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feat(stt): add character insertion bias parameter support #124

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feat(stt): add character insertion bias parameter support #124

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4b7c26f
feat(stt): add character insertion bias parameter
May 16, 2023
5f701db
fix(stt): add character insertion bias parameter to createjoboptions
May 16, 2023
4931039
fix(stt-unit-test): add character insertion bias param to test cases
May 16, 2023
4415b09
fix(stt): add missing set func for character insertion bias on Create…
May 16, 2023
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114 changes: 95 additions & 19 deletions v3/speechtotextv1/speech_to_text_v1.go
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Original file line number Diff line number Diff line change
Expand Up @@ -45,7 +45,6 @@ import (
// names. Broadband and multimedia models have minimum sampling rates of 16 kHz. Narrowband and telephony models have
// minimum sampling rates of 8 kHz. The next-generation models offer high throughput and greater transcription accuracy.
//
//
// Effective 15 March 2022, previous-generation models for all languages other than Arabic and Japanese are deprecated.
// The deprecated models remain available until 15 September 2022, when they will be removed from the service and the
// documentation. You must migrate to the equivalent next-generation model by the end of service date. For more
Expand Down Expand Up @@ -314,7 +313,8 @@ func (speechToText *SpeechToTextV1) GetModelWithContext(ctx context.Context, get
//
// ### Streaming mode
//
// For requests to transcribe live audio as it becomes available, you must set the `Transfer-Encoding` header to
// For requests to transcribe live audio as it becomes available, you must set the `Transfer-Encoding` header to
//
// `chunked` to use streaming mode. In streaming mode, the service closes the connection (status code 408) if it does
// not receive at least 15 seconds of audio (including silence) in any 30-second period. The service also closes the
// connection (status code 400) if it detects no speech for `inactivity_timeout` seconds of streaming audio; use the
Expand All @@ -326,7 +326,8 @@ func (speechToText *SpeechToTextV1) GetModelWithContext(ctx context.Context, get
//
// ### Audio formats (content types)
//
// The service accepts audio in the following formats (MIME types).
// The service accepts audio in the following formats (MIME types).
//
// * For formats that are labeled **Required**, you must use the `Content-Type` header with the request to specify the
// format of the audio.
// * For all other formats, you can omit the `Content-Type` header or specify `application/octet-stream` with the header
Expand Down Expand Up @@ -357,12 +358,14 @@ func (speechToText *SpeechToTextV1) GetModelWithContext(ctx context.Context, get
// minimum required rate, the service down-samples the audio to the appropriate rate. If the sampling rate of the audio
// is lower than the minimum required rate, the request fails.
//
// **See also:** [Supported audio
// **See also:** [Supported audio
//
// formats](https://cloud.ibm.com/docs/speech-to-text?topic=speech-to-text-audio-formats).
//
// ### Next-generation models
//
// The service supports next-generation `Multimedia` (16 kHz) and `Telephony` (8 kHz) models for many languages.
// The service supports next-generation `Multimedia` (16 kHz) and `Telephony` (8 kHz) models for many languages.
//
// Next-generation models have higher throughput than the service's previous generation of `Broadband` and `Narrowband`
// models. When you use next-generation models, the service can return transcriptions more quickly and also provide
// noticeably better transcription accuracy.
Expand All @@ -385,7 +388,8 @@ func (speechToText *SpeechToTextV1) GetModelWithContext(ctx context.Context, get
//
// ### Multipart speech recognition
//
// **Note:** The asynchronous HTTP interface, WebSocket interface, and Watson SDKs do not support multipart speech
// **Note:** The asynchronous HTTP interface, WebSocket interface, and Watson SDKs do not support multipart speech
//
// recognition.
//
// The HTTP `POST` method of the service also supports multipart speech recognition. With multipart requests, you pass
Expand Down Expand Up @@ -504,6 +508,9 @@ func (speechToText *SpeechToTextV1) RecognizeWithContext(ctx context.Context, re
if recognizeOptions.BackgroundAudioSuppression != nil {
builder.AddQuery("background_audio_suppression", fmt.Sprint(*recognizeOptions.BackgroundAudioSuppression))
}
if recognizeOptions.CharacterInsertionBias != nil {
builder.AddQuery("character_insertion_bias", fmt.Sprint(*recognizeOptions.CharacterInsertionBias))
}
if recognizeOptions.LowLatency != nil {
builder.AddQuery("low_latency", fmt.Sprint(*recognizeOptions.LowLatency))
}
Expand Down Expand Up @@ -706,7 +713,8 @@ func (speechToText *SpeechToTextV1) UnregisterCallbackWithContext(ctx context.Co
//
// ### Streaming mode
//
// For requests to transcribe live audio as it becomes available, you must set the `Transfer-Encoding` header to
// For requests to transcribe live audio as it becomes available, you must set the `Transfer-Encoding` header to
//
// `chunked` to use streaming mode. In streaming mode, the service closes the connection (status code 408) if it does
// not receive at least 15 seconds of audio (including silence) in any 30-second period. The service also closes the
// connection (status code 400) if it detects no speech for `inactivity_timeout` seconds of streaming audio; use the
Expand All @@ -718,7 +726,8 @@ func (speechToText *SpeechToTextV1) UnregisterCallbackWithContext(ctx context.Co
//
// ### Audio formats (content types)
//
// The service accepts audio in the following formats (MIME types).
