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<a href=".." class="icon icon-home"> Speech v2.0.0+20210325</a>
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<span>Operations</span>
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<a class="" href="../operations_get">Get</a>
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<span>Speech</span>
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<a class="current" href=".">Longrunningrecognize</a>
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<li class="toctree-l2"><a href="#scopes">Scopes</a></li>
<li class="toctree-l2"><a href="#required-request-value">Required Request Value</a></li>
<li><a class="toctree-l3" href="#about-cursors">About Cursors</a></li>
<li class="toctree-l2"><a href="#optional-output-flags">Optional Output Flags</a></li>
<li class="toctree-l2"><a href="#optional-general-properties">Optional General Properties</a></li>
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<p>Performs asynchronous speech recognition: receive results via the google.longrunning.Operations interface. Returns either an <code>Operation.error</code> or an <code>Operation.response</code> which contains a <code>LongRunningRecognizeResponse</code> message. For more information on asynchronous speech recognition, see the <a href="https://cloud.google.com/speech-to-text/docs/async-recognize">how-to</a>.</p>
<h1 id="scopes">Scopes</h1>
<p>You will need authorization for the <em>https://www.googleapis.com/auth/cloud-platform</em> scope to make a valid call.</p>
<p>If unset, the scope for this method defaults to <em>https://www.googleapis.com/auth/cloud-platform</em>.
You can set the scope for this method like this: <code>speech1 --scope &lt;scope&gt; speech longrunningrecognize ...</code></p>
<h1 id="required-request-value">Required Request Value</h1>
<p>The request value is a data-structure with various fields. Each field may be a simple scalar or another data-structure.
In the latter case it is advised to set the field-cursor to the data-structure's field to specify values more concisely.</p>
<p>For example, a structure like this:</p>
<pre><code>LongRunningRecognizeRequest:
audio:
content: string
uri: string
config:
audio-channel-count: integer
diarization-config:
enable-speaker-diarization: boolean
max-speaker-count: integer
min-speaker-count: integer
speaker-tag: integer
enable-automatic-punctuation: boolean
enable-separate-recognition-per-channel: boolean
enable-word-time-offsets: boolean
encoding: string
language-code: string
max-alternatives: integer
metadata:
audio-topic: string
industry-naics-code-of-audio: integer
interaction-type: string
microphone-distance: string
original-media-type: string
original-mime-type: string
recording-device-name: string
recording-device-type: string
model: string
profanity-filter: boolean
sample-rate-hertz: integer
use-enhanced: boolean
</code></pre>
<p>can be set completely with the following arguments which are assumed to be executed in the given order. Note how the cursor position is adjusted to the respective structures, allowing simple field names to be used most of the time.</p>
<ul>
<li><code>-r .audio content=et</code><ul>
<li>The audio data bytes encoded as specified in <code>RecognitionConfig</code>. Note: as with all bytes fields, proto buffers use a pure binary representation, whereas JSON representations use base64.</li>
</ul>
</li>
<li>
<p><code>uri=magna</code></p>
<ul>
<li>URI that points to a file that contains audio data bytes as specified in <code>RecognitionConfig</code>. The file must not be compressed (for example, gzip). Currently, only Google Cloud Storage URIs are supported, which must be specified in the following format: <code>gs://bucket_name/object_name</code> (other URI formats return google.rpc.Code.INVALID_ARGUMENT). For more information, see <a href="https://cloud.google.com/storage/docs/reference-uris">Request URIs</a>.</li>
</ul>
</li>
<li>
<p><code>..config audio-channel-count=90</code></p>
<ul>
<li>The number of channels in the input audio data. ONLY set this for MULTI-CHANNEL recognition. Valid values for LINEAR16 and FLAC are <code>1</code>-<code>8</code>. Valid values for OGG_OPUS are &#39;1&#39;-&#39;254&#39;. Valid value for MULAW, AMR, AMR_WB and SPEEX_WITH_HEADER_BYTE is only <code>1</code>. If <code>0</code> or omitted, defaults to one channel (mono). Note: We only recognize the first channel by default. To perform independent recognition on each channel set <code>enable_separate_recognition_per_channel</code> to &#39;true&#39;.</li>
</ul>
</li>
<li><code>diarization-config enable-speaker-diarization=true</code><ul>
