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<h1><a href="texttospeech_v1beta1.html">Cloud Text-to-Speech API</a> . <a href="texttospeech_v1beta1.text.html">text</a></h1>
<h2>Instance Methods</h2>
<p class="toc_element">
  <code><a href="#close">close()</a></code></p>
<p class="firstline">Close httplib2 connections.</p>
<p class="toc_element">
  <code><a href="#synthesize">synthesize(body=None, x__xgafv=None)</a></code></p>
<p class="firstline">Synthesizes speech synchronously: receive results after all text input has been processed.</p>
<h3>Method Details</h3>
<div class="method">
    <code class="details" id="close">close()</code>
  <pre>Close httplib2 connections.</pre>
</div>

<div class="method">
    <code class="details" id="synthesize">synthesize(body=None, x__xgafv=None)</code>
  <pre>Synthesizes speech synchronously: receive results after all text input has been processed.

Args:
  body: object, The request body.
    The object takes the form of:

{ # The top-level message sent by the client for the `SynthesizeSpeech` method.
    &quot;input&quot;: { # 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. # Required. The Synthesizer requires either plain text or SSML as input.
      &quot;text&quot;: &quot;A String&quot;, # The raw text to be synthesized.
      &quot;ssml&quot;: &quot;A String&quot;, # 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](https://cloud.google.com/text-to-speech/docs/ssml).
    },
    &quot;audioConfig&quot;: { # Description of audio data to be synthesized. # Required. The configuration of the synthesized audio.
      &quot;speakingRate&quot;: 3.14, # 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 &lt; 0.25 or &gt; 4.0 will return an error.
      &quot;pitch&quot;: 3.14, # 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.
      &quot;sampleRateHertz&quot;: 42, # Optional. The synthesis sample rate (in hertz) for this audio. When this is specified in SynthesizeSpeechRequest, if this is different from the voice&#x27;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.
      &quot;audioEncoding&quot;: &quot;A String&quot;, # Required. The format of the audio byte stream.
      &quot;volumeGainDb&quot;: 3.14, # 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&#x27;s usually no effective increase in loudness for any value greater than that.
      &quot;effectsProfileId&quot;: [ # Optional. Input only. An identifier which selects &#x27;audio effects&#x27; 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](https://cloud.google.com/text-to-speech/docs/audio-profiles) for current supported profile ids.
        &quot;A String&quot;,
      ],
    },
    &quot;enableTimePointing&quot;: [ # Whether and what timepoints are returned in the response.
      &quot;A String&quot;,
    ],
    &quot;voice&quot;: { # Description of which voice to use for a synthesis request. # Required. The desired voice of the synthesized audio.
      &quot;name&quot;: &quot;A String&quot;, # The name of the voice. If not set, the service will choose a voice based on the other parameters such as language_code and gender.
      &quot;languageCode&quot;: &quot;A String&quot;, # Required. The language (and potentially also the region) of the voice expressed as a [BCP-47](https://www.rfc-editor.org/rfc/bcp/bcp47.txt) language tag, e.g. &quot;en-US&quot;. This should not include a script tag (e.g. use &quot;cmn-cn&quot; rather than &quot;cmn-Hant-cn&quot;), 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&#x27;t a Canadian voice available), or even a different language, e.g. using &quot;nb&quot; (Norwegian Bokmal) instead of &quot;no&quot; (Norwegian)&quot;.
      &quot;ssmlGender&quot;: &quot;A String&quot;, # The preferred gender of the voice. If not set, the service will choose a voice based on the other parameters such as language_code 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.
    },
  }

  x__xgafv: string, V1 error format.
    Allowed values
      1 - v1 error format
      2 - v2 error format

Returns:
  An object of the form:

    { # The message returned to the client by the `SynthesizeSpeech` method.
    &quot;audioConfig&quot;: { # Description of audio data to be synthesized. # The audio metadata of `audio_content`.
      &quot;speakingRate&quot;: 3.14, # 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 &lt; 0.25 or &gt; 4.0 will return an error.
      &quot;pitch&quot;: 3.14, # 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.
      &quot;sampleRateHertz&quot;: 42, # Optional. The synthesis sample rate (in hertz) for this audio. When this is specified in SynthesizeSpeechRequest, if this is different from the voice&#x27;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.
      &quot;audioEncoding&quot;: &quot;A String&quot;, # Required. The format of the audio byte stream.
      &quot;volumeGainDb&quot;: 3.14, # 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&#x27;s usually no effective increase in loudness for any value greater than that.
      &quot;effectsProfileId&quot;: [ # Optional. Input only. An identifier which selects &#x27;audio effects&#x27; 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](https://cloud.google.com/text-to-speech/docs/audio-profiles) for current supported profile ids.
        &quot;A String&quot;,
      ],
    },
    &quot;timepoints&quot;: [ # A link between a position in the original request input and a corresponding time in the output audio. It&#x27;s only supported via of SSML input.
      { # This contains a mapping between a certain point in the input text and a corresponding time in the output audio.
        &quot;markName&quot;: &quot;A String&quot;, # Timepoint name as received from the client within tag.
        &quot;timeSeconds&quot;: 3.14, # Time offset in seconds from the start of the synthesized audio.
      },
    ],
    &quot;audioContent&quot;: &quot;A String&quot;, # 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.
  }</pre>
</div>

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