// The service accepts audio in the following formats (MIME types).
//
// * For formats that are labeled **Required**, you must use the `Content-Type` header with the request to specify the
// format of the audio.
// * For all other formats, you can omit the `Content-Type` header or specify `application/octet-stream` with the header
Expand Down Expand Up @@ -749,12 +758,14 @@ func (speechToText *SpeechToTextV1) UnregisterCallbackWithContext(ctx context.Co
// minimum required rate, the service down-samples the audio to the appropriate rate. If the sampling rate of the audio
// is lower than the minimum required rate, the request fails.
//
// **See also:** [Supported audio
// **See also:** [Supported audio
//
// formats](https://cloud.ibm.com/docs/speech-to-text?topic=speech-to-text-audio-formats).
//
// ### Next-generation models
//
// The service supports next-generation `Multimedia` (16 kHz) and `Telephony` (8 kHz) models for many languages.
// The service supports next-generation `Multimedia` (16 kHz) and `Telephony` (8 kHz) models for many languages.
//
// Next-generation models have higher throughput than the service's previous generation of `Broadband` and `Narrowband`
// models. When you use next-generation models, the service can return transcriptions more quickly and also provide
// noticeably better transcription accuracy.
Expand Down Expand Up @@ -1413,10 +1424,10 @@ func (speechToText *SpeechToTextV1) DeleteLanguageModelWithContext(ctx context.C
// model](https://cloud.ibm.com/docs/speech-to-text?topic=speech-to-text-languageCreate#trainModel-language)
// * [Language support for customization](https://cloud.ibm.com/docs/speech-to-text?topic=speech-to-text-custom-support)
//
//
// ### Training failures
//
// Training can fail to start for the following reasons:
// Training can fail to start for the following reasons:
//
// * The service is currently handling another request for the custom model, such as another training request or a
// request to add a corpus or grammar to the model.
// * No training data have been added to the custom model.
Expand Down Expand Up @@ -1859,8 +1870,6 @@ func (speechToText *SpeechToTextV1) GetCorpusWithContext(ctx context.Context, ge
// another corpus or grammar, or they were modified in some way with the [Add custom words](#addwords) or [Add a custom
// word](#addword) method.
//
//
//
// **See also:** [Deleting a corpus from a custom language
// model](https://cloud.ibm.com/docs/speech-to-text?topic=speech-to-text-manageCorpora#deleteCorpus).
func (speechToText *SpeechToTextV1) DeleteCorpus(deleteCorpusOptions *DeleteCorpusOptions) (response *core.DetailedResponse, err error) {
Expand Down Expand Up @@ -2938,7 +2947,8 @@ func (speechToText *SpeechToTextV1) DeleteAcousticModelWithContext(ctx context.C
//
// ### Training failures
//
// Training can fail to start for the following reasons:
// Training can fail to start for the following reasons:
//
// * The service is currently handling another request for the custom model, such as another training request or a
// request to add audio resources to the model.
// * The custom model contains less than 10 minutes or more than 200 hours of audio data.
Expand Down Expand Up @@ -3259,7 +3269,8 @@ func (speechToText *SpeechToTextV1) ListAudioWithContext(ctx context.Context, li
//
// ### Content types for audio-type resources
//
// You can add an individual audio file in any format that the service supports for speech recognition. For an
// You can add an individual audio file in any format that the service supports for speech recognition. For an
//
// audio-type resource, use the `Content-Type` parameter to specify the audio format (MIME type) of the audio file,
// including specifying the sampling rate, channels, and endianness where indicated.
// * `audio/alaw` (Specify the sampling rate (`rate`) of the audio.)
Expand All @@ -3284,12 +3295,14 @@ func (speechToText *SpeechToTextV1) ListAudioWithContext(ctx context.Context, li
// minimum required rate, the service down-samples the audio to the appropriate rate. If the sampling rate of the audio
// is lower than the minimum required rate, the service labels the audio file as `invalid`.
//
// **See also:** [Supported audio
// **See also:** [Supported audio
//
// formats](https://cloud.ibm.com/docs/speech-to-text?topic=speech-to-text-audio-formats).
//
// ### Content types for archive-type resources
//
// You can add an archive file (**.zip** or **.tar.gz** file) that contains audio files in any format that the service
// You can add an archive file (**.zip** or **.tar.gz** file) that contains audio files in any format that the service
//
// supports for speech recognition. For an archive-type resource, use the `Content-Type` parameter to specify the media
// type of the archive file:
// * `application/zip` for a **.zip** file
Expand All @@ -3309,7 +3322,8 @@ func (speechToText *SpeechToTextV1) ListAudioWithContext(ctx context.Context, li
//
// ### Naming restrictions for embedded audio files
//
// The name of an audio file that is contained in an archive-type resource can include a maximum of 128 characters.
// The name of an audio file that is contained in an archive-type resource can include a maximum of 128 characters.
//
// This includes the file extension and all elements of the name (for example, slashes).
func (speechToText *SpeechToTextV1) AddAudio(addAudioOptions *AddAudioOptions) (response *core.DetailedResponse, err error) {
return speechToText.AddAudioWithContext(context.Background(), addAudioOptions)
Expand Down Expand Up @@ -5050,6 +5064,31 @@ type CreateJobOptions struct {
// support](https://cloud.ibm.com/docs/speech-to-text?topic=speech-to-text-detection#detection-support).
BackgroundAudioSuppression *float32 `json:"background_audio_suppression,omitempty"`