<li>If &#39;true&#39;, enables speaker detection for each recognized word in the top alternative of the recognition result using a speaker_tag provided in the WordInfo.</li>
</ul>
</li>
<li><code>max-speaker-count=73</code><ul>
<li>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.</li>
</ul>
</li>
<li><code>min-speaker-count=74</code><ul>
<li>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.</li>
</ul>
</li>
<li>
<p><code>speaker-tag=93</code></p>
<ul>
<li>Output only. Unused.</li>
</ul>
</li>
<li>
<p><code>.. enable-automatic-punctuation=false</code></p>
<ul>
<li>If &#39;true&#39;, 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 &#39;false&#39; value does not add punctuation to result hypotheses.</li>
</ul>
</li>
<li><code>enable-separate-recognition-per-channel=true</code><ul>
<li>This needs to be set to <code>true</code> explicitly and <code>audio_channel_count</code> &gt; 1 to get each channel recognized separately. The recognition result will contain a <code>channel_tag</code> field to state which channel that result belongs to. If this is not true, we will only recognize the first channel. The request is billed cumulatively for all channels recognized: <code>audio_channel_count</code> multiplied by the length of the audio.</li>
</ul>
</li>
<li><code>enable-word-time-offsets=true</code><ul>
<li>If <code>true</code>, the top result includes a list of words and the start and end time offsets (timestamps) for those words. If <code>false</code>, no word-level time offset information is returned. The default is <code>false</code>.</li>
</ul>
</li>
<li><code>encoding=duo</code><ul>
<li>Encoding of audio data sent in all <code>RecognitionAudio</code> messages. This field is optional for <code>FLAC</code> and <code>WAV</code> audio files and required for all other audio formats. For details, see AudioEncoding.</li>
</ul>
</li>
<li><code>language-code=ipsum</code><ul>
<li>Required. The language of the supplied audio as a <a href="https://www.rfc-editor.org/rfc/bcp/bcp47.txt">BCP-47</a> language tag. Example: &#34;en-US&#34;. See <a href="https://cloud.google.com/speech-to-text/docs/languages">Language Support</a> for a list of the currently supported language codes.</li>
</ul>
</li>
<li><code>max-alternatives=39</code><ul>
<li>Maximum number of recognition hypotheses to be returned. Specifically, the maximum number of <code>SpeechRecognitionAlternative</code> messages within each <code>SpeechRecognitionResult</code>. The server may return fewer than <code>max_alternatives</code>. Valid values are <code>0</code>-<code>30</code>. A value of <code>0</code> or <code>1</code> will return a maximum of one. If omitted, will return a maximum of one.</li>
</ul>
</li>
<li><code>metadata audio-topic=lorem</code><ul>
<li>Description of the content. Eg. &#34;Recordings of federal supreme court hearings from 2012&#34;.</li>
</ul>
</li>
<li><code>industry-naics-code-of-audio=89</code><ul>
<li>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/.</li>
</ul>
</li>
<li><code>interaction-type=eos</code><ul>
<li>The use case most closely describing the audio content to be recognized.</li>
</ul>
</li>
<li><code>microphone-distance=dolor</code><ul>
<li>The audio type that most closely describes the audio being recognized.</li>
</ul>
</li>
<li><code>original-media-type=ea</code><ul>
<li>The original media the speech was recorded on.</li>
</ul>
</li>
<li><code>original-mime-type=ipsum</code><ul>
<li>Mime type of the original audio file. For example <code>audio/m4a</code>, <code>audio/x-alaw-basic</code>, <code>audio/mp3</code>, <code>audio/3gpp</code>. A list of possible audio mime types is maintained at http://www.iana.org/assignments/media-types/media-types.xhtml#audio</li>
</ul>
</li>
<li><code>recording-device-name=invidunt</code><ul>
<li>The device used to make the recording. Examples &#39;Nexus 5X&#39; or &#39;Polycom SoundStation IP 6000&#39; or &#39;POTS&#39; or &#39;VoIP&#39; or &#39;Cardioid Microphone&#39;.</li>
</ul>
</li>
<li>
<p><code>recording-device-type=amet</code></p>
<ul>
<li>The type of device the speech was recorded with.</li>
</ul>
</li>
<li>
<p><code>.. model=duo</code></p>
<ul>
<li>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. <em>Model</em> <em>Description</em> command_and_search Best for short queries such as voice commands or voice search. phone_call Best for audio that originated from a phone call (typically recorded at an 8khz sampling rate). video Best for audio that originated from from video or includes multiple speakers. Ideally the audio is recorded at a 16khz or greater sampling rate. This is a premium model that costs more than the standard rate. default Best for audio that is not one of the specific audio models. For example, long-form audio. Ideally the audio is high-fidelity, recorded at a 16khz or greater sampling rate. </li>
