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 Speech v1 API¶

class google.cloud.speech_v1.types.CreateCustomClassRequest(mapping=None, *, ignore_unknown_fields=False, **kwargs)[source]¶

Bases: proto.message.Message

Message sent by the client for the CreateCustomClass method.

parent¶

Required. The parent resource where this custom class will be created. Format:

projects/{project}/locations/{location}/customClasses

Speech-to-Text supports three locations: global, us (US North America), and eu (Europe). If you are calling the speech.googleapis.com endpoint, use the global location. To specify a region, use a regional endpoint with matching us or eu location value.

Type: str

custom_class_id¶

Required. The ID to use for the custom class, which will become the final component of the custom class’ resource name.

This value should restrict to letters, numbers, and hyphens, with the first character a letter, the last a letter or a number, and be 4-63 characters.

Type: str

custom_class¶

Required. The custom class to create.

Type: google.cloud.speech_v1.types.CustomClass

class google.cloud.speech_v1.types.CreatePhraseSetRequest(mapping=None, *, ignore_unknown_fields=False, **kwargs)[source]¶

Bases: proto.message.Message

Message sent by the client for the CreatePhraseSet method.

parent¶

Required. The parent resource where this phrase set will be created. Format:

projects/{project}/locations/{location}

Speech-to-Text supports three locations: global, us (US North America), and eu (Europe). If you are calling the speech.googleapis.com endpoint, use the global location. To specify a region, use a regional endpoint with matching us or eu location value.

Type: str

phrase_set_id¶

Required. The ID to use for the phrase set, which will become the final component of the phrase set’s resource name.

This value should restrict to letters, numbers, and hyphens, with the first character a letter, the last a letter or a number, and be 4-63 characters.

Type: str

phrase_set¶

Required. The phrase set to create.

Type: google.cloud.speech_v1.types.PhraseSet

class google.cloud.speech_v1.types.CustomClass(mapping=None, *, ignore_unknown_fields=False, **kwargs)[source]¶

Bases: proto.message.Message

A set of words or phrases that represents a common concept likely to appear in your audio, for example a list of passenger ship names. CustomClass items can be substituted into placeholders that you set in PhraseSet phrases.

name¶

The resource name of the custom class.

Type: str

custom_class_id¶

If this custom class is a resource, the custom_class_id is the resource id of the CustomClass. Case sensitive.

Type: str

items¶

A collection of class items.

Type: MutableSequence[google.cloud.speech_v1.types.CustomClass.ClassItem]

class ClassItem(mapping=None, *, ignore_unknown_fields=False, **kwargs)[source]¶

Bases: proto.message.Message

An item of the class.

value¶

The class item’s value.

Type: str

class google.cloud.speech_v1.types.DeleteCustomClassRequest(mapping=None, *, ignore_unknown_fields=False, **kwargs)[source]¶

Bases: proto.message.Message

Message sent by the client for the DeleteCustomClass method.

name¶

Required. The name of the custom class to delete. Format:

projects/{project}/locations/{location}/customClasses/{custom_class}

Speech-to-Text supports three locations: global, us (US North America), and eu (Europe). If you are calling the speech.googleapis.com endpoint, use the global location. To specify a region, use a regional endpoint with matching us or eu location value.

Type: str

class google.cloud.speech_v1.types.DeletePhraseSetRequest(mapping=None, *, ignore_unknown_fields=False, **kwargs)[source]¶

Bases: proto.message.Message

Message sent by the client for the DeletePhraseSet method.

name¶

Required. The name of the phrase set to delete. Format:

projects/{project}/locations/{location}/phraseSets/{phrase_set}

Type: str

class google.cloud.speech_v1.types.GetCustomClassRequest(mapping=None, *, ignore_unknown_fields=False, **kwargs)[source]¶

Bases: proto.message.Message

Message sent by the client for the GetCustomClass method.

name¶

Required. The name of the custom class to retrieve. Format:

projects/{project}/locations/{location}/customClasses/{custom_class}

Type: str

class google.cloud.speech_v1.types.GetPhraseSetRequest(mapping=None, *, ignore_unknown_fields=False, **kwargs)[source]¶

Bases: proto.message.Message

Message sent by the client for the GetPhraseSet method.

name¶

Required. The name of the phrase set to retrieve. Format:

projects/{project}/locations/{location}/phraseSets/{phrase_set}

Speech-to-Text supports three locations: global, us (US North America), and eu (Europe). If you are calling the speech.googleapis.com endpoint, use the global location. To specify a region, use a regional endpoint with matching us or eu location value.

Type: str

class google.cloud.speech_v1.types.ListCustomClassesRequest(mapping=None, *, ignore_unknown_fields=False, **kwargs)[source]¶

Bases: proto.message.Message

Message sent by the client for the ListCustomClasses method.

parent¶

Required. The parent, which owns this collection of custom classes. Format:

projects/{project}/locations/{location}/customClasses

Speech-to-Text supports three locations: global, us (US North America), and eu (Europe). If you are calling the speech.googleapis.com endpoint, use the global location. To specify a region, use a regional endpoint with matching us or eu location value.

Type: str

page_size¶

The maximum number of custom classes to return. The service may return fewer than this value. If unspecified, at most 50 custom classes will be returned. The maximum value is 1000; values above 1000 will be coerced to 1000.

Type: int

page_token¶

A page token, received from a previous ListCustomClass call. Provide this to retrieve the subsequent page.

When paginating, all other parameters provided to ListCustomClass must match the call that provided the page token.

Type: str

class google.cloud.speech_v1.types.ListCustomClassesResponse(mapping=None, *, ignore_unknown_fields=False, **kwargs)[source]¶

Bases: proto.message.Message

Message returned to the client by the ListCustomClasses method.

custom_classes¶

The custom classes.

