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- {"ast":null,"code":"/*\n录音\nhttps://github.com/xiangyuecn/Recorder\n*/\n(function (factory) {\n var browser = typeof window == \"object\" && !!window.document;\n var win = browser ? window : Object; //非浏览器环境，Recorder挂载在Object下面\n factory(win, browser);\n //umd returnExports.js\n if (typeof define == 'function' && define.amd) {\n define(function () {\n return win.Recorder;\n });\n }\n ;\n if (typeof module == 'object' && module.exports) {\n module.exports = win.Recorder;\n }\n ;\n})(function (Export, isBrowser) {\n \"use strict\";\n\n var NOOP = function () {};\n var IsNum = function (v) {\n return typeof v == \"number\";\n };\n var ToJson = function (v) {\n return JSON.stringify(v);\n };\n var Recorder = function (set) {\n return new initFn(set);\n };\n var LM = Recorder.LM = \"2024-10-20 22:15\";\n var GitUrl = \"https://github.com/xiangyuecn/Recorder\";\n var RecTxt = \"Recorder\";\n var getUserMediaTxt = \"getUserMedia\";\n var srcSampleRateTxt = \"srcSampleRate\";\n var sampleRateTxt = \"sampleRate\";\n var bitRateTxt = \"bitRate\";\n var CatchTxt = \"catch\";\n var WRec2 = Export[RecTxt]; //重复加载js\n if (WRec2 && WRec2.LM == LM) {\n WRec2.CLog(WRec2.i18n.$T(\"K8zP::重复导入{1}\", 0, RecTxt), 3);\n return;\n }\n ;\n\n //是否已经打开了全局的麦克风录音，所有工作都已经准备好了，就等接收音频数据了\n Recorder.IsOpen = function () {\n var stream = Recorder.Stream;\n if (stream) {\n var tracks = Tracks_(stream),\n track = tracks[0];\n if (track) {\n var state = track.readyState;\n return state == \"live\" || state == track.LIVE;\n }\n ;\n }\n ;\n return false;\n };\n /*H5录音时的AudioContext缓冲大小。会影响H5录音时的onProcess调用速率，相对于AudioContext.sampleRate=48000时，4096接近12帧/s，调节此参数可生成比较流畅的回调动画。\n \t取值256, 512, 1024, 2048, 4096, 8192, or 16384\n \t注意，取值不能过低，2048开始不同浏览器可能回调速率跟不上造成音质问题。\n \t一般无需调整，调整后需要先close掉已打开的录音，再open时才会生效。\n */\n Recorder.BufferSize = 4096;\n //销毁已持有的所有全局资源，当要彻底移除Recorder时需要显式的调用此方法\n Recorder.Destroy = function () {\n CLog(RecTxt + \" Destroy\");\n Disconnect(); //断开可能存在的全局Stream、资源\n\n for (var k in DestroyList) {\n DestroyList[k]();\n }\n ;\n };\n var DestroyList = {};\n //登记一个需要销毁全局资源的处理方法\n Recorder.BindDestroy = function (key, call) {\n DestroyList[key] = call;\n };\n //判断浏览器是否支持录音，随时可以调用。注意：仅仅是检测浏览器支持情况，不会判断和调起用户授权，不会判断是否支持特定格式录音。\n Recorder.Support = function () {\n if (!isBrowser) return false;\n var scope = navigator.mediaDevices || {};\n if (!scope[getUserMediaTxt]) {\n scope = navigator;\n scope[getUserMediaTxt] || (scope[getUserMediaTxt] = scope.webkitGetUserMedia || scope.mozGetUserMedia || scope.msGetUserMedia);\n }\n ;\n if (!scope[getUserMediaTxt]) {\n return false;\n }\n ;\n Recorder.Scope = scope;\n if (!Recorder.GetContext()) {\n return false;\n }\n ;\n return true;\n };\n //获取AudioContext对象，如果浏览器不支持将返回null。tryNew=false时返回全局的Recorder.Ctx，Ctx.state可能是closed，仅限用于解码等操作。tryNew=true时会尝试创建新的ctx（不支持close的老浏览器依旧返回全局的），注意用完必须自己调用CloseNewCtx(ctx)关闭；注意：非用户操作（触摸、点击等）时调用返回的ctx.state可能是suspended状态，需要在用户操作时调用ctx.resume恢复成running状态，参考rec的runningContext配置\n Recorder.GetContext = function (tryNew) {\n if (!isBrowser) return null;\n var AC = window.AudioContext;\n if (!AC) {\n AC = window.webkitAudioContext;\n }\n ;\n if (!AC) {\n return null;\n }\n ;\n var ctx = Recorder.Ctx,\n isNew = 0;\n if (!ctx) {\n //241020版本后不再保留打开状态的ctx，原因是iOS不全部关闭时新的ctx有可能不正常\n //不能反复构造，低版本number of hardware contexts reached maximum (6)\n ctx = Recorder.Ctx = new AC();\n isNew = 1;\n Recorder.NewCtxs = Recorder.NewCtxs || [];\n Recorder.BindDestroy(\"Ctx\", function () {\n var ctx = Recorder.Ctx;\n if (ctx && ctx.close) {\n //能关掉就关掉，关不掉就保留着\n CloseCtx(ctx);\n Recorder.Ctx = 0;\n }\n ;\n var arr = Recorder.NewCtxs;\n Recorder.NewCtxs = [];\n for (var i = 0; i < arr.length; i++) CloseCtx(arr[i]);\n });\n }\n ;\n if (tryNew && ctx.close) {\n //没法关闭的不允许再创建\n if (!isNew) {\n if (!ctx._useC) CloseCtx(ctx); //关闭全局的，不再保留打开状态\n ctx = new AC(); //如果是上面新建的就用上面的，不然就用全新的\n }\n ;\n ctx._useC = 1;\n Recorder.NewCtxs.push(ctx);\n }\n ;\n return ctx;\n };\n //关闭新创建的AudioContext（之前老版本如果是全局的不关闭，241020版本后全部关闭）\n Recorder.CloseNewCtx = function (ctx) {\n if (ctx && ctx.close) {\n CloseCtx(ctx);\n var arr = Recorder.NewCtxs || [],\n L = arr.length;\n for (var i = 0; i < arr.length; i++) {\n if (arr[i] == ctx) {\n arr.splice(i, 1);\n break;\n }\n }\n CLog($T(\"mSxV::剩{1}个GetContext未close\", 0, L + \"-1=\" + arr.length), arr.length ? 3 : 0);\n }\n };\n var CloseCtx = function (ctx) {\n if (ctx && ctx.close && !ctx._isC) {\n ctx._isC = 1;\n if (ctx.state != \"closed\") {\n try {\n ctx.close();\n } catch (e) {\n CLog(\"ctx close err\", 1, e);\n }\n }\n }\n };\n //当AudioContext的状态是suspended时，调用resume恢复状态，但如果没有用户操作resume可能没有回调，封装解决此回调问题；check(count)返回true继续尝试resume，返回false终止任务（不回调False）\n var ResumeCtx = Recorder.ResumeCtx = function (ctx, check, True, False) {\n var isEnd = 0,\n isBind = 0,\n isLsSC = 0,\n runC = 0,\n EL = \"EventListener\",\n Tag = \"ResumeCtx \";\n var end = function (err, ok) {\n if (isBind) {\n bind();\n }\n if (!isEnd) {\n isEnd = 1; //回调结果\n err && False(err, runC);\n ok && True(runC);\n }\n if (ok) {\n //监听后续状态变化\n if (!ctx._LsSC && ctx[\"add\" + EL]) ctx[\"add\" + EL](\"statechange\", run);\n ctx._LsSC = 1;\n isLsSC = 1;\n }\n };\n var bind = function (add) {\n if (add && isBind) return;\n isBind = add ? 1 : 0;\n var types = [\"focus\", \"mousedown\", \"mouseup\", \"touchstart\", \"touchend\"];\n for (var i = 0; i < types.length; i++) window[(add ? \"add\" : \"remove\") + EL](types[i], run, true);\n };\n var run = function () {\n var sVal = ctx.state,\n spEnd = CtxSpEnd(sVal);\n if (!isEnd && !check(spEnd ? ++runC : runC)) return end(); //终止，不回调\n if (spEnd) {\n if (isLsSC) CLog(Tag + \"sc \" + sVal, 3);\n bind(1); //绑定用户事件尝试恢复\n ctx.resume().then(function () {\n //resume回调不可靠\n if (isLsSC) CLog(Tag + \"sc \" + ctx.state);\n end(0, 1);\n })[CatchTxt](function (e) {\n //出错且无法恢复\n CLog(Tag + \"error\", 1, e);\n if (!CtxSpEnd(ctx.state)) {\n end(e.message || \"error\");\n }\n });\n } else if (sVal == \"closed\") {\n if (isLsSC && !ctx._isC) CLog(Tag + \"sc \" + sVal, 1); //无法恢复，打个日志\n end(\"ctx closed\");\n } else {\n end(0, 1);\n }\n ; //running 或老的无state\n };\n run();\n };\n var CtxSpEnd = Recorder.CtxSpEnd = function (v) {\n return v == \"suspended\" || v == \"interrupted\"; //后面这个仅iOS有\n };\n var CtxState = function (ctx) {\n var v = ctx.state,\n msg = \"ctx.state=\" + v;\n if (CtxSpEnd(v)) msg += $T(\"nMIy::（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）\");\n return msg;\n };\n\n /*是否启用MediaRecorder.WebM.PCM来进行音频采集连接（如果浏览器支持的话），默认启用，禁用或者不支持时将使用AudioWorklet或ScriptProcessor来连接；MediaRecorder采集到的音频数据比其他方式更好，几乎不存在丢帧现象，所以音质明显会好很多，建议保持开启*/\n var ConnectEnableWebM = \"ConnectEnableWebM\";\n Recorder[ConnectEnableWebM] = true;\n\n /*是否启用AudioWorklet特性来进行音频采集连接（如果浏览器支持的话），默认禁用，禁用或不支持时将使用过时的ScriptProcessor来连接（如果方法还在的话），当前AudioWorklet的实现在移动端没有ScriptProcessor稳健；ConnectEnableWebM如果启用并且有效时，本参数将不起作用*/\n var ConnectEnableWorklet = \"ConnectEnableWorklet\";\n Recorder[ConnectEnableWorklet] = false;\n\n /*初始化H5音频采集连接。如果自行提供了sourceStream将只进行一次简单的连接处理。如果是普通麦克风录音，此时的Stream是全局的，Safari上断开后就无法再次进行连接使用，表现为静音，因此使用全部使用全局处理避免调用到disconnect；全局处理也有利于屏蔽底层细节，start时无需再调用底层接口，提升兼容、可靠性。*/\n var Connect = function (streamStore) {\n var bufferSize = streamStore.BufferSize || Recorder.BufferSize;\n var stream = streamStore.Stream;\n var ctx = stream._c,\n ctxSR = ctx[sampleRateTxt],\n srChunk = {};\n\n //获取音频流信息\n var tracks = Tracks_(stream),\n track = tracks[0],\n trackSet = null,\n tsMsg = \"\";\n if (track && track.getSettings) {\n trackSet = track.getSettings();\n var trackSR = trackSet[sampleRateTxt];\n if (trackSR && trackSR != ctxSR) {\n tsMsg = $T(\"eS8i::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，\", 0, trackSR, ctxSR);\n }\n }\n stream._ts = trackSet;\n CLog(tsMsg + \"Stream TrackSet: \" + ToJson(trackSet), tsMsg ? 3 : 0);\n var mediaConn = function (node) {\n var media = stream._m = ctx.createMediaStreamSource(stream);\n var ctxDest = ctx.destination,\n cmsdTxt = \"createMediaStreamDestination\";\n if (ctx[cmsdTxt]) {\n ctxDest = stream._d = ctx[cmsdTxt]();\n }\n ;\n media.connect(node);\n node.connect(ctxDest);\n };\n var isWebM,\n isWorklet,\n badInt,\n webMTips = \"\";\n var calls = stream._call;\n\n //浏览器回传的音频数据处理\n var onReceive = function (float32Arr, arrSR) {\n for (var k0 in calls) {\n //has item\n if (arrSR != ctxSR) {\n //MediaRecorder录制的采样率可能和ctx的采样率不同（16k），转换采样率方便统一处理代码也更简单，但音质不会变好，甚至可能会变差一点\n srChunk.index = 0;\n srChunk = Recorder.SampleData([float32Arr], arrSR, ctxSR, srChunk, {\n _sum: 1\n });\n var pcm = srChunk.data;\n var sum = srChunk._sum;\n } else {\n //采样率相同，不需要转换采样率\n srChunk = {};\n var size = float32Arr.length;\n var pcm = new Int16Array(size);\n var sum = 0;\n for (var j = 0; j < size; j++) {\n //floatTo16BitPCM \n var s = Math.max(-1, Math.min(1, float32Arr[j]));\n s = s < 0 ? s * 0x8000 : s * 0x7FFF;\n pcm[j] = s;\n sum += Math.abs(s);\n }\n ;\n }\n ;\n for (var k in calls) {\n calls[k](pcm, sum);\n }\n ;\n return;\n }\n ;\n };\n var scriptProcessor = \"ScriptProcessor\"; //一堆字符串名字，有利于压缩js\n var audioWorklet = \"audioWorklet\";\n var recAudioWorklet = RecTxt + \" \" + audioWorklet;\n var RecProc = \"RecProc\";\n var MediaRecorderTxt = \"MediaRecorder\";\n var MRWebMPCM = MediaRecorderTxt + \".WebM.PCM\";\n\n //===================连接方式三=========================\n //古董级别的 ScriptProcessor 处理，目前所有浏览器均兼容，虽然是过时的方法，但更稳健，移动端性能比AudioWorklet强\n var oldFn = ctx.createScriptProcessor || ctx.createJavaScriptNode;\n var oldIsBest = $T(\"ZGlf::。由于{1}内部1秒375次回调，在移动端可能会有性能问题导致回调丢失录音变短，PC端无影响，暂不建议开启{1}。\", 0, audioWorklet);\n var oldScript = function () {\n isWorklet = stream.isWorklet = false;\n _Disconn_n(stream);\n CLog($T(\"7TU0::Connect采用老的{1}，\", 0, scriptProcessor) + i18n.get(Recorder[ConnectEnableWorklet] ? $T(\"JwCL::但已设置{1}尝试启用{2}\", 2) : $T(\"VGjB::可设置{1}尝试启用{2}\", 2), [RecTxt + \".\" + ConnectEnableWorklet + \"=true\", audioWorklet]) + webMTips + oldIsBest, 3);\n var process = stream._p = oldFn.call(ctx, bufferSize, 1, 1); //单声道，省的数据处理复杂\n mediaConn(process);\n process.onaudioprocess = function (e) {\n var arr = e.inputBuffer.getChannelData(0);\n onReceive(arr, ctxSR);\n };\n };\n\n //===================连接方式二=========================\n var connWorklet = function () {\n //尝试开启AudioWorklet处理\n isWebM = stream.isWebM = false;\n _Disconn_r(stream);\n isWorklet = stream.isWorklet = !oldFn || Recorder[ConnectEnableWorklet];\n var AwNode = window.AudioWorkletNode;\n if (!(isWorklet && ctx[audioWorklet] && AwNode)) {\n oldScript(); //被禁用 或 不支持，直接使用老的\n return;\n }\n ;\n var clazzUrl = function () {\n var xf = function (f) {\n return f.toString().replace(/^function|DEL_/g, \"\").replace(/\\$RA/g, recAudioWorklet);\n };\n var clazz = 'class ' + RecProc + ' extends AudioWorkletProcessor{';\n clazz += \"constructor \" + xf(function (option) {\n DEL_super(option);\n var This = this,\n bufferSize = option.processorOptions.bufferSize;\n This.bufferSize = bufferSize;\n This.buffer = new Float32Array(bufferSize * 2); //乱给size搞乱缓冲区不管\n This.pos = 0;\n This.port.onmessage = function (e) {\n if (e.data.kill) {\n This.kill = true;\n $C.log(\"$RA kill call\");\n }\n };\n $C.log(\"$RA .ctor call\", option);\n });\n\n //https://developer.mozilla.org/en-US/docs/Web/API/AudioWorkletProcessor/process 每次回调128个采样数据，1秒375次回调，高频导致移动端性能问题，结果就是回调次数缺斤少两，进而导致丢失数据，PC端似乎没有性能问题\n clazz += \"process \" + xf(function (input, b, c) {\n //需要等到ctx激活后才会有回调\n var This = this,\n bufferSize = This.bufferSize;\n var buffer = This.buffer,\n pos = This.pos;\n input = (input[0] || [])[0] || [];\n if (input.length) {\n buffer.set(input, pos);\n pos += input.length;\n var len = ~~(pos / bufferSize) * bufferSize;\n if (len) {\n this.port.postMessage({\n val: buffer.slice(0, len)\n });\n var more = buffer.subarray(len, pos);\n buffer = new Float32Array(bufferSize * 2);\n buffer.set(more);\n pos = more.length;\n This.buffer = buffer;\n }\n This.pos = pos;\n }\n return !This.kill;\n });\n clazz += '}' + 'try{' + 'registerProcessor(\"' + RecProc + '\", ' + RecProc + ')' + '}catch(e){$C.error(\"' + recAudioWorklet + ' Reg Error\",e)}';\n clazz = clazz.replace(/\\$C\\./g, \"console.\"); //一些编译器会文本替换日志函数\n //URL.createObjectURL 本地有些浏览器会报 Not allowed to load local resource，直接用dataurl\n return \"data:text/javascript;base64,\" + btoa(unescape(encodeURIComponent(clazz)));\n };\n var awNext = function () {\n //可以继续，没有调用断开\n return isWorklet && stream._na;\n };\n var nodeAlive = stream._na = function () {\n //start时会调用，只要没有收到数据就断定AudioWorklet有问题，恢复用老的\n if (badInt !== \"\") {\n //没有回调过数据\n clearTimeout(badInt);\n badInt = setTimeout(function () {\n badInt = 0;\n if (awNext()) {\n CLog($T(\"MxX1::{1}未返回任何音频，恢复使用{2}\", 0, audioWorklet, scriptProcessor), 3);\n oldFn && oldScript(); //未来没有老的，可能是误判\n }\n ;\n }, 500);\n }\n ;\n };\n var createNode = function () {\n if (!awNext()) return;\n var node = stream._n = new AwNode(ctx, RecProc, {\n processorOptions: {\n bufferSize: bufferSize\n }\n });\n mediaConn(node);\n node.port.onmessage = function (e) {\n if (badInt) {\n clearTimeout(badInt);\n badInt = \"\";\n }\n ;\n if (awNext()) {\n onReceive(e.data.val, ctxSR);\n } else if (!isWorklet) {\n CLog($T(\"XUap::{1}多余回调\", 0, audioWorklet), 3);\n }\n ;\n };\n CLog($T(\"yOta::Connect采用{1}，设置{2}可恢复老式{3}\", 0, audioWorklet, RecTxt + \".\" + ConnectEnableWorklet + \"=false\", scriptProcessor) + webMTips + oldIsBest, 3);\n };\n\n //如果start时的resume和下面的构造node同时进行，将会导致部分浏览器崩溃 (STATUS_ACCESS_VIOLATION)，源码assets中 ztest_chrome_bug_AudioWorkletNode.html 可测试。所以，将所有代码套到resume里面（不管catch），避免出现这个问题\n var ctxOK = function () {\n if (!awNext()) return;\n if (ctx[RecProc]) {\n createNode();\n return;\n }\n ;\n var url = clazzUrl();\n ctx[audioWorklet].addModule(url).then(function (e) {\n if (!awNext()) return;\n ctx[RecProc] = 1;\n createNode();\n if (badInt) {\n //重新计时\n nodeAlive();\n }\n ;\n })[CatchTxt](function (e) {\n //fix 关键字，保证catch压缩时保持字符串形式\n CLog(audioWorklet + \".addModule Error\", 1, e);\n awNext() && oldScript();\n });\n };\n ResumeCtx(ctx, function () {\n return awNext();\n }, ctxOK, ctxOK);\n };\n\n //===================连接方式一=========================\n var connWebM = function () {\n //尝试开启MediaRecorder录制webm+pcm处理\n var MR = window[MediaRecorderTxt];\n var onData = \"ondataavailable\";\n var webmType = \"audio/webm; codecs=pcm\";\n isWebM = stream.isWebM = Recorder[ConnectEnableWebM];\n var supportMR = MR && onData in MR.prototype && MR.isTypeSupported(webmType);\n webMTips = supportMR ? \"\" : $T(\"VwPd::（此浏览器不支持{1}）\", 0, MRWebMPCM);\n if (!isWebM || !supportMR) {\n connWorklet(); //被禁用 或 不支持MediaRecorder 或 不支持webm+pcm\n return;\n }\n var mrNext = function () {\n //可以继续，没有调用断开\n return isWebM && stream._ra;\n };\n stream._ra = function () {\n //start时会调用，只要没有收到数据就断定MediaRecorder有问题，降级处理\n if (badInt !