Method: text.synthesize

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Synthesizes speech synchronously: receive results after all text input has been processed.

HTTP request

POST https://texttospeech.googleapis.com/v1/text:synthesize

The URL uses gRPC Transcoding syntax.

Request body

The request body contains data with the following structure:

JSON representation 
{
  "input": {
    object (SynthesisInput)
  },
  "voice": {
    object (VoiceSelectionParams)
  },
  "audioConfig": {
    object (AudioConfig)
  }
}
Fields
input

object (SynthesisInput)

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

object (VoiceSelectionParams)

Required. The desired voice of the synthesized audio.
audioConfig

object (AudioConfig)

Required. The configuration of the synthesized audio.

Response body

If successful, the response body contains data with the following structure:

The message returned to the client by the text.synthesize method.

JSON representation 
{
  "audioContent": string
}
Fields
audioContent

string (bytes format)

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

A base64-encoded string.

Authorization Scopes

Requires the following OAuth scope:

  • https://www.googleapis.com/auth/cloud-platform

For more information, see the Authentication Overview.

SynthesisInput

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

JSON representation 
{

  // Union field input_source can be only one of the following:
  "text": string,
  "ssml": string
  // End of list of possible types for union field input_source.
}
Fields
Union field input_source. The input source, which is either plain text or SSML. input_source can be only one of the following:
text

string

The raw text to be synthesized.
ssml

string

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

VoiceSelectionParams

Description of which voice to use for a synthesis request.

JSON representation 
{
  "languageCode": string,
  "name": string,
  "ssmlGender": enum (SsmlVoiceGender)
}
Fields
languageCode

string

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

string

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

enum (SsmlVoiceGender)

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

AudioConfig

Description of audio data to be synthesized.

JSON representation 
{
  "audioEncoding": enum (AudioEncoding),
  "speakingRate": number,
  "pitch": number,
  "volumeGainDb": number,
  "sampleRateHertz": integer,
  "effectsProfileId": [
    string
  ]
}
Fields
audioEncoding

enum (AudioEncoding)

Required. The format of the audio byte stream.
speakingRate

number

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

number

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

number

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

integer

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

string

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

AudioEncoding

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

Enums
AUDIO_ENCODING_UNSPECIFIED Not specified. Will return result google.rpc.Code.INVALID_ARGUMENT.
LINEAR16 Uncompressed 16-bit signed little-endian samples (Linear PCM). Audio content returned as LINEAR16 also contains a WAV header.
MP3 MP3 audio at 32kbps.
OGG_OPUS Opus encoded audio wrapped in an ogg container. The result will be a file which can be played natively on Android, and in browsers (at least Chrome and Firefox). The quality of the encoding is considerably higher than MP3 while using approximately the same bitrate.
MULAW 8-bit samples that compand 14-bit audio samples using G.711 PCMU/mu-law. Audio content returned as MULAW also contains a WAV header.
ALAW 8-bit samples that compand 14-bit audio samples using G.711 PCMU/A-law. Audio content returned as ALAW also contains a WAV header.