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 Texttospeech v1beta1 API

class google.cloud.texttospeech_v1beta1.types.AudioConfig(mapping=None, *, ignore_unknown_fields=False, **kwargs)[source]

Bases: proto.message.Message

Description of audio data to be synthesized.

audio_encoding

Required. The format of the audio byte stream.

Type

google.cloud.texttospeech_v1beta1.types.AudioEncoding

speaking_rate

Optional. Input only. Speaking rate/speed, in the range [0.25, 4.0]. 1.0 is the normal native speed supported by the specific voice. 2.0 is twice as fast, and 0.5 is half as fast. If unset(0.0), defaults to the native 1.0 speed. Any other values < 0.25 or > 4.0 will return an error.

Type

float

pitch

Optional. Input only. Speaking pitch, in the range [-20.0, 20.0]. 20 means increase 20 semitones from the original pitch. -20 means decrease 20 semitones from the original pitch.

Type

float

volume_gain_db

Optional. Input only. Volume gain (in dB) of the normal native volume supported by the specific voice, in the range [-96.0, 16.0]. If unset, or set to a value of 0.0 (dB), will play at normal native signal amplitude. A value of -6.0 (dB) will play at approximately half the amplitude of the normal native signal amplitude. A value of +6.0 (dB) will play at approximately twice the amplitude of the normal native signal amplitude. Strongly recommend not to exceed +10 (dB) as there’s usually no effective increase in loudness for any value greater than that.

Type

float

sample_rate_hertz

Optional. The synthesis sample rate (in hertz) for this audio. When this is specified in SynthesizeSpeechRequest, if this is different from the voice’s natural sample rate, then the synthesizer will honor this request by converting to the desired sample rate (which might result in worse audio quality), unless the specified sample rate is not supported for the encoding chosen, in which case it will fail the request and return [google.rpc.Code.INVALID_ARGUMENT][google.rpc.Code.INVALID_ARGUMENT].

Type

int

effects_profile_id

Optional. Input only. An identifier which selects ‘audio effects’ profiles that are applied on (post synthesized) text to speech. Effects are applied on top of each other in the order they are given. See audio profiles for current supported profile ids.

Type

MutableSequence[str]

class google.cloud.texttospeech_v1beta1.types.AudioEncoding(value)[source]

Bases: proto.enums.Enum

Configuration to set up audio encoder. The encoding determines the output audio format that we’d like.

Values:
AUDIO_ENCODING_UNSPECIFIED (0):

Not specified. Will return result [google.rpc.Code.INVALID_ARGUMENT][google.rpc.Code.INVALID_ARGUMENT].

LINEAR16 (1):

Uncompressed 16-bit signed little-endian samples (Linear PCM). Audio content returned as LINEAR16 also contains a WAV header.

MP3 (2):

MP3 audio at 32kbps.

MP3_64_KBPS (4):

MP3 at 64kbps.

OGG_OPUS (3):

Opus encoded audio wrapped in an ogg container. The result will be a file which can be played natively on Android, and in browsers (at least Chrome and Firefox). The quality of the encoding is considerably higher than MP3 while using approximately the same bitrate.

MULAW (5):

8-bit samples that compand 14-bit audio samples using G.711 PCMU/mu-law. Audio content returned as MULAW also contains a WAV header.

ALAW (6):

8-bit samples that compand 14-bit audio samples using G.711 PCMU/A-law. Audio content returned as ALAW also contains a WAV header.

class google.cloud.texttospeech_v1beta1.types.CustomVoiceParams(mapping=None, *, ignore_unknown_fields=False, **kwargs)[source]

Bases: proto.message.Message

Description of the custom voice to be synthesized.

model

Required. The name of the AutoML model that synthesizes the custom voice.

Type

str

reported_usage

Optional. The usage of the synthesized audio to be reported.

Type

google.cloud.texttospeech_v1beta1.types.CustomVoiceParams.ReportedUsage

class ReportedUsage(value)[source]

Bases: proto.enums.Enum

The usage of the synthesized audio. You must report your honest and correct usage of the service as it’s regulated by contract and will cause significant difference in billing.

Values:
REPORTED_USAGE_UNSPECIFIED (0):

Request with reported usage unspecified will be rejected.

REALTIME (1):

For scenarios where the synthesized audio is not downloadable and can only be used once. For example, real-time request in IVR system.