Type: MutableSequence[google.cloud.speech_v1.types.CustomClass]

next_page_token¶

A token, which can be sent as page_token to retrieve the next page. If this field is omitted, there are no subsequent pages.

Type: str

class google.cloud.speech_v1.types.ListPhraseSetRequest(mapping=None, *, ignore_unknown_fields=False, **kwargs)[source]¶

Bases: proto.message.Message

Message sent by the client for the ListPhraseSet method.

parent¶

Required. The parent, which owns this collection of phrase set. Format:

projects/{project}/locations/{location}

Speech-to-Text supports three locations: global, us (US North America), and eu (Europe). If you are calling the speech.googleapis.com endpoint, use the global location. To specify a region, use a regional endpoint with matching us or eu location value.

Type: str

page_size¶

The maximum number of phrase sets to return. The service may return fewer than this value. If unspecified, at most 50 phrase sets will be returned. The maximum value is 1000; values above 1000 will be coerced to 1000.

Type: int

page_token¶

A page token, received from a previous ListPhraseSet call. Provide this to retrieve the subsequent page.

When paginating, all other parameters provided to ListPhraseSet must match the call that provided the page token.

Type: str

class google.cloud.speech_v1.types.ListPhraseSetResponse(mapping=None, *, ignore_unknown_fields=False, **kwargs)[source]¶

Bases: proto.message.Message

Message returned to the client by the ListPhraseSet method.

phrase_sets¶

The phrase set.

Type: MutableSequence[google.cloud.speech_v1.types.PhraseSet]

next_page_token¶

A token, which can be sent as page_token to retrieve the next page. If this field is omitted, there are no subsequent pages.

Type: str

class google.cloud.speech_v1.types.LongRunningRecognizeMetadata(mapping=None, *, ignore_unknown_fields=False, **kwargs)[source]¶

Bases: proto.message.Message

Describes the progress of a long-running LongRunningRecognize call. It is included in the metadata field of the Operation returned by the GetOperation call of the google::longrunning::Operations service.

progress_percent¶

Approximate percentage of audio processed thus far. Guaranteed to be 100 when the audio is fully processed and the results are available.

Type: int

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

uri¶

Output only. The URI of the audio file being transcribed. Empty if the audio was sent as byte content.

Type: str

class google.cloud.speech_v1.types.LongRunningRecognizeRequest(mapping=None, *, ignore_unknown_fields=False, **kwargs)[source]¶

Bases: proto.message.Message

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

config¶

Required. Provides information to the recognizer that specifies how to process the request.

Type: google.cloud.speech_v1.types.RecognitionConfig

audio¶

Required. The audio data to be recognized.

Type: google.cloud.speech_v1.types.RecognitionAudio

output_config¶

Optional. Specifies an optional destination for the recognition results.

Type: google.cloud.speech_v1.types.TranscriptOutputConfig

class google.cloud.speech_v1.types.LongRunningRecognizeResponse(mapping=None, *, ignore_unknown_fields=False, **kwargs)[source]¶

Bases: proto.message.Message

The only message returned to the client by the LongRunningRecognize method. It contains the result as zero or more sequential SpeechRecognitionResult messages. It is included in the result.response field of the Operation returned by the GetOperation call of the google::longrunning::Operations service.

results¶

Sequential list of transcription results corresponding to sequential portions of audio.

Type: MutableSequence[google.cloud.speech_v1.types.SpeechRecognitionResult]

total_billed_time¶

When available, billed audio seconds for the corresponding request.

Type: google.protobuf.duration_pb2.Duration

output_config¶

Original output config if present in the request.

Type: google.cloud.speech_v1.types.TranscriptOutputConfig

output_error¶

If the transcript output fails this field contains the relevant error.

Type: google.rpc.status_pb2.Status

speech_adaptation_info¶

Provides information on speech adaptation behavior in response

Type: google.cloud.speech_v1.types.SpeechAdaptationInfo

request_id¶

The ID associated with the request. This is a unique ID specific only to the given request.

Type: int

class google.cloud.speech_v1.types.PhraseSet(mapping=None, *, ignore_unknown_fields=False, **kwargs)[source]¶

Bases: proto.message.Message

Provides “hints” to the speech recognizer to favor specific words and phrases in the results.

name¶

The resource name of the phrase set.

Type: str

phrases¶

A list of word and phrases.

Type: MutableSequence[google.cloud.speech_v1.types.PhraseSet.Phrase]

boost¶

Hint Boost. Positive value will increase the probability that a specific phrase will be recognized over other similar sounding phrases. The higher the boost, the higher the chance of false positive recognition as well. Negative boost values would correspond to anti-biasing. Anti-biasing is not enabled, so negative boost will simply be ignored. Though boost can accept a wide range of positive values, most use cases are best served with values between 0 (exclusive) and 20. We recommend using a binary search approach to finding the optimal value for your use case as well as adding phrases both with and without boost to your requests.

Type: float

class Phrase(mapping=None, *, ignore_unknown_fields=False, **kwargs)[source]¶

Bases: proto.message.Message

A phrases containing words and phrase “hints” so that the speech recognition is more likely to recognize them. This can be used to improve the accuracy for specific words and phrases, for example, if specific commands are typically spoken by the user. This can also be used to add additional words to the vocabulary of the recognizer. See usage limits.

List items can also include pre-built or custom classes containing groups of words that represent common concepts that occur in natural language. For example, rather than providing a phrase hint for every month of the year (e.g. “i was born in january”, “i was born in febuary”, …), use the pre-built $MONTH class improves the likelihood of correctly transcribing audio that includes months (e.g. “i was born in $month”). To refer to pre-built classes, use the class’ symbol prepended with $ e.g. $MONTH. To refer to custom classes that were defined inline in the request, set the class’s custom_class_id to a string unique to all class resources and inline classes. Then use the class’ id wrapped in ${...} e.g. “${my-months}”. To refer to custom classes resources, use the class’ id wrapped in ${} (e.g. ${my-months}).

