Powered by Universal Speech Solutions LLC

 MRCP

Google SS Plugin

Usage Guide

 

Revision: 10

Created: May 24, 2018

Last updated: March 8, 2021

Author: Arsen Chaloyan


 

Table of Contents

 

1  Overview.. 4

1.1         Installation. 4

1.2         Applicable Versions 4

2  Supported Features. 5

2.1         MRCP Methods. 5

2.2         MRCP Events. 5

2.3         MRCP Header Fields. 5

2.4         Speech Data. 5

3  Supported Voices. 6

4  Configuration Format 7

4.1         Document 7

4.2         Synthesis Settings. 8

4.3         Waveform Manager 10

4.4         SDR Manager 11

4.5         Monitoring Agent 12

4.6         Usage Change Handler 13

4.7         Usage Refresh Handler 13

4.8         License Server 14

5  Configuration Steps. 16

5.1         Using Default Configuration. 16

5.2         Specifying Synthesis Language. 16

5.3         Specifying Sampling Rate. 16

5.4         Specifying Voice Parameters. 16

5.5         Specifying Prosody Parameters. 17

5.6         Specifying Speech Data. 17

5.7         Maintaining Waveforms. 17

5.8         Maintaining Synthesis Details Records. 18

5.9         Using Cache. 18

6  Monitoring Usage Details 20

6.1         Log Usage. 20

6.2         Update Usage. 20

6.3         Dump Channels. 21

7  Usage Examples 22

7.1         SSML. 22

7.2         Plain Text 22

8  Sequence Diagram.. 24

9  References 25

9.1         Google Cloud Platform.. 25

9.2         Specifications. 25

 

 

1       Overview

This guide describes how to configure and use the Google Speech Synthesis (GSS) plugin to the UniMRCP server. The document is intended for users having a certain knowledge of Google Cloud Speech Platform and UniMRCP.

 

1.1      Installation

For installation instructions, use one of the guides below.

·         RPM Package Installation (Red Hat / Cent OS)

·         Deb Package Installation (Debian / Ubuntu)

1.2      Applicable Versions

Instructions provided in this guide are applicable to the following versions.

 

UniMRCP 1.5.0 and above

UniMRCP GSS Plugin 1.0.0 and above

 

2       Supported Features

This is a brief check list of the features currently supported by the UniMRCP server running with the GSS plugin.

2.1      MRCP Methods

ü  SPEAK

ü  STOP

ü  PAUSE

ü  RESUME

ü  BARGE-IN-OCCURRED

ü  SET-PARAMS

ü  GET-PARAMS

2.2      MRCP Events

ü  SPEECH-MARKER

ü  SPEAK-COMPLETE

2.3      MRCP Header Fields

ü  Kill-On-Barge-In

ü  Completion-Cause

ü  Voice-Gender

ü  Voice-Name

ü  Prosody-Rate

ü  Prosody-Volume

ü  Speech-Language

ü  Logging-Tag

ü  Cache-Control

2.4      Speech Data

ü  Plain text (text/plain)

ü  SSML (application/ssml+xml or application/synthesis+ssml)

 

 

3       Supported Voices

All the voices supported by Google Text-to-Speech API are listed in the following page:

 

https://cloud.google.com/text-to-speech/docs/voices

 

4       Configuration Format

The configuration file of the GSS plugin is located in /opt/unimrcp/conf/umsgss.xml. The configuration file is written in XML.

4.1      Document

The root element of the XML document must be <umsgss>.

Attributes

 

Name

      Unit

Description

license-file

File path

Specifies the license file. File name may include patterns containing '*' sign. If multiple files match the pattern, the most recent one gets used.

gapp-credentials-file

File path

Specifies the Google Application Credentials file to use. File name may include patterns containing '*' sign. If multiple files match the pattern, the most recent one gets used.

 

Parent

None.

Children

 

Name

Unit

Description

<synth-settings>

String

Specifies synthesis parameters employed via gRPC.

<waveform-manager>

String

Specifies parameters of the waveform manager. Available since GSS 1.2.0.

<sdr-manager>

String

Specifies parameters of the Synthesis Details Record (SDR) manager. Available since GSS 1.2.0.

<monitoring-agent>

String

Specifies parameters of the monitoring manager.

<license-server>

String

Specifies parameters used to connect to the license server. The use of the license server is optional.

