es.js

/*
Recorder i18n/es.js
https://github.com/xiangyuecn/Recorder

Usage: Recorder.i18n.lang="es"

Desc: Spanish, Español, 西班牙语。Esta traducción proviene principalmente de: traducción de google + traducción de Baidu, traducida del chino al español. 此翻译主要来自：google翻译+百度翻译，由中文翻译成西班牙语。

注意：请勿修改//@@打头的文本行；以下代码结构由/src/package-i18n.js自动生成，只允许在字符串中填写翻译后的文本，请勿改变代码结构；翻译的文本如果需要明确的空值，请填写"=Empty"；文本中的变量用{n}表示（n代表第几个变量），所有变量必须都出现至少一次，如果不要某变量用{n!}表示

Note: Do not modify the text lines starting with //@@; The following code structure is automatically generated by /src/package-i18n.js, only the translated text is allowed to be filled in the string, please do not change the code structure; If the translated text requires an explicit empty value, please fill in "=Empty"; Variables in the text are represented by {n} (n represents the number of variables), all variables must appear at least once, if a variable is not required, it is represented by {n!}
*/
(function(factory){
	var browser=typeof window=="object" && !!window.document;
	var win=browser?window:Object; //非浏览器环境，Recorder挂载在Object下面
	factory(win.Recorder,browser);
}(function(Recorder,isBrowser){
"use strict";
var i18n=Recorder.i18n;

//@@User Code-1 Begin 手写代码放这里 Put the handwritten code here @@

//@@User Code-1 End @@

//@@Exec i18n.lang="es";
Recorder.CLog('Import Recorder i18n lang="es"');

//i18n.alias["other-lang-key"]="es";

var putSet={lang:"es"};

i18n.data["rtl$es"]=false;
i18n.data["desc$es"]="Spanish, Español, 西班牙语。Esta traducción proviene principalmente de: traducción de google + traducción de Baidu, traducida del chino al español. 此翻译主要来自：google翻译+百度翻译，由中文翻译成西班牙语。";
//@@Exec i18n.GenerateDisplayEnglish=true;



//*************** Begin srcFile=recorder-core.js ***************
i18n.put(putSet,
[ //@@PutList 

//@@zh="重复导入{1}"
//@@en="Duplicate import {1}"
//@@Put0
 "K8zP:"+ //args: {1}
       "Importación duplicada {1}"

//@@zh="剩{1}个GetContext未close"
//@@en="There are {1} GetContext unclosed"
,"mSxV:"+ //args: {1}
       "Los {1} GetContext restantes no han sido close"

//@@zh="（注意：ctx不是running状态，rec.open和start至少要有一个在用户操作(触摸、点击等)时进行调用，否则将在rec.start时尝试进行ctx.resume，可能会产生兼容性问题(仅iOS)，请参阅文档中runningContext配置）"
//@@en=" (Note: ctx is not in the running state. At least one of rec.open and start must be called during user operations (touch, click, etc.), otherwise ctx.resume will be attempted during rec.start, which may cause compatibility issues (iOS only), please refer to the runningContext configuration in the documentation) "
,"nMIy:"+ //no args
       " (Nota: ctx no está en estado running. Se debe llamar al menos a uno de rec.open y start durante la operación del usuario (tocar, hacer clic, etc.); de lo contrario, se intentará ctx.resume durante rec.start, lo que puede causar compatibilidad problemas (solo iOS), consulte la configuración de runningContext en la documentación) "

//@@zh="Stream的采样率{1}不等于{2}，将进行采样率转换（注意：音质不会变好甚至可能变差），主要在移动端未禁用回声消除时会产生此现象，浏览器有回声消除时可能只会返回16k采样率的音频数据，"
//@@en="The sampleRate of the Stream {1} is not equal to {2}, so the sampleRate conversion will be performed (note: the sound quality will not improve and may even deteriorate). This phenomenon mainly occurs when echoCancellation is not disabled on the mobile terminal. When the browser has echoCancellation, it may only return audio data with a sampleRate of 16k. "
,"eS8i:"+ //args: {1}-{2}
       "El sampleRate {1} de la transmisión no es igual a {2} y se realizará la conversión de sampleRate (nota: la calidad del sonido no mejorará o incluso puede empeorar). Este fenómeno ocurre principalmente cuando echoCancellation no está desactivado en el terminal móvil. Cuando el navegador tiene echoCancellation, es posible que solo se devuelvan datos de audio con una frecuencia de muestreo de 16k. "

//@@zh="。由于{1}内部1秒375次回调，在移动端可能会有性能问题导致回调丢失录音变短，PC端无影响，暂不建议开启{1}。"
//@@en=". Due to 375 callbacks in 1 second in {1}, there may be performance problems on the mobile side, which may cause the callback to be lost and the recording to be shortened, but it will not affect the PC side. It is not recommended to enable {1} for now."
,"ZGlf:"+ //args: {1}
       ". Debido a las 375 devoluciones de llamadas por segundo dentro de {1}, puede haber problemas de rendimiento en el lado móvil que pueden provocar que se pierdan las devoluciones de llamadas y que la grabación sea más corta. No hay ningún impacto en el lado de la PC. No se recomienda habilitar {1} por el momento."