Speech-to-Text supports three locations: global, us (US North America), and eu (Europe). If you are calling the speech.googleapis.com endpoint, use the global location. To specify a region, use a regional endpoint with matching us or eu location value.

value¶

The phrase itself.

Type: str

boost¶

Hint Boost. Overrides the boost set at the phrase set level. Positive value will increase the probability that a specific phrase will be recognized over other similar sounding phrases. The higher the boost, the higher the chance of false positive recognition as well. Negative boost will simply be ignored. Though boost can accept a wide range of positive values, most use cases are best served with values between 0 and 20. We recommend using a binary search approach to finding the optimal value for your use case as well as adding phrases both with and without boost to your requests.

Type: float

class google.cloud.speech_v1.types.RecognitionAudio(mapping=None, *, ignore_unknown_fields=False, **kwargs)[source]¶

Bases: proto.message.Message

Contains audio data in the encoding specified in the RecognitionConfig. Either content or uri must be supplied. Supplying both or neither returns [google.rpc.Code.INVALID_ARGUMENT][google.rpc.Code.INVALID_ARGUMENT]. See content limits.

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.

content¶

The audio data bytes encoded as specified in RecognitionConfig. Note: as with all bytes fields, proto buffers use a pure binary representation, whereas JSON representations use base64.

This field is a member of oneof audio_source.

Type: bytes

uri¶

URI that points to a file that contains audio data bytes as specified in RecognitionConfig. The file must not be compressed (for example, gzip). Currently, only Google Cloud Storage URIs are supported, which must be specified in the following format: gs://bucket_name/object_name (other URI formats return [google.rpc.Code.INVALID_ARGUMENT][google.rpc.Code.INVALID_ARGUMENT]). For more information, see Request URIs.

This field is a member of oneof audio_source.

Type: str

class google.cloud.speech_v1.types.RecognitionConfig(mapping=None, *, ignore_unknown_fields=False, **kwargs)[source]¶

Bases: proto.message.Message

Provides information to the recognizer that specifies how to process the request.

encoding¶

Encoding of audio data sent in all RecognitionAudio messages. This field is optional for FLAC and WAV audio files and required for all other audio formats. For details, see [AudioEncoding][google.cloud.speech.v1.RecognitionConfig.AudioEncoding].

Type: google.cloud.speech_v1.types.RecognitionConfig.AudioEncoding

sample_rate_hertz¶

Sample rate in Hertz of the audio data sent in all RecognitionAudio messages. 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, see [AudioEncoding][google.cloud.speech.v1.RecognitionConfig.AudioEncoding].

Type: int

audio_channel_count¶

The number of channels in the input audio data. ONLY set this for MULTI-CHANNEL recognition. Valid values for LINEAR16, OGG_OPUS and FLAC are 1-8. Valid value for MULAW, AMR, AMR_WB and SPEEX_WITH_HEADER_BYTE is only 1. If 0 or omitted, defaults to one channel (mono). Note: We only recognize the first channel by default. To perform independent recognition on each channel set enable_separate_recognition_per_channel to ‘true’.

Type: int

enable_separate_recognition_per_channel¶

This needs to be set to true explicitly and audio_channel_count > 1 to get each channel recognized separately. The recognition result will contain a channel_tag field 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_count multiplied by the length of the audio.

Type: bool

language_code¶

Required. The language of the supplied audio as a BCP-47 language tag. Example: “en-US”. See Language Support for a list of the currently supported language codes.

Type: str

alternative_language_codes¶

A list of up to 3 additional BCP-47 language tags, listing possible alternative languages of the supplied audio. See Language Support for a list of the currently supported language codes. If alternative languages are listed, recognition result will contain recognition in the most likely language detected including the main language_code. The recognition result will include the language tag of the language detected in the audio. Note: This feature is only supported for Voice Command and Voice Search use cases and performance may vary for other use cases (e.g., phone call transcription).

Type: MutableSequence[str]

max_alternatives¶

Maximum number of recognition hypotheses to be returned. Specifically, the maximum number of SpeechRecognitionAlternative messages within each SpeechRecognitionResult. The server may return fewer than max_alternatives. Valid values are 0-30. A value of 0 or 1 will return a maximum of one. If omitted, will return a maximum of one.

Type: int

profanity_filter¶

If set to true, 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 to false or omitted, profanities won’t be filtered out.

Type: bool

adaptation¶

Speech adaptation configuration improves the accuracy of speech recognition. For more information, see the speech adaptation documentation. When speech adaptation is set it supersedes the speech_contexts field.

Type: google.cloud.speech_v1.types.SpeechAdaptation

transcript_normalization¶

Optional. Use transcription normalization to automatically replace parts of the transcript with phrases of your choosing. For StreamingRecognize, this normalization only applies to stable partial transcripts (stability > 0.8) and final transcripts.

Type: google.cloud.speech_v1.types.TranscriptNormalization

speech_contexts¶

Array of [SpeechContext][google.cloud.speech.v1.SpeechContext]. A means to provide context to assist the speech recognition. For more information, see speech adaptation.

Type: MutableSequence[google.cloud.speech_v1.types.SpeechContext]

enable_word_time_offsets¶

If true, the top result includes a list of words and the start and end time offsets (timestamps) for those words. If false, no word-level time offset information is returned. The default is false.

Type: bool

enable_word_confidence¶

If true, the top result includes a list of words and the confidence for those words. If false, no word-level confidence information is returned. The default is false.

Type: bool

enable_automatic_punctuation¶

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.

