May contain one or more recognition hypotheses (up to the maximum specified inmax_alternatives). These alternatives are ordered in terms of accuracy, with the top (first) alternative being the most probable, as ranked by the recognizer.
is_final
bool
Iffalse, thisStreamingRecognitionResultrepresents an interim result that may change. Iftrue, this is the final time the speech service will return this particularStreamingRecognitionResult, the recognizer will not return any further hypotheses for this portion of the transcript and corresponding audio.
stability
float
An estimate of the likelihood that the recognizer will not change its guess about this interim result. Values range from 0.0 (completely unstable) to 1.0 (completely stable). This field is only provided for interim results (is_final=false). The default of 0.0 is a sentinel value indicatingstabilitywas not set.
Time offset of the end of this result relative to the beginning of the audio.
channel_tag
int32
For multi-channel audio, this is the channel number corresponding to the recognized result for the audio from that channel. For audio_channel_count = N, its output values can range from '1' to 'N'.
Encoding of audio data sent in allRecognitionAudiomessages. This field is optional forFLACandWAVaudio files and required for all other audio formats. For details, seeAudioEncoding.
sample_rate_hertz
int32
Sample rate in Hertz of the audio data sent in allRecognitionAudiomessages. Valid values are: 8000-48000. 16000 is optimal. For best results, set the sampling rate of the audio source to 16000 Hz. If that's not possible, use the native sample rate of the audio source (instead of re-sampling). This field is optional for FLAC and WAV audio files, but is required for all other audio formats. For details, seeAudioEncoding.
audio_channel_count
int32
The number of channels in the input audio data. ONLY set this for MULTI-CHANNEL recognition. Valid values for LINEAR16 and FLAC are1-8. Valid values for OGG_OPUS are '1'-'254'. Valid value for MULAW, AMR, AMR_WB and SPEEX_WITH_HEADER_BYTE is only1. If0or omitted, defaults to one channel (mono). Note: We only recognize the first channel by default. To perform independent recognition on each channel setenable_separate_recognition_per_channelto 'true'.
enable_separate_recognition_per_channel
bool
This needs to be set totrueexplicitly andaudio_channel_count> 1 to get each channel recognized separately. The recognition result will contain achannel_tagfield to state which channel that result belongs to. If this is not true, we will only recognize the first channel. The request is billed cumulatively for all channels recognized:audio_channel_countmultiplied by the length of the audio.
language_code
string
Required. The language of the supplied audio as aBCP-47language tag. Example: "en-US". SeeLanguage Supportfor a list of the currently supported language codes.
max_alternatives
int32
Maximum number of recognition hypotheses to be returned. Specifically, the maximum number ofSpeechRecognitionAlternativemessages within eachSpeechRecognitionResult. The server may return fewer thanmax_alternatives. Valid values are0-30. A value of0or1will return a maximum of one. If omitted, will return a maximum of one.
profanity_filter
bool
If set totrue, the server will attempt to filter out profanities, replacing all but the initial character in each filtered word with asterisks, e.g. "f***". If set tofalseor omitted, profanities won't be filtered out.
Array ofSpeechContext. A means to provide context to assist the speech recognition. For more information, seespeech adaptation.
enable_word_time_offsets
bool
Iftrue, the top result includes a list of words and the start and end time offsets (timestamps) for those words. Iffalse, no word-level time offset information is returned. The default isfalse.
enable_automatic_punctuation
bool
If 'true', adds punctuation to recognition result hypotheses. This feature is only available in select languages. Setting this for requests in other languages has no effect at all. The default 'false' value does not add punctuation to result hypotheses.
Config to enable speaker diarization and set additional parameters to make diarization better suited for your application. Note: When this is enabled, we send all the words from the beginning of the audio for the top alternative in every consecutive STREAMING responses. This is done in order to improve our speaker tags as our models learn to identify the speakers in the conversation over time. For non-streaming requests, the diarization results will be provided only in the top alternative of the FINAL SpeechRecognitionResult.
Which model to select for the given request. Select the model best suited to your domain to get best results. If a model is not explicitly specified, then we auto-select a model based on the parameters in the RecognitionConfig.
use_enhanced
bool
Set to true to use an enhanced model for speech recognition. Ifuse_enhancedis set to true and themodelfield is not set, then an appropriate enhanced model is chosen if an enhanced model exists for the audio.