//@@zh="Connect采用老的{1}，"
//@@en="Connect uses the old {1}, "
,"7TU0:"+ //args: {1}
       "Connect utiliza el antiguo {1}, "

//@@zh="但已设置{1}尝试启用{2}"
//@@en="But {1} is set trying to enable {2}"
,"JwCL:"+ //args: {1}-{2}
       "Pero {1} está configurado para intentar habilitar {2}"

//@@zh="可设置{1}尝试启用{2}"
//@@en="Can set {1} try to enable {2}"
,"VGjB:"+ //args: {1}-{2}
       "Puedes configurar {1} para intentar habilitar {2}"

//@@zh="{1}未返回任何音频，恢复使用{2}"
//@@en="{1} did not return any audio, reverting to {2}"
,"MxX1:"+ //args: {1}-{2}
       "{1} no devolvió ningún audio, continúe usando {2}"

//@@zh="{1}多余回调"
//@@en="{1} redundant callback"
,"XUap:"+ //args: {1}
       "{1} devolución de llamada redundante"

//@@zh="Connect采用{1}，设置{2}可恢复老式{3}"
//@@en="Connect uses {1}, set {2} to restore old-fashioned {3}"
,"yOta:"+ //args: {1}-{3}
       "Connect usa {1}, configurar {2} puede restaurar el antiguo {3}"

//@@zh="（此浏览器不支持{1}）"
//@@en=" (This browser does not support {1}) "
,"VwPd:"+ //args: {1}
       " (Este navegador no soporta {1}) "

//@@zh="{1}未返回任何音频，降级使用{2}"
//@@en="{1} did not return any audio, downgrade to {2}"
,"vHnb:"+ //args: {1}-{2}
       "{1} no devuelve audio, se ha degradado para usar {2}"

//@@zh="{1}多余回调"
//@@en="{1} redundant callback"
,"O9P7:"+ //args: {1}
       "{1} devolución de llamada redundante"

//@@zh="Connect采用{1}，设置{2}可恢复使用{3}或老式{4}"
//@@en="Connect uses {1}, set {2} to restore to using {3} or old-fashioned {4}"
,"LMEm:"+ //args: {1}-{4}
       "Connect usa {1}, configure {2} para volver a {3} o al antiguo {4}"

//@@zh="{1}的filter采样率变了，重设滤波"
//@@en="The filter sampleRate of {1} has changed, reset the filter"
,"d48C:"+ //args: {1}
       "La frecuencia de muestreo del filtro de {1} ha cambiado y el filtro se ha restablecido"

//@@zh="{1}似乎传入了未重置chunk {2}"
//@@en="{1} seems to have passed in an unreset chunk {2}"
,"tlbC:"+ //args: {1}-{2}
       "{1} parece haber introducido chunk {2} sin restablecer"

//@@zh="{1}和{2}必须是数值"
//@@en="{1} and {2} must be number"
,"VtS4:"+ //args: {1}-{2}
       "{1} y {2} deben ser valores numéricos"

//@@zh="录音open失败："
//@@en="Recording open failed: "
,"5tWi:"+ //no args
       "Error al grabar open: "

//@@zh="open被取消"
//@@en="open cancelled"
,"dFm8:"+ //no args
       "open fue cancelado"

//@@zh="open被中断"
//@@en="open interrupted"
,"VtJO:"+ //no args
       "open fue interrumpido"

//@@zh="，可尝试使用RecordApp解决方案"
//@@en=", you can try to use the RecordApp solution "
,"EMJq:"+ //no args
       ", puedes probar la solución RecordApp"

//@@zh="不能录音："
//@@en="Cannot record: "
,"A5bm:"+ //no args
       "No se puede grabar: "

//@@zh="不支持此浏览器从流中获取录音"
//@@en="This browser does not support obtaining recordings from stream"
,"1iU7:"+ //no args
       "Este navegador no admite la recuperación de grabaciones de transmisiones"

//@@zh="从流中打开录音失败："
//@@en="Failed to open recording from stream: "
,"BTW2:"+ //no args
       "No se pudo abrir la grabación desde la transmisión: "

//@@zh="无权录音(跨域，请尝试给iframe添加麦克风访问策略，如{1})"
//@@en="No permission to record (cross domain, please try adding microphone access policy to iframe, such as: {1})"
,"Nclz:"+ //args: {1}
       "Sin permiso para grabar (entre dominios, intente agregar una política de acceso al micrófono al iframe, como {1})"

//@@zh="，无可用麦克风"
//@@en=", no microphone available"
,"jBa9:"+ //no args
       ", no hay micrófono disponible"

//@@zh="用户拒绝了录音权限"
//@@en="User denied recording permission"
,"gyO5:"+ //no args
       "Usuario denegado permiso de grabación"

//@@zh="浏览器禁止不安全页面录音，可开启https解决"
//@@en="Browser prohibits recording of unsafe pages, which can be resolved by enabling HTTPS"
,"oWNo:"+ //no args
       "El navegador prohíbe el registro de páginas no seguras, lo que se puede solucionar activando https"

//@@zh="此浏览器不支持录音"
//@@en="This browser does not support recording"
,"COxc:"+ //no args
       "Este navegador no admite la grabación"

//@@zh="发现同时多次调用open"
//@@en="It was found that open was called multiple times at the same time"
,"upb8:"+ //no args
       "Descubrí que se llamó a open varias veces al mismo tiempo"

//@@zh="录音功能无效：无音频流"
//@@en="Invalid recording: no audio stream"
,"Q1GA:"+ //no args
       "La función de grabación no funciona: no hay transmisión de audio"

//@@zh="，将尝试禁用回声消除后重试"
//@@en=", will try to disable echoCancellation and try again"
,"KxE2:"+ //no args
       ", intentaré deshabilitar echoCancellation y volveré a intentarlo"

//@@zh="请求录音权限错误"
//@@en="Error requesting recording permission"
,"xEQR:"+ //no args
       "Error al solicitar permiso de grabación"

//@@zh="无法录音："
//@@en="Unable to record: "
,"bDOG:"+ //no args
       "No se puede grabar: "

//@@zh="注意：已配置{1}参数，可能会出现浏览器不能正确选用麦克风、移动端无法启用回声消除等现象"
//@@en="Note: The {1} parameter has been configured, which may cause the browser to not correctly select the microphone, or the mobile terminal to not enable echoCancellation, etc. "
,"IjL3:"+ //args: {1}
       "Nota: Se ha configurado el parámetro {1}, lo que puede provocar que el navegador no seleccione correctamente el micrófono, o que el terminal móvil no habilite echoCancellation, etc. "

