy above ``recognizer_instance.energy_threshold`` (the user has started speaking), and then recording until it encounters ``recognizer_instance.pause_threshold`` seconds of non-speaking or there is no more audio input. The ending silence is not included.
The ``timeout`` parameter is the maximum number of seconds that this will wait for a phrase to start before giving up and throwing an ``speech_recognition.WaitTimeoutError`` exception. If ``timeout`` is ``None``, there will be no wait timeout.
The ``phrase_time_limit`` parameter is the maximum number of seconds that this will allow a phrase to continue before stopping and returning the part of the phrase processed before the time limit was reached. The resulting audio will be the phrase cut off at the time limit. If ``phrase_timeout`` is ``None``, there will be no phrase time limit.
The ``snowboy_configuration`` parameter allows integration with `Snowboy <https://snowboy.kitt.ai/>`__, an offline, high-accuracy, power-efficient hotword recognition engine. When used, this function will pause until Snowboy detects a hotword, after which it will unpause. This parameter should either be ``None`` to turn off Snowboy support, or a tuple of the form ``(SNOWBOY_LOCATION, LIST_OF_HOT_WORD_FILES)``, where ``SNOWBOY_LOCATION`` is the path to the Snowboy root directory, and ``LIST_OF_HOT_WORD_FILES`` is a list of paths to Snowboy hotword configuration files (`*.pmdl` or `*.umdl` format).
This operation will always complete within ``timeout + phrase_timeout`` seconds if both are numbers, either by returning the audio data, or by raising a ``speech_recognition.WaitTimeoutError`` exception.
"""
assert isinstance(source, AudioSource), "Source must be an audio source"
assert source.stream is not None, "Audio source must be entered before listening, see documentation for ``AudioSource``; are you using ``source`` outside of a ``with`` statement?"
assert self.pause_threshold >= self.non_speaking_duration >= 0
if snowboy_configuration is not None:
assert os.path.isfile(os.path.join(snowboy_configuration[0], "snowboydetect.py")), "``snowboy_configuration[0]`` must be a Snowboy root directory containing ``snowboydetect.py``"
for hot_word_file in snowboy_configuration[1]:
assert os.path.isfile(hot_word_file), "``snowboy_configuration[1]`` must be a list of Snowboy hot word configuration files"
seconds_per_buffer = float(source.CHUNK) / source.SAMPLE_RATE
pause_buffer_count = int(math.ceil(self.pause_threshold / seconds_per_buffer)) # number of buffers of non-speaking audio during a phrase, before the phrase should be considered complete
phrase_buffer_count = int(math.ceil(self.phrase_threshold / seconds_per_buffer)) # minimum number of buffers of speaking audio before we consider the speaking audio a phrase
non_speaking_buffer_count = int(math.ceil(self.non_speaking_duration / seconds_per_buffer)) # maximum number of buffers of non-speaking audio to retain before and after a phrase
# read audio input for phrases until there is a phrase that is long enough
elapsed_time = 0 # number of seconds of audio read
buffer = b"" # an empty buffer means that the stream has ended and there is no data left to read
while True:
frames = collections.deque()
if snowboy_configuration is None:
# store audio input until the phrase starts
while True:
# handle waiting too long for phrase by raising an exception
elapsed_time += seconds_per_buffer
if timeout and elapsed_time > timeout:
raise WaitTimeoutError("listening timed out while waiting for phrase to start")
buffer = source.stream.read(source.CHUNK)
if len(buffer) == 0: break # reached end of the stream
frames.append(buffer)
if len(frames) > non_speaking_buffer_count: # ensure we only keep the needed amount of non-speaking buffers
frames.popleft()
# detect whether speaking has started on audio input
energy = audioop.rms(buffer, source.SAMPLE_WIDTH) # energy of the audio signal
if energy > self.energy_threshold: break
# dynamically adjust the energy threshold using asymmetric weighted average
if self.dynamic_energy_threshold:
damping = self.dynamic_energy_adjustment_damping ** seconds_per_buffer # account for different chunk sizes and rates
target_energy = energy * self.dynamic_energy_ratio
self.energy_threshold = self.energy_threshold * damping + target_energy * (1 - damping)
else:
# read audio input until the hotword is said
snowboy_location, snowboy_hot_word_files = snowboy_configuration
buffer, delta_time = self.snowboy_wait_for_hot_word(snowboy_location, snowboy_hot_word_files, source, timeout)
elapsed_time += delta_time
if len(buffer) == 0: break # reached end of the stream
frames.append(buffer)
# read audio input until the phrase ends
pause_count, phrase_count = 0, 0
phrase_start_time = elapsed_time
while True:
# handle phrase being too long by cutting off the audio
elapsed_time += seconds_per_buffer
if phrase_time_limit and elapsed_time - phrase_start_time > phrase_time_limit:
break
buffer = source.stream.read(source.CHUNK)
if len(buffer) == 0: break # reached end of the stream
frames.append(buffer)
phrase_count += 1
# check if speaking has stopped for longer than the pause threshold on the audio input
energy = audioop.rms(buffer, source.SAMPLE_WIDTH) # unit energy of the audio signal within the buffer
if energy > self.energy_threshold:
pause_count = 0
else:
pause_count += 1
if pause_count > pause_buffer_count: # end of the phrase
break
# check how long the detected phrase is, and retry listening if the phrase is too short
phrase_count -= pause_count # exclude the buffers for the pause before the phrase
if phrase_count >= phrase_buffer_count or len(buffer) == 0: break # phrase is long enough or we've reached the end of the stream, so stop listening
# obtain frame data
for i in range(pause_count - non_speaking_buffer_count): frames.pop() # remove extra non-speaking frames at the end
frame_data = b"".join(frames)
return AudioData(frame_data, source.SAMPLE_RATE, source.SAMPLE_WIDTH)
我想获得一种能够采用当前代码并以尽快输出可识别单词的方式实现的软件。就像您在Google翻译中讲话一样。
答案 0 :(得分：1)
您要的是连续语音识别器。这些已经存在，但是语音识别器需要为此应用专门构建，因为它需要非常快速地响应，并且能够正确处理尚未完成的语音。
您使用的Google语音API（https://www.google.com/speech-api/v2/recognize）不是连续的语音识别器。您无法通过修改调用它的代码将其变成连续识别器。
答案 1 :(得分：0)
您看到了吗？ https://speech-to-text-demo.ng.bluemix.net只需单击“录制音频”，您说话时就会在屏幕上看到假设。该演示是开源的，您可以在GitHub中分叉代码。通过使用您喜欢的编程语言使用WebSocket API调用服务，可以实现连续的语音识别效果。
答案 2 :(得分：0)
你可以使用另一个语音引擎的 Pocketsphinx 而他们不会通过互联网连接使用
https://pypi.org/project/pocketsphinx/
from pocketsphinx import LiveSpeech
speech = LiveSpeech(lm=False, keyphrase='forward', kws_threshold=1e-20)
for phrase in speech:
print(phrase.segments(detailed=True))
答案 3 :(得分：-1)
您可以使用threading它是内置的python模块