== \"\") {\n //没有回调过数据\n clearTimeout(badInt);\n badInt = setTimeout(function () {\n //badInt=0; 保留给nodeAlive继续判断\n if (mrNext()) {\n CLog($T(\"vHnb::{1}未返回任何音频，降级使用{2}\", 0, MediaRecorderTxt, audioWorklet), 3);\n connWorklet();\n }\n ;\n }, 500);\n }\n ;\n };\n var mrSet = Object.assign({\n mimeType: webmType\n }, Recorder.ConnectWebMOptions);\n var mr = stream._r = new MR(stream, mrSet);\n var webmData = stream._rd = {};\n mr[onData] = function (e) {\n //提取webm中的pcm数据，提取失败就等着badInt超时降级处理\n var reader = new FileReader();\n reader.onloadend = function () {\n if (mrNext()) {\n var f32arr = WebM_Extract(new Uint8Array(reader.result), webmData);\n if (!f32arr) return;\n if (f32arr == -1) {\n //无法提取，立即降级\n connWorklet();\n return;\n }\n ;\n if (badInt) {\n clearTimeout(badInt);\n badInt = \"\";\n }\n ;\n onReceive(f32arr, webmData.webmSR);\n } else if (!isWebM) {\n CLog($T(\"O9P7::{1}多余回调\", 0, MediaRecorderTxt), 3);\n }\n ;\n };\n reader.readAsArrayBuffer(e.data);\n };\n mr.start(~~(bufferSize / 48)); //按48k时的回调间隔\n CLog($T(\"LMEm::Connect采用{1}，设置{2}可恢复使用{3}或老式{4}\", 0, MRWebMPCM, RecTxt + \".\" + ConnectEnableWebM + \"=false\", audioWorklet, scriptProcessor));\n };\n connWebM();\n };\n var ConnAlive = function (stream) {\n if (stream._na) stream._na(); //检查AudioWorklet连接是否有效，无效就回滚到老的ScriptProcessor\n if (stream._ra) stream._ra(); //检查MediaRecorder连接是否有效，无效就降级处理\n };\n var _Disconn_n = function (stream) {\n stream._na = null;\n if (stream._n) {\n stream._n.port.postMessage({\n kill: true\n });\n stream._n.disconnect();\n stream._n = null;\n }\n ;\n };\n var _Disconn_r = function (stream) {\n stream._ra = null;\n if (stream._r) {\n try {\n stream._r.stop();\n } catch (e) {\n CLog(\"mr stop err\", 1, e);\n }\n stream._r = null;\n }\n ;\n };\n var Disconnect = function (streamStore) {\n streamStore = streamStore || Recorder;\n var isGlobal = streamStore == Recorder;\n var stream = streamStore.Stream;\n if (stream) {\n if (stream._m) {\n stream._m.disconnect();\n stream._m = null;\n }\n ;\n if (!stream._RC && stream._c) {\n //提供的runningContext不处理\n Recorder.CloseNewCtx(stream._c);\n }\n ;\n stream._RC = null;\n stream._c = null;\n if (stream._d) {\n StopS_(stream._d.stream);\n stream._d = null;\n }\n ;\n if (stream._p) {\n stream._p.disconnect();\n stream._p.onaudioprocess = stream._p = null;\n }\n ;\n _Disconn_n(stream);\n _Disconn_r(stream);\n if (isGlobal) {\n //全局的时候，要把流关掉（麦克风），直接提供的流不处理\n StopS_(stream);\n }\n ;\n }\n ;\n streamStore.Stream = 0;\n };\n //关闭一个音频流\n var StopS_ = Recorder.StopS_ = function (stream) {\n var tracks = Tracks_(stream);\n for (var i = 0; i < tracks.length; i++) {\n var track = tracks[i];\n track.stop && track.stop();\n }\n ;\n stream.stop && stream.stop();\n };\n //获取流中的所有轨道\n var Tracks_ = function (stream) {\n var arr1 = 0,\n arr2 = 0,\n arr = [];\n //stream.getTracks() 得到的sourceStream还得去排序 不然第一个可能是video\n if (stream.getAudioTracks) {\n arr1 = stream.getAudioTracks();\n arr2 = stream.getVideoTracks();\n }\n if (!arr1) {\n arr1 = stream.audioTracks;\n arr2 = stream.videoTracks;\n }\n for (var i = 0, L = arr1 ? arr1.length : 0; i < L; i++) arr.push(arr1[i]); //音频放前面，方便取[0]\n for (var i = 0, L = arr2 ? arr2.length : 0; i < L; i++) arr.push(arr2[i]);\n return arr;\n };\n\n /*对pcm数据的采样率进行转换\n pcmDatas: [[Int16,...]] pcm片段列表，二维数组里面是Int16Array，也可传Float32Array（会转成Int16Array）\n pcmSampleRate:48000 pcm数据的采样率\n newSampleRate:16000 需要转换成的采样率，241020版本后支持转成任意采样率，之前老版本newSampleRate>=pcmSampleRate时不会进行任何处理，小于时会进行重新采样\n prevChunkInfo:{} 可选，上次调用时的返回值，用于连续转换，本次调用将从上次结束位置开始进行处理。或可自行定义一个ChunkInfo从pcmDatas指定的位置开始进行转换\n option:{ 可选，配置项\n \t\tframeSize:123456 帧大小，每帧的PCM Int16的数量，采样率转换后的pcm长度为frameSize的整数倍，用于连续转换。目前仅在mp3格式时才有用，frameSize取值为1152，这样编码出来的mp3时长和pcm的时长完全一致，否则会因为mp3最后一帧录音不够填满时添加填充数据导致mp3的时长变长。\n \t\tframeType:\"\" 帧类型，一般为rec.set.type，提供此参数时无需提供frameSize，会自动使用最佳的值给frameSize赋值，目前仅支持mp3=1152(MPEG1 Layer3的每帧采采样数)，其他类型=1。\n \t\t\t以上两个参数用于连续转换时使用，最多使用一个，不提供时不进行帧的特殊处理，提供时必须同时提供prevChunkInfo才有作用。最后一段数据处理时无需提供帧大小以便输出最后一丁点残留数据。\n \t}\n \n 返回ChunkInfo:{\n \t//可定义，从指定位置开始转换到结尾\n \tindex:0 pcmDatas已处理到的索引；比如每次都是单个pcm需要连续处理时，可每次调用前重置成0，pcmDatas仅需传入`[pcm]`固定一个元素\n \toffset:0.0 已处理到的index对应的pcm中的偏移的下一个位置（提升采样率时为结果的pcm）\n \traisePrev:null 提升采样率时的前一个pcm结果采样值\n \t\n \t//可定义，指定的一个滤波配置：默认使用Recorder.IIRFilter低通滤波（可有效抑制混叠产生的杂音，小采样率大于高采样率的75%时不默认滤波），如果提供了配置但fn为null时将不滤波；sr、srn为此滤波函数对应的初始化采样率，当采样率和参数的不一致时将重新设为默认函数\n \tfilter:null||{fn:fn(sample),sr:pcmSampleRate,srn:newSampleRate}\n \t\n \t//仅作为返回值\n \tframeNext:null||[Int16,...] 下一帧的部分数据，frameSize设置了的时候才可能会有\n \tsampleRate:16000 结果的采样率=newSampleRate，老版本<=newSampleRate\n \tdata:[Int16,...] 转换后的PCM结果；如果是连续转换，并且pcmDatas中并没有新数据时，data的长度可能为0\n }\n */\n Recorder.SampleData = function (pcmDatas, pcmSampleRate, newSampleRate, prevChunkInfo, option) {\n var Txt = \"SampleData\";\n prevChunkInfo || (prevChunkInfo = {});\n var index = prevChunkInfo.index || 0;\n var offset = prevChunkInfo.offset || 0;\n var raisePrev = prevChunkInfo.raisePrev || 0;\n var filter = prevChunkInfo.filter;\n if (filter && filter.fn && (filter.sr && filter.sr != pcmSampleRate || filter.srn && filter.srn != newSampleRate)) {\n filter = null;\n CLog($T(\"d48C::{1}的filter采样率变了，重设滤波\", 0, Txt), 3);\n }\n ;\n if (!filter) {\n //采样率差距比较大才开启低通滤波\n if (newSampleRate <= pcmSampleRate) {\n //降低采样率或不变，最高频率用新采样率频率的3/4\n var freq = newSampleRate > pcmSampleRate * 3 / 4 ? 0 : newSampleRate / 2 * 3 / 4;\n filter = {\n fn: freq ? Recorder.IIRFilter(true, pcmSampleRate, freq) : 0\n };\n } else {\n //提升采样率，最高频率用原始采样率频率的3/4\n var freq = pcmSampleRate > newSampleRate * 3 / 4 ? 0 : pcmSampleRate / 2 * 3 / 4;\n filter = {\n fn: freq ? Recorder.IIRFilter(true, newSampleRate, freq) : 0\n };\n }\n ;\n }\n ;\n filter.sr = pcmSampleRate;\n filter.srn = newSampleRate;\n var filterFn = filter.fn;\n var frameNext = prevChunkInfo.frameNext || [];\n option || (option = {});\n var frameSize = option.frameSize || 1;\n if (option.frameType) {\n frameSize = option.frameType == \"mp3\" ? 1152 : 1;\n }\n ;\n var useSum = option._sum,\n _sum = 0; //内部用的，sum不考虑配置了frame\n\n var nLen = pcmDatas.length;\n if (index > nLen + 1) {\n CLog($T(\"tlbC::{1}似乎传入了未重置chunk {2}\", 0, Txt, index + \">\" + nLen), 3);\n }\n ;\n var size = 0;\n for (var i = index; i < nLen; i++) {\n size += pcmDatas[i].length;\n }\n ;\n\n //采样 https://www.cnblogs.com/blqw/p/3782420.html\n var step = pcmSampleRate / newSampleRate;\n if (step > 1) {\n //新采样低于录音采样，进行抽样\n size = Math.max(0, size - Math.floor(offset));\n size = Math.floor(size / step);\n } else if (step < 1) {\n //新采样高于录音采样，插值处理\n var raiseStep = 1 / step;\n size = Math.floor(size * raiseStep);\n }\n ;\n size += frameNext.length;\n var res = new Int16Array(size);\n var idx = 0;\n //添加上一次不够一帧的剩余数据\n for (var i = 0; i < frameNext.length; i++) {\n res[idx] = frameNext[i];\n idx++;\n }\n ;\n //处理数据\n for (; index < nLen; index++) {\n var o = pcmDatas[index],\n isF32 = o instanceof Float32Array;\n var i = offset,\n il = o.length;\n var F = filterFn && filterFn.Embed,\n F1 = 0,\n F2 = 0,\n Fx = 0,\n Fy = 0; //低通滤波后的数据\n\n if (step < 1) {\n //提升采样率\n var idx1 = idx + i,\n prev = raisePrev;\n for (var i0 = 0; i0 < il; i0++) {\n var oVal = o[i0];\n if (isF32) {\n //floatTo16BitPCM \n oVal = Math.max(-1, Math.min(1, oVal));\n oVal = oVal < 0 ? oVal * 0x8000 : oVal * 0x7FFF;\n }\n var pos = Math.floor(idx1);\n idx1 += raiseStep;\n var end = Math.floor(idx1);\n\n //简单的从prev平滑填充到cur，有效减少转换引入的杂音\n var n = (oVal - prev) / (end - pos);\n for (var j = 1; pos < end; pos++, j++) {\n var s = Math.floor(prev + j * n);\n if (F) {\n //IIRFilter代码内置，比函数调用快4倍\n Fx = s;\n Fy = F.b0 * Fx + F.b1 * F.x1 + F.b0 * F.x2 - F.a1 * F.y1 - F.a2 * F.y2;\n F.x2 = F.x1;\n F.x1 = Fx;\n F.y2 = F.y1;\n F.y1 = Fy;\n s = Fy;\n } else {\n s = filterFn ? filterFn(s) : s;\n }\n if (s > 0x7FFF) s = 0x7FFF;else if (s < -0x8000) s = -0x8000; //Int16越界处理\n if (useSum) _sum += Math.abs(s);\n res[pos] = s;\n idx++;\n }\n ;\n prev = raisePrev = oVal;\n i += raiseStep; //插值\n }\n offset = i % 1;\n continue;\n }\n ;\n //降低采样率或不变\n for (var i0 = 0, i2 = 0; i0 < il; i0++, i2++) {\n if (i2 < il) {\n var oVal = o[i2];\n if (isF32) {\n //floatTo16BitPCM \n oVal = Math.max(-1, Math.min(1, oVal));\n oVal = oVal < 0 ? oVal * 0x8000 : oVal * 0x7FFF;\n }\n if (F) {\n //IIRFilter代码内置，比函数调用快4倍\n Fx = oVal;\n Fy = F.b0 * Fx + F.b1 * F.x1 + F.b0 * F.x2 - F.a1 * F.y1 - F.a2 * F.y2;\n F.x2 = F.x1;\n F.x1 = Fx;\n F.y2 = F.y1;\n F.y1 = Fy;\n } else {\n Fy = filterFn ? filterFn(oVal) : oVal;\n }\n }\n F1 = F2;\n F2 = Fy;\n if (i2 == 0) {\n i0--;\n continue;\n } //首次只计算o[0]\n\n //https://www.cnblogs.com/xiaoqi/p/6993912.html\n //当前点=当前点+到后面一个点之间的增量，音质比直接简单抽样好些\n var before = Math.floor(i);\n if (i0 != before) continue;\n var after = Math.ceil(i);\n var atPoint = i - before;\n var beforeVal = F1;\n var afterVal = after < il ? F2 : beforeVal; //后个点越界了，忽略不计\n var val = beforeVal + (afterVal - beforeVal) * atPoint;\n if (val > 0x7FFF) val = 0x7FFF;else if (val < -0x8000) val = -0x8000; //Int16越界处理\n if (useSum) _sum += Math.abs(val);\n res[idx] = val;\n idx++;\n i += step; //抽样\n }\n ;\n offset = Math.max(0, i - il); //不太可能出现负数\n }\n ;\n if (step < 1 && idx + 1 == size) {\n //提升采样率时可能缺1个，直接删除结尾1个\n size--;\n res = new Int16Array(res.buffer.slice(0, size * 2));\n }\n ;\n if (idx - 1 != size && idx != size) CLog(Txt + \" idx:\" + idx + \" != size:\" + size, 3); //越界1个\n\n //帧处理\n frameNext = null;\n var frameNextSize = size % frameSize;\n if (frameNextSize > 0) {\n var u8Pos = (size - frameNextSize) * 2;\n frameNext = new Int16Array(res.buffer.slice(u8Pos));\n res = new Int16Array(res.buffer.slice(0, u8Pos));\n }\n ;\n var obj = {\n index: index,\n offset: offset,\n raisePrev: raisePrev,\n filter: filter,\n frameNext: frameNext,\n sampleRate: newSampleRate,\n data: res\n };\n if (useSum) obj._sum = _sum;\n return obj;\n };\n\n /*IIR低通、高通滤波，移植自：https://gitee.com/52jian/digital-audio-filter AudioFilter.java\n \tuseLowPass: true或false，true为低通滤波，false为高通滤波\n \tsampleRate: 待处理pcm的采样率\n \tfreq: 截止频率Hz，最大频率为sampleRate/2，低通时会切掉高于此频率的声音，高通时会切掉低于此频率的声音，注意滤波并非100%的切掉不需要的声音，而是减弱频率对应的声音，离截止频率越远对应声音减弱越厉害，离截止频率越近声音就几乎无衰减\n \t返回的是一个函数，用此函数对pcm的每个采样值按顺序进行处理即可（不同pcm不可共用）；注意此函数返回值可能会越界超过Int16范围，自行限制一下即可：Math.min(Math.max(val,-0x8000),0x7FFF)\n 可重新赋值一个函数，来改变Recorder的默认行为，比如SampleData中的低通滤波*/\n Recorder.IIRFilter = function (useLowPass, sampleRate, freq) {\n var ov = 2 * Math.PI * freq / sampleRate;\n var sn = Math.sin(ov);\n var cs = Math.cos(ov);\n var alpha = sn / 2;\n var a0 = 1 + alpha;\n var a1 = -2 * cs / a0;\n var a2 = (1 - alpha) / a0;\n if (useLowPass) {\n var b0 = (1 - cs) / 2 / a0;\n var b1 = (1 - cs) / a0;\n } else {\n var b0 = (1 + cs) / 2 / a0;\n var b1 = -(1 + cs) / a0;\n }\n var x1 = 0,\n x2 = 0,\n y = 0,\n y1 = 0,\n y2 = 0;\n var fn = function (x) {\n y = b0 * x + b1 * x1 + b0 * x2 - a1 * y1 - a2 * y2;\n x2 = x1;\n x1 = x;\n y2 = y1;\n y1 = y;\n return y;\n };\n fn.Embed = {\n x1: 0,\n x2: 0,\n y1: 0,\n y2: 0,\n b0: b0,\n b1: b1,\n a1: a1,\n a2: a2\n };\n return fn;\n };\n\n /*计算音量百分比的一个方法\n pcmAbsSum: pcm Int16所有采样的绝对值的和\n pcmLength: pcm长度\n 返回值：0-100，主要当做百分比用\n 注意：这个不是分贝，因此没用volume当做名称*/\n Recorder.PowerLevel = function (pcmAbsSum, pcmLength) {\n /*计算音量 https://blog.csdn.net/jody1989/article/details/73480259\n 更高灵敏度算法:\n \t限定最大感应值10000\n \t\t线性曲线：低音量不友好\n \t\t\tpower/10000*100 \n \t\t对数曲线：低音量友好，但需限定最低感应值\n \t\t\t(1+Math.log10(power/10000))*100\n */\n var power = pcmAbsSum / pcmLength || 0; //NaN\n var level;\n if (power < 1251) {\n //1250的结果10%，更小的音量采用线性取值\n level = Math.round(power / 1250 * 10);\n } else {\n level = Math.round(Math.min(100, Math.max(0, (1 + Math.log(power / 10000) / Math.log(10)) * 100)));\n }\n ;\n return level;\n };\n\n /*计算音量，单位dBFS（满刻度相对电平）\n maxSample: 为16位pcm采样的绝对值中最大的一个（计算峰值音量），或者为pcm中所有采样的绝对值的平局值\n 返回值：-100~0 （最大值0dB，最小值-100代替-∞）\n */\n Recorder.PowerDBFS = function (maxSample) {\n var val = Math.max(0.1, maxSample || 0),\n Pref = 0x7FFF;\n val = Math.min(val, Pref);\n //https://www.logiclocmusic.com/can-you-tell-the-decibel/\n //https://blog.csdn.net/qq_17256689/article/details/120442510\n val = 20 * Math.log(val / Pref) / Math.log(10);\n return Math.max(-100, Math.round(val));\n };\n\n //带时间的日志输出，可设为一个空函数来屏蔽日志输出\n //CLog(msg,errOrLogMsg, logMsg...) err为数字时代表日志类型1:error 2:log默认 3:warn，否则当做内容输出，第一个参数不能是对象因为要拼接时间，后面可以接无数个输出参数\n Recorder.CLog = function (msg, err) {\n if (typeof console != \"object\") return;\n var now = new Date();\n var t = (\"0\" + now.getMinutes()).substr(-2) + \":\" + (\"0\" + now.getSeconds()).substr(-2) + \".\" + (\"00\" + now.getMilliseconds()).substr(-3);\n var recID = this && this.envIn && this.envCheck && this.id;\n var arr = [\"[\" + t + \" \" + RecTxt + (recID ? \":\" + recID : \"\") + \"]\" + msg];\n var a = arguments,\n cwe = Recorder.CLog;\n var i = 2,\n fn = cwe.log || console.log;\n if (IsNum(err)) {\n fn = err == 1 ? cwe.error || console.error : err == 3 ? cwe.warn || console.warn : fn;\n } else {\n i = 1;\n }\n ;\n for (; i < a.length; i++) {\n arr.push(a[i]);\n }\n ;\n if (IsLoser) {\n //古董浏览器，仅保证基本的可执行不代码异常\n fn && fn(\"[IsLoser]\" + arr[0], arr.length > 1 ? arr : \"\");\n } else {\n fn.apply(console, arr);\n }\n ;\n };\n var CLog = function () {\n Recorder.CLog.apply(this, arguments);\n };\n var IsLoser = true;\n try {\n IsLoser = !console.log.apply;\n } catch (e) {}\n ;\n var ID = 0;\n function initFn(set) {\n var This = this;\n This.id = ++ID;\n\n //如果开启了流量统计，这里将发送一个图片请求\n Traffic();\n var o = {\n type: \"mp3\" //输出类型：mp3,wav，wav输出文件尺寸超大不推荐使用，但mp3编码支持会导致js文件超大，如果不需支持mp3可以使js文件大幅减小\n //,bitRate:16 //比特率 wav:16或8位，MP3：8kbps 1k/s，8kbps 2k/s 录音文件很小\n\n //,sampleRate:16000 //采样率，wav格式大小=sampleRate*时间；mp3此项对低比特率有影响，高比特率几乎无影响。\n //wav任意值，mp3取值范围：48000, 44100, 32000, 24000, 22050, 16000, 12000, 11025, 8000\n //采样率参考https://www.cnblogs.com/devin87/p/mp3-recorder.html\n ,\n\n onProcess: NOOP //fn(buffers,powerLevel,bufferDuration,bufferSampleRate,newBufferIdx,asyncEnd) buffers=[[Int16,...],...]