 

Example

This is an example of a bare document.

 

< umsgss license-file="umsgss_*.lic" gapp-credentials-file="*.json">

</ umsgss>

 

4.2      Synthesis Settings

This element specifies synthesis parameters.

Attributes

 

Name

Unit

Description

language

String

Specifies the default language to use, if not set by the client.

bypass-ssml

Boolean

Specifies whether to transparently bypass or parse received content in order to determine voice parameters set in SSML. Available since GSS 1.1.0.

normalize-ssml

Boolean

Specifies whether to normalize SSML. The parameter is observed only when bypass-ssml is set to false. Available since GSS 1.4.0.

voice-name

String

Specifies the default voice name. Can be overridden by client. Available since GSS 1.1.0.

voice-gender

String

Specifies the default voice gender. Can be overridden by client. Available since GSS 1.1.0.

effects-profile

String

Specifies the audio effects profile identifier.

https://cloud.google.com/text-to-speech/docs/audio-profiles

Available since GSS 1.7.0/

http-proxy

String

Specifies the URI of HTTP proxy, if used. Available since GSS 1.4.0.

grpc-log-redirection

Boolean

Specifies whether to enable gRPC log redirection. Available since GSS 1.6.0.

grpc-log-verbosity

String

Specifies gRPC logging verbosity. One of DEBUG, INFO, ERROR. See GRPC_VERBOSITY for more info. Available since GSS 1.6.0.

grpc-log-trace

String

Specifies a comma separated list of tracers producing gRPC logs. Use 'all' to turn all tracers on. See GRPC_TRACE for more info. Available since GSS 1.6.0.

caching

Boolean

Specifies whether to enable caching of synthesized waveforms. Available since GSS 1.8.0.

prosody-rate

Double

Specifies the default prosody rate (speaking_rate) in the range [0.25, 4.0]. Can be overridden by client. Available since GSS 1.9.0.

prosody-volume

Double

Specifies the default prosody volume (volume_gain_db) in the range [-96.0, 16.0]. Can be overridden by client. Available since GSS 1.9.0.

prosody-pitch

Double

Specifies the default prosody pitch in the range [-20.0, 20.0]. Available since GSS 1.9.0.

deadline

Time interval [msec]

Specifies the gRPC call deadline. Defaults to 0 (disabled). Available since GSS 1.10.0.

reattempt

Boolean

Specifies whether to reattempt the gRPC call if the original attempt fails. Disabled by default. Available since GSS 1.10.0.

service-uri

String

Specifies the service endpoint and defaults to texttospeech.googleapis.com:443. Available since GSS 1.11.0.

 

Parent

<umsgss>

Children

None.

Example

This is an example of synthesis parameters.

 

   <synth-settings

      language="en-US"

   bypass-ssml="true"

   normalize-ssml="true"

      voice-name=""

      voice-gender=""

   />

4.3      Waveform Manager

This element specifies parameters of the waveform manager.

Availability

>= GSS 1.2.0.

Attributes

 

Name

Unit

Description

save-waveforms

Boolean

Specifies whether to save waveforms or not.

purge-existing

Boolean

Specifies whether to delete existing records on start-up.

max-file-age

Time interval [min]

Specifies a time interval in minutes after expiration of which a waveform is deleted. Set 0 for infinite.

max-file-count

Integer

Specifies the max number of waveforms to store. If reached, the oldest waveform is deleted. Set 0 for infinite.

waveform-folder

Dir path

Specifies a folder the waveforms should be stored in.

file-prefix

String

Specifies a prefix used to compose the name of the file to be stored. Defaults to 'umsgss-', if not specified.

use-logging-tag

Boolean

Specifies whether to use the MRCP header field Logging-Tag, if present, to compose the name of the file to be stored. Available since GSS 1.6.0.

 

Parent

<umsgss>

Children

None.

Example

The example below defines a typical utterance manager having the default parameters set.

 

   <waveform-manager

      save-waveforms="false"

      purge-existing="false"

      max-file-age="60"

      max-file-count="100"     

      waveform-folder=""

   />

 

4.4      SDR Manager

This element specifies parameters of the Synthesis Details Record (SDR) manager.

Availability

>= GSS 1.2.0.

