mirror of
https://github.com/FAUSheppy/ths-speech
synced 2025-12-07 02:51:37 +01:00
working google recog
This commit is contained in:
1060
python-server/audiosegment_wrapper.py
Normal file
1060
python-server/audiosegment_wrapper.py
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File diff suppressed because it is too large
Load Diff
@@ -1,6 +1,6 @@
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import base64
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import base64
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import os.path
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import os.path
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from pydub import AudioSegment
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import audiosegment_wrapper as AudioSegment
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def save_audio(filename, base64_string):
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def save_audio(filename, base64_string):
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decoded = None
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decoded = None
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@@ -15,7 +15,9 @@ def save_audio(filename, base64_string):
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# return b"ERROR_FILE_EXISTS"
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# return b"ERROR_FILE_EXISTS"
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with open(orig_filename,"wb") as f:
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with open(orig_filename,"wb") as f:
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f.write(decoded)
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f.write(decoded)
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AudioSegment.from_file(orig_filename).export(filename,format="wav")
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seg = AudioSegment.from_file(orig_filename)
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seg = seg.resample(sample_rate_Hz=32000, sample_width=2, channels=1)
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seg.export(filename,format="wav")
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return b"SUCCESS"
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return b"SUCCESS"
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def save_audio_chain(file_str_tupels):
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def save_audio_chain(file_str_tupels):
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@@ -37,6 +39,7 @@ def save_audio_chain(file_str_tupels):
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if not completeAudio:
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if not completeAudio:
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return b"ERROR_AUDIO_CONCAT_FAILED"
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return b"ERROR_AUDIO_CONCAT_FAILED"
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else:
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else:
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completeAudio = completeAudio.resample(sample_rate_Hz=32000, sample_width=2, channels=1)
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completeAudio.export(file_str_tupels[0][0],format="wav")
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completeAudio.export(file_str_tupels[0][0],format="wav")
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return b"SUCCESS"
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return b"SUCCESS"
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@@ -3,6 +3,7 @@ import multiprocessing as mp
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import os.path
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import os.path
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import filesystem
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import filesystem
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import log
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import log
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import transcribe_async
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USE_FREE=False
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USE_FREE=False
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USE_PAID=True
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USE_PAID=True
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@@ -17,26 +18,24 @@ def create_and_save_transcript(filename):
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def analyse(filename):
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def analyse(filename):
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''' returns the transcripted audio, or None if the analysis fails '''
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''' returns the transcripted audio, or None if the analysis fails '''
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try:
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if USE_FREE:
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recognizer = spr.Recognizer()
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recognizer = spr.Recognizer()
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with spr.AudioFile(filename) as source:
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with spr.AudioFile(filename) as source:
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audio = recognizer.record(source)
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audio = recognizer.record(source)
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try:
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if USE_FREE:
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string = free_google_backend(recognizer, audio)
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string = free_google_backend(recognizer, audio)
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elif USE_PAID:
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elif USE_PAID:
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string = paid_google_backend(recognizer,audio)
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string = paid_google_backend(filename)
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except spr.UnknownValueError:
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except spr.UnknownValueError:
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log.log("Audio file is broken or not an audio file")
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log.log("Audio file is broken or not an audio file")
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return "ERROR_AUDIO_FILE_INVALID"
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return "ERROR_AUDIO_FILE_INVALID"
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except spr.RequestError as e:
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except spr.RequestError as e:
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log.log("Could not connect to google API: {}".format(e))
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log.log("Could not connect to google API: {}".format(e))
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return "ERROR_API_FAILURE"
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return "ERROR_API_FAILURE"
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return string
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return string
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def free_google_backend(recognizer, audio):
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def free_google_backend(recognizer, audio):
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return recognizer.recognize_google(audio,language="de-DE")
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return recognizer.recognize_google(audio,language="de-DE")
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def paid_google_backend(recognizer, audio):
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def paid_google_backend(filename):
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pass
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return transcribe_async.transcribe_file(filename)
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@@ -24,68 +24,50 @@ Example usage:
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import argparse
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import argparse
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import io
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import io
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from gcloud import storage
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from google.cloud import speech
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from google.cloud.speech import enums
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from google.cloud.speech import types
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# [START speech_transcribe_async]
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# [START speech_transcribe_async]
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def transcribe_file(speech_file):
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def transcribe_file(speech_file):
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"""Transcribe the given audio file asynchronously."""