//@@zh="，未配置 {1} 时浏览器可能会自动启用回声消除，移动端未禁用回声消除时可能会降低系统播放音量（关闭录音后可恢复）和仅提供16k采样率的音频流（不需要回声消除时可明确配置成禁用来获得48k高音质的流），请参阅文档中{2}配置"
//@@en=", when {1} is not configured, the browser may automatically enable echoCancellation. When echoCancellation is not disabled on the mobile terminal, the system playback volume may be reduced (can be restored after closing the recording) and only 16k sampleRate audio stream is provided (when echoCancellation is not required, it can be explicitly configured to disable to obtain 48k high-quality stream). Please refer to the {2} configuration in the document"
,"RiWe:"+ //args: {1}-{2}
       ", cuando no se configura {1}, el navegador puede habilitar automáticamente la cancelación de eco. Cuando la cancelación de eco no está deshabilitada en el terminal móvil, el volumen de reproducción del sistema puede reducirse (se puede restaurar después de cerrar la grabación) y solo se proporciona una transmisión de audio con una frecuencia de muestreo de 16 k (cuando no se requiere la cancelación de eco, se puede configurar explícitamente para que se deshabilite y se obtenga una transmisión de alta calidad de 48 k). Consulte la configuración {2} en el documento"

//@@zh="close被忽略（因为同时open了多个rec，只有最后一个会真正close）"
//@@en="close is ignored (because multiple recs are open at the same time, only the last one will actually close)"
,"hWVz:"+ //no args
       "close se ignora (debido a que se abren varios recs al mismo tiempo, solo el último será realmente close)"

//@@zh="忽略"
//@@en="ignore"
,"UHvm:"+ //no args
       "descuido"

//@@zh="不支持{1}架构"
//@@en="{1} architecture not supported"
,"Essp:"+ //args: {1}
       "No es compatible con la arquitectura {1}"

//@@zh="{1}类型不支持设置takeoffEncodeChunk"
//@@en="{1} type does not support setting takeoffEncodeChunk"
,"2XBl:"+ //args: {1}
       "El tipo {1} no admite la configuración de takeoffEncodeChunk"

//@@zh="(未加载编码器)"
//@@en="(Encoder not loaded)"
,"LG7e:"+ //no args
       "(sin codificador cargado)"

//@@zh="{1}环境不支持实时处理"
//@@en="{1} environment does not support real-time processing"
,"7uMV:"+ //args: {1}
       "El entorno {1} no admite el procesamiento en tiempo real"

//@@zh="补偿{1}ms"
//@@en="Compensation {1}ms"
,"4Kfd:"+ //args: {1}
       "Compensación {1}ms"

//@@zh="未补偿{1}ms"
//@@en="Uncompensated {1}ms"
,"bM5i:"+ //args: {1}
       "{1} ms sin compensar"

//@@zh="回调出错是不允许的，需保证不会抛异常"
//@@en="Callback error is not allowed, you need to ensure that no exception will be thrown"
,"gFUF:"+ //no args
       "No se permiten errores en las devoluciones de llamada y se debe garantizar que no se produzcan excepciones"

//@@zh="低性能，耗时{1}ms"
//@@en="Low performance, took {1}ms"
,"2ghS:"+ //args: {1}
       "Bajo rendimiento, tarda {1} ms"

//@@zh="未进入异步前不能清除buffers"
//@@en="Buffers cannot be cleared before entering async"
,"ufqH:"+ //no args
       "Los buffers no se pueden borrar antes de entrar asíncrono"

//@@zh="start失败：未open"
//@@en="start failed: not open"
,"6WmN:"+ //no args
       "start falló: no open"

//@@zh="start 开始录音，"
//@@en="start recording, "
,"kLDN:"+ //no args
       "start, comenzar a grabar, "

//@@zh="start被中断"
//@@en="start was interrupted"
,"Bp2y:"+ //no args
       "start fue interrumpido"

//@@zh="，可能无法录音："
//@@en=", may fail to record: "
,"upkE:"+ //no args
       ", es posible que no sea posible grabar: "

//@@zh="stop 和start时差:"
//@@en="Stop and start time difference: "
,"Xq4s:"+ //no args
       "stop, diferencia horaria con start: "

//@@zh="补偿:"
//@@en="compensate: "
,"3CQP:"+ //no args
       "compensar: "

//@@zh="结束录音失败："
//@@en="Failed to stop recording: "
,"u8JG:"+ //no args
       "No se pudo finalizar la grabación: "

//@@zh="，请设置{1}"
//@@en=", please set {1}"
,"1skY:"+ //args: {1}
       ", por favor establece {1}"

//@@zh="结束录音 编码花{1}ms 音频时长{2}ms 文件大小{3}b"
//@@en="Stop recording, encoding takes {1}ms, audio duration {2}ms, file size {3}b"
,"Wv7l:"+ //args: {1}-{3}
       "Finalizar la grabación. La codificación tarda {1} ms. La duración del audio es de {2} ms. El tamaño del archivo es {3}b"

//@@zh="{1}编码器返回的不是{2}"
//@@en="{1} encoder returned not {2}"
,"Vkbd:"+ //args: {1}-{2}
       "El codificador {1} no devuelve {2}"

//@@zh="启用takeoffEncodeChunk后stop返回的blob长度为0不提供音频数据"
//@@en="After enabling takeoffEncodeChunk, the length of the blob returned by stop is 0 and no audio data is provided"
,"QWnr:"+ //no args
       "Después de habilitar takeoffEncodeChunk, la longitud del blob devuelta por stop es 0 y no se proporcionan datos de audio"

//@@zh="生成的{1}无效"
//@@en="Invalid generated {1}"
,"Sz2H:"+ //args: {1}
       "El {1} generado no es válido"

//@@zh="未开始录音"
//@@en="Recording not started"
,"wf9t:"+ //no args
       "Grabación no iniciada"

//@@zh="，开始录音前无用户交互导致AudioContext未运行"
//@@en=", No user interaction before starting recording, resulting in AudioContext not running"
,"Dl2c:"+ //no args
       ", no hay interacción del usuario antes de comenzar a grabar, lo que hace que AudioContext no se ejecute"

//@@zh="未采集到录音"
//@@en="Recording not captured"
,"Ltz3:"+ //no args
       "No se recopiló ninguna grabación"