Attributes

 

Name

Unit

Description

save-records

Boolean

Specifies whether to save recognition details records or not.

purge-existing

Boolean

Specifies whether to delete existing records on start-up.

max-file-age

Time interval [min]

Specifies a time interval in minutes after expiration of which a record is deleted. Set 0 for infinite.

max-file-count

Integer

Specifies the max number of records to store. If reached, the oldest record is deleted. Set 0 for infinite.

record-folder

Dir path

Specifies a folder to store recognition details records in. Defaults to ${UniMRCPInstallDir}/var.

file-prefix

String

Specifies a prefix used to compose the name of the file to be stored. Defaults to 'umsgss-', if not specified.

use-logging-tag

Boolean

Specifies whether to use the MRCP header field Logging-Tag, if present, to compose the name of the file to be stored. Available since GSS 1.6.0.

 

Parent

<umsgss>

Children

None.

Example

The example below defines a typical utterance manager having the default parameters set.

 

   <sdr-manager

      save-records="false"

      purge-existing="false"

      max-file-age="60"

      max-file-count="100"     

      waveform-folder=""

   />

 

4.5      Monitoring Agent

This element specifies parameters of the monitoring agent.

Attributes

 

Name

Unit

Description

refresh-period

Time interval [sec]

Specifies a time interval in seconds used to periodically refresh usage details. See <usage-refresh-handler>.

 

Parent

<umsgss>

Children

<usage-change-handler>

<usage-refresh-handler>

Example

The example below defines a monitoring agent with usage change and refresh handlers.

 

   <monitoring-agent refresh-period="60">

 

      <usage-change-handler>

        <log-usage enable="true" priority="NOTICE"/>

      </usage-change-handler>

 

      <usage-refresh-handler>

        <dump-channels enable="true" status-file="umsgss-channels.status"/>

      </usage-refresh-handler >

 

   </monitoring-agent>

 

4.6      Usage Change Handler

This element specifies an event handler called on every usage change.

Attributes

None.

Parent

<monitoring-agent>

Children

<log-usage>

<update-usage>

<dump-channels>

Example

This is an example of the usage change event handler.

 

      <usage-change-handler>

        <log-usage enable="true" priority="NOTICE"/>

        <update-usage enable="false" status-file="umsgss-usage.status"/>

        <dump-channels enable="false" status-file="umsgss-channels.status"/>

      </usage-change-handler>

 

4.7       Usage Refresh Handler

This element specifies an event handler called periodically to update usage details.

Attributes

None.

Parent

<monitoring-agent>

Children

<log-usage>

<update-usage>

<dump-channels>

Example

This is an example of the usage change event handler.

 

      <usage-refresh-handler>

        <log-usage enable="true" priority="NOTICE"/>

        <update-usage enable="false" status-file="umsgss-usage.status"/>

        <dump-channels enable="false" status-file="umsgss-channels.status"/>

      </usage-refresh-handler>

 

4.8       License Server

This element specifies parameters used to connect to the license server.

Attributes

 

Name

Unit

Description

enable

Boolean

Specifies whether the use of license server is enabled or not. If enabled, the license-file attribute is not honored.

server-address

String

Specifies the IP address or host name of the license server.

certificate-file

File path

Specifies the client certificate used to connect to the license server. File name may include patterns containing a '*' sign. If multiple files match the pattern, the most recent one gets used.

ca-file

File path

Specifies the certificate authority used to validate the license server.

channel-count

Integer

Specifies the number of channels to check out from the license server. If not specified or set to 0, either all available channels or a pool of channels will be checked based on the configuration of the license server.

http-proxy-address

String

Specifies the IP address or host name of the HTTP proxy server, if used. Available since GSS 1.6.0.

http-proxy-port

Integer

Specifies the port number of the HTTP proxy server, if used. Available since GSS 1.6.0.

 

Parent

<umsgss>

Children

None.

Example

The example below defines a typical configuration which can be used to connect to a license server located, for example, at 10.0.0.1.

 

   <license-server

      enable="true"

      server-address="10.0.0.1"

      certificate-file="unilic_client_*.crt"

      ca-file="unilic_ca.crt"

   />

 

For further reference to the license server, visit

 

http://unimrcp.org/licserver

 

 

5       Configuration Steps

This section outlines common configuration steps.

5.1      Using Default Configuration

The default configuration should be sufficient for the general use.

