- JSON representation
- AudioEncoding
- SpeechAdaptation
- ABNFGrammar
- TranscriptNormalization
- Entry
- SpeechContext
- SpeakerDiarizationConfig
- RecognitionMetadata
- InteractionType
- MicrophoneDistance
- OriginalMediaType
- RecordingDeviceType
Provides information to the recognizer that specifies how to process the request.
JSON representation |
---|
{ "encoding": enum ( |
Fields | |||||||||||||||||||
---|---|---|---|---|---|---|---|---|---|---|---|---|---|---|---|---|---|---|---|
encoding |
Encoding of audio data sent in all |
||||||||||||||||||
sampleRateHertz |
Sample rate in Hertz of the audio data sent in all |
||||||||||||||||||
audioChannelCount |
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 |
||||||||||||||||||
enableSeparateRecognitionPerChannel |
This needs to be set to |
||||||||||||||||||
languageCode |
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. |
||||||||||||||||||
alternativeLanguageCodes[] |
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 languageCode. 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). |
||||||||||||||||||
maxAlternatives |
Maximum number of recognition hypotheses to be returned. Specifically, the maximum number of |
||||||||||||||||||
profanityFilter |
If set to |
||||||||||||||||||
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 |
||||||||||||||||||
transcriptNormalization |
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. |
||||||||||||||||||
speechContexts[] |
Array of |
||||||||||||||||||
enableWordTimeOffsets |
If |
||||||||||||||||||
enableWordConfidence |
If |
||||||||||||||||||
enableAutomaticPunctuation |
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. |
||||||||||||||||||
enableSpokenPunctuation |
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. |
||||||||||||||||||
enableSpokenEmojis |
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. |
||||||||||||||||||
enableSpeakerDiarization |
If 'true', enables speaker detection for each recognized word in the top alternative of the recognition result using a speakerTag provided in the WordInfo. Note: Use diarizationConfig instead. |
||||||||||||||||||
diarizationSpeakerCount |
If set, specifies the estimated number of speakers in the conversation. Defaults to '2'. Ignored unless enableSpeakerDiarization is set to true. Note: Use diarizationConfig instead. |
||||||||||||||||||
diarizationConfig |
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. |
||||||||||||||||||
metadata |
Metadata regarding this request. |
||||||||||||||||||
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.
|
||||||||||||||||||
useEnhanced |
Set to true to use an enhanced model for speech recognition. If If |
AudioEncoding
The encoding of the audio data sent in the request.
All encodings support only 1 channel (mono) audio, unless the audioChannelCount
and enableSeparateRecognitionPerChannel
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
error code.
Enums | |
---|---|
ENCODING_UNSPECIFIED |
Not specified. |
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 . FLAC stream encoding supports 16-bit and 24-bit samples, however, not all fields in STREAMINFO are supported. |
MULAW |
8-bit samples that compand 14-bit audio samples using G.711 PCMU/mu-law. |
AMR |
Adaptive Multi-Rate Narrowband codec. sampleRateHertz must be 8000. |
AMR_WB |
Adaptive Multi-Rate Wideband codec. sampleRateHertz must be 16000. |
OGG_OPUS |
Opus encoded audio frames in Ogg container (OggOpus). sampleRateHertz must be one of 8000, 12000, 16000, 24000, or 48000. |
SPEEX_WITH_HEADER_BYTE |
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. sampleRateHertz must be 16000. |
MP3 |
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, sampleRateHertz has to match the sample rate of the file being used. |
WEBM_OPUS |
Opus encoded audio frames in WebM container (OggOpus). sampleRateHertz must be one of 8000, 12000, 16000, 24000, or 48000. |
SpeechAdaptation
Speech adaptation configuration.
JSON representation |
---|
{ "phraseSets": [ { object ( |
Fields | |
---|---|
phraseSets[] |
A collection of phrase sets. To specify the hints inline, leave the phrase set's |
phraseSetReferences[] |
A collection of phrase set resource names to use. |
customClasses[] |
A collection of custom classes. To specify the classes inline, leave the class' |
abnfGrammar |
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 |
ABNFGrammar
JSON representation |
---|
{ "abnfStrings": [ string ] } |
Fields | |
---|---|
abnfStrings[] |
All declarations and rules of an ABNF grammar broken up into multiple strings that will end up concatenated. |
TranscriptNormalization
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.