//@@zh="未加载{1}编码器，请尝试到{2}的src/engine内找到{1}的编码器并加载"
//@@en="The {1} encoder is not loaded. Please try to find the {1} encoder in the src/engine directory of the {2} and load it"
,"xGuI:"+ //args: {1}-{2}
       "El codificador de {1} no está cargado. Intente encontrar el codificador de {1} en src/engine de {2} y cárguelo"

//@@zh="录音错误："
//@@en="Recording error: "
,"AxOH:"+ //no args
       "Error de grabación: "

//@@zh="音频buffers被释放"
//@@en="Audio buffers are released"
,"xkKd:"+ //no args
       "Se liberan los buffers de audio"

//@@zh="采样:{1} 花:{2}ms"
//@@en="Sampled: {1}, took: {2}ms"
,"CxeT:"+ //args: {1}-{2}
       "Muestra: {1} Flor: {2}ms"

//@@zh="非浏览器环境，不支持{1}"
//@@en="Non-browser environment, does not support {1}"
,"NonBrowser-1:"+ //args: {1}
       "Entorno sin navegador, no es compatible con {1}"

//@@zh="参数错误：{1}"
//@@en="Illegal argument: {1}"
,"IllegalArgs-1:"+ //args: {1}
       "Error de parámetro: {1}"

//@@zh="调用{1}需要先导入{2}"
//@@en="Calling {1} needs to import {2} first"
,"NeedImport-2:"+ //args: {1}-{2}
       "Para llamar a {1}, primero debes importar {2}"

//@@zh="不支持：{1}"
//@@en="Not support: {1}"
,"NotSupport-1:"+ //args: {1}
       "No compatible: {1}"

//@@zh="覆盖导入{1}"
//@@en="Override import {1}"
,"8HO5:"+ //args: {1}
       "Anular importación {1}"

]);
//*************** End srcFile=recorder-core.js ***************



//*************** Begin srcFile=engine/beta-amr.js ***************
i18n.put(putSet,
[ //@@PutList 

//@@zh="AMR-NB(NarrowBand)，采样率设置无效（只提供8000hz），比特率范围：{1}（默认12.2kbps），一帧20ms、{2}字节；浏览器一般不支持播放amr格式，可用Recorder.amr2wav()转码成wav播放"
//@@en="AMR-NB (NarrowBand), sampleRate setting is invalid (only 8000hz is provided), bitRate range: {1} (default 12.2kbps), one frame 20ms, {2} bytes; browsers generally do not support playing amr format, available Recorder.amr2wav() transcoding into wav playback"
//@@Put0
 "b2mN:"+ //args: {1}-{2}
       "AMR-NB (NarrowBand), la configuración sampleRate no es válida (solo se proporcionan 8000 hz), rango bitRate: {1} (predeterminado 12.2 kbps), un cuadro de 20 ms, {2} bytes; los navegadores generalmente no admiten la reproducción en formato amr, disponible Recorder.amr2wav() Transcodifica a wav para reproducción"

//@@zh="AMR Info: 和设置的不匹配{1}，已更新成{2}"
//@@en="AMR Info: does not match the set {1}, has been updated to {2}"
,"tQBv:"+ //args: {1}-{2}
       "AMR Info: no coincide con el conjunto {1}, se ha actualizado a {2}"

//@@zh="数据采样率低于{1}"
//@@en="Data sampleRate lower than {1}"
,"q12D:"+ //args: {1}
       "Los datos sampleRate están por debajo de {1}"

//@@zh="当前浏览器版本太低，无法实时处理"
//@@en="The current browser version is too low to process in real time"
,"TxjV:"+ //no args
       "La versión actual del navegador es demasiado baja y no se puede procesar en tiempo real"

//@@zh="takeoffEncodeChunk接管AMR编码器输出的二进制数据，只有首次回调数据（首帧）包含AMR头；在合并成AMR文件时，如果没有把首帧数据包含进去，则必须在文件开头添加上AMR头：Recorder.AMR.AMR_HEADER（转成二进制），否则无法播放"
//@@en="takeoffEncodeChunk takes over the binary data output by the AMR encoder, and only the first callback data (the first frame) contains the AMR header; when merging into an AMR file, if the first frame data is not included, the AMR header must be added at the beginning of the file: Recorder.AMR.AMR_HEADER (converted to binary), otherwise it cannot be played"
,"Q7p7:"+ //no args
       "takeoffEncodeChunk se hace cargo de la salida de datos binarios del codificador AMR. Solo los primeros datos de devolución de llamada (primer cuadro) contienen el encabezado AMR; al fusionarlos en un archivo AMR, si los datos del primer cuadro no están incluidos, el encabezado AMR debe agregarse en el comienzo del archivo: Recorder.AMR.AMR_HEADER (convertido a binario), de lo contrario no se puede reproducir"

//@@zh="当前环境不支持Web Worker，amr实时编码器运行在主线程中"
//@@en="The current environment does not support Web Worker, and the amr real-time encoder runs in the main thread"
,"6o9Z:"+ //no args
       "El entorno actual no es compatible con Web Worker y el codificador en tiempo real amr se ejecuta en el hilo principal"

//@@zh="amr worker剩{1}个未stop"
//@@en="amr worker left {1} unstopped"
,"yYWs:"+ //args: {1}
       "a amr worker le quedan {1} no stop"

//@@zh="amr编码器未start"
//@@en="amr encoder not started"
,"jOi8:"+ //no args
       "codificador amr no start"

]);
//*************** End srcFile=engine/beta-amr.js ***************



//*************** Begin srcFile=engine/beta-ogg.js ***************
i18n.put(putSet,
[ //@@PutList 

//@@zh="Ogg Vorbis，比特率取值16-100kbps，采样率取值无限制"
//@@en="Ogg Vorbis, bitRate 16-100kbps, sampleRate unlimited"
//@@Put0
 "O8Gn:"+ //no args
       "Ogg Vorbis, el valor de bitRate es de 16-100 kbps, el valor de sampleRate es ilimitado"

//@@zh="当前浏览器版本太低，无法实时处理"
//@@en="The current browser version is too low to process in real time"
,"5si6:"+ //no args
       "La versión actual del navegador es demasiado baja y no se puede procesar en tiempo real"