OFFLINE (2):

For scenarios where the synthesized audio is downloadable and can be reused. For example, the synthesized audio is downloaded, stored in customer service system and played repeatedly.

class google.cloud.texttospeech_v1beta1.types.ListVoicesRequest(mapping=None, *, ignore_unknown_fields=False, **kwargs)[source]

Bases: proto.message.Message

The top-level message sent by the client for the ListVoices method.

language_code

Optional. Recommended. BCP-47 language tag. If not specified, the API will return all supported voices. If specified, the ListVoices call will only return voices that can be used to synthesize this language_code. For example, if you specify "en-NZ", all "en-NZ" voices will be returned. If you specify "no", both "no-\*" (Norwegian) and "nb-\*" (Norwegian Bokmal) voices will be returned.

Type

str

class google.cloud.texttospeech_v1beta1.types.ListVoicesResponse(mapping=None, *, ignore_unknown_fields=False, **kwargs)[source]

Bases: proto.message.Message

The message returned to the client by the ListVoices method.

voices

The list of voices.

Type

MutableSequence[google.cloud.texttospeech_v1beta1.types.Voice]

class google.cloud.texttospeech_v1beta1.types.SsmlVoiceGender(value)[source]

Bases: proto.enums.Enum

Gender of the voice as described in SSML voice element.

Values:
SSML_VOICE_GENDER_UNSPECIFIED (0):

An unspecified gender. In VoiceSelectionParams, this means that the client doesn’t care which gender the selected voice will have. In the Voice field of ListVoicesResponse, this may mean that the voice doesn’t fit any of the other categories in this enum, or that the gender of the voice isn’t known.

MALE (1):

A male voice.

FEMALE (2):

A female voice.

NEUTRAL (3):

A gender-neutral voice. This voice is not yet supported.

class google.cloud.texttospeech_v1beta1.types.SynthesisInput(mapping=None, *, ignore_unknown_fields=False, **kwargs)[source]

Bases: proto.message.Message

Contains text input to be synthesized. Either text or ssml must be supplied. Supplying both or neither returns [google.rpc.Code.INVALID_ARGUMENT][google.rpc.Code.INVALID_ARGUMENT]. The input size is limited to 5000 bytes.

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.

text

The raw text to be synthesized.

This field is a member of oneof input_source.

Type

str

ssml

The SSML document to be synthesized. The SSML document must be valid and well-formed. Otherwise the RPC will fail and return [google.rpc.Code.INVALID_ARGUMENT][google.rpc.Code.INVALID_ARGUMENT]. For more information, see SSML.

This field is a member of oneof input_source.

Type

str

class google.cloud.texttospeech_v1beta1.types.SynthesizeLongAudioMetadata(mapping=None, *, ignore_unknown_fields=False, **kwargs)[source]

Bases: proto.message.Message

Metadata for response returned by the SynthesizeLongAudio method.

start_time

Time when the request was received.

Type

google.protobuf.timestamp_pb2.Timestamp

last_update_time

Time of the most recent processing update.

Type

google.protobuf.timestamp_pb2.Timestamp

progress_percentage

The progress of the most recent processing update in percentage, ie. 70.0%.

Type

float

class google.cloud.texttospeech_v1beta1.types.SynthesizeLongAudioRequest(mapping=None, *, ignore_unknown_fields=False, **kwargs)[source]

Bases: proto.message.Message

The top-level message sent by the client for the SynthesizeLongAudio method.

parent

The resource states of the request in the form of projects/*/locations/*.

Type

str

input

Required. The Synthesizer requires either plain text or SSML as input. While Long Audio is in preview, SSML is temporarily unsupported.

Type

google.cloud.texttospeech_v1beta1.types.SynthesisInput

audio_config

Required. The configuration of the synthesized audio.

Type

google.cloud.texttospeech_v1beta1.types.AudioConfig

output_gcs_uri

Required. Specifies a Cloud Storage URI for the synthesis results. Must be specified in the format: gs://bucket_name/object_name, and the bucket must already exist.

Type

str

voice

Required. The desired voice of the synthesized audio.

Type

google.cloud.texttospeech_v1beta1.types.VoiceSelectionParams

class google.cloud.texttospeech_v1beta1.types.SynthesizeLongAudioResponse(mapping=None, *, ignore_unknown_fields=False, **kwargs)[source]

Bases: proto.message.Message

The message returned to the client by the SynthesizeLongAudio method.

class google.cloud.texttospeech_v1beta1.types.SynthesizeSpeechRequest(mapping=None, *, ignore_unknown_fields=False, **kwargs)[source]

Bases: proto.message.Message

The top-level message sent by the client for the SynthesizeSpeech method.

input

Required. The Synthesizer requires either plain text or SSML as input.