：缓冲的PCM数据，为从开始录音到现在的所有pcm片段；powerLevel：当前缓冲的音量级别0-100，bufferDuration：已缓冲时长，bufferSampleRate：缓冲使用的采样率（当type支持边录边转码(Worker)时，此采样率和设置的采样率相同，否则不一定相同）；newBufferIdx:本次回调新增的buffer起始索引；asyncEnd:fn() 如果onProcess是异步的(返回值为true时)，处理完成时需要调用此回调，如果不是异步的请忽略此参数，此方法回调时必须是真异步（不能真异步时需用setTimeout包裹）。onProcess返回值：如果返回true代表开启异步模式，在某些大量运算的场合异步是必须的，必须在异步处理完成时调用asyncEnd(不能真异步时需用setTimeout包裹)，在onProcess执行后新增的buffer会全部替换成空数组，因此本回调开头应立即将newBufferIdx到本次回调结尾位置的buffer全部保存到另外一个数组内，处理完成后写回buffers中本次回调的结尾位置。\n\n //*******高级设置******\n //,sourceStream:MediaStream Object\n //可选直接提供一个媒体流，从这个流中录制、实时处理音频数据（当前Recorder实例独享此流）；不提供时为普通的麦克风录音，由getUserMedia提供音频流（所有Recorder实例共享同一个流）\n //比如：audio、video标签dom节点的captureStream方法（实验特性，不同浏览器支持程度不高）返回的流；WebRTC中的remote流；自己创建的流等\n //注意：流内必须至少存在一条音轨(Audio Track)，比如audio标签必须等待到可以开始播放后才会有音轨，否则open会失败\n\n //,runningContext:AudioContext\n //可选提供一个state为running状态的AudioContext对象(ctx)；默认会在rec.open时自动创建一个新的ctx，无用户操作（触摸、点击等）时调用rec.open的ctx.state可能为suspended，会在rec.start时尝试进行ctx.resume，如果也无用户操作ctx.resume可能不会恢复成running状态（目前仅iOS上有此兼容性问题），导致无法去读取媒体流，这时请提前在用户操作时调用Recorder.GetContext(true)来得到一个running状态AudioContext（用完需调用CloseNewCtx(ctx)关闭）\n\n //,audioTrackSet:{ deviceId:\"\",groupId:\"\", autoGainControl:true, echoCancellation:true, noiseSuppression:true }\n //普通麦克风录音时getUserMedia方法的audio配置参数，比如指定设备id，回声消除、降噪开关；注意：提供的任何配置值都不一定会生效\n //由于麦克风是全局共享的，所以新配置后需要close掉以前的再重新open\n //同样可配置videoTrackSet，更多参考: https://developer.mozilla.org/en-US/docs/Web/API/MediaTrackConstraints\n\n //,disableEnvInFix:false 内部参数，禁用设备卡顿时音频输入丢失补偿功能\n\n //,takeoffEncodeChunk:NOOP //fn(chunkBytes) chunkBytes=[Uint8,...]：实时编码环境下接管编码器输出，当编码器实时编码出一块有效的二进制音频数据时实时回调此方法；参数为二进制的Uint8Array，就是编码出来的音频数据片段，所有的chunkBytes拼接在一起即为完整音频。本实现的想法最初由QQ2543775048提出\n //当提供此回调方法时，将接管编码器的数据输出，编码器内部将放弃存储生成的音频数据；如果当前编码器或环境不支持实时编码处理，将在open时直接走fail逻辑\n //因此提供此回调后调用stop方法将无法获得有效的音频数据，因为编码器内没有音频数据，因此stop时返回的blob将是一个字节长度为0的blob\n //大部分录音格式编码器都支持实时编码（边录边转码），比如mp3格式：会实时的将编码出来的mp3片段通过此方法回调，所有的chunkBytes拼接到一起即为完整的mp3，此种拼接的结果比mock方法实时生成的音质更加，因为天然避免了首尾的静默\n //不支持实时编码的录音格式不可以提供此回调（wav格式不支持，因为wav文件头中需要提供文件最终长度），提供了将在open时直接走fail逻辑\n };\n for (var k in set) {\n o[k] = set[k];\n }\n ;\n This.set = o;\n var vB = o[bitRateTxt],\n vS = o[sampleRateTxt]; //校验配置参数\n if (vB && !IsNum(vB) || vS && !IsNum(vS)) {\n This.CLog($T.G(\"IllegalArgs-1\", [$T(\"VtS4::{1}和{2}必须是数值\", 0, sampleRateTxt, bitRateTxt)]), 1, set);\n }\n ;\n o[bitRateTxt] = +vB || 16;\n o[sampleRateTxt] = +vS || 16000;\n This.state = 0; //运行状态，0未录音 1录音中 2暂停 3等待ctx激活\n This._S = 9; //stop同步锁，stop可以阻止open过程中还未运行的start\n This.Sync = {\n O: 9,\n C: 9\n }; //和Recorder.Sync一致，只不过这个是非全局的，仅用来简化代码逻辑，无实际作用\n }\n ;\n //同步锁，控制对Stream的竞争；用于close时中断异步的open；一个对象open如果变化了都要阻止close，Stream的控制权交个新的对象\n Recorder.Sync = {\n /*open*/O: 9,\n /*close*/C: 9\n };\n Recorder.prototype = initFn.prototype = {\n CLog: CLog\n\n //流相关的数据存储在哪个对象里面；如果提供了sourceStream，数据直接存储在当前对象中，否则存储在全局\n ,\n _streamStore: function () {\n if (this.set.sourceStream) {\n return this;\n } else {\n return Recorder;\n }\n }\n //当前实例用到的Stream，可能是全局的，也可能是独享的\n ,\n _streamGet: function () {\n return this._streamStore().Stream;\n }\n //当前实例用到的AudioContext，可能是全局的，也可能是独享的\n ,\n _streamCtx: function () {\n var m = this._streamGet();\n return m && m._c;\n }\n\n //打开录音资源True(),False(msg,isUserNotAllow)，需要调用close。注意：此方法是异步的；一般使用时打开，用完立即关闭；可重复调用，可用来测试是否能录音；open和start至少有一个应当在用户操作（触摸、点击等）下进行调用，原因参考runningContext配置\n ,\n open: function (True, False) {\n var This = this,\n set = This.set,\n streamStore = This._streamStore(),\n newCtx = 0;\n True = True || NOOP;\n var failCall = function (errMsg, isUserNotAllow) {\n isUserNotAllow = !!isUserNotAllow;\n This.CLog($T(\"5tWi::录音open失败：\") + errMsg + \",isUserNotAllow:\" + isUserNotAllow, 1);\n if (newCtx) Recorder.CloseNewCtx(newCtx);\n False && False(errMsg, isUserNotAllow);\n };\n This._streamTag = getUserMediaTxt;\n var ok = function () {\n This.CLog(\"open ok, id:\" + This.id + \" stream:\" + This._streamTag);\n True();\n This._SO = 0; //解除stop对open中的start调用的阻止\n };\n\n //同步锁\n var Lock = streamStore.Sync;\n var lockOpen = ++Lock.O,\n lockClose = Lock.C;\n This._O = This._O_ = lockOpen; //记住当前的open，如果变化了要阻止close，这里假定了新对象已取代当前对象并且不再使用\n This._SO = This._S; //记住open过程中的stop，中途任何stop调用后都不能继续open中的start\n var lockFail = function () {\n //允许多次open，但不允许任何一次close，或者自身已经调用了关闭\n if (lockClose != Lock.C || !This._O) {\n var err = $T(\"dFm8::open被取消\");\n if (lockOpen == Lock.O) {\n //无新的open，已经调用了close进行取消，此处应让上次的close明确生效\n This.close();\n } else {\n err = $T(\"VtJO::open被中断\");\n }\n ;\n failCall(err);\n return true;\n }\n ;\n };\n\n //环境配置检查\n if (!isBrowser) {\n failCall($T.G(\"NonBrowser-1\", [\"open\"]) + $T(\"EMJq::，可尝试使用RecordApp解决方案\") + \"(\" + GitUrl + \"/tree/master/app-support-sample)\");\n return;\n }\n ;\n var checkMsg = This.envCheck({\n envName: \"H5\",\n canProcess: true\n });\n if (checkMsg) {\n failCall($T(\"A5bm::不能录音：\") + checkMsg);\n return;\n }\n ;\n\n //尽量先创建好ctx，不然异步下创建可能不是running状态\n var ctx;\n var getCtx = function () {\n ctx = set.runningContext;\n if (!ctx) ctx = newCtx = Recorder.GetContext(true); //2023-06 尽量创建新的ctx，免得Safari再次连接无回调\n };\n\n //***********已直接提供了音频流************\n if (set.sourceStream) {\n This._streamTag = \"set.sourceStream\";\n if (!Recorder.GetContext()) {\n failCall($T(\"1iU7::不支持此浏览器从流中获取录音\"));\n return;\n }\n ;\n getCtx();\n Disconnect(streamStore); //可能已open过，直接先尝试断开\n var stream = This.Stream = set.sourceStream;\n stream._c = ctx;\n stream._RC = set.runningContext;\n stream._call = {};\n try {\n Connect(streamStore);\n } catch (e) {\n Disconnect(streamStore);\n failCall($T(\"BTW2::从流中打开录音失败：\") + e.message);\n return;\n }\n ok();\n return;\n }\n ;\n\n //***********打开麦克风得到全局的音频流************\n var codeFail = function (code, msg) {\n try {\n //跨域的优先检测一下\n window.top.a;\n } catch (e) {\n failCall($T(\"Nclz::无权录音(跨域，请尝试给iframe添加麦克风访问策略，如{1})\", 0, 'allow=\"camera;microphone\"'));\n return;\n }\n ;\n if (codeErr1(1, code)) {\n if (/Found/i.test(code)) {\n //可能是非安全环境导致的没有设备\n failCall(msg + $T(\"jBa9::，无可用麦克风\"));\n } else {\n failCall(msg);\n }\n ;\n }\n ;\n };\n var codeErr1 = function (call, code) {\n //排除几个明确原因的错误\n if (/Permission|Allow/i.test(code)) {\n if (call) failCall($T(\"gyO5::用户拒绝了录音权限\"), true);\n } else if (window.isSecureContext === false) {\n if (call) failCall($T(\"oWNo::浏览器禁止不安全页面录音，可开启https解决\"));\n } else {\n return 1;\n }\n ;\n };\n\n //如果已打开并且有效就不要再打开了\n if (Recorder.IsOpen()) {\n ok();\n return;\n }\n ;\n if (!Recorder.Support()) {\n codeFail(\"\", $T(\"COxc::此浏览器不支持录音\"));\n return;\n }\n ;\n getCtx();\n\n //请求权限，如果从未授权，一般浏览器会弹出权限请求弹框\n var f1 = function (stream) {\n //请求成功回调\n //https://github.com/xiangyuecn/Recorder/issues/14 获取到的track.readyState!=\"live\"，刚刚回调时可能是正常的，但过一下可能就被关掉了，原因不明。延迟一下保证真异步。对正常浏览器不影响\n setTimeout(function () {\n stream._call = {};\n var oldStream = Recorder.Stream;\n if (oldStream) {\n Disconnect(); //直接断开已存在的，旧的Connect未完成会自动终止\n stream._call = oldStream._call;\n }\n ;\n Recorder.Stream = stream;\n stream._c = ctx;\n stream._RC = set.runningContext;\n if (lockFail()) return;\n if (Recorder.IsOpen()) {\n if (oldStream) This.CLog($T(\"upb8::发现同时多次调用open\"), 1);\n Connect(streamStore);\n ok(); //只连接，因为AudioContext不一定在运行，无法知道是否有数据回调\n } else {\n failCall($T(\"Q1GA::录音功能无效：无音频流\"));\n }\n ;\n }, 100);\n };\n var f2 = function (e) {\n //请求失败回调\n var code = e.name || e.message || e.code + \":\" + e;\n var tryMsg = \"\";\n if (callUmCount == 1 && codeErr1(0, code)) {\n tryMsg = $T(\"KxE2::，将尝试禁用回声消除后重试\");\n }\n This.CLog($T(\"xEQR::请求录音权限错误\") + tryMsg + \"|\" + e, tryMsg ? 3 : 1, e);\n if (tryMsg) {\n //重试\n callUserMedia(1);\n } else {\n codeFail(code, $T(\"bDOG::无法录音：\") + e);\n }\n ;\n };\n var callUmCount = 0;\n var callUserMedia = function (retry) {\n callUmCount++;\n var atsTxt = \"audioTrackSet\";\n var t_AGC = \"autoGainControl\",\n t_AEC = \"echoCancellation\",\n t_ANS = \"noiseSuppression\";\n var atsTxtJs = atsTxt + \":{\" + t_AEC + \",\" + t_ANS + \",\" + t_AGC + \"}\";\n var trackSet = JSON.parse(ToJson(set[atsTxt] || true)); //true 跟 {} 兼容性？\n This.CLog(\"open... \" + callUmCount + \" \" + atsTxt + \":\" + ToJson(trackSet));\n if (retry) {\n //回声消除有些浏览器可能导致无法打开录音，尝试明确禁用来保证能最基础的录\n if (typeof trackSet != \"object\") trackSet = {}; //默认true\n trackSet[t_AGC] = false;\n trackSet[t_AEC] = false;\n trackSet[t_ANS] = false;\n }\n ;\n //这里指明采样率，虽然可以解决手机上MediaRecorder采样率16k的问题，是回声消除导致的只能获取到16k的流，禁用回声消除可恢复48k。(issues#230)而且会导致乱用音频输入设备，MediaStreamTrack的applyConstraints也无法修改采样率\n //trackSet[sampleRateTxt]=ctx[sampleRateTxt];\n if (trackSet[sampleRateTxt]) {\n This.CLog($T(\"IjL3::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\", 0, atsTxt + \".\" + sampleRateTxt), 3);\n }\n ;\n var mSet = {\n audio: trackSet,\n video: set.videoTrackSet || false\n };\n try {\n var pro = Recorder.Scope[getUserMediaTxt](mSet, f1, f2);\n } catch (e) {\n //不能设置trackSet就算了\n This.CLog(getUserMediaTxt, 3, e);\n mSet = {\n audio: true,\n video: false\n };\n pro = Recorder.Scope[getUserMediaTxt](mSet, f1, f2);\n }\n ;\n This.CLog(getUserMediaTxt + \"(\" + ToJson(mSet) + \") \" + CtxState(ctx) + $T(\"RiWe::，未配置 {1} 时浏览器可能会自动启用回声消除，移动端未禁用回声消除时可能会降低系统播放音量（关闭录音后可恢复）和仅提供16k采样率的音频流（不需要回声消除时可明确配置成禁用来获得48k高音质的流），请参阅文档中{2}配置\", 0, atsTxtJs, atsTxt) + \"(\" + GitUrl + \") LM:\" + LM + \" UA:\" + navigator.userAgent);\n if (pro && pro.then) {\n pro.then(f1)[CatchTxt](f2); //fix 关键字，保证catch压缩时保持字符串形式\n }\n ;\n };\n callUserMedia();\n }\n //关闭释放录音资源\n ,\n close: function (call) {\n call = call || NOOP;\n var This = this,\n streamStore = This._streamStore();\n This._stop();\n var sTag = \" stream:\" + This._streamTag;\n var Lock = streamStore.Sync;\n This._O = 0;\n if (This._O_ != Lock.O) {\n //唯一资源Stream的控制权已交给新对象，这里不能关闭。此处在每次都弹权限的浏览器内可能存在泄漏，新对象被拒绝权限可能不会调用close，忽略这种不处理\n This.CLog($T(\"hWVz::close被忽略（因为同时open了多个rec，只有最后一个会真正close）\") + sTag, 3);\n call();\n return;\n }\n ;\n Lock.C++; //获得控制权\n\n Disconnect(streamStore);\n This.CLog(\"close,\" + sTag);\n call();\n }\n\n /*模拟一段录音数据，后面可以调用stop进行编码，需提供pcm数据[1,2,3...]，pcm的采样率*/,\n mock: function (pcmData, pcmSampleRate) {\n var This = this;\n This._stop(); //清理掉已有的资源\n\n This.isMock = 1;\n This.mockEnvInfo = null;\n This.buffers = [pcmData];\n This.recSize = pcmData.length;\n This._setSrcSR(pcmSampleRate);\n This._streamTag = \"mock\";\n return This;\n },\n _setSrcSR: function (sampleRate) {\n var This = this,\n set = This.set;\n var setSr = set[sampleRateTxt];\n if (setSr > sampleRate) {\n set[sampleRateTxt] = sampleRate;\n } else {\n setSr = 0;\n }\n This[srcSampleRateTxt] = sampleRate;\n This.CLog(srcSampleRateTxt + \": \" + sampleRate + \" set.\" + sampleRateTxt + \": \" + set[sampleRateTxt] + (setSr ? \" \" + $T(\"UHvm::忽略\") + \": \" + setSr : \"\"), setSr ? 3 : 0);\n },\n envCheck: function (envInfo) {\n //平台环境下的可用性检查，任何时候都可以调用检查，返回errMsg:\"\"正常，\"失败原因\"\n //envInfo={envName:\"H5\",canProcess:true}\n var errMsg,\n This = this,\n set = This.set;\n\n //检测CPU的数字字节序，TypedArray字节序是个迷，直接拒绝罕见的大端模式，因为找不到这种CPU进行测试\n var tag = \"CPU_BE\";\n if (!errMsg && !Recorder[tag] && typeof Int8Array == \"function\" && !new Int8Array(new Int32Array([1]).buffer)[0]) {\n Traffic(tag); //如果开启了流量统计，这里将发送一个图片请求\n errMsg = $T(\"Essp::不支持{1}架构\", 0, tag);\n }\n ;\n\n //编码器检查环境下配置是否可用\n if (!errMsg) {\n var type = set.type,\n hasFn = This[type + \"_envCheck\"];\n if (set.takeoffEncodeChunk) {\n //需要实时编码返回数据，此时需要检查环境是否有实时特性、和是否可实时编码\n if (!hasFn) {\n errMsg = $T(\"2XBl::{1}类型不支持设置takeoffEncodeChunk\", 0, type) + (This[type] ? \"\" : $T(\"LG7e::(未加载编码器)\"));\n } else if (!envInfo.canProcess) {\n errMsg = $T(\"7uMV::{1}环境不支持实时处理\", 0, envInfo.envName);\n }\n ;\n }\n ;\n if (!errMsg && hasFn) {\n //编码器已实现环境检查\n errMsg = This[type + \"_envCheck\"](envInfo, set);\n }\n ;\n }\n ;\n return errMsg || \"\";\n },\n envStart: function (mockEnvInfo, sampleRate) {\n //平台环境相关的start调用\n var This = this,\n set = This.set;\n This.isMock = mockEnvInfo ? 1 : 0; //非H5环境需要启用mock，并提供envCheck需要的环境信息\n This.mockEnvInfo = mockEnvInfo;\n This.buffers = []; //数据缓冲\n This.recSize = 0; //数据大小\n if (mockEnvInfo) {\n This._streamTag = \"env$\" + mockEnvInfo.envName;\n }\n ;\n This.state = 1; //运行状态，0未录音 1录音中 2暂停 3等待ctx激活\n This.envInLast = 0; //envIn接收到最后录音内容的时间\n This.envInFirst = 0; //envIn接收到的首个录音内容的录制时间\n This.envInFix = 0; //补偿的总时间\n This.envInFixTs = []; //补偿计数列表\n\n //engineCtx需要提前确定最终的采样率\n This._setSrcSR(sampleRate);\n This.engineCtx = 0;\n //此类型有边录边转码(Worker)支持\n if (This[set.type + \"_start\"]) {\n var engineCtx = This.engineCtx = This[set.type + \"_start\"](set);\n if (engineCtx) {\n engineCtx.pcmDatas = [];\n engineCtx.pcmSize = 0;\n }\n ;\n }\n ;\n },\n envResume: function () {\n //和平台环境无关的恢复录音\n //重新开始计数\n this.envInFixTs = [];\n },\n envIn: function (pcm, sum) {\n //和平台环境无关的pcm[Int16]输入\n var This = this,\n set = This.set,\n engineCtx = This.engineCtx;\n if (This.state != 1) {\n if (!This.state) This.CLog(\"envIn at state=0\", 3);\n return;\n }\n ;\n var bufferSampleRate = This[srcSampleRateTxt];\n var size = pcm.length;\n var powerLevel = Recorder.PowerLevel(sum, size);\n var buffers = This.buffers;\n var bufferFirstIdx = buffers.length; //之前的buffer都是经过onProcess处理好的，不允许再修改\n buffers.push(pcm);\n\n //有engineCtx时会被覆盖，这里保存一份\n var buffersThis = buffers;\n var bufferFirstIdxThis = bufferFirstIdx;\n\n //卡顿丢失补偿：因为设备很卡的时候导致H5接收到的数据量不够造成播放时候变速，结果比实际的时长要短，此处保证了不会变短，但不能修复丢失的音频数据造成音质变差。当前算法采用输入时间侦测下一帧是否需要添加补偿帧，需要(6次输入||超过1秒)以上才会开始侦测，如果滑动窗口内丢失超过1/3就会进行补偿\n var now = Date.now();\n var pcmTime = Math.round(size / bufferSampleRate * 1000);\n This.envInLast = now;\n if (This.buffers.length == 1) {\n //记下首个录音数据的录制时间\n This.envInFirst = now - pcmTime;\n }\n ;\n var envInFixTs = This.envInFixTs;\n envInFixTs.splice(0, 0, {\n t: now,\n d: pcmTime\n });\n //保留3秒的计数滑动窗口，另外超过3秒的停顿不补偿\n var tsInStart = now,\n tsPcm = 0;\n for (var i = 0; i < envInFixTs.length; i++) {\n var o = envInFixTs[i];\n if (now - o.t > 3000) {\n envInFixTs.length = i;\n break;\n }\n ;\n tsInStart = o.t;\n tsPcm += o.d;\n }\n ;\n //达到需要的数据量，开始侦测是否需要补偿\n var tsInPrev = envInFixTs[1];\n var tsIn = now - tsInStart;\n var lost = tsIn - tsPcm;\n if (lost > tsIn / 3 && (tsInPrev && tsIn > 1000 || envInFixTs.length >= 6)) {\n //丢失过多，开始执行补偿\n var addTime = now - tsInPrev.t - pcmTime; //距离上次输入丢失这么多ms\n if (addTime > pcmTime / 5) {\n //丢失超过本帧的1/5\n var fixOpen = !set.disableEnvInFix;\n This.CLog(\"[\" + now + \"]\" + i18n.get(fixOpen ? $T(\"4Kfd::补偿{1}ms\", 1) : $T(\"bM5i::未补偿{1}ms\", 1), [addTime]), 3);\n This.envInFix += addTime;\n\n //用静默进行补偿\n if (fixOpen) {\n var addPcm = new Int16Array(addTime * bufferSampleRate / 1000);\n size += addPcm.length;\n buffers.push(addPcm);\n }\n ;\n }\n ;\n }\n ;\n var sizeOld = This.recSize,\n addSize = size;\n var bufferSize = sizeOld + addSize;\n This.recSize = bufferSize; //此值在onProcess后需要修正，可能新数据被修改\n\n //此类型有边录边转码(Worker)支持，开启实时转码\n if (engineCtx) {\n //转换成set的采样率\n var chunkInfo = Recorder.SampleData(buffers, bufferSampleRate, set[sampleRateTxt], engineCtx.chunkInfo);\n engineCtx.chunkInfo = chunkInfo;\n sizeOld = engineCtx.pcmSize;\n addSize = chunkInfo.data.length;\n bufferSize = sizeOld + addSize;\n engineCtx.pcmSize = bufferSize; //此值在onProcess后需要修正，可能新数据被修改\n\n buffers = engineCtx.pcmDatas;\n bufferFirstIdx = buffers.length;\n buffers.push(chunkInfo.data);\n bufferSampleRate = chunkInfo[sampleRateTxt];\n }\n ;\n var duration = Math.round(bufferSize / bufferSampleRate * 1000);\n var bufferNextIdx = buffers.length;\n var bufferNextIdxThis = buffersThis.length;\n\n //允许异步处理buffer数据\n var asyncEnd = function () {\n //重新计算size，异步的早已减去添加的，同步的需去掉本次添加的然后重新计算\n var num = asyncBegin ? 