// The character_insertion_bias parameter controls the service's bias for competing strings of different lengths
// during speech recognition. With next-generation models, the service parses audio character by character.
// As it does, it establishes hypotheses of previous character strings to help determine viable next characters.
// During this process, it collects candidate strings of different lengths.
//
// By default, each model uses a default character_insertion_bias of 0.0.
// This value is optimized to produce the best balance between hypotheses with different numbers of characters.
// The default is typically adequate for most speech recognition.
// However, certain use cases might benefit from favoring hypotheses with shorter or longer strings of characters.
// In such cases, specifying a change from the default can improve speech recognition.
//
// You can use the character_insertion_bias parameter to indicate that the service is to favor shorter or longer
// strings as it considers subsequent characters for its hypotheses.
// The value you provide depends on the characteristics of your audio.
// The range of acceptable values is from -1.0 to 1.0:
//
// Negative values cause the service to prefer hypotheses with shorter strings of characters.
// Positive values cause the service to prefer hypotheses with longer strings of characters.
// As your value approaches -1.0 or 1.0, the impact of the parameter becomes more pronounced.
// To determine the most effective value for your scenario, start by setting the value of the parameter
// to a small increment, such as -0.1, -0.05, 0.05, or 0.1, and assess how the value impacts the transcription results.
//
// The parameter is not available for previous-generation models.
CharacterInsertionBias *float32 `json:"character_insertion_bias,omitempty"`