//@@zh="takeoffEncodeChunk接管OggVorbis编码器输出的二进制数据，Ogg由数据页组成，一页包含多帧音频数据（含几秒的音频，一页数据无法单独解码和播放），此编码器每次输出都是完整的一页数据，因此实时性会比较低；在合并成完整ogg文件时，必须将输出的所有数据合并到一起，否则可能无法播放，不支持截取中间一部分单独解码和播放"
//@@en="takeoffEncodeChunk takes over the binary data output by the OggVorbis encoder. Ogg is composed of data pages. One page contains multiple frames of audio data (including a few seconds of audio, and one page of data cannot be decoded and played alone). This encoder outputs a complete page of data each time, so the real-time performance will be relatively low; when merging into a complete ogg file, all the output data must be merged together, otherwise it may not be able to play, and it does not support intercepting the middle part to decode and play separately"
,"R8yz:"+ //no args
       "takeoffEncodeChunk se hace cargo de la salida de datos binarios del codificador OggVorbis. Ogg se compone de páginas de datos. Una página contiene múltiples fotogramas de datos de audio (incluidos varios segundos de audio. Una página de datos no se puede decodificar ni reproducir por separado). Cada salida de este codificador está completo. Una página de datos, por lo que el rendimiento en tiempo real será relativamente bajo; al fusionar en un archivo ogg completo, todos los datos de salida deben fusionarse; de ​​lo contrario, es posible que no se reproduzca y no se admita intercepte la parte media y decodifíquela y reprodúzcala por separado"

//@@zh="当前环境不支持Web Worker，OggVorbis实时编码器运行在主线程中"
//@@en="The current environment does not support Web Worker, and the OggVorbis real-time encoder runs in the main thread"
,"hB9D:"+ //no args
       "El entorno actual no admite Web Workers y el codificador en tiempo real OggVorbis se ejecuta en el hilo principal"

//@@zh="ogg worker剩{1}个未stop"
//@@en="There are {1} unstopped ogg workers"
,"oTiy:"+ //args: {1}
       "a ogg worker le quedan {1} no stop"

//@@zh="ogg编码器未start"
//@@en="ogg encoder not started"
,"dIpw:"+ //no args
       "codificador ogg no start"

]);
//*************** End srcFile=engine/beta-ogg.js ***************



//*************** Begin srcFile=engine/beta-webm.js ***************
i18n.put(putSet,
[ //@@PutList 

//@@zh="此浏览器不支持进行webm编码，未实现MediaRecorder"
//@@en="This browser does not support webm encoding, MediaRecorder is not implemented"
//@@Put0
 "L49q:"+ //no args
       "Este navegador no admite la codificación webm y MediaRecorder no está implementado"

//@@zh="只有比较新的浏览器支持，压缩率和mp3差不多。由于未找到对已有pcm数据进行快速编码的方法，只能按照类似边播放边收听形式把数据导入到MediaRecorder，有几秒就要等几秒。输出音频虽然可以通过比特率来控制文件大小，但音频文件中的比特率并非设定比特率，采样率由于是我们自己采样的，到这个编码器随他怎么搞"
//@@en="Only newer browsers support it, and the compression rate is similar to mp3. Since there is no way to quickly encode the existing pcm data, the data can only be imported into MediaRecorder in a similar manner while playing and listening, and it takes a few seconds to wait for a few seconds. Although the output audio can control the file size through the bitRate, the bitRate in the audio file is not the set bitRate. Since the sampleRate is sampled by ourselves, we can do whatever we want with this encoder."
,"tsTW:"+ //no args
       "Sólo los navegadores más nuevos lo admiten y la tasa de compresión es similar a la de mp3. Dado que no hay forma de codificar rápidamente los datos pcm existentes, los datos sólo se pueden importar a MediaRecorder de forma similar a la reproducción y escucha, y hay que esperar unos segundos. Aunque el tamaño del archivo de audio de salida se puede controlar mediante la velocidad de bits, la velocidad de bits en el archivo de audio no es la velocidad de bits establecida. Dado que la frecuencia de muestreo la probamos nosotros mismos, podemos hacer lo que queramos con este codificador"

//@@zh="此浏览器不支持把录音转成webm格式"
//@@en="This browser does not support converting recordings to webm format"
,"aG4z:"+ //no args
       "Este navegador no admite la conversión de grabaciones al formato webm"

//@@zh="转码webm出错：{1}"
//@@en="Error encoding webm: {1}"
,"PIX0:"+ //args: {1}
       "Error al transcodificar webm: {1}"

]);
//*************** End srcFile=engine/beta-webm.js ***************



//*************** Begin srcFile=engine/g711x.js ***************
i18n.put(putSet,
[ //@@PutList 

//@@zh="{1}；{2}音频文件无法直接播放，可用Recorder.{2}2wav()转码成wav播放；采样率比特率设置无效，固定为8000hz采样率、16位，每个采样压缩成8位存储，音频文件大小为8000字节/秒；如需任意采样率支持，请使用Recorder.{2}_encode()方法"
//@@en="{1}; {2} audio files cannot be played directly, and can be transcoded into wav by Recorder.{2}2wav(); the sampleRate bitRate setting is invalid, fixed at 8000hz sampleRate, 16 bits, each sample is compressed into 8 bits for storage, and the audio file size is 8000 bytes/second; if you need any sampleRate support, please use Recorder.{2}_encode() Method"
//@@Put0
 "d8YX:"+ //args: {1}-{2}
       "{1}; {2} El archivo de audio no se puede reproducir directamente. Puede utilizar Recorder.{2}2wav() para transcodificarlo a wav para su reproducción; la configuración de velocidad de bits de frecuencia de muestreo no es válida y está fijada en 8000 hz de muestreo velocidad, 16 bits, y cada muestra está comprimida. en un almacenamiento de 8 bits, el tamaño del archivo de audio es 8000 bytes/segundo; si necesita compatibilidad con cualquier frecuencia de muestreo, utilice el método Recorder.{2}_encode()"

//@@zh="数据采样率低于{1}"
//@@en="Data sampleRate lower than {1}"
,"29UK:"+ //args: {1}
       "Los datos sampleRate están por debajo de {1}"