JSON representation |
---|
{
"entries": [
{
object ( |
Fields | |
---|---|
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. |
Entry
A single replacement configuration.
JSON representation |
---|
{ "search": string, "replace": string, "caseSensitive": boolean } |
Fields | |
---|---|
search |
What to replace. Max length is 100 characters. |
replace |
What to replace with. Max length is 100 characters. |
caseSensitive |
Whether the search is case sensitive. |
SpeechContext
Provides "hints" to the speech recognizer to favor specific words and phrases in the results.
JSON representation |
---|
{ "phrases": [ string ], "boost": number } |
Fields | |
---|---|
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. |
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 |
SpeakerDiarizationConfig
Config to enable speaker diarization.
JSON representation |
---|
{ "enableSpeakerDiarization": boolean, "minSpeakerCount": integer, "maxSpeakerCount": integer, "speakerTag": integer } |
Fields | |
---|---|
enableSpeakerDiarization |
If 'true', enables speaker detection for each recognized word in the top alternative of the recognition result using a speakerTag provided in the WordInfo. |
minSpeakerCount |
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. |
maxSpeakerCount |
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. |
speakerTag |
Output only. Unused. |
RecognitionMetadata
Description of audio data to be recognized.
JSON representation |
---|
{ "interactionType": enum ( |
Fields | |
---|---|
interactionType |
The use case most closely describing the audio content to be recognized. |
industryNaicsCodeOfAudio |
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/. |
microphoneDistance |
The audio type that most closely describes the audio being recognized. |
originalMediaType |
The original media the speech was recorded on. |
recordingDeviceType |
The type of device the speech was recorded with. |
recordingDeviceName |
The device used to make the recording. Examples 'Nexus 5X' or 'Polycom SoundStation IP 6000' or 'POTS' or 'VoIP' or 'Cardioid Microphone'. |
originalMimeType |
Mime type of the original audio file. For example |
obfuscatedId |
Obfuscated (privacy-protected) ID of the user, to identify number of unique users using the service. |
audioTopic |
Description of the content. Eg. "Recordings of federal supreme court hearings from 2012". |
InteractionType
Use case categories that the audio recognition request can be described by.
Enums | |
---|---|
INTERACTION_TYPE_UNSPECIFIED |
Use case is either unknown or is something other than one of the other values below. |
DISCUSSION |
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 |
One or more persons lecturing or presenting to others, mostly uninterrupted. |
PHONE_CALL |
A phone-call or video-conference in which two or more people, who are not in the same room, are actively participating. |
VOICEMAIL |
A recorded message intended for another person to listen to. |
PROFESSIONALLY_PRODUCED |
Professionally produced audio (eg. TV Show, Podcast). |
VOICE_SEARCH |
Transcribe spoken questions and queries into text. |
VOICE_COMMAND |
Transcribe voice commands, such as for controlling a device. |
DICTATION |
Transcribe speech to text to create a written document, such as a text-message, email or report. |
MicrophoneDistance
Enumerates the types of capture settings describing an audio file.
Enums | |
---|---|
MICROPHONE_DISTANCE_UNSPECIFIED |
Audio type is not known. |
NEARFIELD |
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 |
The speaker if within 3 meters of the microphone. |
FARFIELD |
The speaker is more than 3 meters away from the microphone. |
OriginalMediaType
The original media the speech was recorded on.
Enums | |
---|---|
ORIGINAL_MEDIA_TYPE_UNSPECIFIED |
Unknown original media type. |
AUDIO |
The speech data is an audio recording. |
VIDEO |
The speech data originally recorded on a video. |
RecordingDeviceType
The type of device the speech was recorded with.
Enums | |
---|---|
RECORDING_DEVICE_TYPE_UNSPECIFIED |
The recording device is unknown. |
SMARTPHONE |
Speech was recorded on a smartphone. |
PC |
Speech was recorded using a personal computer or tablet. |
PHONE_LINE |
Speech was recorded over a phone line. |
VEHICLE |
Speech was recorded in a vehicle. |
OTHER_OUTDOOR_DEVICE |
Speech was recorded outdoors. |
OTHER_INDOOR_DEVICE |
Speech was recorded indoors. |