Type: bool

enable_spoken_punctuation¶

The spoken punctuation behavior for the call If not set, uses default behavior based on model of choice e.g. command_and_search will enable spoken punctuation by default If ‘true’, replaces spoken punctuation with the corresponding symbols in the request. For example, “how are you question mark” becomes “how are you?”. See https://cloud.google.com/speech-to-text/docs/spoken-punctuation for support. If ‘false’, spoken punctuation is not replaced.

Type: google.protobuf.wrappers_pb2.BoolValue

enable_spoken_emojis¶

The spoken emoji behavior for the call If not set, uses default behavior based on model of choice If ‘true’, adds spoken emoji formatting for the request. This will replace spoken emojis with the corresponding Unicode symbols in the final transcript. If ‘false’, spoken emojis are not replaced.

Type: google.protobuf.wrappers_pb2.BoolValue

diarization_config¶

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.

Type: google.cloud.speech_v1.types.SpeakerDiarizationConfig

metadata¶

Metadata regarding this request.

Type: google.cloud.speech_v1.types.RecognitionMetadata

model¶

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.

Model	Description
`latest_long`	Best for long form content like media or conversation.
`latest_short`	Best for short form content like commands or single shot directed speech.
`command_and_search`	Best for short queries such as voice commands or voice search.
`phone_call`	Best for audio that originated from a phone call (typically recorded at an 8khz sampling rate).
`video`	Best for audio that originated from video or includes multiple speakers. Ideally the audio is recorded at a 16khz or greater sampling rate. This is a premium model that costs more than the standard rate.
`default`	Best for audio that is not one of the specific audio models. For example, long-form audio. Ideally the audio is high-fidelity, recorded at a 16khz or greater sampling rate.
`medical_conversation`	Best for audio that originated from a conversation between a medical provider and patient.
`medical_dictation`	Best for audio that originated from dictation notes by a medical provider.

Type: str

use_enhanced¶

Set to true to use an enhanced model for speech recognition. If use_enhanced is set to true and the model field is not set, then an appropriate enhanced model is chosen if an enhanced model exists for the audio.

If use_enhanced is 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.

Type: bool

class AudioEncoding(value)[source]¶

Bases: proto.enums.Enum

The encoding of the audio data sent in the request.

All encodings support only 1 channel (mono) audio, unless the audio_channel_count and enable_separate_recognition_per_channel fields are set.

For best results, the audio source should be captured and transmitted using a lossless encoding (FLAC or LINEAR16). The accuracy of the speech recognition can be reduced if lossy codecs are used to capture or transmit audio, particularly if background noise is present. Lossy codecs include MULAW, AMR, AMR_WB, OGG_OPUS, SPEEX_WITH_HEADER_BYTE, MP3, and WEBM_OPUS.

The FLAC and WAV audio file formats include a header that describes the included audio content. You can request recognition for WAV files that contain either LINEAR16 or MULAW encoded audio. If you send FLAC or WAV audio file format in your request, you do not need to specify an AudioEncoding; the audio encoding format is determined from the file header. If you specify an AudioEncoding when you send send FLAC or WAV audio, the encoding configuration must match the encoding described in the audio header; otherwise the request returns an [google.rpc.Code.INVALID_ARGUMENT][google.rpc.Code.INVALID_ARGUMENT] error code.

Values:

ENCODING_UNSPECIFIED (0):: Not specified.
LINEAR16 (1):: Uncompressed 16-bit signed little-endian samples (Linear PCM).
FLAC (2):: 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. FLAC stream encoding supports 16-bit and 24-bit samples, however, not all fields in STREAMINFO are supported.
MULAW (3):: 8-bit samples that compand 14-bit audio samples using G.711 PCMU/mu-law.
AMR (4):: Adaptive Multi-Rate Narrowband codec. sample_rate_hertz must be 8000.
AMR_WB (5):: Adaptive Multi-Rate Wideband codec. sample_rate_hertz must be 16000.
OGG_OPUS (6):: Opus encoded audio frames in Ogg container (OggOpus). sample_rate_hertz must be one of 8000, 12000, 16000, 24000, or 48000.
SPEEX_WITH_HEADER_BYTE (7):: Although the use of lossy encodings is not recommended, if a very low bitrate encoding is required, OGG_OPUS is highly preferred over Speex encoding. The Speex encoding supported by Cloud Speech API has a header byte in each block, as in MIME type audio/x-speex-with-header-byte. It is a variant of the RTP Speex encoding defined in RFC 5574. The stream is a sequence of blocks, one block per RTP packet. Each block starts with a byte containing the length of the block, in bytes, followed by one or more frames of Speex data, padded to an integral number of bytes (octets) as specified in RFC 5574. In other words, each RTP header is replaced with a single byte containing the block length. Only Speex wideband is supported. sample_rate_hertz must be 16000.
MP3 (8):: MP3 audio. MP3 encoding is a Beta feature and only available in v1p1beta1. 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.
WEBM_OPUS (9):: Opus encoded audio frames in WebM container (OggOpus). sample_rate_hertz must be one of 8000, 12000, 16000, 24000, or 48000.

class google.cloud.speech_v1.types.RecognitionMetadata(mapping=None, *, ignore_unknown_fields=False, **kwargs)[source]¶

Bases: proto.message.Message

Description of audio data to be recognized.

interaction_type¶

The use case most closely describing the audio content to be recognized.

Type: google.cloud.speech_v1.types.RecognitionMetadata.InteractionType

industry_naics_code_of_audio¶

The industry vertical to which this speech recognition request most closely applies. This is most indicative of the topics contained in the audio. Use the 6-digit NAICS code to identify the industry vertical - see https://www.naics.com/search/.

Type: int

microphone_distance¶

The audio type that most closely describes the audio being recognized.