//@@zh="{1}编码器未start"
//@@en="{1} encoder not started"
,"quVJ:"+ //args: {1}
       "codificador {1} no start"

]);
//*************** End srcFile=engine/g711x.js ***************



//*************** Begin srcFile=engine/mp3.js ***************
i18n.put(putSet,
[ //@@PutList 

//@@zh="采样率范围：{1}；比特率范围：{2}（不同比特率支持的采样率范围不同，小于32kbps时采样率需小于32000）"
//@@en="sampleRate range: {1}; bitRate range: {2} (the sampleRate range supported by different bitRate is different, when the bitRate is less than 32kbps, the sampleRate must be less than 32000)"
//@@Put0
 "Zm7L:"+ //args: {1}-{2}
       "rango sampleRate: {1}; rango bitRate: {2} (diferentes bitRate admiten diferentes rangos sampleRate. Cuando es inferior a 32 kbps, sampleRate debe ser inferior a 32000)"

//@@zh="{1}不在mp3支持的取值范围：{2}"
//@@en="{1} is not in the value range supported by mp3: {2}"
,"eGB9:"+ //args: {1}-{2}
       "{1} no está en el rango de valores soportado por mp3: {2}"

//@@zh="sampleRate已更新为{1}，因为{2}不在mp3支持的取值范围：{3}"
//@@en="sampleRate has been updated to {1}, because {2} is not in the value range supported by mp3: {3}"
,"zLTa:"+ //args: {1}-{3}
       "sampleRate se ha actualizado a {1} porque {2} no está en el rango de valores admitido por mp3: {3}"

//@@zh="当前浏览器版本太低，无法实时处理"
//@@en="The current browser version is too low to process in real time"
,"yhUs:"+ //no args
       "La versión actual del navegador es demasiado baja y no se puede procesar en tiempo real"

//@@zh="当前环境不支持Web Worker，mp3实时编码器运行在主线程中"
//@@en="The current environment does not support Web Worker, and the mp3 real-time encoder runs in the main thread"
,"k9PT:"+ //no args
       "El entorno actual no es compatible con Web Worker y el codificador en tiempo real mp3 se ejecuta en el hilo principal"

//@@zh="mp3 worker剩{1}个未stop"
//@@en="There are {1} unstopped mp3 workers left"
,"fT6M:"+ //args: {1}
       "a mp3 worker le quedan {1} no stop"

//@@zh="mp3编码器未start"
//@@en="mp3 encoder not started"
,"mPxH:"+ //no args
       "codificador mp3 no start"

//@@zh="和设置的不匹配{1}，已更新成{2}"
//@@en="Does not match the set {1}, has been updated to {2}"
,"uY9i:"+ //args: {1}-{2}
       "No coincide con la configuración {1}, se ha actualizado a {2}"

//@@zh="Fix移除{1}帧"
//@@en="Fix remove {1} frame"
,"iMSm:"+ //args: {1}
       "Fix elimina {1} fotogramas"

//@@zh="移除帧数过多"
//@@en="Remove too many frames"
,"b9zm:"+ //no args
       "Eliminar demasiados fotogramas"

]);
//*************** End srcFile=engine/mp3.js ***************



//*************** Begin srcFile=engine/pcm.js ***************
i18n.put(putSet,
[ //@@PutList 

//@@zh="pcm为未封装的原始音频数据，pcm音频文件无法直接播放，可用Recorder.pcm2wav()转码成wav播放；支持位数8位、16位（填在比特率里面），采样率取值无限制"
//@@en="pcm is unencapsulated original audio data, pcm audio files cannot be played directly, and can be transcoded into wav by Recorder.pcm2wav(); it supports 8-bit and 16-bit bits (fill in the bitRate), and the sampleRate is unlimited"
//@@Put0
 "fWsN:"+ //no args
       "pcm son datos de audio originales no encapsulados. Los archivos de audio Pcm no se pueden reproducir directamente. Recorder.pcm2wav() se puede utilizar para transcodificar a wav para su reproducción. Admite dígitos de 8 y 16 bits (rellene bitRate) y el valor de sampleRate es ilimitado"

//@@zh="PCM Info: 不支持{1}位，已更新成{2}位"
//@@en="PCM Info: {1} bit is not supported, has been updated to {2} bit"
,"uMUJ:"+ //args: {1}-{2}
       "PCM Info: El bit {1} no es compatible y se ha actualizado al bit {2}"

//@@zh="pcm2wav必须提供sampleRate和bitRate"
//@@en="pcm2wav must provide sampleRate and bitRate"
,"KmRz:"+ //no args
       "pcm2wav debe proporcionar sampleRate y bitRate"

//@@zh="pcm编码器未start"
//@@en="pcm encoder not started"
,"sDkA:"+ //no args
       "codificador pcm no start"

]);
//*************** End srcFile=engine/pcm.js ***************



//*************** Begin srcFile=engine/wav.js ***************
i18n.put(putSet,
[ //@@PutList 

//@@zh="支持位数8位、16位（填在比特率里面），采样率取值无限制；此编码器仅在pcm数据前加了一个44字节的wav头，编码出来的16位wav文件去掉开头的44字节即可得到pcm（注：其他wav编码器可能不是44字节）"
//@@en="Supports 8-bit and 16-bit bits (fill in the bitRate), and the sampleRate is unlimited; this encoder only adds a 44-byte wav header before the pcm data, and the encoded 16-bit wav file removes the beginning 44 bytes to get pcm (note: other wav encoders may not be 44 bytes)"
//@@Put0
 "gPSE:"+ //no args
       "Admite dígitos de 8 y 16 bits (completados en bitRate) y el valor de sampleRate es ilimitado; este codificador solo agrega un encabezado wav de 44 bytes antes de los datos pcm, y el archivo wav codificado de 16 bits elimina los 44 bits iniciales. Bytes para obtener pcm (nota: es posible que otros codificadores WAV no tengan 44 bytes)"