0 : -addSize;\n var hasClear = buffers[0] == null;\n for (var i = bufferFirstIdx; i < bufferNextIdx; i++) {\n var buffer = buffers[i];\n if (buffer == null) {\n //已被主动释放内存，比如长时间实时传输录音时\n hasClear = 1;\n } else {\n num += buffer.length;\n\n //推入后台边录边转码\n if (engineCtx && buffer.length) {\n This[set.type + \"_encode\"](engineCtx, buffer);\n }\n ;\n }\n ;\n }\n ;\n\n //同步清理This.buffers，不管buffers到底清了多少个，buffersThis是使用不到的进行全清\n if (hasClear && engineCtx) {\n var i = bufferFirstIdxThis;\n if (buffersThis[0]) {\n i = 0;\n }\n ;\n for (; i < bufferNextIdxThis; i++) {\n buffersThis[i] = null;\n }\n ;\n }\n ;\n\n //统计修改后的size，如果异步发生clear要原样加回来，同步的无需操作\n if (hasClear) {\n num = asyncBegin ? addSize : 0;\n buffers[0] = null; //彻底被清理\n }\n ;\n if (engineCtx) {\n engineCtx.pcmSize += num;\n } else {\n This.recSize += num;\n }\n ;\n };\n //实时回调处理数据，允许修改或替换上次回调以来新增的数据 ，但是不允许修改已处理过的，不允许增删第一维数组 ，允许将第二维数组任意修改替换成空数组也可以\n var asyncBegin = 0,\n procTxt = \"rec.set.onProcess\";\n try {\n asyncBegin = set.onProcess(buffers, powerLevel, duration, bufferSampleRate, bufferFirstIdx, asyncEnd);\n } catch (e) {\n //此错误显示不要用CLog，这样控制台内相同内容不会重复打印\n console.error(procTxt + $T(\"gFUF::回调出错是不允许的，需保证不会抛异常\"), e);\n }\n ;\n var slowT = Date.now() - now;\n if (slowT > 10 && This.envInFirst - now > 1000) {\n //1秒后开始onProcess性能监测\n This.CLog(procTxt + $T(\"2ghS::低性能，耗时{1}ms\", 0, slowT), 3);\n }\n ;\n if (asyncBegin === true) {\n //开启了异步模式，onProcess已接管buffers新数据，立即清空，避免出现未处理的数据\n var hasClear = 0;\n for (var i = bufferFirstIdx; i < bufferNextIdx; i++) {\n if (buffers[i] == null) {\n //已被主动释放内存，比如长时间实时传输录音时 ，但又要开启异步模式，此种情况是非法的\n hasClear = 1;\n } else {\n buffers[i] = new Int16Array(0);\n }\n ;\n }\n ;\n if (hasClear) {\n This.CLog($T(\"ufqH::未进入异步前不能清除buffers\"), 3);\n } else {\n //还原size，异步结束后再统计仅修改后的size，如果发生clear要原样加回来\n if (engineCtx) {\n engineCtx.pcmSize -= addSize;\n } else {\n This.recSize -= addSize;\n }\n ;\n }\n ;\n } else {\n asyncEnd();\n }\n ;\n }\n\n //开始录音，需先调用open；只要open成功时，调用此方法是安全的，如果未open强行调用导致的内部错误将不会有任何提示，stop时自然能得到错误；注意：open和start至少有一个应当在用户操作（触摸、点击等）下进行调用，原因参考runningContext配置\n ,\n start: function () {\n var This = this;\n var isOpen = 1;\n if (This.set.sourceStream) {\n //直接提供了流，仅判断是否调用了open\n if (!This.Stream) {\n isOpen = 0;\n }\n } else if (!Recorder.IsOpen()) {\n //监测全局麦克风是否打开并且有效\n isOpen = 0;\n }\n ;\n if (!isOpen) {\n This.CLog($T(\"6WmN::start失败：未open\"), 1);\n return;\n }\n ;\n var ctx = This._streamCtx();\n This.CLog($T(\"kLDN::start 开始录音，\") + CtxState(ctx) + \" stream:\" + This._streamTag);\n This._stop();\n This.envStart(null, ctx[sampleRateTxt]);\n This.state = 3; //0未录音 1录音中 2暂停 3等待ctx激活\n\n //检查open过程中stop是否已经调用过\n if (This._SO && This._SO + 1 != This._S) {\n //上面调用过一次 _stop\n //open未完成就调用了stop，此种情况终止start。也应尽量避免出现此情况\n This.CLog($T(\"Bp2y::start被中断\"), 3);\n return;\n }\n ;\n This._SO = 0;\n var end = function () {\n if (This.state == 3) {\n This.state = 1;\n This.resume();\n }\n };\n var tag = \"AudioContext resume: \";\n\n //如果有数据回调，就不等待ctx resume\n var stream = This._streamGet();\n stream._call[This.id] = function () {\n This.CLog(tag + ctx.state + '|stream ok');\n end();\n };\n ResumeCtx(ctx, function (runC) {\n runC && This.CLog(tag + \"wait...\");\n return This.state == 3;\n }, function (runC) {\n runC && This.CLog(tag + ctx.state);\n end();\n }, function (err) {\n //比较少见，可能对录音没有影响\n This.CLog(tag + ctx.state + $T(\"upkE::，可能无法录音：\") + err, 1);\n end();\n });\n }\n\n /*暂停录音*/,\n pause: function () {\n var This = this,\n stream = This._streamGet();\n if (This.state) {\n This.state = 2;\n This.CLog(\"pause\");\n if (stream) delete stream._call[This.id];\n }\n ;\n }\n /*恢复录音*/,\n resume: function () {\n var This = this,\n stream = This._streamGet();\n var tag = \"resume\",\n tag3 = tag + \"(wait ctx)\";\n if (This.state == 3) {\n //start还在等ctx恢复\n This.CLog(tag3);\n } else if (This.state) {\n This.state = 1;\n This.CLog(tag);\n This.envResume();\n if (stream) {\n stream._call[This.id] = function (pcm, sum) {\n if (This.state == 1) {\n This.envIn(pcm, sum);\n }\n ;\n };\n ConnAlive(stream); //AudioWorklet只会在ctx激活后运行\n }\n ;\n var ctx = This._streamCtx();\n if (ctx) {\n //AudioContext如果被暂停，尽量恢复\n ResumeCtx(ctx, function (runC) {\n runC && This.CLog(tag3 + \"...\");\n return This.state == 1;\n }, function (runC) {\n runC && This.CLog(tag3 + ctx.state);\n ConnAlive(stream);\n }, function (err) {\n This.CLog(tag3 + ctx.state + \"[err]\" + err, 1);\n });\n }\n ;\n }\n ;\n },\n _stop: function (keepEngine) {\n var This = this,\n set = This.set;\n if (!This.isMock) {\n This._S++;\n }\n ;\n if (This.state) {\n This.pause();\n This.state = 0;\n }\n ;\n if (!keepEngine && This[set.type + \"_stop\"]) {\n This[set.type + \"_stop\"](This.engineCtx);\n This.engineCtx = 0;\n }\n ;\n }\n /*\n 结束录音并返回录音数据blob对象\n \tTrue(blob,duration,mime)\n \t\tblob：录音数据audio/mp3|wav格式；默认是Blob对象，可设置rec.dataType=\"arraybuffer\"改成ArrayBuffer\n \t\tduration：录音时长，单位毫秒\n \t\tmime：\"auido/mp3\" blob数据的类型，方便ArrayBuffer时区分类型\n \tFalse(msg)\n \tautoClose:false 可选，是否自动调用close，默认为false\n */,\n stop: function (True, False, autoClose) {\n var This = this,\n set = This.set,\n t1;\n var envInMS = This.envInLast - This.envInFirst,\n envInLen = envInMS && This.buffers.length; //可能未start\n This.CLog($T(\"Xq4s::stop 和start时差:\") + (envInMS ? envInMS + \"ms \" + $T(\"3CQP::补偿:\") + This.envInFix + \"ms\" + \" envIn:\" + envInLen + \" fps:\" + (envInLen / envInMS * 1000).toFixed(1) : \"-\") + \" stream:\" + This._streamTag + \" (\" + GitUrl + \") LM:\" + LM);\n var end = function () {\n This._stop(); //彻底关掉engineCtx\n if (autoClose) {\n This.close();\n }\n ;\n };\n var err = function (msg) {\n This.CLog($T(\"u8JG::结束录音失败：\") + msg, 1);\n False && False(msg);\n end();\n };\n var ok = function (blob, mime, duration) {\n var tBlob = \"blob\",\n tABuf = \"arraybuffer\",\n tDT = \"dataType\",\n tDDT = \"DefaultDataType\";\n var dType = This[tDT] || Recorder[tDDT] || tBlob,\n dTag = tDT + \"=\" + dType;\n var isAB = blob instanceof ArrayBuffer,\n dErr = 0;\n var dLen = isAB ? blob.byteLength : blob.size;\n if (dType == tABuf) {\n if (!isAB) dErr = 1;\n } else if (dType == tBlob) {\n if (typeof Blob != \"function\") {\n dErr = $T.G(\"NonBrowser-1\", [dTag]) + $T(\"1skY::，请设置{1}\", 0, RecTxt + \".\" + tDDT + '=\"' + tABuf + '\"');\n } else {\n if (isAB) blob = new Blob([blob], {\n type: mime\n });\n if (!(blob instanceof Blob)) dErr = 1;\n mime = blob.type || mime;\n }\n } else {\n dErr = $T.G(\"NotSupport-1\", [dTag]);\n }\n ;\n This.CLog($T(\"Wv7l::结束录音 编码花{1}ms 音频时长{2}ms 文件大小{3}b\", 0, Date.now() - t1, duration, dLen) + \" \" + dTag + \",\" + mime);\n if (dErr) {\n err(dErr != 1 ? dErr : $T(\"Vkbd::{1}编码器返回的不是{2}\", 0, set.type, dType) + \", \" + dTag);\n return;\n }\n ;\n if (set.takeoffEncodeChunk) {\n //接管了输出，此时blob长度为0\n This.CLog($T(\"QWnr::启用takeoffEncodeChunk后stop返回的blob长度为0不提供音频数据\"), 3);\n } else if (dLen < Math.max(50, duration / 5)) {\n //1秒小于0.2k？\n err($T(\"Sz2H::生成的{1}无效\", 0, set.type));\n return;\n }\n ;\n True && True(blob, duration, mime);\n end();\n };\n if (!This.isMock) {\n var isCtxWait = This.state == 3;\n if (!This.state || isCtxWait) {\n err($T(\"wf9t::未开始录音\") + (isCtxWait ? $T(\"Dl2c::，开始录音前无用户交互导致AudioContext未运行\") : \"\"));\n return;\n }\n ;\n }\n ;\n This._stop(true); //将录音状态改成未录音\n var size = This.recSize;\n if (!size) {\n err($T(\"Ltz3::未采集到录音\"));\n return;\n }\n ;\n if (!This[set.type]) {\n err($T(\"xGuI::未加载{1}编码器，请尝试到{2}的src/engine内找到{1}的编码器并加载\", 0, set.type, RecTxt));\n return;\n }\n ;\n\n //环境配置检查，此处仅针对mock调用，因为open已经检查过了\n if (This.isMock) {\n var checkMsg = This.envCheck(This.mockEnvInfo || {\n envName: \"mock\",\n canProcess: false\n }); //没有提供环境信息的mock时没有onProcess回调\n if (checkMsg) {\n err($T(\"AxOH::录音错误：\") + checkMsg);\n return;\n }\n ;\n }\n ;\n\n //此类型有边录边转码(Worker)支持\n var engineCtx = This.engineCtx;\n if (This[set.type + \"_complete\"] && engineCtx) {\n var duration = Math.round(engineCtx.pcmSize / set[sampleRateTxt] * 1000); //采用后的数据长度和buffers的长度可能微小的不一致，是采样率连续转换的精度问题\n\n t1 = Date.now();\n This[set.type + \"_complete\"](engineCtx, function (blob, mime) {\n ok(blob, mime, duration);\n }, err);\n return;\n }\n ;\n\n //标准UI线程转码，调整采样率\n t1 = Date.now();\n if (!This.buffers[0]) {\n err($T(\"xkKd::音频buffers被释放\"));\n return;\n }\n ;\n var chunk = Recorder.SampleData(This.buffers, This[srcSampleRateTxt], set[sampleRateTxt]);\n set[sampleRateTxt] = chunk[sampleRateTxt];\n var res = chunk.data;\n var duration = Math.round(res.length / set[sampleRateTxt] * 1000);\n This.CLog($T(\"CxeT::采样:{1} 花:{2}ms\", 0, size + \"->\" + res.length, Date.now() - t1));\n setTimeout(function () {\n t1 = Date.now();\n This[set.type](res, function (blob, mime) {\n ok(blob, mime, duration);\n }, function (msg) {\n err(msg);\n });\n });\n }\n };\n\n //=======从WebM字节流中提取pcm数据，提取成功返回Float32Array，失败返回null||-1=====\n var WebM_Extract = function (inBytes, scope) {\n if (!scope.pos) {\n scope.pos = [0];\n scope.tracks = {};\n scope.bytes = [];\n }\n ;\n var tracks = scope.tracks,\n position = [scope.pos[0]];\n var endPos = function () {\n scope.pos[0] = position[0];\n };\n var sBL = scope.bytes.length;\n var bytes = new Uint8Array(sBL + inBytes.length);\n bytes.set(scope.bytes);\n bytes.set(inBytes, sBL);\n scope.bytes = bytes;\n\n //先读取文件头和Track信息\n if (!scope._ht) {\n readMatroskaVInt(bytes, position); //EBML Header\n readMatroskaBlock(bytes, position); //跳过EBML Header内容\n if (!BytesEq(readMatroskaVInt(bytes, position), [0x18, 0x53, 0x80, 0x67])) {\n return; //未识别到Segment\n }\n readMatroskaVInt(bytes, position); //跳过Segment长度值\n while (position[0] < bytes.length) {\n var eid0 = readMatroskaVInt(bytes, position);\n var bytes0 = readMatroskaBlock(bytes, position);\n var pos0 = [0],\n audioIdx = 0;\n if (!bytes0) return; //数据不全，等待缓冲\n //Track完整数据，循环读取TrackEntry\n if (BytesEq(eid0, [0x16, 0x54, 0xAE, 0x6B])) {\n while (pos0[0] < bytes0.length) {\n var eid1 = readMatroskaVInt(bytes0, pos0);\n var bytes1 = readMatroskaBlock(bytes0, pos0);\n var pos1 = [0],\n track = {\n channels: 0,\n sampleRate: 0\n };\n if (BytesEq(eid1, [0xAE])) {\n //TrackEntry\n while (pos1[0] < bytes1.length) {\n var eid2 = readMatroskaVInt(bytes1, pos1);\n var bytes2 = readMatroskaBlock(bytes1, pos1);\n var pos2 = [0];\n if (BytesEq(eid2, [0xD7])) {\n //Track Number\n var val = BytesInt(bytes2);\n track.number = val;\n tracks[val] = track;\n } else if (BytesEq(eid2, [0x83])) {\n //Track Type\n var val = BytesInt(bytes2);\n if (val == 1) track.type = \"video\";else if (val == 2) {\n track.type = \"audio\";\n if (!audioIdx) scope.track0 = track;\n track.idx = audioIdx++;\n } else track.type = \"Type-\" + val;\n } else if (BytesEq(eid2, [0x86])) {\n //Track Codec\n var str = \"\";\n for (var i = 0; i < bytes2.length; i++) {\n str += String.fromCharCode(bytes2[i]);\n }\n track.codec = str;\n } else if (BytesEq(eid2, [0xE1])) {\n while (pos2[0] < bytes2.length) {\n //循环读取 Audio 属性\n var eid3 = readMatroskaVInt(bytes2, pos2);\n var bytes3 = readMatroskaBlock(bytes2, pos2);\n //采样率、位数、声道数\n if (BytesEq(eid3, [0xB5])) {\n var val = 0,\n arr = new Uint8Array(bytes3.reverse()).buffer;\n if (bytes3.length == 4) val = new Float32Array(arr)[0];else if (bytes3.length == 8) val = new Float64Array(arr)[0];else CLog(\"WebM Track !Float\", 1, bytes3);\n track[sampleRateTxt] = Math.round(val);\n } else if (BytesEq(eid3, [0x62, 0x64])) track.bitDepth = BytesInt(bytes3);else if (BytesEq(eid3, [0x9F])) track.channels = BytesInt(bytes3);\n }\n }\n }\n }\n }\n ;\n scope._ht = 1;\n CLog(\"WebM Tracks\", tracks);\n endPos();\n break;\n }\n }\n }\n\n //校验音频参数信息，如果不符合代码要求，统统拒绝处理\n var track0 = scope.track0;\n if (!track0) return;\n var trackSR = track0[sampleRateTxt];\n scope.webmSR = trackSR;\n if (track0.bitDepth == 16 && /FLOAT/i.test(track0.codec)) {\n track0.bitDepth = 32; //chrome v66 实际为浮点数\n CLog(\"WebM 16->32 bit\", 3);\n }\n if (trackSR < 8000 || track0.bitDepth != 32 || track0.channels < 1 || !/(\\b|_)PCM\\b/i.test(track0.codec)) {\n scope.bytes = []; //格式非预期 无法处理，清空缓冲数据\n if (!scope.bad) CLog(\"WebM Track Unexpected\", 3, scope);\n scope.bad = 1;\n return -1;\n }\n\n //循环读取Cluster内的SimpleBlock\n var datas = [],\n dataLen = 0;\n while (position[0] < bytes.length) {\n var eid1 = readMatroskaVInt(bytes, position);\n var bytes1 = readMatroskaBlock(bytes, position);\n if (!bytes1) break; //数据不全，等待缓冲\n if (BytesEq(eid1, [0xA3])) {\n //SimpleBlock完整数据\n var trackNo = bytes1[0] & 0xf;\n var track = tracks[trackNo];\n if (!track) {\n //不可能没有，数据出错？