5.2      Specifying Synthesis Language

Synthesis language can be specified by the client per MRCP session by means of the header field Speech-Language set in a SET-PARAMS or SPEAK request, or inline in the SSML data. Otherwise, the parameter language set in the configuration file umsgss.xml is used. The parameter defaults to en-US.

5.3      Specifying Sampling Rate

Sampling rate is determined based on the SDP negotiation. Refer to the configuration guide of the UniMRCP server on how to specify supported encodings and sampling rates to be used in communication between the client and server. Either 8 or 16 kHz can be used by Google Cloud Text-to-Speech API for synthesis.

5.4      Specifying Voice Parameters

Global Settings

The default voice name and gender can be specified from the configuration file umsgss.xml using the voice-name and voice-gender attributes of the synth-settings element. This functionality is available since GSS 1.1.0 release.

MRCP Header Fields

The voice name and gender can be specified by the MRCP client in SET-PARAMS and SPEAK requests.

·         Voice-Name

This is an optional parameter indicating the name of the voice to use for synthesis.

·         Voice-Gender

This is an optional parameter indicating the preferred gender of the voice to use for synthesis, which can be set to either male or female or neutral.

SSML Content

The voice name and gender can also be specified using the corresponding attributes of the voice element in SSML content. In order to parse and determine the parameters and pass them forward to Google Text-to-Speech API accordingly, the bypass-ssml attribute of the synth-settings element must be set to false in the configuration file umsgss.xml. This functionality is available since GSS 1.1.0 release.

 

Since GSS 1.1.0 release, if the bypass-ssml attribute is set to false and the normalize-ssml attribute is set to true, then the voice element, if present, is stripped off from the SSML content passed to the service in order to conform to the subset of SSML supported by Google Text-to-Speech API.

5.5      Specifying Prosody Parameters

The following prosody parameters can be specified by the MRCP client in SET-PARAMS and SPEAK requests.

·         Prosody-Rate

This is an optional parameter indicating the speaking rate, which can be set to one of the following labels: x-slow, slow, medium, fast, x-fast, default.

·         Prosody-Volume

This is an optional parameter indicating the speaking volume, which can be set to one of the following labels: silent, x-soft, soft, medium, loud, x-loud, default.

5.6      Specifying Speech Data

Speech data can be specified by the MRCP client in SPEAK requests using one of the following content types:

·         plain/text

·         application/ssml+xml (or application/synthesis+ssml)

5.7      Maintaining Waveforms

Collection of waveforms is not required for regular operation and is disabled by default. However, enabling this functionality allows to save synthesized speech received from the Google Cloud Speech service and later listen to them offline.

The relevant settings can be specified via the element waveform-manager.

·         save-waveforms

Utterances can optionally be recorded and stored if the configuration parameter save-waveforms is set to true.

·         purge-existing

This parameter specifies whether to delete existing waveforms on start-up.

·         max-file-age

This parameter specifies a time interval in minutes after expiration of which a waveform is deleted. If set to 0, there is no expiration time specified.

·         max-file-count

This parameter specifies the maximum number of waveforms to store. If the specified number is reached, the oldest waveform is deleted. If set to 0, there is no limit specified.

·         waveform-folder

This parameter specifies a path to the directory used to store waveforms in. The directory defaults to ${UniMRCPInstallDir}/var.

5.8      Maintaining Synthesis Details Records

Collection of synthesis details records (SDR) is not required for regular operation and is disabled by default. However, enabling this functionality allows to store details of each synthesis attempt in a separate file and analyze them later offline. The SDRs ate stored in the JSON format.

The relevant settings can be specified via the element sdr-manager.

·         save-records

This parameter specifies whether to save synthesis details records or not.

·         purge-existing

This parameter specifies whether to delete existing records on start-up.

·         max-file-age

This parameter specifies a time interval in minutes after expiration of which a record is deleted. If set to 0, there is no expiration time specified.

·         max-file-count

This parameter specifies the maximum number of records to store. If the specified number is reached, the oldest record is deleted. If set to 0, there is no limit specified.

·         record-folder

This parameter specifies a path to the directory used to store records in. The directory defaults to ${UniMRCPInstallDir}/var.

 

5.9      Using Cache

Since GSS 1.8.0, synthesized waveforms can be stored and re-used for consecutive speech synthesis requests, when applicable. In order to use this functionality, the attribute caching of the element synth-settings must be set to true. The attribute defaults to false.