//@@zh="WAV Info: 不支持{1}位，已更新成{2}位"
//@@en="WAV Info: {1} bit is not supported, has been updated to {2} bit"
,"wyw9:"+ //args: {1}-{2}
       "WAV Info: El bit {1} no es compatible y se ha actualizado al bit {2}"

]);
//*************** End srcFile=engine/wav.js ***************



//*************** Begin srcFile=extensions/buffer_stream.player.js ***************
i18n.put(putSet,
[ //@@PutList 

//@@zh="getAudioSrc方法已过时：请直接使用getMediaStream然后赋值给audio.srcObject，仅允许在不支持srcObject的浏览器中调用本方法赋值给audio.src以做兼容"
//@@en="The getAudioSrc method is obsolete: please use getMediaStream directly and then assign it to audio.srcObject, it is only allowed to call this method in browsers that do not support srcObject and assign it to audio.src for compatibility"
//@@Put0
 "0XYC:"+ //no args
       "El método getAudioSrc está obsoleto: utilice getMediaStream directamente y asígnelo a audio.srcObject. Solo se permite llamar a este método en navegadores que no admiten srcObject y asignarlo a audio.src por compatibilidad"

//@@zh="start被stop终止"
//@@en="start is terminated by stop"
,"6DDt:"+ //no args
       "start es cancelado por stop"

//@@zh="{1}多次start"
//@@en="{1} repeat start"
,"I4h4:"+ //args: {1}
       "{1} se repite start"

//@@zh="浏览器不支持打开{1}"
//@@en="The browser does not support opening {1}"
,"P6Gs:"+ //args: {1}
       "El navegador no admite la apertura de {1}"

//@@zh="（注意：ctx不是running状态，start需要在用户操作(触摸、点击等)时进行调用，否则会尝试进行ctx.resume，可能会产生兼容性问题(仅iOS)，请参阅文档中runningContext配置）"
//@@en=" (Note: ctx is not in the running state, start needs to be called when the user operates (touch, click, etc.), otherwise it will try to perform ctx.resume, which may cause compatibility issues (only iOS), please refer to the runningContext configuration in the document) "
,"JwDm:"+ //no args
       " (Nota: ctx no está en estado running. Es necesario llamar a start cuando el usuario opera (tocar, hacer clic, etc.); de lo contrario, se intentará ctx.resume, lo que puede causar problemas de compatibilidad (solo iOS). Consulte la configuración runningContext en el documento) "

//@@zh="此浏览器的AudioBuffer实现不支持动态特性，采用兼容模式"
//@@en="The AudioBuffer implementation of this browser does not support dynamic features, use compatibility mode"
,"qx6X:"+ //no args
       "La implementación AudioBuffer de este navegador no admite funciones dinámicas y utiliza el modo de compatibilidad"

//@@zh="环境检测超时"
//@@en="Environment detection timeout"
,"cdOx:"+ //no args
       "Tiempo de espera de detección del entorno"

//@@zh="可能无法播放：{1}"
//@@en="Could not play: {1}"
,"S2Bu:"+ //args: {1}
       "No puede jugar: {1}"

//@@zh="input调用失败：非pcm[Int16,...]输入时，必须解码或者使用transform转换"
//@@en="input call failed: non-pcm[Int16,...] input must be decoded or converted using transform"
,"ZfGG:"+ //no args
       "Falló la llamada input: no PCM [int16,...] al ingresar, se debe decodificar o usar la conversión transform"

//@@zh="input调用失败：未提供sampleRate"
//@@en="input call failed: sampleRate not provided"
,"N4ke:"+ //no args
       "Falló la llamada input: no se proporcionó sampleRate"

//@@zh="input调用失败：data的sampleRate={1}和之前的={2}不同"
//@@en="input call failed: sampleRate={1} of data is different from previous={2}"
,"IHZd:"+ //args: {1}-{2}
       "Falló la llamada a input: sampleRate={1} de los datos es diferente de la anterior ={2}"

//@@zh="{1}未调用start方法"
//@@en="{1} did not call the start method"
,"TZPq:"+ //args: {1}
       "{1} no se llama al método start"

//@@zh="浏览器不支持音频解码"
//@@en="Browser does not support audio decoding"
,"iCFC:"+ //no args
       "El navegador no admite decodificación de audio"

//@@zh="音频解码数据必须是ArrayBuffer"
//@@en="Audio decoding data must be ArrayBuffer"
,"wE2k:"+ //no args
       "Los datos de decodificación de audio deben ser ArrayBuffer"

//@@zh="音频解码失败：{1}"
//@@en="Audio decoding failed: {1}"
,"mOaT:"+ //args: {1}
       "Falló la decodificación de audio: {1}"

]);
//*************** End srcFile=extensions/buffer_stream.player.js ***************



//*************** Begin srcFile=extensions/create-audio.nmn2pcm.js ***************
i18n.put(putSet,
[ //@@PutList 

//@@zh="符号[{1}]无效：{2}"
//@@en="Invalid symbol [{1}]: {2}"
//@@Put0
 "3RBa:"+ //args: {1}-{2}
       "Símbolo [{1}] no válido: {2}"

//@@zh="音符[{1}]无效：{2}"
//@@en="Invalid note [{1}]: {2}"
,"U212:"+ //args: {1}-{2}
       "Nota [{1}] no válido: {2}"

//@@zh="多个音时必须对齐，相差{1}ms"
//@@en="Multiple tones must be aligned, with a difference of {1}ms"
,"7qAD:"+ //args: {1}
       "Hay que alinearse a la hora de múltiples sonidos, con diferencias {1}ms"

//@@zh="祝你生日快乐"
//@@en="Happy Birthday to You"
,"QGsW:"+ //no args
       "Happy Birthday to You"

//@@zh="致爱丽丝"
//@@en="For Elise"
,"emJR:"+ //no args
       "For Elise"

//@@zh="卡农-右手简谱"
//@@en="Canon - Right Hand Notation"
,"GsYy:"+ //no args
       "Canon - símbolo de la mano derecha"

//@@zh="卡农"
//@@en="Canon"
,"bSFZ:"+ //no args
       "Canon"