Type: google.cloud.speech_v1.types.RecognitionMetadata.MicrophoneDistance

original_media_type¶

The original media the speech was recorded on.

Type: google.cloud.speech_v1.types.RecognitionMetadata.OriginalMediaType

recording_device_type¶

The type of device the speech was recorded with.

Type: google.cloud.speech_v1.types.RecognitionMetadata.RecordingDeviceType

recording_device_name¶

The device used to make the recording. Examples ‘Nexus 5X’ or ‘Polycom SoundStation IP 6000’ or ‘POTS’ or ‘VoIP’ or ‘Cardioid Microphone’.

Type: str

original_mime_type¶

Mime type of the original audio file. For example audio/m4a, audio/x-alaw-basic, audio/mp3, audio/3gpp. A list of possible audio mime types is maintained at http://www.iana.org/assignments/media-types/media-types.xhtml#audio

Type: str

audio_topic¶

Description of the content. Eg. “Recordings of federal supreme court hearings from 2012”.

Type: str

class InteractionType(value)[source]¶

Bases: proto.enums.Enum

Use case categories that the audio recognition request can be described by.

Values:

INTERACTION_TYPE_UNSPECIFIED (0):: Use case is either unknown or is something other than one of the other values below.
DISCUSSION (1):: Multiple people in a conversation or discussion. For example in a meeting with two or more people actively participating. Typically all the primary people speaking would be in the same room (if not, see PHONE_CALL)
PRESENTATION (2):: One or more persons lecturing or presenting to others, mostly uninterrupted.
PHONE_CALL (3):: A phone-call or video-conference in which two or more people, who are not in the same room, are actively participating.
VOICEMAIL (4):: A recorded message intended for another person to listen to.
PROFESSIONALLY_PRODUCED (5):: Professionally produced audio (eg. TV Show, Podcast).
VOICE_SEARCH (6):: Transcribe spoken questions and queries into text.
VOICE_COMMAND (7):: Transcribe voice commands, such as for controlling a device.
DICTATION (8):: Transcribe speech to text to create a written document, such as a text-message, email or report.

class MicrophoneDistance(value)[source]¶

Bases: proto.enums.Enum

Enumerates the types of capture settings describing an audio file.

Values:

MICROPHONE_DISTANCE_UNSPECIFIED (0):: Audio type is not known.
NEARFIELD (1):: The audio was captured from a closely placed microphone. Eg. phone, dictaphone, or handheld microphone. Generally if there speaker is within 1 meter of the microphone.
MIDFIELD (2):: The speaker if within 3 meters of the microphone.
FARFIELD (3):: The speaker is more than 3 meters away from the microphone.

class OriginalMediaType(value)[source]¶

Bases: proto.enums.Enum

The original media the speech was recorded on.

Values:

ORIGINAL_MEDIA_TYPE_UNSPECIFIED (0):: Unknown original media type.
AUDIO (1):: The speech data is an audio recording.
VIDEO (2):: The speech data originally recorded on a video.

class RecordingDeviceType(value)[source]¶

Bases: proto.enums.Enum

The type of device the speech was recorded with.

Values:

RECORDING_DEVICE_TYPE_UNSPECIFIED (0):: The recording device is unknown.
SMARTPHONE (1):: Speech was recorded on a smartphone.
PC (2):: Speech was recorded using a personal computer or tablet.
PHONE_LINE (3):: Speech was recorded over a phone line.
VEHICLE (4):: Speech was recorded in a vehicle.
OTHER_OUTDOOR_DEVICE (5):: Speech was recorded outdoors.
OTHER_INDOOR_DEVICE (6):: Speech was recorded indoors.

class google.cloud.speech_v1.types.RecognizeRequest(mapping=None, *, ignore_unknown_fields=False, **kwargs)[source]¶

Bases: proto.message.Message

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

config¶

Required. Provides information to the recognizer that specifies how to process the request.

Type: google.cloud.speech_v1.types.RecognitionConfig

audio¶

Required. The audio data to be recognized.

Type: google.cloud.speech_v1.types.RecognitionAudio

class google.cloud.speech_v1.types.RecognizeResponse(mapping=None, *, ignore_unknown_fields=False, **kwargs)[source]¶

Bases: proto.message.Message

The only message returned to the client by the Recognize method. It contains the result as zero or more sequential SpeechRecognitionResult messages.

results¶

Sequential list of transcription results corresponding to sequential portions of audio.

Type: MutableSequence[google.cloud.speech_v1.types.SpeechRecognitionResult]

total_billed_time¶

When available, billed audio seconds for the corresponding request.

Type: google.protobuf.duration_pb2.Duration

speech_adaptation_info¶

Provides information on adaptation behavior in response

Type: google.cloud.speech_v1.types.SpeechAdaptationInfo

request_id¶

The ID associated with the request. This is a unique ID specific only to the given request.

Type: int

class google.cloud.speech_v1.types.SpeakerDiarizationConfig(mapping=None, *, ignore_unknown_fields=False, **kwargs)[source]¶

Bases: proto.message.Message

Config to enable speaker diarization.

enable_speaker_diarization¶

If ‘true’, enables speaker detection for each recognized word in the top alternative of the recognition result using a speaker_label provided in the WordInfo.

Type: bool

min_speaker_count¶

Minimum number of speakers in the conversation. This range gives you more flexibility by allowing the system to automatically determine the correct number of speakers. If not set, the default value is 2.

Type: int

max_speaker_count¶

Maximum number of speakers in the conversation. This range gives you more flexibility by allowing the system to automatically determine the correct number of speakers. If not set, the default value is 6.

Type: int

speaker_tag¶

Output only. Unused.