The lifetime and size of cached records are controlled by the attributes max-file-age and max-file-count of the element waveform-manager.

The cached records are persistent and populated on initial loading, unless the attribute purge-existing of the element waveform-manager is set to true.

The following speech synthesis parameters are observed while searching for a cached record.

·         language

·         voice-name

·         voice-gender

·         sampling-rate

·         prosody-rate

·         prosody-volume

·         content

The following cache control directives are observed while searching for a cached record.

·         max-age

·         min-fresh

The cache control directives can be specified by the client per individual speech synthesis request via the MRCP header field Cache-Control. By default, no cache control directives are applied.

 

 

6       Monitoring Usage Details

The number of in-use and total licensed channels can be monitored in several alternate ways. There is a set of actions which can take place on certain events. The behavior is configurable via the element monitoring-agent, which contains two event handlers: usage-change-handler and usage-refresh-handler.

 

While the usage-change-handler is invoked on every acquisition and release of a licensed channel, the usage-refresh-handler is invoked periodically on expiration of a timeout specified by the attribute refresh-period.

 

The following actions can be specified for either of the two handlers.

6.1      Log Usage

The action log-usage logs the following data in the order specified.

·         The number of currently in-use channels.

·         The maximum number of channels used concurrently. Available since GSS 1.2.0.

·         The total number of licensed channels.

The following is a sample log statement, indicating 0 in-use, 0 max-used and 2 total channels.

 

[NOTICE] GSS Usage: 0/0/2

 

6.2      Update Usage

The action update-usage writes the following data to a status file umsgss-usage.status, located by default in the directory ${UniMRCPInstallDir}/var/status.

·         The number of currently in-use channels.

·         The maximum number of channels used concurrently. Available since GSS 1.2.0.

·         The total number of licensed channels.

·         The current status of the license permit.

·         The license server alarm. Set to on, if the license server is not available for more than one hour; otherwise, set to off. This parameter is maintained only if the license server is used. Available since GSS 1.4.0.

The following is a sample content of the status file.

 

in-use channels: 0

max used channels: 0

total channels: 2

license permit: true

licserver alarm: off

 

6.3      Dump Channels

The action dump-channels writes the identifiers of in-use channels to a status file umsgss-channels.status, located by default in the directory ${UniMRCPInstallDir}/var/status.

7       Usage Examples

7.1      SSML

This examples demonstrates how to perform speech synthesis by using a SPEAK request with an SSML content.

 

C->S:

 

MRCP/2.0 309 SPEAK 1

Channel-Identifier: 4dde51f37d1a9546@speechsynth

Content-Type: application/ssml+xml

Voice-Age: 28

Content-Length: 163

 

<?xml version="1.0"?>

<speak version="1.0" xml:lang="en-US" xmlns="http://www.w3.org/2001/10/synthesis">

  <p>

    <s>Welcome to Uni MRCP.</s>

  </p>

</speak>

 

S->C:

 

MRCP/2.0 83 1 200 IN-PROGRESS

Channel-Identifier: 4dde51f37d1a9546@speechsynth

 

 

S->C:

 

MRCP/2.0 122 SPEAK-COMPLETE 1 COMPLETE

Channel-Identifier: 4dde51f37d1a9546@speechsynth

Completion-Cause: 000 normal

 

 

7.2      Plain Text

This example demonstrates how to perform speech synthesis by using a SPEAK request with a plain text content.

 

C->S:

 

MRCP/2.0 155 SPEAK 1

Channel-Identifier: 85667d0efbf95345@speechsynth

Content-Type: text/plain

Voice-Age: 28

Content-Length: 20

 

Welcome to Uni MRCP.

 

S->C:

 

MRCP/2.0 83 1 200 IN-PROGRESS

Channel-Identifier: 85667d0efbf95345@speechsynth

 

 

S->C:

 

MRCP/2.0 122 SPEAK-COMPLETE 1 COMPLETE

Channel-Identifier: 85667d0efbf95345@speechsynth

Completion-Cause: 000 normal

 

8       Sequence Diagram

The following sequence diagram outlines common interactions between all the main components involved in a typical synthesis session performed over MRCPv2.

9       References

9.1      Google Cloud Platform

·         Text-to-Speech API

·         How-to Guides

·         Best Practices

9.2      Specifications

·         Speech Synthesizer Resource

·         SSML