\n CLog(\"WebM !Track\" + trackNo, 1, tracks);\n } else if (track.idx === 0) {\n var u8arr = new Uint8Array(bytes1.length - 4);\n for (var i = 4; i < bytes1.length; i++) {\n u8arr[i - 4] = bytes1[i];\n }\n datas.push(u8arr);\n dataLen += u8arr.length;\n }\n }\n endPos();\n }\n if (dataLen) {\n var more = new Uint8Array(bytes.length - scope.pos[0]);\n more.set(bytes.subarray(scope.pos[0]));\n scope.bytes = more; //清理已读取了的缓冲数据\n scope.pos[0] = 0;\n var u8arr = new Uint8Array(dataLen); //已获取的音频数据\n for (var i = 0, i2 = 0; i < datas.length; i++) {\n u8arr.set(datas[i], i2);\n i2 += datas[i].length;\n }\n var arr = new Float32Array(u8arr.buffer);\n if (track0.channels > 1) {\n //多声道，提取一个声道\n var arr2 = [];\n for (var i = 0; i < arr.length;) {\n arr2.push(arr[i]);\n i += track0.channels;\n }\n arr = new Float32Array(arr2);\n }\n ;\n return arr;\n }\n };\n //两个字节数组内容是否相同\n var BytesEq = function (bytes1, bytes2) {\n if (!bytes1 || bytes1.length != bytes2.length) return false;\n if (bytes1.length == 1) return bytes1[0] == bytes2[0];\n for (var i = 0; i < bytes1.length; i++) {\n if (bytes1[i] != bytes2[i]) return false;\n }\n return true;\n };\n //字节数组BE转成int数字\n var BytesInt = function (bytes) {\n var s = \"\"; //0-8字节，js位运算只支持4字节\n for (var i = 0; i < bytes.length; i++) {\n var n = bytes[i];\n s += (n < 16 ? \"0\" : \"\") + n.toString(16);\n }\n ;\n return parseInt(s, 16) || 0;\n };\n //读取一个可变长数值字节数组\n var readMatroskaVInt = function (arr, pos, trim) {\n var i = pos[0];\n if (i >= arr.length) return;\n var b0 = arr[i],\n b2 = (\"0000000\" + b0.toString(2)).substr(-8);\n var m = /^(0*1)(\\d*)$/.exec(b2);\n if (!m) return;\n var len = m[1].length,\n val = [];\n if (i + len > arr.length) return;\n for (var i2 = 0; i2 < len; i2++) {\n val[i2] = arr[i];\n i++;\n }\n if (trim) val[0] = parseInt(m[2] || '0', 2);\n pos[0] = i;\n return val;\n };\n //读取一个自带长度的内容字节数组\n var readMatroskaBlock = function (arr, pos) {\n var lenVal = readMatroskaVInt(arr, pos, 1);\n if (!lenVal) return;\n var len = BytesInt(lenVal);\n var i = pos[0],\n val = [];\n if (len < 0x7FFFFFFF) {\n //超大值代表没有长度\n if (i + len > arr.length) return;\n for (var i2 = 0; i2 < len; i2++) {\n val[i2] = arr[i];\n i++;\n }\n }\n pos[0] = i;\n return val;\n };\n //=====End WebM读取=====\n\n //=====i18n 简版国际化多语言支持=====\n var i18n = Recorder.i18n = {\n lang: \"zh-CN\" //默认中文\n ,\n alias: {\n \"zh-CN\": \"zh\",\n \"en-US\": \"en\"\n } //别名配置，映射到一个语言实例\n ,\n locales: {} //语言实例：zh:{key:\"text\"}\n ,\n data: {} //各种数据，desc$xx：语言描述，rtl$xx：文本方向是否从右到左 rtl$zh:false rtl$ar:true\n //添加语言文本内容，set={lang:\"\",overwrite:true}，texts=[\"key:text{1}\",...]\n ,\n put: function (set, texts) {\n var tag = RecTxt + \".i18n.put: \";\n var overwrite = set.overwrite;\n overwrite = overwrite == null || overwrite;\n var lang = set.lang;\n lang = i18n.alias[lang] || lang;\n if (!lang) throw new Error(tag + \"set.lang?\");\n var locale = i18n.locales[lang];\n if (!locale) {\n locale = {};\n i18n.locales[lang] = locale;\n }\n ;\n var exp = /^([\\w\\-]+):/,\n m;\n for (var i = 0; i < texts.length; i++) {\n var v = texts[i];\n m = exp.exec(v);\n if (!m) {\n CLog(tag + \"'key:'? \" + v, 3, set);\n continue;\n }\n var key = m[1],\n v = v.substr(key.length + 1);\n if (!overwrite && locale[key]) continue;\n locale[key] = v;\n }\n }\n //获取key对应的文本，如果没有对应文本，将返回en的，en的也没有将返回zh的\n ,\n get: function /*key,args,lang*/\n () {\n return i18n.v_G.apply(null, arguments);\n },\n v_G: function (key, args, lang) {\n //全局可重写get\n args = args || [];\n lang = lang || i18n.lang;\n lang = i18n.alias[lang] || lang;\n var locale = i18n.locales[lang];\n var val = locale && locale[key] || \"\";\n if (!val && lang != \"zh\") {\n if (lang == \"en\") return i18n.v_G(key, args, \"zh\");\n return i18n.v_G(key, args, \"en\");\n }\n i18n.lastLang = lang;\n if (val == \"=Empty\") return \"\";\n return val.replace(/\\{(\\d+)(\\!?)\\}/g, function (v, a, b) {\n a = +a || 0;\n v = args[a - 1];\n if (a < 1 || a > args.length) {\n v = \"{?}\";\n CLog(\"i18n[\" + key + \"] no {\" + a + \"}: \" + val, 3);\n }\n return b ? \"\" : v;\n });\n }\n /**返回一个国际化文本，返回的文本取决于i18n.lang。\n \t调用参数：T(\"key:[lang]:中文{1}\",\"[lang]:英文{1}\",...,0,\"args1\",\"args2\")，除了key:，其他都是可选的，文本如果在语言实例中不存在会push进去，第一个文本无lang时默认zh，第二个无lang时默认en，文本中用{1-n}来插入args\n \t第一个args前面这个参数必须是0；也可以不提供args，这个参数填args的数量，此时不返回文本，只返回key，再用i18n.get提供参数获取文本\n \t本函数调用，第一个文本需中文，调用尽量简单，不要换行，方便后续自动提取翻译列表\n \targs如果旧的参数位置发生了变更，应当使用新的key，让旧的翻译失效，增加args无需更换key\n \tkey的生成使用随机字符串，不同源码里面可以搞个不同前缀:\n \t\ts=\"\";L=4; for(var i=0,n;i<L;i++){ n=~~(Math.random()*62);s+=n<10?n:String.fromCharCode(n<36?55+n:61+n); }; s\n **/,\n $T: function () {\n return i18n.v_T.apply(null, arguments);\n },\n v_T: function () {\n //全局可重写$T\n var a = arguments,\n key = \"\",\n args = [],\n isArgs = 0,\n tag = RecTxt + \".i18n.$T:\";\n var exp = /^([\\w\\-]*):/,\n m;\n for (var i = 0; i < a.length; i++) {\n var v = a[i];\n if (i == 0) {\n m = exp.exec(v);\n key = m && m[1];\n if (!key) throw new Error(tag + \"0 'key:'?\");\n v = v.substr(key.length + 1);\n }\n if (isArgs === -1) args.push(v);else if (isArgs) throw new Error(tag + \" bad args\");else if (v === 0) isArgs = -1;else if (IsNum(v)) {\n if (v < 1) throw new Error(tag + \" bad args\");\n isArgs = v;\n } else {\n var lang = i == 1 ? \"en\" : i ? \"\" : \"zh\";\n m = exp.exec(v);\n if (m) {\n lang = m[1] || lang;\n v = v.substr(m[1].length + 1);\n }\n if (!m || !lang) throw new Error(tag + i + \" 'lang:'?\");\n i18n.put({\n lang: lang,\n overwrite: false\n }, [key + \":\" + v]);\n }\n }\n if (!key) return \"\";\n if (isArgs > 0) return key;\n return i18n.v_G(key, args);\n }\n };\n var $T = i18n.$T;\n $T.G = i18n.get;\n //预定义文本\n $T(\"NonBrowser-1::非浏览器环境，不支持{1}\", 1);\n $T(\"IllegalArgs-1::参数错误：{1}\", 1);\n $T(\"NeedImport-2::调用{1}需要先导入{2}\", 2);\n $T(\"NotSupport-1::不支持：{1}\", 1);\n //=====End i18n=====\n\n //流量统计用1像素图片地址，设置为空将不参与统计\n Recorder.TrafficImgUrl = \"//ia.51.la/go1?id=20469973&pvFlag=1\";\n var Traffic = Recorder.Traffic = function (report) {\n if (!isBrowser) return;\n report = report ? \"/\" + RecTxt + \"/Report/\" + report : \"\";\n var imgUrl = Recorder.TrafficImgUrl;\n if (imgUrl) {\n var data = Recorder.Traffic;\n var m = /^(https?:..[^\\/#]*\\/?)[^#]*/i.exec(location.href) || [];\n var host = m[1] || \"http://file/\";\n var idf = (m[0] || host) + report;\n if (imgUrl.indexOf(\"//\") == 0) {\n //给url加上http前缀，如果是file协议下，不加前缀没法用\n if (/^https:/i.test(idf)) {\n imgUrl = \"https:\" + imgUrl;\n } else {\n imgUrl = \"http:\" + imgUrl;\n }\n ;\n }\n ;\n if (report) {\n imgUrl = imgUrl + \"&cu=\" + encodeURIComponent(host + report);\n }\n ;\n if (!data[idf]) {\n data[idf] = 1;\n var img = new Image();\n img.src = imgUrl;\n CLog(\"Traffic Analysis Image: \" + (report || RecTxt + \".TrafficImgUrl=\" + Recorder.TrafficImgUrl));\n }\n ;\n }\n ;\n };\n if (WRec2) {\n CLog($T(\"8HO5::覆盖导入{1}\", 0, RecTxt), 1);\n WRec2.Destroy();\n }\n ;\n Export[RecTxt] = 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browser=typeof window==\"object\" && !!window.document;\n\tvar win=browser?window:Object; //非浏览器环境，Recorder挂载在Object下面\n\tfactory(win,browser);\n\t//umd returnExports.js\n\tif(typeof(define)=='function' && define.amd){\n\t\tdefine(function(){\n\t\t\treturn win.Recorder;\n\t\t});\n\t};\n\tif(typeof(module)=='object' && module.exports){\n\t\tmodule.exports=win.Recorder;\n\t};\n}(function(Export,isBrowser){\n\"use strict\";\n\nvar NOOP=function(){};\nvar IsNum=function(v){return typeof v==\"number\"};\nvar ToJson=function(v){return JSON.stringify(v)};\n\nvar Recorder=function(set){\n\treturn new initFn(set);\n};\nvar LM=Recorder.LM=\"2024-10-20 22:15\";\nvar GitUrl=\"https://github.com/xiangyuecn/Recorder\";\nvar RecTxt=\"Recorder\";\nvar getUserMediaTxt=\"getUserMedia\";\nvar srcSampleRateTxt=\"srcSampleRate\";\nvar sampleRateTxt=\"sampleRate\";\nvar bitRateTxt=\"bitRate\";\nvar CatchTxt=\"catch\";\n\nvar WRec2=Export[RecTxt];//重复加载js\nif(WRec2&&WRec2.LM==LM){\n\tWRec2.CLog(WRec2.i18n.$T(\"K8zP::重复导入{1}\",0,RecTxt),3);\n\treturn;\n};\n\n\n//是否已经打开了全局的麦克风录音，所有工作都已经准备好了，就等接收音频数据了\nRecorder.IsOpen=function(){\n\tvar stream=Recorder.Stream;\n\tif(stream){\n\t\tvar tracks=Tracks_(stream), track=tracks[0];\n\t\tif(track){\n\t\t\tvar state=track.readyState;\n\t\t\treturn state==\"live\"||state==track.LIVE;\n\t\t};\n\t};\n\treturn false;\n};\n/*H5录音时的AudioContext缓冲大小。会影响H5录音时的onProcess调用速率，相对于AudioContext.sampleRate=48000时，4096接近12帧/s，调节此参数可生成比较流畅的回调动画。\n\t取值256, 512, 1024, 2048, 4096, 8192, or 16384\n\t注意，取值不能过低，2048开始不同浏览器可能回调速率跟不上造成音质问题。\n\t一般无需调整，调整后需要先close掉已打开的录音，再open时才会生效。\n*/\nRecorder.BufferSize=4096;\n//销毁已持有的所有全局资源，当要彻底移除Recorder时需要显式的调用此方法\nRecorder.Destroy=function(){\n\tCLog(RecTxt+\" Destroy\");\n\tDisconnect();//断开可能存在的全局Stream、资源\n\t\n\tfor(var k in DestroyList){\n\t\tDestroyList[k]();\n\t};\n};\nvar DestroyList={};\n//登记一个需要销毁全局资源的处理方法\nRecorder.BindDestroy=function(key,call){\n\tDestroyList[key]=call;\n};\n//判断浏览器是否支持录音，随时可以调用。注意：仅仅是检测浏览器支持情况，不会判断和调起用户授权，不会判断是否支持特定格式录音。\nRecorder.Support=function(){\n\tif(!isBrowser) return false;\n\tvar scope=navigator.mediaDevices||{};\n\tif(!scope[getUserMediaTxt]){\n\t\tscope=navigator;\n\t\tscope[getUserMediaTxt]||(scope[getUserMediaTxt]=scope.webkitGetUserMedia||scope.mozGetUserMedia||scope.msGetUserMedia);\n\t};\n\tif(!scope[getUserMediaTxt]){\n\t\treturn false;\n\t};\n\tRecorder.Scope=scope;\n\t\n\tif(!Recorder.GetContext()){\n\t\treturn false;\n\t};\n\treturn true;\n};\n//获取AudioContext对象，如果浏览器不支持将返回null。tryNew=false时返回全局的Recorder.Ctx，Ctx.state可能是closed，仅限用于解码等操作。tryNew=true时会尝试创建新的ctx（不支持close的老浏览器依旧返回全局的），注意用完必须自己调用CloseNewCtx(ctx)关闭；注意：非用户操作（触摸、点击等）时调用返回的ctx.state可能是suspended状态，需要在用户操作时调用ctx.resume恢复成running状态，参考rec的runningContext配置\nRecorder.GetContext=function(tryNew){\n\tif(!isBrowser) return null;\n\tvar AC=window.AudioContext;\n\tif(!AC){\n\t\tAC=window.webkitAudioContext;\n\t};\n\tif(!AC){\n\t\treturn null;\n\t};\n\t\n\tvar ctx=Recorder.Ctx, isNew=0;\n\tif(!ctx){ //241020版本后不再保留打开状态的ctx，原因是iOS不全部关闭时新的ctx有可能不正常\n\t\t//不能反复构造，低版本number of hardware contexts reached maximum (6)\n\t\tctx=Recorder.Ctx=new AC(); isNew=1;\n\t\tRecorder.NewCtxs=Recorder.NewCtxs||[];\n\t\t\n\t\tRecorder.BindDestroy(\"Ctx\",function(){\n\t\t\tvar ctx=Recorder.Ctx;\n\t\t\tif(ctx&&ctx.close){//能关掉就关掉，关不掉就保留着\n\t\t\t\tCloseCtx(ctx);\n\t\t\t\tRecorder.Ctx=0;\n\t\t\t};\n\t\t\tvar arr=Recorder.NewCtxs; Recorder.NewCtxs=[];\n\t\t\tfor(var i=0;i<arr.length;i++)CloseCtx(arr[i]);\n\t\t});\n\t};\n\tif(tryNew && ctx.close){//没法关闭的不允许再创建\n\t\tif(!isNew){\n\t\t\tif(!ctx._useC) CloseCtx(ctx); //关闭全局的，不再保留打开状态\n\t\t\tctx=new AC(); //如果是上面新建的就用上面的，不然就用全新的\n\t\t};\n\t\tctx._useC=1;\n\t\tRecorder.NewCtxs.push(ctx);\n\t};\n\treturn ctx;\n};\n//关闭新创建的AudioContext（之前老版本如果是全局的不关闭，241020版本后全部关闭）\nRecorder.CloseNewCtx=function(ctx){\n\tif(ctx && ctx.close){\n\t\tCloseCtx(ctx);\n\t\tvar arr=Recorder.NewCtxs||[],L=arr.length;\n\t\tfor(var i=0;i<arr.length;i++){\n\t\t\tif(arr[i]==ctx){ arr.splice(i,1); break; }\n\t\t}\n\t\tCLog($T(\"mSxV::剩{1}个GetContext未close\",0,L+\"-1=\"+arr.length),arr.length?3:0);\n\t}\n};\nvar CloseCtx=function(ctx){\n\tif(ctx && ctx.close && !ctx._isC){\n\t\tctx._isC=1;\n\t\tif(ctx.state!=\"closed\"){\n\t\t\ttry{ ctx.close() }catch(e){ CLog(\"ctx close err\",1,e) }\n\t\t}\n\t}\n};\n//当AudioContext的状态是suspended时，调用resume恢复状态，但如果没有用户操作resume可能没有回调，封装解决此回调问题；check(count)返回true继续尝试resume，返回false终止任务（不回调False）\nvar ResumeCtx=Recorder.ResumeCtx=function(ctx,check,True,False){\n\tvar isEnd=0,isBind=0,isLsSC=0,runC=0,EL=\"EventListener\",Tag=\"ResumeCtx \";\n\tvar end=function(err,ok){\n\t\tif(isBind){ bind() }\n\t\tif(!isEnd){ isEnd=1; //回调结果\n\t\t\terr&&False(err,runC);\n\t\t\tok&&True(runC);\n\t\t}\n\t\tif(ok){ //监听后续状态变化\n\t\t\tif(!ctx._LsSC && ctx[\"add\"+EL]) ctx[\"add\"+EL](\"statechange\",run);\n\t\t\tctx._LsSC=1; isLsSC=1;\n\t\t}\n\t};\n\tvar bind=function(add){\n\t\tif(add && isBind) return; isBind=add?1:0;\n\t\tvar types=[\"focus\",\"mousedown\",\"mouseup\",\"touchstart\",\"touchend\"];\n\t\tfor(var i=0;i<types.length;i++)\n\t\t\twindow[(add?\"add\":\"remove\")+EL](types[i],run,true);\n\t};\n\tvar run=function(){\n\t\tvar sVal=ctx.state,spEnd=CtxSpEnd(sVal);\n\t\tif(!isEnd && !check(spEnd?++runC:runC))return end(); //终止，不回调\n\t\tif(spEnd){\n\t\t\tif(isLsSC)CLog(Tag+\"sc \"+sVal,3);\n\t\t\tbind(1); //绑定用户事件尝试恢复\n\t\t\tctx.resume().then(function(){ //resume回调不可靠\n\t\t\t\tif(isLsSC)CLog(Tag+\"sc \"+ctx.state);\n\t\t\t\tend(0,1);\n\t\t\t})[CatchTxt](function(e){ //出错且无法恢复\n\t\t\t\tCLog(Tag+\"error\",1,e);\n\t\t\t\tif(!CtxSpEnd(ctx.state)){\n\t\t\t\t\tend(e.message||\"error\");\n\t\t\t\t}\n\t\t\t});\n\t\t}else if(sVal==\"closed\"){\n\t\t\tif(isLsSC && !ctx._isC)CLog(Tag+\"sc \"+sVal,1); //无法恢复，打个日志\n\t\t\tend(\"ctx closed\");\n\t\t}else{ end(0,1) }; //running 或老的无state\n\t};\n\trun();\n};\nvar CtxSpEnd=Recorder.CtxSpEnd=function(v){\n\treturn v==\"suspended\"||v==\"interrupted\"; //后面这个仅iOS有\n};\nvar CtxState=function(ctx){\n\tvar v=ctx.state,msg=\"ctx.state=\"+v;\n\tif(CtxSpEnd(v))msg+=$T(\"nMIy::（注意：ctx不是running状态，rec.open和start至少要有一个在用户操作(触摸、点击等)时进行调用，否则将在rec.start时尝试进行ctx.resume，可能会产生兼容性问题(仅iOS)，请参阅文档中runningContext配置）\");\n\treturn msg;\n};\n\n\n/*是否启用MediaRecorder.WebM.PCM来进行音频采集连接（如果浏览器支持的话），默认启用，禁用或者不支持时将使用AudioWorklet或ScriptProcessor来连接；MediaRecorder采集到的音频数据比其他方式更好，几乎不存在丢帧现象，所以音质明显会好很多，建议保持开启*/\nvar ConnectEnableWebM=\"ConnectEnableWebM\";\nRecorder[ConnectEnableWebM]=true;\n\n/*是否启用AudioWorklet特性来进行音频采集连接（如果浏览器支持的话），默认禁用，禁用或不支持时将使用过时的ScriptProcessor来连接（如果方法还在的话），当前AudioWorklet的实现在移动端没有ScriptProcessor稳健；ConnectEnableWebM如果启用并且有效时，本参数将不起作用*/\nvar ConnectEnableWorklet=\"ConnectEnableWorklet\";\nRecorder[ConnectEnableWorklet]=false;\n\n/*初始化H5音频采集连接。如果自行提供了sourceStream将只进行一次简单的连接处理。如果是普通麦克风录音，此时的Stream是全局的，Safari上断开后就无法再次进行连接使用，表现为静音，因此使用全部使用全局处理避免调用到disconnect；全局处理也有利于屏蔽底层细节，start时无需再调用底层接口，提升兼容、可靠性。*/\nvar Connect=function(streamStore){\n\tvar bufferSize=streamStore.BufferSize||Recorder.BufferSize;\n\t\n\tvar stream=streamStore.Stream;\n\tvar ctx=stream._c, ctxSR=ctx[sampleRateTxt], srChunk={};\n\t\n\t//获取音频流信息\n\tvar tracks=Tracks_(stream),track=tracks[0],trackSet=null,tsMsg=\"\";\n\tif(track && track.getSettings){\n\t\ttrackSet=track.getSettings();\n\t\tvar trackSR=trackSet[sampleRateTxt];\n\t\tif(trackSR && trackSR!=ctxSR){\n\t\t\ttsMsg=$T(\"eS8i::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，\",0,trackSR,ctxSR);\n\t\t}\n\t}\n\tstream._ts=trackSet;\n\tCLog(tsMsg+\"Stream TrackSet: \"+ToJson(trackSet), tsMsg?3:0);\n\t\n\tvar mediaConn=function(node){\n\t\tvar media=stream._m=ctx.createMediaStreamSource(stream);\n\t\tvar ctxDest=ctx.destination,cmsdTxt=\"createMediaStreamDestination\";\n\t\tif(ctx[cmsdTxt]){\n\t\t\tctxDest=stream._d=ctx[cmsdTxt]();\n\t\t};\n\t\tmedia.connect(node);\n\t\tnode.connect(ctxDest);\n\t}\n\tvar isWebM,isWorklet,badInt,webMTips=\"\";\n\tvar calls=stream._call;\n\t\n\t//浏览器回传的音频数据处理\n\tvar onReceive=function(float32Arr, arrSR){\n\t\tfor(var k0 in calls){//has item\n\t\t\tif(arrSR!