Type: int

class google.cloud.speech_v1.types.SpeechAdaptation(mapping=None, *, ignore_unknown_fields=False, **kwargs)[source]¶

Bases: proto.message.Message

Speech adaptation configuration.

phrase_sets¶

A collection of phrase sets. To specify the hints inline, leave the phrase set’s name blank and fill in the rest of its fields. Any phrase set can use any custom class.

Type: MutableSequence[google.cloud.speech_v1.types.PhraseSet]

phrase_set_references¶

A collection of phrase set resource names to use.

Type: MutableSequence[str]

custom_classes¶

A collection of custom classes. To specify the classes inline, leave the class’ name blank and fill in the rest of its fields, giving it a unique custom_class_id. Refer to the inline defined class in phrase hints by its custom_class_id.

Type: MutableSequence[google.cloud.speech_v1.types.CustomClass]

abnf_grammar¶

Augmented Backus-Naur form (ABNF) is a standardized grammar notation comprised by a set of derivation rules. See specifications: https://www.w3.org/TR/speech-grammar

Type: google.cloud.speech_v1.types.SpeechAdaptation.ABNFGrammar

class ABNFGrammar(mapping=None, *, ignore_unknown_fields=False, **kwargs)[source]¶

Bases: proto.message.Message

abnf_strings¶

All declarations and rules of an ABNF grammar broken up into multiple strings that will end up concatenated.

Type: MutableSequence[str]

class google.cloud.speech_v1.types.SpeechAdaptationInfo(mapping=None, *, ignore_unknown_fields=False, **kwargs)[source]¶

Bases: proto.message.Message

Information on speech adaptation use in results

adaptation_timeout¶

Whether there was a timeout when applying speech adaptation. If true, adaptation had no effect in the response transcript.

Type: bool

timeout_message¶

If set, returns a message specifying which part of the speech adaptation request timed out.

Type: str

class google.cloud.speech_v1.types.SpeechContext(mapping=None, *, ignore_unknown_fields=False, **kwargs)[source]¶

Bases: proto.message.Message

Provides “hints” to the speech recognizer to favor specific words and phrases in the results.

phrases¶

A list of strings containing words and phrases “hints” so that the speech recognition is more likely to recognize them. This can be used to improve the accuracy for specific words and phrases, for example, if specific commands are typically spoken by the user. This can also be used to add additional words to the vocabulary of the recognizer. See usage limits.

List items can also be set to classes for groups of words that represent common concepts that occur in natural language. For example, rather than providing phrase hints for every month of the year, using the $MONTH class improves the likelihood of correctly transcribing audio that includes months.

Type: MutableSequence[str]

boost¶

Hint Boost. Positive value will increase the probability that a specific phrase will be recognized over other similar sounding phrases. The higher the boost, the higher the chance of false positive recognition as well. Negative boost values would correspond to anti-biasing. Anti-biasing is not enabled, so negative boost will simply be ignored. Though boost can accept a wide range of positive values, most use cases are best served with values between 0 and 20. We recommend using a binary search approach to finding the optimal value for your use case.

Type: float

class google.cloud.speech_v1.types.SpeechRecognitionAlternative(mapping=None, *, ignore_unknown_fields=False, **kwargs)[source]¶

Bases: proto.message.Message

Alternative hypotheses (a.k.a. n-best list).

transcript¶

Transcript text representing the words that the user spoke. In languages that use spaces to separate words, the transcript might have a leading space if it isn’t the first result. You can concatenate each result to obtain the full transcript without using a separator.

Type: str

confidence¶

The confidence estimate between 0.0 and 1.0. A higher number indicates an estimated greater likelihood that the recognized words are correct. This field is set only for the top alternative of a non-streaming result or, of a streaming result where is_final=true. This field is not guaranteed to be accurate and users should not rely on it to be always provided. The default of 0.0 is a sentinel value indicating confidence was not set.

Type: float

words¶

A list of word-specific information for each recognized word. Note: When enable_speaker_diarization is true, you will see all the words from the beginning of the audio.

Type: MutableSequence[google.cloud.speech_v1.types.WordInfo]

class google.cloud.speech_v1.types.SpeechRecognitionResult(mapping=None, *, ignore_unknown_fields=False, **kwargs)[source]¶

Bases: proto.message.Message

A speech recognition result corresponding to a portion of the audio.

alternatives¶

May contain one or more recognition hypotheses (up to the maximum specified in max_alternatives). These alternatives are ordered in terms of accuracy, with the top (first) alternative being the most probable, as ranked by the recognizer.

Type: MutableSequence[google.cloud.speech_v1.types.SpeechRecognitionAlternative]

channel_tag¶

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’.

Type: int

result_end_time¶

Time offset of the end of this result relative to the beginning of the audio.

Type: google.protobuf.duration_pb2.Duration

language_code¶

Output only. The BCP-47 language tag of the language in this result. This language code was detected to have the most likelihood of being spoken in the audio.

Type: str

class google.cloud.speech_v1.types.StreamingRecognitionConfig(mapping=None, *, ignore_unknown_fields=False, **kwargs)[source]¶

Bases: proto.message.Message

Provides information to the recognizer that specifies how to process the request.

config¶

Required. Provides information to the recognizer that specifies how to process the request.

Type: google.cloud.speech_v1.types.RecognitionConfig

single_utterance¶

If false or omitted, the recognizer will perform continuous recognition (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 StreamingRecognitionResults with the is_final flag set to true.

If true, the recognizer 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 recognition. It will return no more than one StreamingRecognitionResult with the is_final flag set to true.

The single_utterance field can only be used with specified models, otherwise an error is thrown. The model field in [RecognitionConfig][] must be set to:

command_and_search
phone_call AND additional field useEnhanced=true
The model field is left undefined. In this case the API auto-selects a model based on any other parameters that you set in RecognitionConfig.