Type

google.cloud.texttospeech_v1beta1.types.SynthesisInput

voice

Required. The desired voice of the synthesized audio.

Type

google.cloud.texttospeech_v1beta1.types.VoiceSelectionParams

audio_config

Required. The configuration of the synthesized audio.

Type

google.cloud.texttospeech_v1beta1.types.AudioConfig

enable_time_pointing

Whether and what timepoints are returned in the response.

Type

MutableSequence[google.cloud.texttospeech_v1beta1.types.SynthesizeSpeechRequest.TimepointType]

class TimepointType(value)[source]

Bases: proto.enums.Enum

The type of timepoint information that is returned in the response.

Values:
TIMEPOINT_TYPE_UNSPECIFIED (0):

Not specified. No timepoint information will be returned.

SSML_MARK (1):

Timepoint information of <mark> tags in SSML input will be returned.

class google.cloud.texttospeech_v1beta1.types.SynthesizeSpeechResponse(mapping=None, *, ignore_unknown_fields=False, **kwargs)[source]

Bases: proto.message.Message

The message returned to the client by the SynthesizeSpeech method.

audio_content

The audio data bytes encoded as specified in the request, including the header for encodings that are wrapped in containers (e.g. MP3, OGG_OPUS). For LINEAR16 audio, we include the WAV header. Note: as with all bytes fields, protobuffers use a pure binary representation, whereas JSON representations use base64.

Type

bytes

timepoints

A link between a position in the original request input and a corresponding time in the output audio. It’s only supported via <mark> of SSML input.

Type

MutableSequence[google.cloud.texttospeech_v1beta1.types.Timepoint]

audio_config

The audio metadata of audio_content.

Type

google.cloud.texttospeech_v1beta1.types.AudioConfig

class google.cloud.texttospeech_v1beta1.types.Timepoint(mapping=None, *, ignore_unknown_fields=False, **kwargs)[source]

Bases: proto.message.Message

This contains a mapping between a certain point in the input text and a corresponding time in the output audio.

mark_name

Timepoint name as received from the client within <mark> tag.

Type

str

time_seconds

Time offset in seconds from the start of the synthesized audio.

Type

float

class google.cloud.texttospeech_v1beta1.types.Voice(mapping=None, *, ignore_unknown_fields=False, **kwargs)[source]

Bases: proto.message.Message

Description of a voice supported by the TTS service.

language_codes

The languages that this voice supports, expressed as BCP-47 language tags (e.g. “en-US”, “es-419”, “cmn-tw”).

Type

MutableSequence[str]

name

The name of this voice. Each distinct voice has a unique name.

Type

str

ssml_gender

The gender of this voice.

Type

google.cloud.texttospeech_v1beta1.types.SsmlVoiceGender

natural_sample_rate_hertz

The natural sample rate (in hertz) for this voice.

Type

int

class google.cloud.texttospeech_v1beta1.types.VoiceSelectionParams(mapping=None, *, ignore_unknown_fields=False, **kwargs)[source]

Bases: proto.message.Message

Description of which voice to use for a synthesis request.

language_code

Required. The language (and potentially also the region) of the voice expressed as a BCP-47 language tag, e.g. “en-US”. This should not include a script tag (e.g. use “cmn-cn” rather than “cmn-Hant-cn”), because the script will be inferred from the input provided in the SynthesisInput. The TTS service will use this parameter to help choose an appropriate voice. Note that the TTS service may choose a voice with a slightly different language code than the one selected; it may substitute a different region (e.g. using en-US rather than en-CA if there isn’t a Canadian voice available), or even a different language, e.g. using “nb” (Norwegian Bokmal) instead of “no” (Norwegian)”.

Type

str

name

The name of the voice. If not set, the service will choose a voice based on the other parameters such as language_code and gender.

Type

str

ssml_gender

The preferred gender of the voice. If not set, the service will choose a voice based on the other parameters such as language_code and name. Note that this is only a preference, not requirement; if a voice of the appropriate gender is not available, the synthesizer should substitute a voice with a different gender rather than failing the request.

Type

google.cloud.texttospeech_v1beta1.types.SsmlVoiceGender

custom_voice

The configuration for a custom voice. If [CustomVoiceParams.model] is set, the service will choose the custom voice matching the specified configuration.

Type

google.cloud.texttospeech_v1beta1.types.CustomVoiceParams