=ctxSR){ //MediaRecorder录制的采样率可能和ctx的采样率不同（16k），转换采样率方便统一处理代码也更简单，但音质不会变好，甚至可能会变差一点\n\t\t\t\tsrChunk.index=0;\n\t\t\t\tsrChunk=Recorder.SampleData([float32Arr],arrSR,ctxSR,srChunk,{_sum:1});\n\t\t\t\tvar pcm=srChunk.data;\n\t\t\t\tvar sum=srChunk._sum;\n\t\t\t}else{ //采样率相同，不需要转换采样率\n\t\t\t\tsrChunk={};\n\t\t\t\tvar size=float32Arr.length;\n\t\t\t\t\n\t\t\t\tvar pcm=new Int16Array(size);\n\t\t\t\tvar sum=0;\n\t\t\t\tfor(var j=0;j<size;j++){//floatTo16BitPCM \n\t\t\t\t\tvar s=Math.max(-1,Math.min(1,float32Arr[j]));\n\t\t\t\t\ts=s<0?s*0x8000:s*0x7FFF;\n\t\t\t\t\tpcm[j]=s;\n\t\t\t\t\tsum+=Math.abs(s);\n\t\t\t\t};\n\t\t\t};\n\t\t\t\n\t\t\tfor(var k in calls){\n\t\t\t\tcalls[k](pcm,sum);\n\t\t\t};\n\t\t\t\n\t\t\treturn;\n\t\t};\n\t};\n\t\n\tvar scriptProcessor=\"ScriptProcessor\";//一堆字符串名字，有利于压缩js\n\tvar audioWorklet=\"audioWorklet\";\n\tvar recAudioWorklet=RecTxt+\" \"+audioWorklet;\n\tvar RecProc=\"RecProc\";\n\tvar MediaRecorderTxt=\"MediaRecorder\";\n\tvar MRWebMPCM=MediaRecorderTxt+\".WebM.PCM\";\n\n\n//===================连接方式三=========================\n\t//古董级别的 ScriptProcessor 处理，目前所有浏览器均兼容，虽然是过时的方法，但更稳健，移动端性能比AudioWorklet强\n\tvar oldFn=ctx.createScriptProcessor||ctx.createJavaScriptNode;\n\tvar oldIsBest=$T(\"ZGlf::。由于{1}内部1秒375次回调，在移动端可能会有性能问题导致回调丢失录音变短，PC端无影响，暂不建议开启{1}。\",0,audioWorklet);\n\tvar oldScript=function(){\n\t\tisWorklet=stream.isWorklet=false;\n\t\t_Disconn_n(stream);\n\t\tCLog($T(\"7TU0::Connect采用老的{1}，\",0,scriptProcessor)\n\t\t\t+i18n.get(Recorder[ConnectEnableWorklet]?\n\t\t\t\t$T(\"JwCL::但已设置{1}尝试启用{2}\",2)\n\t\t\t\t:$T(\"VGjB::可设置{1}尝试启用{2}\",2)\n\t\t\t\t,[RecTxt+\".\"+ConnectEnableWorklet+\"=true\",audioWorklet]\n\t\t\t)+webMTips+oldIsBest,3);\n\t\t\n\t\tvar process=stream._p=oldFn.call(ctx,bufferSize,1,1);//单声道，省的数据处理复杂\n\t\tmediaConn(process);\n\t\t\n\t\tprocess.onaudioprocess=function(e){\n\t\t\tvar arr=e.inputBuffer.getChannelData(0);\n\t\t\tonReceive(arr, ctxSR);\n\t\t};\n\t};\n\n\n//===================连接方式二=========================\nvar connWorklet=function(){\n\t//尝试开启AudioWorklet处理\n\tisWebM=stream.isWebM=false;\n\t_Disconn_r(stream);\n\t\n\tisWorklet=stream.isWorklet=!oldFn || Recorder[ConnectEnableWorklet];\n\tvar AwNode=window.AudioWorkletNode;\n\tif(!(isWorklet && ctx[audioWorklet] && AwNode)){\n\t\toldScript();//被禁用 或 不支持，直接使用老的\n\t\treturn;\n\t};\n\tvar clazzUrl=function(){\n\t\tvar xf=function(f){return f.toString().replace(/^function|DEL_/g,\"\").replace(/\\$RA/g,recAudioWorklet)};\n\t\tvar clazz='class '+RecProc+' extends AudioWorkletProcessor{';\n\t\t\tclazz+=\"constructor \"+xf(function(option){\n\t\t\t\tDEL_super(option);\n\t\t\t\tvar This=this,bufferSize=option.processorOptions.bufferSize;\n\t\t\t\tThis.bufferSize=bufferSize;\n\t\t\t\tThis.buffer=new Float32Array(bufferSize*2);//乱给size搞乱缓冲区不管\n\t\t\t\tThis.pos=0;\n\t\t\t\tThis.port.onmessage=function(e){\n\t\t\t\t\tif(e.data.kill){\n\t\t\t\t\t\tThis.kill=true;\n\t\t\t\t\t\t$C.log(\"$RA kill call\");\n\t\t\t\t\t}\n\t\t\t\t};\n\t\t\t\t$C.log(\"$RA .ctor call\", option);\n\t\t\t});\n\t\t\t\n\t\t\t//https://developer.mozilla.org/en-US/docs/Web/API/AudioWorkletProcessor/process 每次回调128个采样数据，1秒375次回调，高频导致移动端性能问题，结果就是回调次数缺斤少两，进而导致丢失数据，PC端似乎没有性能问题\n\t\t\tclazz+=\"process \"+xf(function(input,b,c){//需要等到ctx激活后才会有回调\n\t\t\t\tvar This=this,bufferSize=This.bufferSize;\n\t\t\t\tvar buffer=This.buffer,pos=This.pos;\n\t\t\t\tinput=(input[0]||[])[0]||[];\n\t\t\t\tif(input.length){\n\t\t\t\t\tbuffer.set(input,pos);\n\t\t\t\t\tpos+=input.length;\n\t\t\t\t\t\n\t\t\t\t\tvar len=~~(pos/bufferSize)*bufferSize;\n\t\t\t\t\tif(len){\n\t\t\t\t\t\tthis.port.postMessage({ val: buffer.slice(0,len) });\n\t\t\t\t\t\t\n\t\t\t\t\t\tvar more=buffer.subarray(len,pos);\n\t\t\t\t\t\tbuffer=new Float32Array(bufferSize*2);\n\t\t\t\t\t\tbuffer.set(more);\n\t\t\t\t\t\tpos=more.length;\n\t\t\t\t\t\tThis.buffer=buffer;\n\t\t\t\t\t}\n\t\t\t\t\tThis.pos=pos;\n\t\t\t\t}\n\t\t\t\treturn !This.kill;\n\t\t\t});\n\t\tclazz+='}'\n\t\t\t+'try{'\n\t\t\t\t+'registerProcessor(\"'+RecProc+'\", '+RecProc+')'\n\t\t\t+'}catch(e){$C.error(\"'+recAudioWorklet+' Reg Error\",e)}';\n\t\tclazz=clazz.replace(/\\$C\\./g,\"console.\");//一些编译器会文本替换日志函数\n\t\t//URL.createObjectURL 本地有些浏览器会报 Not allowed to load local resource，直接用dataurl\n\t\treturn \"data:text/javascript;base64,\"+btoa(unescape(encodeURIComponent(clazz)));\n\t};\n\t\n\tvar awNext=function(){//可以继续，没有调用断开\n\t\treturn isWorklet && stream._na;\n\t};\n\tvar nodeAlive=stream._na=function(){\n\t\t//start时会调用，只要没有收到数据就断定AudioWorklet有问题，恢复用老的\n\t\tif(badInt!==\"\"){//没有回调过数据\n\t\t\tclearTimeout(badInt);\n\t\t\tbadInt=setTimeout(function(){\n\t\t\t\tbadInt=0;\n\t\t\t\tif(awNext()){\n\t\t\t\t\tCLog($T(\"MxX1::{1}未返回任何音频，恢复使用{2}\",0,audioWorklet,scriptProcessor),3);\n\t\t\t\t\toldFn&&oldScript();//未来没有老的，可能是误判\n\t\t\t\t};\n\t\t\t},500);\n\t\t};\n\t};\n\tvar createNode=function(){\n\t\tif(!awNext())return;\n\t\tvar node=stream._n=new AwNode(ctx, RecProc, {\n\t\t\tprocessorOptions:{bufferSize:bufferSize}\n\t\t});\n\t\tmediaConn(node);\n\t\tnode.port.onmessage=function(e){\n\t\t\tif(badInt){\n\t\t\t\tclearTimeout(badInt);badInt=\"\";\n\t\t\t};\n\t\t\tif(awNext()){\n\t\t\t\tonReceive(e.data.val, ctxSR);\n\t\t\t}else if(!isWorklet){\n\t\t\t\tCLog($T(\"XUap::{1}多余回调\",0,audioWorklet),3);\n\t\t\t};\n\t\t};\n\t\tCLog($T(\"yOta::Connect采用{1}，设置{2}可恢复老式{3}\",0,audioWorklet,RecTxt+\".\"+ConnectEnableWorklet+\"=false\",scriptProcessor)+webMTips+oldIsBest,3);\n\t};\n\t\n\t//如果start时的resume和下面的构造node同时进行，将会导致部分浏览器崩溃 (STATUS_ACCESS_VIOLATION)，源码assets中 ztest_chrome_bug_AudioWorkletNode.html 可测试。所以，将所有代码套到resume里面（不管catch），避免出现这个问题\n\tvar ctxOK=function(){\n\t\tif(!awNext())return;\n\t\tif(ctx[RecProc]){\n\t\t\tcreateNode();\n\t\t\treturn;\n\t\t};\n\t\tvar url=clazzUrl();\n\t\tctx[audioWorklet].addModule(url).then(function(e){\n\t\t\tif(!awNext())return;\n\t\t\tctx[RecProc]=1;\n\t\t\tcreateNode();\n\t\t\tif(badInt){//重新计时\n\t\t\t\tnodeAlive();\n\t\t\t};\n\t\t})[CatchTxt](function(e){ //fix 关键字，保证catch压缩时保持字符串形式\n\t\t\tCLog(audioWorklet+\".addModule Error\",1,e);\n\t\t\tawNext()&&oldScript();\n\t\t});\n\t};\n\tResumeCtx(ctx,function(){ return awNext() } ,ctxOK,ctxOK);\n};\n\n\n//===================连接方式一=========================\nvar connWebM=function(){\n\t//尝试开启MediaRecorder录制webm+pcm处理\n\tvar MR=window[MediaRecorderTxt];\n\tvar onData=\"ondataavailable\";\n\tvar webmType=\"audio/webm; codecs=pcm\";\n\tisWebM=stream.isWebM=Recorder[ConnectEnableWebM];\n\t\n\tvar supportMR=MR && (onData in MR.prototype) && MR.isTypeSupported(webmType);\n\twebMTips=supportMR?\"\":$T(\"VwPd::（此浏览器不支持{1}）\",0,MRWebMPCM);\n\tif(!isWebM || !supportMR){\n\t\tconnWorklet(); //被禁用 或 不支持MediaRecorder 或 不支持webm+pcm\n\t\treturn;\n\t}\n\t\n\tvar mrNext=function(){//可以继续，没有调用断开\n\t\treturn isWebM && stream._ra;\n\t};\n\tstream._ra=function(){\n\t\t//start时会调用，只要没有收到数据就断定MediaRecorder有问题，降级处理\n\t\tif(badInt!==\"\"){//没有回调过数据\n\t\t\tclearTimeout(badInt);\n\t\t\tbadInt=setTimeout(function(){\n\t\t\t\t//badInt=0; 保留给nodeAlive继续判断\n\t\t\t\tif(mrNext()){\n\t\t\t\t\tCLog($T(\"vHnb::{1}未返回任何音频，降级使用{2}\",0,MediaRecorderTxt,audioWorklet),3);\n\t\t\t\t\tconnWorklet();\n\t\t\t\t};\n\t\t\t},500);\n\t\t};\n\t};\n\t\n\tvar mrSet=Object.assign({mimeType:webmType}, Recorder.ConnectWebMOptions);\n\tvar mr=stream._r=new MR(stream, mrSet);\n\tvar webmData=stream._rd={};\n\tmr[onData]=function(e){\n\t\t//提取webm中的pcm数据，提取失败就等着badInt超时降级处理\n\t\tvar reader=new FileReader();\n\t\treader.onloadend=function(){\n\t\t\tif(mrNext()){\n\t\t\t\tvar f32arr=WebM_Extract(new Uint8Array(reader.result),webmData);\n\t\t\t\tif(!f32arr)return;\n\t\t\t\tif(f32arr==-1){//无法提取，立即降级\n\t\t\t\t\tconnWorklet();\n\t\t\t\t\treturn;\n\t\t\t\t};\n\t\t\t\t\n\t\t\t\tif(badInt){\n\t\t\t\t\tclearTimeout(badInt);badInt=\"\";\n\t\t\t\t};\n\t\t\t\tonReceive(f32arr, webmData.webmSR);\n\t\t\t}else if(!isWebM){\n\t\t\t\tCLog($T(\"O9P7::{1}多余回调\",0,MediaRecorderTxt),3);\n\t\t\t};\n\t\t};\n\t\treader.readAsArrayBuffer(e.data);\n\t};\n\tmr.start(~~(bufferSize/48));//按48k时的回调间隔\n\tCLog($T(\"LMEm::Connect采用{1}，设置{2}可恢复使用{3}或老式{4}\",0,MRWebMPCM,RecTxt+\".\"+ConnectEnableWebM+\"=false\",audioWorklet,scriptProcessor));\n};\n\n\tconnWebM();\n};\nvar ConnAlive=function(stream){\n\tif(stream._na) stream._na(); //检查AudioWorklet连接是否有效，无效就回滚到老的ScriptProcessor\n\tif(stream._ra) stream._ra(); //检查MediaRecorder连接是否有效，无效就降级处理\n};\nvar _Disconn_n=function(stream){\n\tstream._na=null;\n\tif(stream._n){\n\t\tstream._n.port.postMessage({kill:true});\n\t\tstream._n.disconnect();\n\t\tstream._n=null;\n\t};\n};\nvar _Disconn_r=function(stream){\n\tstream._ra=null;\n\tif(stream._r){\n\t\ttry{ stream._r.stop() }catch(e){ CLog(\"mr stop err\",1,e) }\n\t\tstream._r=null;\n\t};\n};\nvar Disconnect=function(streamStore){\n\tstreamStore=streamStore||Recorder;\n\tvar isGlobal=streamStore==Recorder;\n\t\n\tvar stream=streamStore.Stream;\n\tif(stream){\n\t\tif(stream._m){\n\t\t\tstream._m.disconnect();\n\t\t\tstream._m=null;\n\t\t};\n\t\tif(!stream._RC && stream._c){//提供的runningContext不处理\n\t\t\tRecorder.CloseNewCtx(stream._c);\n\t\t};\n\t\tstream._RC=null; stream._c=null;\n\t\tif(stream._d){\n\t\t\tStopS_(stream._d.stream);\n\t\t\tstream._d=null;\n\t\t};\n\t\tif(stream._p){\n\t\t\tstream._p.disconnect();\n\t\t\tstream._p.onaudioprocess=stream._p=null;\n\t\t};\n\t\t_Disconn_n(stream);\n\t\t_Disconn_r(stream);\n\t\t\n\t\tif(isGlobal){//全局的时候，要把流关掉（麦克风），直接提供的流不处理\n\t\t\tStopS_(stream);\n\t\t};\n\t};\n\tstreamStore.Stream=0;\n};\n//关闭一个音频流\nvar StopS_=Recorder.StopS_=function(stream){\n\tvar tracks=Tracks_(stream);\n\tfor(var i=0;i<tracks.length;i++){\n\t\tvar track=tracks[i];\n\t\ttrack.stop&&track.stop();\n\t};\n\tstream.stop&&stream.stop();\n};\n//获取流中的所有轨道\nvar Tracks_=function(stream){\n\tvar arr1=0,arr2=0,arr=[];\n\t//stream.getTracks() 得到的sourceStream还得去排序 不然第一个可能是video\n\tif(stream.getAudioTracks){ arr1=stream.getAudioTracks(); arr2=stream.getVideoTracks(); }\n\tif(!arr1){ arr1=stream.audioTracks; arr2=stream.videoTracks; }\n\tfor(var i=0,L=arr1?arr1.length:0;i<L;i++)arr.push(arr1[i]); //音频放前面，方便取[0]\n\tfor(var i=0,L=arr2?arr2.length:0;i<L;i++)arr.push(arr2[i]);\n\treturn arr;\n};\n\n/*对pcm数据的采样率进行转换\npcmDatas: [[Int16,...]] pcm片段列表，二维数组里面是Int16Array，也可传Float32Array（会转成Int16Array）\npcmSampleRate:48000 pcm数据的采样率\nnewSampleRate:16000 需要转换成的采样率，241020版本后支持转成任意采样率，之前老版本newSampleRate>=pcmSampleRate时不会进行任何处理，小于时会进行重新采样\nprevChunkInfo:{} 可选，上次调用时的返回值，用于连续转换，本次调用将从上次结束位置开始进行处理。或可自行定义一个ChunkInfo从pcmDatas指定的位置开始进行转换\noption:{ 可选，配置项\n\t\tframeSize:123456 帧大小，每帧的PCM Int16的数量，采样率转换后的pcm长度为frameSize的整数倍，用于连续转换。目前仅在mp3格式时才有用，frameSize取值为1152，这样编码出来的mp3时长和pcm的时长完全一致，否则会因为mp3最后一帧录音不够填满时添加填充数据导致mp3的时长变长。\n\t\tframeType:\"\" 帧类型，一般为rec.set.type，提供此参数时无需提供frameSize，会自动使用最佳的值给frameSize赋值，目前仅支持mp3=1152(MPEG1 Layer3的每帧采采样数)，其他类型=1。\n\t\t\t以上两个参数用于连续转换时使用，最多使用一个，不提供时不进行帧的特殊处理，提供时必须同时提供prevChunkInfo才有作用。最后一段数据处理时无需提供帧大小以便输出最后一丁点残留数据。\n\t}\n\n返回ChunkInfo:{\n\t//可定义，从指定位置开始转换到结尾\n\tindex:0 pcmDatas已处理到的索引；比如每次都是单个pcm需要连续处理时，可每次调用前重置成0，pcmDatas仅需传入`[pcm]`固定一个元素\n\toffset:0.0 已处理到的index对应的pcm中的偏移的下一个位置（提升采样率时为结果的pcm）\n\traisePrev:null 提升采样率时的前一个pcm结果采样值\n\t\n\t//可定义，指定的一个滤波配置：默认使用Recorder.IIRFilter低通滤波（可有效抑制混叠产生的杂音，小采样率大于高采样率的75%时不默认滤波），如果提供了配置但fn为null时将不滤波；sr、srn为此滤波函数对应的初始化采样率，当采样率和参数的不一致时将重新设为默认函数\n\tfilter:null||{fn:fn(sample),sr:pcmSampleRate,srn:newSampleRate}\n\t\n\t//仅作为返回值\n\tframeNext:null||[Int16,...] 下一帧的部分数据，frameSize设置了的时候才可能会有\n\tsampleRate:16000 结果的采样率=newSampleRate，老版本<=newSampleRate\n\tdata:[Int16,...] 转换后的PCM结果；如果是连续转换，并且pcmDatas中并没有新数据时，data的长度可能为0\n}\n*/\nRecorder.SampleData=function(pcmDatas,pcmSampleRate,newSampleRate,prevChunkInfo,option){\n\tvar Txt=\"SampleData\";\n\tprevChunkInfo||(prevChunkInfo={});\n\tvar index=prevChunkInfo.index||0;\n\tvar offset=prevChunkInfo.offset||0;\n\tvar raisePrev=prevChunkInfo.raisePrev||0;\n\t\n\tvar filter=prevChunkInfo.filter;\n\tif(filter&&filter.fn&&(filter.sr&&filter.sr!=pcmSampleRate || filter.srn&&filter.srn!=newSampleRate)){\n\t\tfilter=null; CLog($T(\"d48C::{1}的filter采样率变了，重设滤波\",0,Txt),3);\n\t};\n\tif(!filter){ //采样率差距比较大才开启低通滤波\n\t\tif(newSampleRate<=pcmSampleRate){ //降低采样率或不变，最高频率用新采样率频率的3/4\n\t\t\tvar freq=newSampleRate>pcmSampleRate*3/4?0: newSampleRate/2 *3/4;\n\t\t\tfilter={fn:freq?Recorder.IIRFilter(true,pcmSampleRate,freq):0};\n\t\t}else{ //提升采样率，最高频率用原始采样率频率的3/4\n\t\t\tvar freq=pcmSampleRate>newSampleRate*3/4?0: pcmSampleRate/2 *3/4;\n\t\t\tfilter={fn:freq?Recorder.IIRFilter(true,newSampleRate,freq):0};\n\t\t};\n\t};\n\tfilter.sr=pcmSampleRate;\n\tfilter.srn=newSampleRate;\n\tvar filterFn=filter.fn;\n\t\n\tvar frameNext=prevChunkInfo.frameNext||[];\n\toption||(option={});\n\tvar frameSize=option.frameSize||1;\n\tif(option.frameType){\n\t\tframeSize=option.frameType==\"mp3\"?1152:1;\n\t};\n\tvar useSum=option._sum, _sum=0; //内部用的，sum不考虑配置了frame\n\t\n\tvar nLen=pcmDatas.length;\n\tif(index>nLen+1){\n\t\tCLog($T(\"tlbC::{1}似乎传入了未重置chunk {2}\",0,Txt,index+\">\"+nLen),3);\n\t};\n\tvar size=0;\n\tfor(var i=index;i<nLen;i++){\n\t\tsize+=pcmDatas[i].length;\n\t};\n\t\n\t//采样 https://www.cnblogs.com/blqw/p/3782420.html\n\tvar step=pcmSampleRate/newSampleRate;\n\tif(step>1){//新采样低于录音采样，进行抽样\n\t\tsize=Math.max(0,size-Math.floor(offset));\n\t\tsize=Math.floor(size/step);\n\t}else if(step<1){//新采样高于录音采样，插值处理\n\t\tvar raiseStep=1/step;\n\t\tsize=Math.floor(size*raiseStep);\n\t};\n\t\n\tsize+=frameNext.length;\n\tvar res=new Int16Array(size);\n\tvar idx=0;\n\t//添加上一次不够一帧的剩余数据\n\tfor(var i=0;i<frameNext.length;i++){\n\t\tres[idx]=frameNext[i];\n\t\tidx++;\n\t};\n\t//处理数据\n\tfor (;index<nLen;index++) {\n\t\tvar o=pcmDatas[index], isF32=o instanceof Float32Array;\n\t\tvar i=offset,il=o.length;\n\t\tvar F=filterFn&&filterFn.Embed,F1=0,F2=0,Fx=0,Fy=0;//低通滤波后的数据\n\t\t\n\t\tif(step<1){ //提升采样率\n\t\t\tvar idx1=idx+i, prev=raisePrev;\n\t\t\tfor(var i0=0;i0<il;i0++){\n\t\t\t\tvar oVal=o[i0];\n\t\t\t\tif(isF32){//floatTo16BitPCM \n\t\t\t\t\toVal=Math.max(-1,Math.min(1,oVal));\n\t\t\t\t\toVal=oVal<0?oVal*0x8000:oVal*0x7FFF;\n\t\t\t\t}\n\t\t\t\t\n\t\t\t\tvar pos=Math.floor(idx1);\n\t\t\t\tidx1+=raiseStep;\n\t\t\t\tvar end=Math.floor(idx1);\n\t\t\t\n\t\t\t\t//简单的从prev平滑填充到cur，有效减少转换引入的杂音\n\t\t\t\tvar n=(oVal-prev)/(end-pos);\n\t\t\t\tfor(var j=1;pos<end;pos++,j++){\n\t\t\t\t\tvar s=Math.floor(prev+(j*n));\n\t\t\t\t\tif(F){//IIRFilter代码内置，比函数调用快4倍\n\t\t\t\t\t\tFx=s;\n\t\t\t\t\t\tFy=F.b0 * Fx + F.b1 * F.x1 + F.b0 * F.x2 - F.a1 * F.y1 - F.a2 * F.y2;\n\t\t\t\t\t\tF.x2 = F.x1; F.x1 = Fx; F.y2 = F.y1; F.y1 = Fy;\n\t\t\t\t\t\ts=Fy;\n\t\t\t\t\t}else{ s=filterFn?filterFn(s):s; }\n\t\t\t\t\t\n\t\t\t\t\tif(s>0x7FFF) s=0x7FFF; else if(s<-0x8000) s=-0x8000; //Int16越界处理\n\t\t\t\t\tif(useSum) _sum+=Math.abs(s);\n\t\t\t\t\tres[pos]=s;\n\t\t\t\t\tidx++;\n\t\t\t\t};\n\t\t\t\t\n\t\t\t\tprev=raisePrev=oVal;\n\t\t\t\ti+=raiseStep;//插值\n\t\t\t}\n\t\t\toffset=i%1;\n\t\t\tcontinue;\n\t\t};\n\t\t//降低采样率或不变\n\t\tfor(var i0=0,i2=0;i0<il;i0++,i2++){\n\t\t\tif(i2<il){\n\t\t\t\tvar oVal=o[i2];\n\t\t\t\tif(isF32){//floatTo16BitPCM \n\t\t\t\t\toVal=Math.max(-1,Math.min(1,oVal));\n\t\t\t\t\toVal=oVal<0?oVal*0x8000:oVal*0x7FFF;\n\t\t\t\t}\n\t\t\t\tif(F){//IIRFilter代码内置，比函数调用快4倍\n\t\t\t\t\tFx=oVal;\n\t\t\t\t\tFy=F.b0 * Fx + F.b1 * F.x1 + F.b0 * F.x2 - F.a1 * F.y1 - F.a2 * F.y2;\n\t\t\t\t\tF.x2 = F.x1; F.x1 = Fx; F.y2 = F.y1; F.y1 = Fy;\n\t\t\t\t}else{ Fy=filterFn?filterFn(oVal):oVal; }\n\t\t\t}\n\t\t\tF1=F2; F2=Fy;\n\t\t\tif(i2==0){ i0--; continue; } //首次只计算o[0]\n\t\t\t\n\t\t\t//https://www.cnblogs.com/xiaoqi/p/6993912.html\n\t\t\t//当前点=当前点+到后面一个点之间的增量，音质比直接简单抽样好些\n\t\t\tvar before = Math.floor(i);\n\t\t\tif(i0!