Type: bool

interim_results¶

If true, interim results (tentative hypotheses) may be returned as they become available (these interim results are indicated with the is_final=false flag). If false or omitted, only is_final=true result(s) are returned.

Type: bool

enable_voice_activity_events¶

If true, responses with voice activity speech events will be returned as they are detected.

Type: bool

voice_activity_timeout¶

If set, the server will automatically close the stream after the specified duration has elapsed after the last VOICE_ACTIVITY speech event has been sent. The field voice_activity_events must also be set to true.

Type: google.cloud.speech_v1.types.StreamingRecognitionConfig.VoiceActivityTimeout

class VoiceActivityTimeout(mapping=None, *, ignore_unknown_fields=False, **kwargs)[source]¶

Bases: proto.message.Message

Events that a timeout can be set on for voice activity.

speech_start_timeout¶

Duration to timeout the stream if no speech begins.

Type: google.protobuf.duration_pb2.Duration

speech_end_timeout¶

Duration to timeout the stream after speech ends.

Type: google.protobuf.duration_pb2.Duration

class google.cloud.speech_v1.types.StreamingRecognitionResult(mapping=None, *, ignore_unknown_fields=False, **kwargs)[source]¶

Bases: proto.message.Message

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

alternatives¶

May contain one or more recognition hypotheses (up to the maximum specified in max_alternatives). These alternatives are ordered in terms of accuracy, with the top (first) alternative being the most probable, as ranked by the recognizer.

Type: MutableSequence[google.cloud.speech_v1.types.SpeechRecognitionAlternative]

is_final¶

If false, this StreamingRecognitionResult represents an interim result that may change. If true, this is the final time the speech service will return this particular StreamingRecognitionResult, the recognizer will not return any further hypotheses for this portion of the transcript and corresponding audio.

Type: bool

stability¶

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 indicating stability was not set.

Type: float

result_end_time¶

Time offset of the end of this result relative to the beginning of the audio.

Type: google.protobuf.duration_pb2.Duration

channel_tag¶

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’.

Type: int

language_code¶

Output only. The BCP-47 language tag of the language in this result. This language code was detected to have the most likelihood of being spoken in the audio.

Type: str

class google.cloud.speech_v1.types.StreamingRecognizeRequest(mapping=None, *, ignore_unknown_fields=False, **kwargs)[source]¶

Bases: proto.message.Message

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

This field is a member of oneof streaming_request.

Type: google.cloud.speech_v1.types.StreamingRecognitionConfig

audio_content¶

The audio data to be recognized. Sequential chunks of audio data are sent in sequential StreamingRecognizeRequest messages. The first StreamingRecognizeRequest message must not contain audio_content data and all subsequent StreamingRecognizeRequest messages must contain audio_content data. The audio bytes must be encoded as specified in RecognitionConfig. Note: as with all bytes fields, proto buffers use a pure binary representation (not base64). See content limits.

This field is a member of oneof streaming_request.

Type: bytes

class google.cloud.speech_v1.types.StreamingRecognizeResponse(mapping=None, *, ignore_unknown_fields=False, **kwargs)[source]¶

Bases: proto.message.Message

StreamingRecognizeResponse is the only message returned to the client by StreamingRecognize. A series of zero or more StreamingRecognizeResponse messages are streamed back to the client. If there is no recognizable audio, and single_utterance is set to false, then no messages are streamed back to the client.

Here’s an example of a series of StreamingRecognizeResponses that might be returned while processing audio:

results { alternatives { transcript: “tube” } stability: 0.01 }
results { alternatives { transcript: “to be a” } stability: 0.01 }
results { alternatives { transcript: “to be” } stability: 0.9 } results { alternatives { transcript: ” or not to be” } stability: 0.01 }
results { alternatives { transcript: “to be or not to be” confidence: 0.92 } alternatives { transcript: “to bee or not to bee” } is_final: true }
results { alternatives { transcript: ” that’s” } stability: 0.01 }
results { alternatives { transcript: ” that is” } stability: 0.9 } results { alternatives { transcript: ” the question” } stability: 0.01 }
results { alternatives { transcript: ” that is the question” confidence: 0.98 } alternatives { transcript: ” that was the question” } is_final: true }

Notes:

Only two of the above responses #4 and #7 contain final results; they are indicated by is_final: true. Concatenating these together generates the full transcript: “to be or not to be that is the question”.
The others contain interim results. #3 and #6 contain two interim results: the first portion has a high stability and is less likely to change; the second portion has a low stability and is very likely to change. A UI designer might choose to show only high stability results.
The specific stability and confidence values shown above are only for illustrative purposes. Actual values may vary.
In each response, only one of these fields will be set: error, speech_event_type, or one or more (repeated) results.

error¶

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

Type: google.rpc.status_pb2.Status

results¶

This repeated list contains zero or more results that correspond to consecutive portions of the audio currently being processed. It contains zero or one is_final=true result (the newly settled portion), followed by zero or more is_final=false results (the interim results).

Type: MutableSequence[google.cloud.speech_v1.types.StreamingRecognitionResult]

speech_event_type¶

Indicates the type of speech event.

Type: google.cloud.speech_v1.types.StreamingRecognizeResponse.SpeechEventType

speech_event_time¶

Time offset between the beginning of the audio and event emission.

Type: google.protobuf.duration_pb2.Duration

total_billed_time¶

When available, billed audio seconds for the stream. Set only if this is the last response in the stream.

Type: google.protobuf.duration_pb2.Duration

speech_adaptation_info¶

Provides information on adaptation behavior in response

Type: google.cloud.speech_v1.types.SpeechAdaptationInfo

request_id¶

The ID associated with the request. This is a unique ID specific only to the given request.

Type: int

class SpeechEventType(value)[source]¶

Bases: proto.enums.Enum

Indicates the type of speech event.