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url = upload_file(speech_file)
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from google.cloud import speech
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print(url)
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from google.cloud.speech import enums
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return transcribe_gcs("gs://"+url)
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from google.cloud.speech import types
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client = speech.SpeechClient()
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# [START speech_python_migration_async_request]
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def upload_file(filename):
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with io.open(speech_file, 'rb') as audio_file:
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bukket = "ths-speech-audio/"
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content = audio_file.read()
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client = storage.Client()
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cb = client.get_bucket("ths-speech-audio")
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blob = cb.blob(filename)
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blob.upload_from_filename(filename)
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return bukket + filename
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audio = types.RecognitionAudio(content=content)
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config = types.RecognitionConfig(
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encoding=enums.RecognitionConfig.AudioEncoding.LINEAR16,
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sample_rate_hertz=16000,
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language_code='en-US')
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# [START speech_python_migration_async_response]
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operation = client.long_running_recognize(config, audio)
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# [END speech_python_migration_async_request]
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print('Waiting for operation to complete...')
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response = operation.result(timeout=90)
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# Each result is for a consecutive portion of the audio. Iterate through
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# them to get the transcripts for the entire audio file.
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for result in response.results:
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# The first alternative is the most likely one for this portion.
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print(u'Transcript: {}'.format(result.alternatives[0].transcript))
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print('Confidence: {}'.format(result.alternatives[0].confidence))
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# [END speech_python_migration_async_response]
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# [END speech_transcribe_async]
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# [START speech_transcribe_async_gcs]
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def transcribe_gcs(gcs_uri):
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def transcribe_gcs(gcs_uri):
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"""Asynchronously transcribes the audio file specified by the gcs_uri."""
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"""Asynchronously transcribes the audio file specified by the gcs_uri."""
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from google.cloud import speech
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from google.cloud.speech import enums
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from google.cloud.speech import types
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client = speech.SpeechClient()
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client = speech.SpeechClient()
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audio = types.RecognitionAudio(uri=gcs_uri)
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audio = types.RecognitionAudio(uri=gcs_uri)
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config = types.RecognitionConfig(
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config = types.RecognitionConfig(
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#encoding=enums.RecognitionConfig.AudioEncoding.FLAC,
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encoding=enums.RecognitionConfig.AudioEncoding.LINEAR16,
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#sample_rate_hertz=16000,
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sample_rate_hertz=32000,
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language_code='de-DE')
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language_code='de-DE')
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operation = client.long_running_recognize(config, audio)
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operation = client.long_running_recognize(config, audio)
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print('Waiting for operation to complete...')
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print('Waiting for operation to complete...')
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response = operation.result(timeout=90)
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response = operation.result(timeout=900)
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# Each result is for a consecutive portion of the audio. Iterate through
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# Each result is for a consecutive portion of the audio. Iterate through
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# them to get the transcripts for the entire audio file.
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# them to get the transcripts for the entire audio file.
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ret = ""
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for result in response.results:
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for result in response.results:
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# The first alternative is the most likely one for this portion.
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# The first alternative is the most likely one for this portion.
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ret += result.alternatives[0].transcript
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print(u'Transcript: {}'.format(result.alternatives[0].transcript))
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print(u'Transcript: {}'.format(result.alternatives[0].transcript))
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print('Confidence: {}'.format(result.alternatives[0].confidence))
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print('Confidence: {}'.format(result.alternatives[0].confidence))
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return ret
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# [END speech_transcribe_async_gcs]
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# [END speech_transcribe_async_gcs]
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