</ul>
</li>
<li><code>profanity-filter=true</code><ul>
<li>If set to <code>true</code>, the server will attempt to filter out profanities, replacing all but the initial character in each filtered word with asterisks, e.g. &#34;f***&#34;. If set to <code>false</code> or omitted, profanities won&#39;t be filtered out.</li>
</ul>
</li>
<li><code>sample-rate-hertz=8</code><ul>
<li>Sample rate in Hertz of the audio data sent in all <code>RecognitionAudio</code> messages. Valid values are: 8000-48000. 16000 is optimal. For best results, set the sampling rate of the audio source to 16000 Hz. If that&#39;s not possible, use the native sample rate of the audio source (instead of re-sampling). This field is optional for FLAC and WAV audio files, but is required for all other audio formats. For details, see AudioEncoding.</li>
</ul>
</li>
<li><code>use-enhanced=true</code><ul>
<li>Set to true to use an enhanced model for speech recognition. If <code>use_enhanced</code> is set to true and the <code>model</code> field is not set, then an appropriate enhanced model is chosen if an enhanced model exists for the audio. If <code>use_enhanced</code> is true and an enhanced version of the specified model does not exist, then the speech is recognized using the standard version of the specified model.</li>
</ul>
</li>
</ul>
<h3 id="about-cursors">About Cursors</h3>
<p>The cursor position is key to comfortably set complex nested structures. The following rules apply:</p>
<ul>
<li>The cursor position is always set relative to the current one, unless the field name starts with the <code>.</code> character. Fields can be nested such as in <code>-r f.s.o</code> .</li>
<li>The cursor position is set relative to the top-level structure if it starts with <code>.</code>, e.g. <code>-r .s.s</code></li>
<li>You can also set nested fields without setting the cursor explicitly. For example, to set a value relative to the current cursor position, you would specify <code>-r struct.sub_struct=bar</code>.</li>
<li>You can move the cursor one level up by using <code>..</code>. Each additional <code>.</code> moves it up one additional level. E.g. <code>...</code> would go three levels up.</li>
</ul>
<h1 id="optional-output-flags">Optional Output Flags</h1>
<p>The method's return value a JSON encoded structure, which will be written to standard output by default.</p>
<ul>
<li><strong>-o out</strong><ul>
<li><em>out</em> specifies the <em>destination</em> to which to write the server's result to.
It will be a JSON-encoded structure.
The <em>destination</em> may be <code>-</code> to indicate standard output, or a filepath that is to contain the received bytes.
If unset, it defaults to standard output.</li>
</ul>
</li>
</ul>
<h1 id="optional-general-properties">Optional General Properties</h1>
<p>The following properties can configure any call, and are not specific to this method.</p>
<ul>
<li>
<p><strong>-p $-xgafv=string</strong></p>
<ul>
<li>V1 error format.</li>
</ul>
</li>
<li>
<p><strong>-p access-token=string</strong></p>
<ul>
<li>OAuth access token.</li>
</ul>
</li>
<li>
<p><strong>-p alt=string</strong></p>
<ul>
<li>Data format for response.</li>
</ul>
</li>
<li>
<p><strong>-p callback=string</strong></p>
<ul>
<li>JSONP</li>
</ul>
</li>
<li>
<p><strong>-p fields=string</strong></p>
<ul>
<li>Selector specifying which fields to include in a partial response.</li>
</ul>
</li>
<li>
<p><strong>-p key=string</strong></p>
<ul>
<li>API key. Your API key identifies your project and provides you with API access, quota, and reports. Required unless you provide an OAuth 2.0 token.</li>
</ul>
</li>
<li>
<p><strong>-p oauth-token=string</strong></p>
<ul>
<li>OAuth 2.0 token for the current user.</li>
</ul>
</li>
<li>
<p><strong>-p pretty-print=boolean</strong></p>
<ul>
<li>Returns response with indentations and line breaks.</li>
</ul>
</li>
<li>
<p><strong>-p quota-user=string</strong></p>
<ul>
<li>Available to use for quota purposes for server-side applications. Can be any arbitrary string assigned to a user, but should not exceed 40 characters.</li>
</ul>
</li>
<li>
<p><strong>-p upload-type=string</strong></p>
<ul>
<li>Legacy upload protocol for media (e.g. &#34;media&#34;, &#34;multipart&#34;).</li>
</ul>
</li>
<li>
<p><strong>-p upload-protocol=string</strong></p>
<ul>
<li>Upload protocol for media (e.g. &#34;raw&#34;, &#34;multipart&#34;).</li>
</ul>
</li>
</ul>
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