]);
//*************** End srcFile=extensions/create-audio.nmn2pcm.js ***************



//*************** Begin srcFile=extensions/sonic.js ***************
i18n.put(putSet,
[ //@@PutList 

//@@zh="当前环境不支持Web Worker，不支持调用Sonic.Async"
//@@en="The current environment does not support Web Worker and does not support calling Sonic.Async"
//@@Put0
 "Ikdz:"+ //no args
       "El entorno actual no admite Web Worker, no admite llamadas a Sonic.Async"

//@@zh="sonic worker剩{1}个未flush"
//@@en="There are {1} unflushed sonic workers left"
,"IC5Y:"+ //args: {1}
       "sonic worker deja {1} sin flush"

]);
//*************** End srcFile=extensions/sonic.js ***************



//*************** Begin srcFile=app-support/app-native-support.js ***************
i18n.put(putSet,
[ //@@PutList 

//@@zh="{1}中的{2}方法未实现，请在{3}文件中或配置文件中实现此方法"
//@@en="The {2} method in {1} is not implemented, please implement this method in the {3} file or configuration file"
//@@Put0
 "WWoj:"+ //args: {1}-{3}
       "El método {2} en {1} no se implementa, por favor implemente este método en el archivo {3} o en el archivo de configuración"

//@@zh="未开始录音，但收到Native PCM数据"
//@@en="Recording does not start, but Native PCM data is received"
,"rCAM:"+ //no args
       "No se inició la grabación, pero se recibieron los datos de Native PCM"

//@@zh="检测到跨域iframe，NativeRecordReceivePCM无法注入到顶层，已监听postMessage转发兼容传输数据，请自行实现将top层接收到数据转发到本iframe（不限层），不然无法接收到录音数据"
//@@en="A cross-domain iframe is detected. NativeRecordReceivePCM cannot be injected into the top layer. It has listened to postMessage to be compatible with data transmission. Please implement it by yourself to forward the data received by the top layer to this iframe (no limit on layer), otherwise the recording data cannot be received."
,"t2OF:"+ //no args
       "Detectado iframe entre dominios, NativeRecordReceivePCM no se puede inyectar en el nivel superior, se han monitoreado los datos de transmisión compatibles con el reenvío de postMessage, por favor, realice por sí mismo el reenvío de los datos recibidos en la capa superior a este iframe (capa ilimitada), de lo contrario no se pueden recibir los datos de grabación"

//@@zh="未开始录音"
//@@en="Recording not started"
,"Z2y2:"+ //no args
       "Grabación no iniciada"

]);
//*************** End srcFile=app-support/app-native-support.js ***************



//*************** Begin srcFile=app-support/app.js ***************
i18n.put(putSet,
[ //@@PutList 

//@@zh="重复导入{1}"
//@@en="Duplicate import {1}"
//@@Put0
 "uXtA:"+ //args: {1}
       "Importación duplicada {1}"

//@@zh="注意：因为并发调用了其他录音相关方法，当前 {1} 的调用结果已被丢弃且不会有回调"
//@@en="Note: Because other recording-related methods are called concurrently, the current call result of {1} has been discarded and there will be no callback"
,"kIBu:"+ //args: {1}
       "Nota: Debido a que se llaman simultáneamente otros métodos relacionados con la grabación, el resultado de la llamada actual de {1} se ha descartado y no habrá devolución de llamada"

//@@zh="重复注册{1}"
//@@en="Duplicate registration {1}"
,"ha2K:"+ //args: {1}
       "Doble registro {1}"

//@@zh="仅清理资源"
//@@en="Clean resources only"
,"wpTL:"+ //no args
       "Solo limpiar recursos"

//@@zh="未开始录音"
//@@en="Recording not started"
,"bpvP:"+ //no args
       "Grabación no iniciada"

//@@zh="当前环境不支持实时回调，无法进行{1}"
//@@en="The current environment does not support real-time callback and cannot be performed {1}"
,"fLJD:"+ //args: {1}
       "El entorno actual no admite devoluciones de llamada en tiempo real y no se puede realizar {1}"

//@@zh="录音权限请求失败："
//@@en="Recording permission request failed: "
,"YnzX:"+ //no args
       "La solicitud de permiso de grabación falló: "

//@@zh="需先调用{1}"
//@@en="Need to call {1} first"
,"nwKR:"+ //args: {1}
       "Primero hay que llamar a {1}"

//@@zh="当前不是浏览器环境，需引入针对此平台的支持文件（{1}），或调用{2}自行实现接入"
//@@en="This is not a browser environment. You need to import support files for this platform ({1}), or call {2} to implement the access yourself."
,"citA:"+ //args: {1}-{2}
       "Actualmente no es un entorno de navegador, es necesario introducir un archivo de soporte para esta plataforma ({1}), o llamar al {2} para lograr su propio acceso."

//@@zh="开始录音失败："
//@@en="Failed to start recording: "
,"ecp9:"+ //no args
       "Falló al iniciar la grabación: "

//@@zh="不能录音："
//@@en="Cannot record: "
,"EKmS:"+ //no args
       "No se puede grabar: "

//@@zh="已开始录音"
//@@en="Recording started"
,"k7Qo:"+ //no args
       "Se ha iniciado la grabación"

//@@zh="结束录音失败："
//@@en="Failed to stop recording: "
,"Douz:"+ //no args
       "Falló al finalizar la grabación: "

//@@zh="和Start时差：{1}ms"
//@@en="Time difference from Start: {1}ms"
,"wqSH:"+ //args: {1}
       "Diferencia horaria con start: {1}ms"

//@@zh="结束录音 耗时{1}ms 音频时长{2}ms 文件大小{3}b {4}"
//@@en="Stop recording, takes {1}ms, audio duration {2}ms, file size {3}b, {4}"
,"g3VX:"+ //args: {1}-{4}
       "Terminar la grabación lleva tiempo {1}ms , duración del audio {2}ms , tamaño del archivo {3}b , {4}"

]);
//*************** End srcFile=app-support/app.js ***************

//@@User Code-2 Begin 手写代码放这里 Put the handwritten code here @@

//@@User Code-2 End @@

}));