// If `true` for next-generation `Multimedia` and `Telephony` models that support low latency, directs the service to
// produce results even more quickly than it usually does. Next-generation models produce transcription results faster
// than previous-generation models. The `low_latency` parameter causes the models to produce results even more quickly,
Expand Down Expand Up @@ -5356,6 +5395,12 @@ func (_options *CreateJobOptions) SetBackgroundAudioSuppression(backgroundAudioS
return _options
}

// SetCharacterInsertionBias : Allow user to set CharacterInsertionBias
func (_options *CreateJobOptions) SetCharacterInsertionBias(characterInsertionBias float32) *CreateJobOptions {
_options.CharacterInsertionBias = core.Float32Ptr(characterInsertionBias)
return _options
}

// SetLowLatency : Allow user to set LowLatency
func (_options *CreateJobOptions) SetLowLatency(lowLatency bool) *CreateJobOptions {
_options.LowLatency = core.BoolPtr(lowLatency)
Expand Down Expand Up @@ -7212,6 +7257,31 @@ type RecognizeOptions struct {
// support](https://cloud.ibm.com/docs/speech-to-text?topic=speech-to-text-detection#detection-support).
BackgroundAudioSuppression *float32 `json:"background_audio_suppression,omitempty"`

// The character_insertion_bias parameter controls the service's bias for competing strings of different lengths
// during speech recognition. With next-generation models, the service parses audio character by character.
// As it does, it establishes hypotheses of previous character strings to help determine viable next characters.
// During this process, it collects candidate strings of different lengths.
//
// By default, each model uses a default character_insertion_bias of 0.0.
// This value is optimized to produce the best balance between hypotheses with different numbers of characters.
// The default is typically adequate for most speech recognition.
// However, certain use cases might benefit from favoring hypotheses with shorter or longer strings of characters.
// In such cases, specifying a change from the default can improve speech recognition.
//
// You can use the character_insertion_bias parameter to indicate that the service is to favor shorter or longer
// strings as it considers subsequent characters for its hypotheses.
// The value you provide depends on the characteristics of your audio.
// The range of acceptable values is from -1.0 to 1.0:
//
// Negative values cause the service to prefer hypotheses with shorter strings of characters.
// Positive values cause the service to prefer hypotheses with longer strings of characters.
// As your value approaches -1.0 or 1.0, the impact of the parameter becomes more pronounced.
// To determine the most effective value for your scenario, start by setting the value of the parameter
// to a small increment, such as -0.1, -0.05, 0.05, or 0.1, and assess how the value impacts the transcription results.
//
// The parameter is not available for previous-generation models.
CharacterInsertionBias *float32 `json:"character_insertion_bias,omitempty"`

// If `true` for next-generation `Multimedia` and `Telephony` models that support low latency, directs the service to
// produce results even more quickly than it usually does. Next-generation models produce transcription results faster
// than previous-generation models. The `low_latency` parameter causes the models to produce results even more quickly,
Expand Down Expand Up @@ -7459,6 +7529,12 @@ func (_options *RecognizeOptions) SetBackgroundAudioSuppression(backgroundAudioS
return _options
}

// SetCharacterInsertionBias : Allow user to set CharacterInsertionBias
func (_options *RecognizeOptions) SetCharacterInsertionBias(characterInsertionBias float32) *RecognizeOptions {
_options.CharacterInsertionBias = core.Float32Ptr(characterInsertionBias)
return _options
}

// SetLowLatency : Allow user to set LowLatency
func (_options *RecognizeOptions) SetLowLatency(lowLatency bool) *RecognizeOptions {
_options.LowLatency = core.BoolPtr(lowLatency)
Expand Down
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