Ifuse_enhancedis true and an enhanced version of the specified model does not exist, then the speech is recognized using the standard version of the specified model.
from google.cloud import speech
client = speech.SpeechClient()
with open(speech_file, "rb") as audio_file:
content = audio_file.read()
audio = speech.RecognitionAudio(content=content)
config = speech.RecognitionConfig(
encoding=speech.RecognitionConfig.AudioEncoding.LINEAR16,
sample_rate_hertz=44100,
language_code="en-US",
audio_channel_count=2,
enable_separate_recognition_per_channel=True,
)
response = client.recognize(config=config, audio=audio)
for i, result in enumerate(response.results):
alternative = result.alternatives[0]
print("-" * 20)
print("First alternative of result {}".format(i))
print(u"Transcript: {}".format(alternative.transcript))
print(u"Channel Tag: {}".format(result.channel_tag))
google.api_core.exceptions.InvalidArgument: 400 Request payload size exceeds the limit: 10485760 bytes.
일련의 오디오 파일을 텍스트로 변환하는 프로젝트에 처음으로 GCS Speech API를 사용하고 있습니다.각 파일은 약 60 분이 소요되며 전체 시간 동안 지속적으로 말하는 사람입니다.GC SDK를 설치했으며 다음과 같이 요청을 수행하는 데 사용하고 있습니다.gcloud ml speech recognize-long-running \ "/path/to/file/audio.flac" \ --language-code="pt-PT" --async
API가 최대 180 분까지 파일을 처리 할 수있는 경우 최대10,000자의 음성을출력 할 방법이 없기 때문에 매우 어려운 제한 인 것 같습니다. 오디오 파일을 더 작은 조각으로 나누려고했고 최대 4 개의 15 분 샘플에 도달했지만 동일한 오류가 발생했습니다.게다가, 그것이 효과가 있더라도 여기에서 내가 만드는 모든 새로운 녹음을 앞으로 나누는 것은 매우 지루하고 비실용적 일 것입니다.
(env) C:\__STT>
(env) C:\__STT>
(env) C:\__STT>python transcribe_async.py test_Linda_audio_converter.flac
Traceback (most recent call last):
File "C:\__STT\env\lib\site-packages\google\api_core\grpc_helpers.py", line 57, in error_remapped_callable
return callable_(*args, **kwargs)
File "C:\__STT\env\lib\site-packages\grpc\_channel.py", line 923, in __call__
return _end_unary_response_blocking(state, call, False, None)
File "C:\__STT\env\lib\site-packages\grpc\_channel.py", line 826, in _end_unary_response_blocking
raise _InactiveRpcError(state)
grpc._channel._InactiveRpcError: <_InactiveRpcError of RPC that terminated with:
status = StatusCode.INVALID_ARGUMENT
details = "Request payload size exceeds the limit: 10485760 bytes."
debug_error_string = "{"created":"@1605235850.871000000","description":"Error received from peer ipv4:216.58.220.138:443","file":"src/core/lib/surface/call.cc","file_line":1062,"grpc_message":"Request payload size exceeds the limit: 10485760 bytes.","grpc_status":3}"
>
The above exception was the direct cause of the following exception:
Traceback (most recent call last):
File "transcribe_async.py", line 117, in <module>
transcribe_file(args.path)
File "transcribe_async.py", line 54, in transcribe_file
request={"config": config, "audio": audio}
File "C:\__STT\env\lib\site-packages\google\cloud\speech_v1\services\speech\client.py", line 425, in long_running_recognize
response = rpc(request, retry=retry, timeout=timeout, metadata=metadata,)
File "C:\__STT\env\lib\site-packages\google\api_core\gapic_v1\method.py", line 145, in __call__
return wrapped_func(*args, **kwargs)
File "C:\__STT\env\lib\site-packages\google\api_core\grpc_helpers.py", line 59, in error_remapped_callable
six.raise_from(exceptions.from_grpc_error(exc), exc)
File "<string>", line 3, in raise_from
google.api_core.exceptions.InvalidArgument: 400 Request payload size exceeds the limit: 10485760 bytes.
(env) C:\__STT>
(env) C:\__STT>
Google Cloud 지원팀과 이야기를 나눈 후 무료 평가판 구독 제한과 파일 크기 (~ 60 분) 때문이라는 결론에 도달했습니다.
유료 구독으로 업그레이드하고 내 파일을 Google Cloud Storage에 업로드 한 후 트랜스 크립 션에서 페이로드를받을 수있었습니다.