=before)continue;\n\t\t\tvar after = Math.ceil(i);\n\t\t\tvar atPoint = i - before;\n\t\t\t\n\t\t\tvar beforeVal=F1;\n\t\t\tvar afterVal=after<il ? F2 : beforeVal; //后个点越界了，忽略不计\n\t\t\tvar val=beforeVal+(afterVal-beforeVal)*atPoint;\n\t\t\t\n\t\t\tif(val>0x7FFF) val=0x7FFF; else if(val<-0x8000) val=-0x8000; //Int16越界处理\n\t\t\tif(useSum) _sum+=Math.abs(val);\n\t\t\tres[idx]=val;\n\t\t\t\n\t\t\tidx++;\n\t\t\ti+=step;//抽样\n\t\t};\n\t\toffset=Math.max(0, i-il); //不太可能出现负数\n\t};\n\tif(step<1 && idx+1==size){ //提升采样率时可能缺1个，直接删除结尾1个\n\t\tsize--; res=new Int16Array(res.buffer.slice(0, size*2));\n\t};\n\tif(idx-1!=size && idx!=size)CLog(Txt+\" idx:\"+idx+\" != size:\"+size,3); //越界1个\n\t\n\t//帧处理\n\tframeNext=null;\n\tvar frameNextSize=size%frameSize;\n\tif(frameNextSize>0){\n\t\tvar u8Pos=(size-frameNextSize)*2;\n\t\tframeNext=new Int16Array(res.buffer.slice(u8Pos));\n\t\tres=new Int16Array(res.buffer.slice(0,u8Pos));\n\t};\n\t\n\tvar obj={\n\t\tindex:index\n\t\t,offset:offset\n\t\t,raisePrev:raisePrev\n\t\t,filter:filter\n\t\t\n\t\t,frameNext:frameNext\n\t\t,sampleRate:newSampleRate\n\t\t,data:res\n\t};\n\tif(useSum) obj._sum=_sum;\n\treturn obj;\n};\n\n/*IIR低通、高通滤波，移植自：https://gitee.com/52jian/digital-audio-filter AudioFilter.java\n\tuseLowPass: true或false，true为低通滤波，false为高通滤波\n\tsampleRate: 待处理pcm的采样率\n\tfreq: 截止频率Hz，最大频率为sampleRate/2，低通时会切掉高于此频率的声音，高通时会切掉低于此频率的声音，注意滤波并非100%的切掉不需要的声音，而是减弱频率对应的声音，离截止频率越远对应声音减弱越厉害，离截止频率越近声音就几乎无衰减\n\t返回的是一个函数，用此函数对pcm的每个采样值按顺序进行处理即可（不同pcm不可共用）；注意此函数返回值可能会越界超过Int16范围，自行限制一下即可：Math.min(Math.max(val,-0x8000),0x7FFF)\n可重新赋值一个函数，来改变Recorder的默认行为，比如SampleData中的低通滤波*/\nRecorder.IIRFilter=function(useLowPass, sampleRate, freq){\n\tvar ov = 2 * Math.PI * freq / sampleRate;\n\tvar sn = Math.sin(ov);\n\tvar cs = Math.cos(ov);\n\tvar alpha = sn / 2;\n\t\n\tvar a0 = 1 + alpha;\n\tvar a1 = (-2 * cs) / a0;\n\tvar a2 = (1 - alpha) / a0;\n\tif(useLowPass){\n\t\tvar b0 = (1 - cs) / 2 / a0;\n\t\tvar b1 = (1 - cs) / a0;\n\t}else{\n\t\tvar b0 = (1 + cs) / 2 / a0;\n\t\tvar b1 = -(1 + cs) / a0;\n\t}\n\t\n\tvar x1=0,x2=0,y=0,y1=0,y2=0;\n\tvar fn=function(x){\n\t\ty = b0 * x + b1 * x1 + b0 * x2 - a1 * y1 - a2 * y2;\n\t\tx2 = x1; x1 = x;\n\t\ty2 = y1; y1 = y;\n\t\treturn y;\n\t};\n\tfn.Embed={x1:0,x2:0,y1:0,y2:0,b0:b0,b1:b1,a1:a1,a2:a2};\n\treturn fn;\n};\n\n\n/*计算音量百分比的一个方法\npcmAbsSum: pcm Int16所有采样的绝对值的和\npcmLength: pcm长度\n返回值：0-100，主要当做百分比用\n注意：这个不是分贝，因此没用volume当做名称*/\nRecorder.PowerLevel=function(pcmAbsSum,pcmLength){\n\t/*计算音量 https://blog.csdn.net/jody1989/article/details/73480259\n\t更高灵敏度算法:\n\t\t限定最大感应值10000\n\t\t\t线性曲线：低音量不友好\n\t\t\t\tpower/10000*100 \n\t\t\t对数曲线：低音量友好，但需限定最低感应值\n\t\t\t\t(1+Math.log10(power/10000))*100\n\t*/\n\tvar power=(pcmAbsSum/pcmLength) || 0;//NaN\n\tvar level;\n\tif(power<1251){//1250的结果10%，更小的音量采用线性取值\n\t\tlevel=Math.round(power/1250*10);\n\t}else{\n\t\tlevel=Math.round(Math.min(100,Math.max(0,(1+Math.log(power/10000)/Math.log(10))*100)));\n\t};\n\treturn level;\n};\n\n/*计算音量，单位dBFS（满刻度相对电平）\nmaxSample: 为16位pcm采样的绝对值中最大的一个（计算峰值音量），或者为pcm中所有采样的绝对值的平局值\n返回值：-100~0 （最大值0dB，最小值-100代替-∞）\n*/\nRecorder.PowerDBFS=function(maxSample){\n\tvar val=Math.max(0.1, maxSample||0),Pref=0x7FFF;\n\tval=Math.min(val,Pref);\n\t//https://www.logiclocmusic.com/can-you-tell-the-decibel/\n\t//https://blog.csdn.net/qq_17256689/article/details/120442510\n\tval=20*Math.log(val/Pref)/Math.log(10);\n\treturn Math.max(-100,Math.round(val));\n};\n\n\n\n\n//带时间的日志输出，可设为一个空函数来屏蔽日志输出\n//CLog(msg,errOrLogMsg, logMsg...) err为数字时代表日志类型1:error 2:log默认 3:warn，否则当做内容输出，第一个参数不能是对象因为要拼接时间，后面可以接无数个输出参数\nRecorder.CLog=function(msg,err){\n\tif(typeof console!=\"object\")return;\n\tvar now=new Date();\n\tvar t=(\"0\"+now.getMinutes()).substr(-2)\n\t\t+\":\"+(\"0\"+now.getSeconds()).substr(-2)\n\t\t+\".\"+(\"00\"+now.getMilliseconds()).substr(-3);\n\tvar recID=this&&this.envIn&&this.envCheck&&this.id;\n\tvar arr=[\"[\"+t+\" \"+RecTxt+(recID?\":\"+recID:\"\")+\"]\"+msg];\n\tvar a=arguments,cwe=Recorder.CLog;\n\tvar i=2,fn=cwe.log||console.log;\n\tif(IsNum(err)){\n\t\tfn=err==1?cwe.error||console.error:err==3?cwe.warn||console.warn:fn;\n\t}else{\n\t\ti=1;\n\t};\n\tfor(;i<a.length;i++){\n\t\tarr.push(a[i]);\n\t};\n\tif(IsLoser){//古董浏览器，仅保证基本的可执行不代码异常\n\t\tfn&&fn(\"[IsLoser]\"+arr[0],arr.length>1?arr:\"\");\n\t}else{\n\t\tfn.apply(console,arr);\n\t};\n};\nvar CLog=function(){ Recorder.CLog.apply(this,arguments); };\nvar IsLoser=true;try{IsLoser=!console.log.apply;}catch(e){};\n\n\n\n\nvar ID=0;\nfunction initFn(set){\n\tvar This=this; This.id=++ID;\n\t\n\t//如果开启了流量统计，这里将发送一个图片请求\n\tTraffic();\n\t\n\t\n\tvar o={\n\t\ttype:\"mp3\" //输出类型：mp3,wav，wav输出文件尺寸超大不推荐使用，但mp3编码支持会导致js文件超大，如果不需支持mp3可以使js文件大幅减小\n\t\t//,bitRate:16 //比特率 wav:16或8位，MP3：8kbps 1k/s，8kbps 2k/s 录音文件很小\n\t\t\n\t\t//,sampleRate:16000 //采样率，wav格式大小=sampleRate*时间；mp3此项对低比特率有影响，高比特率几乎无影响。\n\t\t\t\t\t//wav任意值，mp3取值范围：48000, 44100, 32000, 24000, 22050, 16000, 12000, 11025, 8000\n\t\t\t\t\t//采样率参考https://www.cnblogs.com/devin87/p/mp3-recorder.html\n\t\t\n\t\t,onProcess:NOOP //fn(buffers,powerLevel,bufferDuration,bufferSampleRate,newBufferIdx,asyncEnd) buffers=[[Int16,...],...]：缓冲的PCM数据，为从开始录音到现在的所有pcm片段；powerLevel：当前缓冲的音量级别0-100，bufferDuration：已缓冲时长，bufferSampleRate：缓冲使用的采样率（当type支持边录边转码(Worker)时，此采样率和设置的采样率相同，否则不一定相同）；newBufferIdx:本次回调新增的buffer起始索引；asyncEnd:fn() 如果onProcess是异步的(返回值为true时)，处理完成时需要调用此回调，如果不是异步的请忽略此参数，此方法回调时必须是真异步（不能真异步时需用setTimeout包裹）。onProcess返回值：如果返回true代表开启异步模式，在某些大量运算的场合异步是必须的，必须在异步处理完成时调用asyncEnd(不能真异步时需用setTimeout包裹)，在onProcess执行后新增的buffer会全部替换成空数组，因此本回调开头应立即将newBufferIdx到本次回调结尾位置的buffer全部保存到另外一个数组内，处理完成后写回buffers中本次回调的结尾位置。\n\t\t\n\t\t//*******高级设置******\n\t\t//,sourceStream:MediaStream Object\n\t\t\t\t//可选直接提供一个媒体流，从这个流中录制、实时处理音频数据（当前Recorder实例独享此流）；不提供时为普通的麦克风录音，由getUserMedia提供音频流（所有Recorder实例共享同一个流）\n\t\t\t\t//比如：audio、video标签dom节点的captureStream方法（实验特性，不同浏览器支持程度不高）返回的流；WebRTC中的remote流；自己创建的流等\n\t\t\t\t//注意：流内必须至少存在一条音轨(Audio Track)，比如audio标签必须等待到可以开始播放后才会有音轨，否则open会失败\n\t\t\n\t\t//,runningContext:AudioContext\n\t\t\t\t//可选提供一个state为running状态的AudioContext对象(ctx)；默认会在rec.open时自动创建一个新的ctx，无用户操作（触摸、点击等）时调用rec.open的ctx.state可能为suspended，会在rec.start时尝试进行ctx.resume，如果也无用户操作ctx.resume可能不会恢复成running状态（目前仅iOS上有此兼容性问题），导致无法去读取媒体流，这时请提前在用户操作时调用Recorder.GetContext(true)来得到一个running状态AudioContext（用完需调用CloseNewCtx(ctx)关闭）\n\t\t\n\t\t//,audioTrackSet:{ deviceId:\"\",groupId:\"\", autoGainControl:true, echoCancellation:true, noiseSuppression:true }\n\t\t\t\t//普通麦克风录音时getUserMedia方法的audio配置参数，比如指定设备id，回声消除、降噪开关；注意：提供的任何配置值都不一定会生效\n\t\t\t\t//由于麦克风是全局共享的，所以新配置后需要close掉以前的再重新open\n\t\t\t\t//同样可配置videoTrackSet，更多参考: https://developer.mozilla.org/en-US/docs/Web/API/MediaTrackConstraints\n\t\t\n\t\t//,disableEnvInFix:false 内部参数，禁用设备卡顿时音频输入丢失补偿功能\n\t\t\n\t\t//,takeoffEncodeChunk:NOOP //fn(chunkBytes) chunkBytes=[Uint8,...]：实时编码环境下接管编码器输出，当编码器实时编码出一块有效的二进制音频数据时实时回调此方法；参数为二进制的Uint8Array，就是编码出来的音频数据片段，所有的chunkBytes拼接在一起即为完整音频。本实现的想法最初由QQ2543775048提出\n\t\t\t\t//当提供此回调方法时，将接管编码器的数据输出，编码器内部将放弃存储生成的音频数据；如果当前编码器或环境不支持实时编码处理，将在open时直接走fail逻辑\n\t\t\t\t//因此提供此回调后调用stop方法将无法获得有效的音频数据，因为编码器内没有音频数据，因此stop时返回的blob将是一个字节长度为0的blob\n\t\t\t\t//大部分录音格式编码器都支持实时编码（边录边转码），比如mp3格式：会实时的将编码出来的mp3片段通过此方法回调，所有的chunkBytes拼接到一起即为完整的mp3，此种拼接的结果比mock方法实时生成的音质更加，因为天然避免了首尾的静默\n\t\t\t\t//不支持实时编码的录音格式不可以提供此回调（wav格式不支持，因为wav文件头中需要提供文件最终长度），提供了将在open时直接走fail逻辑\n\t};\n\t\n\tfor(var k in set){\n\t\to[k]=set[k];\n\t};\n\tThis.set=o;\n\t\n\tvar vB=o[bitRateTxt],vS=o[sampleRateTxt]; //校验配置参数\n\tif(vB&&!IsNum(vB) || vS&&!IsNum(vS)){\n\t\tThis.CLog($T.G(\"IllegalArgs-1\",[$T(\"VtS4::{1}和{2}必须是数值\",0,sampleRateTxt,bitRateTxt)]),1,set);\n\t};\n\to[bitRateTxt]=+vB||16;\n\to[sampleRateTxt]=+vS||16000;\n\t\n\tThis.state=0;//运行状态，0未录音 1录音中 2暂停 3等待ctx激活\n\tThis._S=9;//stop同步锁，stop可以阻止open过程中还未运行的start\n\tThis.Sync={O:9,C:9};//和Recorder.Sync一致，只不过这个是非全局的，仅用来简化代码逻辑，无实际作用\n};\n//同步锁，控制对Stream的竞争；用于close时中断异步的open；一个对象open如果变化了都要阻止close，Stream的控制权交个新的对象\nRecorder.Sync={/*open*/O:9,/*close*/C:9};\n\nRecorder.prototype=initFn.prototype={\n\tCLog:CLog\n\t\n\t//流相关的数据存储在哪个对象里面；如果提供了sourceStream，数据直接存储在当前对象中，否则存储在全局\n\t,_streamStore:function(){\n\t\tif(this.set.sourceStream){\n\t\t\treturn this;\n\t\t}else{\n\t\t\treturn Recorder;\n\t\t}\n\t}\n\t//当前实例用到的Stream，可能是全局的，也可能是独享的\n\t,_streamGet:function(){\n\t\treturn this._streamStore().Stream;\n\t}\n\t//当前实例用到的AudioContext，可能是全局的，也可能是独享的\n\t,_streamCtx:function(){\n\t\tvar m=this._streamGet();\n\t\treturn m&&m._c;\n\t}\n\t\n\t//打开录音资源True(),False(msg,isUserNotAllow)，需要调用close。注意：此方法是异步的；一般使用时打开，用完立即关闭；可重复调用，可用来测试是否能录音；open和start至少有一个应当在用户操作（触摸、点击等）下进行调用，原因参考runningContext配置\n\t,open:function(True,False){\n\t\tvar This=this,set=This.set,streamStore=This._streamStore(),newCtx=0;\n\t\tTrue=True||NOOP;\n\t\tvar failCall=function(errMsg,isUserNotAllow){\n\t\t\tisUserNotAllow=!!isUserNotAllow;\n\t\t\tThis.CLog($T(\"5tWi::录音open失败：\")+errMsg+\",isUserNotAllow:\"+isUserNotAllow,1);\n\t\t\tif(newCtx)Recorder.CloseNewCtx(newCtx);\n\t\t\tFalse&&False(errMsg,isUserNotAllow);\n\t\t};\n\t\t\n\t\tThis._streamTag=getUserMediaTxt;\n\t\tvar ok=function(){\n\t\t\tThis.CLog(\"open ok, id:\"+This.id+\" stream:\"+This._streamTag);\n\t\t\tTrue();\n\t\t\t\n\t\t\tThis._SO=0;//解除stop对open中的start调用的阻止\n\t\t};\n\t\t\n\t\t\n\t\t//同步锁\n\t\tvar Lock=streamStore.Sync;\n\t\tvar lockOpen=++Lock.O,lockClose=Lock.C;\n\t\tThis._O=This._O_=lockOpen;//记住当前的open，如果变化了要阻止close，这里假定了新对象已取代当前对象并且不再使用\n\t\tThis._SO=This._S;//记住open过程中的stop，中途任何stop调用后都不能继续open中的start\n\t\tvar lockFail=function(){\n\t\t\t//允许多次open，但不允许任何一次close，或者自身已经调用了关闭\n\t\t\tif(lockClose!=Lock.C || !This._O){\n\t\t\t\tvar err=$T(\"dFm8::open被取消\");\n\t\t\t\tif(lockOpen==Lock.O){\n\t\t\t\t\t//无新的open，已经调用了close进行取消，此处应让上次的close明确生效\n\t\t\t\t\tThis.close();\n\t\t\t\t}else{\n\t\t\t\t\terr=$T(\"VtJO::open被中断\");\n\t\t\t\t};\n\t\t\t\tfailCall(err);\n\t\t\t\treturn true;\n\t\t\t};\n\t\t};\n\t\t\n\t\t//环境配置检查\n\t\tif(!isBrowser){\n\t\t\tfailCall($T.G(\"NonBrowser-1\",[\"open\"])+$T(\"EMJq::，可尝试使用RecordApp解决方案\")+\"(\"+GitUrl+\"/tree/master/app-support-sample)\");\n\t\t\treturn;\n\t\t};\n\t\tvar checkMsg=This.envCheck({envName:\"H5\",canProcess:true});\n\t\tif(checkMsg){\n\t\t\tfailCall($T(\"A5bm::不能录音：\")+checkMsg);\n\t\t\treturn;\n\t\t};\n\t\t\n\t\t//尽量先创建好ctx，不然异步下创建可能不是running状态\n\t\tvar ctx;\n\t\tvar getCtx=function(){\n\t\t\tctx=set.runningContext;\n\t\t\tif(!ctx)ctx=newCtx=Recorder.GetContext(true); //2023-06 尽量创建新的ctx，免得Safari再次连接无回调\n\t\t};\n\t\t\n\t\t//***********已直接提供了音频流************\n\t\tif(set.sourceStream){\n\t\t\tThis._streamTag=\"set.sourceStream\";\n\t\t\tif(!Recorder.GetContext()){\n\t\t\t\tfailCall($T(\"1iU7::不支持此浏览器从流中获取录音\"));\n\t\t\t\treturn;\n\t\t\t};\n\t\t\tgetCtx();\n\t\t\t\n\t\t\tDisconnect(streamStore);//可能已open过，直接先尝试断开\n\t\t\tvar stream=This.Stream=set.sourceStream;\n\t\t\tstream._c=ctx;\n\t\t\tstream._RC=set.runningContext;\n\t\t\tstream._call={};\n\t\t\t\n\t\t\ttry{\n\t\t\t\tConnect(streamStore);\n\t\t\t}catch(e){\n\t\t\t\tDisconnect(streamStore);\n\t\t\t\tfailCall($T(\"BTW2::从流中打开录音失败：\")+e.message);\n\t\t\t\treturn;\n\t\t\t}\n\t\t\tok();\n\t\t\treturn;\n\t\t};\n\t\t\n\t\t\n\t\t//***********打开麦克风得到全局的音频流************\n\t\tvar codeFail=function(code,msg){\n\t\t\ttry{//跨域的优先检测一下\n\t\t\t\twindow.top.a;\n\t\t\t}catch(e){\n\t\t\t\tfailCall($T(\"Nclz::无权录音(跨域，请尝试给iframe添加麦克风访问策略，如{1})\",0,'allow=\"camera;microphone\"'));\n\t\t\t\treturn;\n\t\t\t};\n\t\t\t\n\t\t\tif(codeErr1(1,code)){\n\t\t\t\tif(/Found/i.test(code)){//可能是非安全环境导致的没有设备\n\t\t\t\t\tfailCall(msg+$T(\"jBa9::，无可用麦克风\"));\n\t\t\t\t}else{\n\t\t\t\t\tfailCall(msg);\n\t\t\t\t};\n\t\t\t};\n\t\t};\n\t\tvar codeErr1=function(call,code){ //排除几个明确原因的错误\n\t\t\tif(/Permission|Allow/i.test(code)){\n\t\t\t\tif(call) failCall($T(\"gyO5::用户拒绝了录音权限\"),true);\n\t\t\t}else if(window.isSecureContext===false){\n\t\t\t\tif(call) failCall($T(\"oWNo::浏览器禁止不安全页面录音，可开启https解决\"));\n\t\t\t}else{\n\t\t\t\treturn 1;\n\t\t\t};\n\t\t};\n\t\t\n\t\t\n\t\t//如果已打开并且有效就不要再打开了\n\t\tif(Recorder.IsOpen()){\n\t\t\tok();\n\t\t\treturn;\n\t\t};\n\t\tif(!Recorder.Support()){\n\t\t\tcodeFail(\"\",$T(\"COxc::此浏览器不支持录音\"));\n\t\t\treturn;\n\t\t};\n\t\tgetCtx();\n\t\t\n\t\t//请求权限，如果从未授权，一般浏览器会弹出权限请求弹框\n\t\tvar f1=function(stream){ //请求成功回调\n\t\t\t//https://github.com/xiangyuecn/Recorder/issues/14 获取到的track.readyState!=\"live\"，刚刚回调时可能是正常的，但过一下可能就被关掉了，原因不明。延迟一下保证真异步。对正常浏览器不影响\n\t\t\tsetTimeout(function(){\n\t\t\t\tstream._call={};\n\t\t\t\tvar oldStream=Recorder.Stream;\n\t\t\t\tif(oldStream){\n\t\t\t\t\tDisconnect(); //直接断开已存在的，旧的Connect未完成会自动终止\n\t\t\t\t\tstream._call=oldStream._call;\n\t\t\t\t};\n\t\t\t\tRecorder.Stream=stream;\n\t\t\t\tstream._c=ctx;\n\t\t\t\tstream._RC=set.runningContext;\n\t\t\t\tif(lockFail())return;\n\t\t\t\t\n\t\t\t\tif(Recorder.IsOpen()){\n\t\t\t\t\tif(oldStream)This.CLog($T(\"upb8::发现同时多次调用open\"),1);\n\t\t\t\t\t\n\t\t\t\t\tConnect(streamStore);\n\t\t\t\t\tok(); //只连接，因为AudioContext不一定在运行，无法知道是否有数据回调\n\t\t\t\t}else{\n\t\t\t\t\tfailCall($T(\"Q1GA::录音功能无效：无音频流\"));\n\t\t\t\t};\n\t\t\t},100);\n\t\t};\n\t\tvar f2=function(e){ //请求失败回调\n\t\t\tvar code=e.name||e.message||e.code+\":\"+e;\n\t\t\tvar tryMsg=\"\";\n\t\t\tif(callUmCount==1 && codeErr1(0,code)){ \n\t\t\t\ttryMsg=$T(\"KxE2::，将尝试禁用回声消除后重试\");\n\t\t\t}\n\t\t\tThis.CLog($T(\"xEQR::请求录音权限错误\")+tryMsg+\"|\"+e,tryMsg?3:1,e);\n\t\t\t\n\t\t\tif(tryMsg){//重试\n\t\t\t\tcallUserMedia(1);\n\t\t\t}else{\n\t\t\t\tcodeFail(code,$T(\"bDOG::无法录音：\")+e);\n\t\t\t};\n\t\t};\n\t\t\n\t\tvar callUmCount=0;\n\t\tvar callUserMedia=function(retry){\n\t\t\tcallUmCount++;\n\t\t\tvar atsTxt=\"audioTrackSet\";\n\t\t\tvar t_AGC=\"autoGainControl\",t_AEC=\"echoCancellation\",t_ANS=\"noiseSuppression\";\n\t\t\tvar atsTxtJs=atsTxt+\":{\"+t_AEC+\",\"+t_ANS+\",\"+t_AGC+\"}\";\n\t\t\tvar trackSet=JSON.parse(ToJson(set[atsTxt]||true)); //true 跟 {} 兼容性？