Values:

SPEECH_EVENT_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). The client should stop sending additional audio data, half-close the gRPC connection, and wait for any additional results until the server closes the gRPC connection. This event is only sent if single_utterance was set to true, and is not used otherwise.
SPEECH_ACTIVITY_BEGIN (2):: This event indicates that the server has detected the beginning of human voice activity in the stream. This event can be returned multiple times if speech starts and stops repeatedly throughout the stream. This event is only sent if voice_activity_events is set to true.
SPEECH_ACTIVITY_END (3):: This event indicates that the server has detected the end of human voice activity in the stream. This event can be returned multiple times if speech starts and stops repeatedly throughout the stream. This event is only sent if voice_activity_events is set to true.
SPEECH_ACTIVITY_TIMEOUT (4):: This event indicates that the user-set timeout for speech activity begin or end has exceeded. Upon receiving this event, the client is expected to send a half close. Further audio will not be processed.

class google.cloud.speech_v1.types.TranscriptNormalization(mapping=None, *, ignore_unknown_fields=False, **kwargs)[source]¶

Bases: proto.message.Message

Transcription normalization configuration. Use transcription normalization to automatically replace parts of the transcript with phrases of your choosing. For StreamingRecognize, this normalization only applies to stable partial transcripts (stability > 0.8) and final transcripts.

entries¶

A list of replacement entries. We will perform replacement with one entry at a time. For example, the second entry in [“cat” => “dog”, “mountain cat” => “mountain dog”] will never be applied because we will always process the first entry before it. At most 100 entries.

Type: MutableSequence[google.cloud.speech_v1.types.TranscriptNormalization.Entry]

class Entry(mapping=None, *, ignore_unknown_fields=False, **kwargs)[source]¶

Bases: proto.message.Message

A single replacement configuration.

search¶

What to replace. Max length is 100 characters.

Type: str

replace¶

What to replace with. Max length is 100 characters.

Type: str

case_sensitive¶

Whether the search is case sensitive.

Type: bool

class google.cloud.speech_v1.types.TranscriptOutputConfig(mapping=None, *, ignore_unknown_fields=False, **kwargs)[source]¶

Bases: proto.message.Message

Specifies an optional destination for the recognition results.

gcs_uri¶

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

This field is a member of oneof output_type.

Type: str

class google.cloud.speech_v1.types.UpdateCustomClassRequest(mapping=None, *, ignore_unknown_fields=False, **kwargs)[source]¶

Bases: proto.message.Message

Message sent by the client for the UpdateCustomClass method.

custom_class¶

Required. The custom class to update.

The custom class’s name field is used to identify the custom class to be updated. Format:

projects/{project}/locations/{location}/customClasses/{custom_class}

Speech-to-Text supports three locations: global, us (US North America), and eu (Europe). If you are calling the speech.googleapis.com endpoint, use the global location. To specify a region, use a regional endpoint with matching us or eu location value.

Type: google.cloud.speech_v1.types.CustomClass

update_mask¶

The list of fields to be updated.

Type: google.protobuf.field_mask_pb2.FieldMask

class google.cloud.speech_v1.types.UpdatePhraseSetRequest(mapping=None, *, ignore_unknown_fields=False, **kwargs)[source]¶

Bases: proto.message.Message

Message sent by the client for the UpdatePhraseSet method.

phrase_set¶

Required. The phrase set to update.

The phrase set’s name field is used to identify the set to be updated. Format:

projects/{project}/locations/{location}/phraseSets/{phrase_set}

Speech-to-Text supports three locations: global, us (US North America), and eu (Europe). If you are calling the speech.googleapis.com endpoint, use the global location. To specify a region, use a regional endpoint with matching us or eu location value.

Type: google.cloud.speech_v1.types.PhraseSet

update_mask¶

The list of fields to be updated.

Type: google.protobuf.field_mask_pb2.FieldMask

class google.cloud.speech_v1.types.WordInfo(mapping=None, *, ignore_unknown_fields=False, **kwargs)[source]¶

Bases: proto.message.Message

Word-specific information for recognized words.

start_time¶

Time offset relative to the beginning of the audio, and corresponding to the start of the spoken word. This field is only set if enable_word_time_offsets=true and only in the top hypothesis. This is an experimental feature and the accuracy of the time offset can vary.

Type: google.protobuf.duration_pb2.Duration

end_time¶

Time offset relative to the beginning of the audio, and corresponding to the end of the spoken word. This field is only set if enable_word_time_offsets=true and only in the top hypothesis. This is an experimental feature and the accuracy of the time offset can vary.

Type: google.protobuf.duration_pb2.Duration

word¶

The word corresponding to this set of information.

Type: str

confidence¶

The confidence estimate between 0.0 and 1.0. A higher number indicates an estimated greater likelihood that the recognized words are correct. This field is set only for the top alternative of a non-streaming result or, of a streaming result where is_final=true. This field is not guaranteed to be accurate and users should not rely on it to be always provided. The default of 0.0 is a sentinel value indicating confidence was not set.

Type: float

speaker_tag¶

Output only. A distinct integer value is assigned for every speaker within the audio. This field specifies which one of those speakers was detected to have spoken this word. Value ranges from ‘1’ to diarization_speaker_count. speaker_tag is set if enable_speaker_diarization = ‘true’ and only for the top alternative. Note: Use speaker_label instead.

Type: int

speaker_label¶

Output only. A label value assigned for every unique speaker within the audio. This field specifies which speaker was detected to have spoken this word. For some models, like medical_conversation this can be actual speaker role, for example “patient” or “provider”, but generally this would be a number identifying a speaker. This field is only set if enable_speaker_diarization = ‘true’ and only for the top alternative.

Type: str

Types for Google Cloud Speech v1 API¶

google-cloud-speech

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