As of January 1, 2020 this library no longer supports Python 2 on the latest released version. Library versions released prior to that date will continue to be available. For more information please visit Python 2 support on Google Cloud.

Types for Google Cloud Mediatranslation v1beta1 API

class google.cloud.mediatranslation_v1beta1.types.StreamingTranslateSpeechConfig(mapping=None, *, ignore_unknown_fields=False, **kwargs)

Bases: proto.message.Message

Config used for streaming translation.

audio_config

Required. The common config for all the following audio contents.

Type

google.cloud.mediatranslation_v1beta1.types.TranslateSpeechConfig

single_utterance

Optional. If false or omitted, the system performs continuous translation (continuing to wait for and process audio even if the user pauses speaking) until the client closes the input stream (gRPC API) or until the maximum time limit has been reached. May return multiple StreamingTranslateSpeechResults with the is_final flag set to true.

If true, the speech translator will detect a single spoken utterance. When it detects that the user has paused or stopped speaking, it will return an END_OF_SINGLE_UTTERANCE event and cease translation. When the client receives ‘END_OF_SINGLE_UTTERANCE’ event, the client should stop sending the requests. However, clients should keep receiving remaining responses until the stream is terminated. To construct the complete sentence in a streaming way, one should override (if ‘is_final’ of previous response is false), or append (if ‘is_final’ of previous response is true).

Type

bool

class google.cloud.mediatranslation_v1beta1.types.StreamingTranslateSpeechRequest(mapping=None, *, ignore_unknown_fields=False, **kwargs)

Bases: proto.message.Message

The top-level message sent by the client for the StreamingTranslateSpeech method. Multiple StreamingTranslateSpeechRequest messages are sent. The first message must contain a streaming_config message and must not contain audio_content data. All subsequent messages must contain audio_content data and must not contain a streaming_config message.

This message has oneof fields (mutually exclusive fields). For each oneof, at most one member field can be set at the same time. Setting any member of the oneof automatically clears all other members.

streaming_config

Provides information to the recognizer that specifies how to process the request. The first StreamingTranslateSpeechRequest message must contain a streaming_config message.

This field is a member of oneof streaming_request.

Type

google.cloud.mediatranslation_v1beta1.types.StreamingTranslateSpeechConfig

audio_content

The audio data to be translated. Sequential chunks of audio data are sent in sequential StreamingTranslateSpeechRequest messages. The first StreamingTranslateSpeechRequest message must not contain audio_content data and all subsequent StreamingTranslateSpeechRequest messages must contain audio_content data. The audio bytes must be encoded as specified in StreamingTranslateSpeechConfig. Note: as with all bytes fields, protobuffers use a pure binary representation (not base64).

This field is a member of oneof streaming_request.

Type

bytes

class google.cloud.mediatranslation_v1beta1.types.StreamingTranslateSpeechResponse(mapping=None, *, ignore_unknown_fields=False, **kwargs)

Bases: proto.message.Message

A streaming speech translation response corresponding to a portion of the audio currently processed.

error

Output only. If set, returns a [google.rpc.Status][google.rpc.Status] message that specifies the error for the operation.

Type

google.rpc.status_pb2.Status

result

Output only. The translation result that is currently being processed (is_final could be true or false).

Type

google.cloud.mediatranslation_v1beta1.types.StreamingTranslateSpeechResult

speech_event_type

Output only. Indicates the type of speech event.

Type

google.cloud.mediatranslation_v1beta1.types.StreamingTranslateSpeechResponse.SpeechEventType

class SpeechEventType(value)

Bases: proto.enums.Enum

Indicates the type of speech event.

Values:

SPEECH_EVENT_TYPE_UNSPECIFIED (0):

No speech event specified.

END_OF_SINGLE_UTTERANCE (1):

This event indicates that the server has detected the end of the user’s speech utterance and expects no additional speech. Therefore, the server will not process additional audio (although it may subsequently return additional results). When the client receives ‘END_OF_SINGLE_UTTERANCE’ event, the client should stop sending the requests. However, clients should keep receiving remaining responses until the stream is terminated. To construct the complete sentence in a streaming way, one should override (if ‘is_final’ of previous response is false), or append (if ‘is_final’ of previous response is true). This event is only sent if single_utterance was set to true, and is not used otherwise.

class google.cloud.mediatranslation_v1beta1.types.StreamingTranslateSpeechResult(mapping=None, *, ignore_unknown_fields=False, **kwargs)

Bases: proto.message.Message

A streaming speech translation result corresponding to a portion of the audio that is currently being processed.

text_translation_result

Text translation result.

This field is a member of oneof result.

Type

google.cloud.mediatranslation_v1beta1.types.StreamingTranslateSpeechResult.TextTranslationResult

class TextTranslationResult(mapping=None, *, ignore_unknown_fields=False, **kwargs)

Bases: proto.message.Message

Text translation result.

translation

Output only. The translated sentence.

Type

str

is_final

Output only. If false, this StreamingTranslateSpeechResult represents an interim result that may change. If true, this is the final time the translation service will return this particular StreamingTranslateSpeechResult, the streaming translator will not return any further hypotheses for this portion of the transcript and corresponding audio.

Type

bool

class google.cloud.mediatranslation_v1beta1.types.TranslateSpeechConfig(mapping=None, *, ignore_unknown_fields=False, **kwargs)

Bases: proto.message.Message

Provides information to the speech translation that specifies how to process the request.

audio_encoding

Required. Encoding of audio data. Supported formats:

• linear16

  Uncompressed 16-bit signed little-endian samples (Linear PCM).

• flac

  flac (Free Lossless Audio Codec) is the recommended encoding because it is lossless–therefore recognition is not compromised–and requires only about half the bandwidth of linear16.

• mulaw

  8-bit samples that compand 14-bit audio samples using G.711 PCMU/mu-law.

• amr

  Adaptive Multi-Rate Narrowband codec. sample_rate_hertz must be 8000.

• amr-wb

  Adaptive Multi-Rate Wideband codec. sample_rate_hertz must be 16000.

• ogg-opus

  Opus encoded audio frames in Ogg container. sample_rate_hertz must be one of 8000, 12000, 16000, 24000, or 48000.

• mp3

  MP3 audio. Support all standard MP3 bitrates (which range from 32-320 kbps). When using this encoding, sample_rate_hertz has to match the sample rate of the file being used.

Type

str

source_language_code

Required. Source language code (BCP-47) of the input audio.

Type

str

target_language_code

Required. Target language code (BCP-47) of the output.

Type

str

sample_rate_hertz

Optional. Sample rate in Hertz of the audio data. 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).

Type

int

model

Optional. google-provided-model/video and google-provided-model/enhanced-phone-call are premium models. google-provided-model/phone-call is not premium model.

Type

str