\n\t\t\tThis.CLog(\"open... \"+callUmCount+\" \"+atsTxt+\":\"+ToJson(trackSet));\n\t\t\t\n\t\t\tif(retry){ //回声消除有些浏览器可能导致无法打开录音，尝试明确禁用来保证能最基础的录\n\t\t\t\tif(typeof(trackSet)!=\"object\")trackSet={}; //默认true\n\t\t\t\ttrackSet[t_AGC]=false;\n\t\t\t\ttrackSet[t_AEC]=false;\n\t\t\t\ttrackSet[t_ANS]=false;\n\t\t\t};\n\t\t\t//这里指明采样率，虽然可以解决手机上MediaRecorder采样率16k的问题，是回声消除导致的只能获取到16k的流，禁用回声消除可恢复48k。(issues#230)而且会导致乱用音频输入设备，MediaStreamTrack的applyConstraints也无法修改采样率\n\t\t\t//trackSet[sampleRateTxt]=ctx[sampleRateTxt];\n\t\t\tif(trackSet[sampleRateTxt]){\n\t\t\t\tThis.CLog($T(\"IjL3::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\",0,atsTxt+\".\"+sampleRateTxt),3);\n\t\t\t};\n\t\t\t\n\t\t\tvar mSet={audio:trackSet, video:set.videoTrackSet||false};\n\t\t\ttry{\n\t\t\t\tvar pro=Recorder.Scope[getUserMediaTxt](mSet,f1,f2);\n\t\t\t}catch(e){//不能设置trackSet就算了\n\t\t\t\tThis.CLog(getUserMediaTxt,3,e);\n\t\t\t\tmSet={audio:true, video:false};\n\t\t\t\tpro=Recorder.Scope[getUserMediaTxt](mSet,f1,f2);\n\t\t\t};\n\t\t\tThis.CLog(getUserMediaTxt+\"(\"+ToJson(mSet)+\") \"+CtxState(ctx)\n\t\t\t\t+$T(\"RiWe::，未配置 {1} 时浏览器可能会自动启用回声消除，移动端未禁用回声消除时可能会降低系统播放音量（关闭录音后可恢复）和仅提供16k采样率的音频流（不需要回声消除时可明确配置成禁用来获得48k高音质的流），请参阅文档中{2}配置\",0,atsTxtJs,atsTxt)\n\t\t\t\t+\"(\"+GitUrl+\") LM:\"+LM+\" UA:\"+navigator.userAgent);\n\t\t\tif(pro&&pro.then){\n\t\t\t\tpro.then(f1)[CatchTxt](f2); //fix 关键字，保证catch压缩时保持字符串形式\n\t\t\t};\n\t\t};\n\t\tcallUserMedia();\n\t}\n\t//关闭释放录音资源\n\t,close:function(call){\n\t\tcall=call||NOOP;\n\t\t\n\t\tvar This=this,streamStore=This._streamStore();\n\t\tThis._stop();\n\t\tvar sTag=\" stream:\"+This._streamTag;\n\t\t\n\t\tvar Lock=streamStore.Sync;\n\t\tThis._O=0;\n\t\tif(This._O_!=Lock.O){\n\t\t\t//唯一资源Stream的控制权已交给新对象，这里不能关闭。此处在每次都弹权限的浏览器内可能存在泄漏，新对象被拒绝权限可能不会调用close，忽略这种不处理\n\t\t\tThis.CLog($T(\"hWVz::close被忽略（因为同时open了多个rec，只有最后一个会真正close）\")+sTag,3);\n\t\t\tcall();\n\t\t\treturn;\n\t\t};\n\t\tLock.C++;//获得控制权\n\t\t\n\t\tDisconnect(streamStore);\n\t\t\n\t\tThis.CLog(\"close,\"+sTag);\n\t\tcall();\n\t}\n\t\n\t\n\t\n\t\n\t\n\t/*模拟一段录音数据，后面可以调用stop进行编码，需提供pcm数据[1,2,3...]，pcm的采样率*/\n\t,mock:function(pcmData,pcmSampleRate){\n\t\tvar This=this;\n\t\tThis._stop();//清理掉已有的资源\n\t\t\n\t\tThis.isMock=1;\n\t\tThis.mockEnvInfo=null;\n\t\tThis.buffers=[pcmData];\n\t\tThis.recSize=pcmData.length;\n\t\tThis._setSrcSR(pcmSampleRate);\n\t\tThis._streamTag=\"mock\";\n\t\treturn This;\n\t}\n\t,_setSrcSR:function(sampleRate){\n\t\tvar This=this,set=This.set;\n\t\tvar setSr=set[sampleRateTxt];\n\t\tif(setSr>sampleRate){\n\t\t\tset[sampleRateTxt]=sampleRate;\n\t\t}else{ setSr=0 }\n\t\tThis[srcSampleRateTxt]=sampleRate;\n\t\tThis.CLog(srcSampleRateTxt+\": \"+sampleRate+\" set.\"+sampleRateTxt+\": \"+set[sampleRateTxt]+(setSr?\" \"+$T(\"UHvm::忽略\")+\": \"+setSr:\"\"), setSr?3:0);\n\t}\n\t,envCheck:function(envInfo){//平台环境下的可用性检查，任何时候都可以调用检查，返回errMsg:\"\"正常，\"失败原因\"\n\t\t//envInfo={envName:\"H5\",canProcess:true}\n\t\tvar errMsg,This=this,set=This.set;\n\t\t\n\t\t//检测CPU的数字字节序，TypedArray字节序是个迷，直接拒绝罕见的大端模式，因为找不到这种CPU进行测试\n\t\tvar tag=\"CPU_BE\";\n\t\tif(!errMsg && !Recorder[tag] && typeof Int8Array==\"function\" && !new Int8Array(new Int32Array([1]).buffer)[0]){\n\t\t\tTraffic(tag); //如果开启了流量统计，这里将发送一个图片请求\n\t\t\terrMsg=$T(\"Essp::不支持{1}架构\",0,tag);\n\t\t};\n\t\t\n\t\t//编码器检查环境下配置是否可用\n\t\tif(!errMsg){\n\t\t\tvar type=set.type,hasFn=This[type+\"_envCheck\"];\n\t\t\tif(set.takeoffEncodeChunk){//需要实时编码返回数据，此时需要检查环境是否有实时特性、和是否可实时编码\n\t\t\t\tif(!hasFn){\n\t\t\t\t\terrMsg=$T(\"2XBl::{1}类型不支持设置takeoffEncodeChunk\",0,type)+(This[type]?\"\":$T(\"LG7e::(未加载编码器)\"));\n\t\t\t\t}else if(!envInfo.canProcess){\n\t\t\t\t\terrMsg=$T(\"7uMV::{1}环境不支持实时处理\",0,envInfo.envName);\n\t\t\t\t};\n\t\t\t};\n\t\t\t\n\t\t\tif(!errMsg && hasFn){//编码器已实现环境检查\n\t\t\t\terrMsg=This[type+\"_envCheck\"](envInfo,set);\n\t\t\t};\n\t\t};\n\t\t\n\t\treturn errMsg||\"\";\n\t}\n\t,envStart:function(mockEnvInfo,sampleRate){//平台环境相关的start调用\n\t\tvar This=this,set=This.set;\n\t\tThis.isMock=mockEnvInfo?1:0;//非H5环境需要启用mock，并提供envCheck需要的环境信息\n\t\tThis.mockEnvInfo=mockEnvInfo;\n\t\tThis.buffers=[];//数据缓冲\n\t\tThis.recSize=0;//数据大小\n\t\tif(mockEnvInfo){\n\t\t\tThis._streamTag=\"env$\"+mockEnvInfo.envName;\n\t\t};\n\t\t\n\t\tThis.state=1;//运行状态，0未录音 1录音中 2暂停 3等待ctx激活\n\t\tThis.envInLast=0;//envIn接收到最后录音内容的时间\n\t\tThis.envInFirst=0;//envIn接收到的首个录音内容的录制时间\n\t\tThis.envInFix=0;//补偿的总时间\n\t\tThis.envInFixTs=[];//补偿计数列表\n\t\t\n\t\t//engineCtx需要提前确定最终的采样率\n\t\tThis._setSrcSR(sampleRate);\n\t\t\n\t\tThis.engineCtx=0;\n\t\t//此类型有边录边转码(Worker)支持\n\t\tif(This[set.type+\"_start\"]){\n\t\t\tvar engineCtx=This.engineCtx=This[set.type+\"_start\"](set);\n\t\t\tif(engineCtx){\n\t\t\t\tengineCtx.pcmDatas=[];\n\t\t\t\tengineCtx.pcmSize=0;\n\t\t\t};\n\t\t};\n\t}\n\t,envResume:function(){//和平台环境无关的恢复录音\n\t\t//重新开始计数\n\t\tthis.envInFixTs=[];\n\t}\n\t,envIn:function(pcm,sum){//和平台环境无关的pcm[Int16]输入\n\t\tvar This=this,set=This.set,engineCtx=This.engineCtx;\n\t\tif(This.state!=1){\n\t\t\tif(!This.state)This.CLog(\"envIn at state=0\",3);\n\t\t\treturn;\n\t\t};\n\t\tvar bufferSampleRate=This[srcSampleRateTxt];\n\t\tvar size=pcm.length;\n\t\tvar powerLevel=Recorder.PowerLevel(sum,size);\n\t\t\n\t\tvar buffers=This.buffers;\n\t\tvar bufferFirstIdx=buffers.length;//之前的buffer都是经过onProcess处理好的，不允许再修改\n\t\tbuffers.push(pcm);\n\t\t\n\t\t//有engineCtx时会被覆盖，这里保存一份\n\t\tvar buffersThis=buffers;\n\t\tvar bufferFirstIdxThis=bufferFirstIdx;\n\t\t\n\t\t//卡顿丢失补偿：因为设备很卡的时候导致H5接收到的数据量不够造成播放时候变速，结果比实际的时长要短，此处保证了不会变短，但不能修复丢失的音频数据造成音质变差。当前算法采用输入时间侦测下一帧是否需要添加补偿帧，需要(6次输入||超过1秒)以上才会开始侦测，如果滑动窗口内丢失超过1/3就会进行补偿\n\t\tvar now=Date.now();\n\t\tvar pcmTime=Math.round(size/bufferSampleRate*1000);\n\t\tThis.envInLast=now;\n\t\tif(This.buffers.length==1){//记下首个录音数据的录制时间\n\t\t\tThis.envInFirst=now-pcmTime;\n\t\t};\n\t\tvar envInFixTs=This.envInFixTs;\n\t\tenvInFixTs.splice(0,0,{t:now,d:pcmTime});\n\t\t//保留3秒的计数滑动窗口，另外超过3秒的停顿不补偿\n\t\tvar tsInStart=now,tsPcm=0;\n\t\tfor(var i=0;i<envInFixTs.length;i++){\n\t\t\tvar o=envInFixTs[i];\n\t\t\tif(now-o.t>3000){\n\t\t\t\tenvInFixTs.length=i;\n\t\t\t\tbreak;\n\t\t\t};\n\t\t\ttsInStart=o.t;\n\t\t\ttsPcm+=o.d;\n\t\t};\n\t\t//达到需要的数据量，开始侦测是否需要补偿\n\t\tvar tsInPrev=envInFixTs[1];\n\t\tvar tsIn=now-tsInStart;\n\t\tvar lost=tsIn-tsPcm;\n\t\tif( lost>tsIn/3 && (tsInPrev&&tsIn>1000 || envInFixTs.length>=6) ){\n\t\t\t//丢失过多，开始执行补偿\n\t\t\tvar addTime=now-tsInPrev.t-pcmTime;//距离上次输入丢失这么多ms\n\t\t\tif(addTime>pcmTime/5){//丢失超过本帧的1/5\n\t\t\t\tvar fixOpen=!set.disableEnvInFix;\n\t\t\t\tThis.CLog(\"[\"+now+\"]\"+i18n.get(fixOpen?$T(\"4Kfd::补偿{1}ms\",1):$T(\"bM5i::未补偿{1}ms\",1),[addTime]),3);\n\t\t\t\tThis.envInFix+=addTime;\n\t\t\t\t\n\t\t\t\t//用静默进行补偿\n\t\t\t\tif(fixOpen){\n\t\t\t\t\tvar addPcm=new Int16Array(addTime*bufferSampleRate/1000);\n\t\t\t\t\tsize+=addPcm.length;\n\t\t\t\t\tbuffers.push(addPcm);\n\t\t\t\t};\n\t\t\t};\n\t\t};\n\t\t\n\t\t\n\t\tvar sizeOld=This.recSize,addSize=size;\n\t\tvar bufferSize=sizeOld+addSize;\n\t\tThis.recSize=bufferSize;//此值在onProcess后需要修正，可能新数据被修改\n\t\t\n\t\t\n\t\t//此类型有边录边转码(Worker)支持，开启实时转码\n\t\tif(engineCtx){\n\t\t\t//转换成set的采样率\n\t\t\tvar chunkInfo=Recorder.SampleData(buffers,bufferSampleRate,set[sampleRateTxt],engineCtx.chunkInfo);\n\t\t\tengineCtx.chunkInfo=chunkInfo;\n\t\t\t\n\t\t\tsizeOld=engineCtx.pcmSize;\n\t\t\taddSize=chunkInfo.data.length;\n\t\t\tbufferSize=sizeOld+addSize;\n\t\t\tengineCtx.pcmSize=bufferSize;//此值在onProcess后需要修正，可能新数据被修改\n\t\t\t\n\t\t\tbuffers=engineCtx.pcmDatas;\n\t\t\tbufferFirstIdx=buffers.length;\n\t\t\tbuffers.push(chunkInfo.data);\n\t\t\tbufferSampleRate=chunkInfo[sampleRateTxt];\n\t\t};\n\t\t\n\t\tvar duration=Math.round(bufferSize/bufferSampleRate*1000);\n\t\tvar bufferNextIdx=buffers.length;\n\t\tvar bufferNextIdxThis=buffersThis.length;\n\t\t\n\t\t//允许异步处理buffer数据\n\t\tvar asyncEnd=function(){\n\t\t\t//重新计算size，异步的早已减去添加的，同步的需去掉本次添加的然后重新计算\n\t\t\tvar num=asyncBegin?0:-addSize;\n\t\t\tvar hasClear=buffers[0]==null;\n\t\t\tfor(var i=bufferFirstIdx;i<bufferNextIdx;i++){\n\t\t\t\tvar buffer=buffers[i];\n\t\t\t\tif(buffer==null){//已被主动释放内存，比如长时间实时传输录音时\n\t\t\t\t\thasClear=1;\n\t\t\t\t}else{\n\t\t\t\t\tnum+=buffer.length;\n\t\t\t\t\t\n\t\t\t\t\t//推入后台边录边转码\n\t\t\t\t\tif(engineCtx&&buffer.length){\n\t\t\t\t\t\tThis[set.type+\"_encode\"](engineCtx,buffer);\n\t\t\t\t\t};\n\t\t\t\t};\n\t\t\t};\n\t\t\t\n\t\t\t//同步清理This.buffers，不管buffers到底清了多少个，buffersThis是使用不到的进行全清\n\t\t\tif(hasClear && engineCtx){\n\t\t\t\tvar i=bufferFirstIdxThis;\n\t\t\t\tif(buffersThis[0]){\n\t\t\t\t\ti=0;\n\t\t\t\t};\n\t\t\t\tfor(;i<bufferNextIdxThis;i++){\n\t\t\t\t\tbuffersThis[i]=null;\n\t\t\t\t};\n\t\t\t};\n\t\t\t\n\t\t\t//统计修改后的size，如果异步发生clear要原样加回来，同步的无需操作\n\t\t\tif(hasClear){\n\t\t\t\tnum=asyncBegin?addSize:0;\n\t\t\t\t\n\t\t\t\tbuffers[0]=null;//彻底被清理\n\t\t\t};\n\t\t\tif(engineCtx){\n\t\t\t\tengineCtx.pcmSize+=num;\n\t\t\t}else{\n\t\t\t\tThis.recSize+=num;\n\t\t\t};\n\t\t};\n\t\t//实时回调处理数据，允许修改或替换上次回调以来新增的数据 ，但是不允许修改已处理过的，不允许增删第一维数组 ，允许将第二维数组任意修改替换成空数组也可以\n\t\tvar asyncBegin=0,procTxt=\"rec.set.onProcess\";\n\t\ttry{\n\t\t\tasyncBegin=set.onProcess(buffers,powerLevel,duration,bufferSampleRate,bufferFirstIdx,asyncEnd);\n\t\t}catch(e){\n\t\t\t//此错误显示不要用CLog，这样控制台内相同内容不会重复打印\n\t\t\tconsole.error(procTxt+$T(\"gFUF::回调出错是不允许的，需保证不会抛异常\"),e);\n\t\t};\n\t\t\n\t\tvar slowT=Date.now()-now;\n\t\tif(slowT>10 && This.envInFirst-now>1000){ //1秒后开始onProcess性能监测\n\t\t\tThis.CLog(procTxt+$T(\"2ghS::低性能，耗时{1}ms\",0,slowT),3);\n\t\t};\n\t\t\n\t\tif(asyncBegin===true){\n\t\t\t//开启了异步模式，onProcess已接管buffers新数据，立即清空，避免出现未处理的数据\n\t\t\tvar hasClear=0;\n\t\t\tfor(var i=bufferFirstIdx;i<bufferNextIdx;i++){\n\t\t\t\tif(buffers[i]==null){//已被主动释放内存，比如长时间实时传输录音时 ，但又要开启异步模式，此种情况是非法的\n\t\t\t\t\thasClear=1;\n\t\t\t\t}else{\n\t\t\t\t\tbuffers[i]=new Int16Array(0);\n\t\t\t\t};\n\t\t\t};\n\t\t\t\n\t\t\tif(hasClear){\n\t\t\t\tThis.CLog($T(\"ufqH::未进入异步前不能清除buffers\"),3);\n\t\t\t}else{\n\t\t\t\t//还原size，异步结束后再统计仅修改后的size，如果发生clear要原样加回来\n\t\t\t\tif(engineCtx){\n\t\t\t\t\tengineCtx.pcmSize-=addSize;\n\t\t\t\t}else{\n\t\t\t\t\tThis.recSize-=addSize;\n\t\t\t\t};\n\t\t\t};\n\t\t}else{\n\t\t\tasyncEnd();\n\t\t};\n\t}\n\t\n\t\n\t\n\t\n\t//开始录音，需先调用open；只要open成功时，调用此方法是安全的，如果未open强行调用导致的内部错误将不会有任何提示，stop时自然能得到错误；注意：open和start至少有一个应当在用户操作（触摸、点击等）下进行调用，原因参考runningContext配置\n\t,start:function(){\n\t\tvar This=this;\n\t\t\n\t\tvar isOpen=1;\n\t\tif(This.set.sourceStream){//直接提供了流，仅判断是否调用了open\n\t\t\tif(!This.Stream){\n\t\t\t\tisOpen=0;\n\t\t\t}\n\t\t}else if(!Recorder.IsOpen()){//监测全局麦克风是否打开并且有效\n\t\t\tisOpen=0;\n\t\t};\n\t\tif(!isOpen){\n\t\t\tThis.CLog($T(\"6WmN::start失败：未open\"),1);\n\t\t\treturn;\n\t\t};\n\t\tvar ctx=This._streamCtx();\n\t\tThis.CLog($T(\"kLDN::start 开始录音，\")+CtxState(ctx)+\" stream:\"+This._streamTag);\n\t\t\n\t\tThis._stop();\n\t\tThis.envStart(null, ctx[sampleRateTxt]);\n\t\tThis.state=3;//0未录音 1录音中 2暂停 3等待ctx激活\n\t\t\n\t\t//检查open过程中stop是否已经调用过\n\t\tif(This._SO&&This._SO+1!=This._S){//上面调用过一次 _stop\n\t\t\t//open未完成就调用了stop，此种情况终止start。也应尽量避免出现此情况\n\t\t\tThis.CLog($T(\"Bp2y::start被中断\"),3);\n\t\t\treturn;\n\t\t};\n\t\tThis._SO=0;\n\t\t\n\t\tvar end=function(){\n\t\t\tif(This.state==3){\n\t\t\t\tThis.state=1;\n\t\t\t\tThis.resume();\n\t\t\t}\n\t\t};\n\t\tvar tag=\"AudioContext resume: \";\n\t\t\n\t\t//如果有数据回调，就不等待ctx resume\n\t\tvar stream=This._streamGet();\n\t\tstream._call[This.id]=function(){\n\t\t\tThis.CLog(tag+ctx.state+'|stream ok');\n\t\t\tend();\n\t\t};\n\t\tResumeCtx(ctx,function(runC){\n\t\t\trunC&&This.CLog(tag+\"wait...\");\n\t\t\treturn This.state==3;\n\t\t},function(runC){\n\t\t\trunC&&This.CLog(tag+ctx.state);\n\t\t\tend();\n\t\t},function(err){ //比较少见，可能对录音没有影响\n\t\t\tThis.CLog(tag+ctx.state+$T(\"upkE::，可能无法录音：\")+err,1);\n\t\t\tend();\n\t\t});\n\t}\n\t\n\t\n\t\n\t/*暂停录音*/\n\t,pause:function(){\n\t\tvar This=this,stream=This._streamGet();\n\t\tif(This.state){\n\t\t\tThis.state=2;\n\t\t\tThis.CLog(\"pause\");\n\t\t\tif(stream) delete stream._call[This.id];\n\t\t};\n\t}\n\t/*恢复录音*/\n\t,resume:function(){\n\t\tvar This=this,stream=This._streamGet();\n\t\tvar tag=\"resume\",tag3=tag+\"(wait ctx)\";\n\t\tif(This.state==3){ //start还在等ctx恢复\n\t\t\tThis.CLog(tag3);\n\t\t}else if(This.state){\n\t\t\tThis.state=1;\n\t\t\tThis.CLog(tag);\n\t\t\tThis.envResume();\n\t\t\t\n\t\t\tif(stream){\n\t\t\t\tstream._call[This.id]=function(pcm,sum){\n\t\t\t\t\tif(This.state==1){\n\t\t\t\t\t\tThis.envIn(pcm,sum);\n\t\t\t\t\t};\n\t\t\t\t};\n\t\t\t\tConnAlive(stream);//AudioWorklet只会在ctx激活后运行\n\t\t\t};\n\t\t\t\n\t\t\tvar ctx=This._streamCtx();\n\t\t\tif(ctx){ //AudioContext如果被暂停，尽量恢复\n\t\t\t\tResumeCtx(ctx,function(runC){\n\t\t\t\t\trunC&&This.CLog(tag3+\"...\");\n\t\t\t\t\treturn This.state==1;\n\t\t\t\t},function(runC){\n\t\t\t\t\trunC&&This.CLog(tag3+ctx.state);\n\t\t\t\t\tConnAlive(stream);\n\t\t\t\t},function(err){\n\t\t\t\t\tThis.CLog(tag3+ctx.state+\"[err]\"+err,1);\n\t\t\t\t});\n\t\t\t};\n\t\t};\n\t}\n\t\n\t\n\t\n\t\n\t,_stop:function(keepEngine){\n\t\tvar This=this,set=This.set;\n\t\tif(!This.isMock){\n\t\t\tThis._S++;\n\t\t};\n\t\tif(This.state){\n\t\t\tThis.pause();\n\t\t\tThis.state=0;\n\t\t};\n\t\tif(!keepEngine && This[set.type+\"_stop\"]){\n\t\t\tThis[set.type+\"_stop\"](This.engineCtx);\n\t\t\tThis.engineCtx=0;\n\t\t};\n\t}\n\t/*\n\t结束录音并返回录音数据blob对象\n\t\tTrue(blob,duration,mime)\n\t\t\tblob：录音数据audio/mp3|wav格式；默认是Blob对象，可设置rec.dataType=\"arraybuffer\"改成ArrayBuffer\n\t\t\tduration：录音时长，单位毫秒\n\t\t\tmime：\"auido/mp3\" blob数据的类型，方便ArrayBuffer时区分类型\n\t\tFalse(msg)\n\t\tautoClose:false 可选，是否自动调用close，默认为false\n\t*/\n\t,stop:function(True,False,autoClose){\n\t\tvar This=this,set=This.set,t1;\n\t\tvar envInMS=This.envInLast-This.envInFirst, envInLen=envInMS&&This.buffers.length; //可能未start\n\t\tThis.CLog($T(\"Xq4s::stop 和start时差:\")\n\t\t\t+(envInMS?envInMS+\"ms \"+$T(\"3CQP::补偿:\")+This.envInFix+\"ms\"\n\t\t\t\t+\" envIn:\"+envInLen+\" fps:\"+(envInLen/envInMS*1000).toFixed(1)\n\t\t\t:\"-\")+\" stream:\"+This._streamTag+\" (\"+GitUrl+\") LM:\"+LM);\n\t\t\n\t\tvar end=function(){\n\t\t\tThis._stop();//彻底关掉engineCtx\n\t\t\tif(autoClose){\n\t\t\t\tThis.close();\n\t\t\t};\n\t\t};\n\t\tvar err=function(msg){\n\t\t\tThis.CLog($T(\"u8JG::结束录音失败：\")+msg,1);\n\t\t\tFalse&&False(msg);\n\t\t\tend();\n\t\t};\n\t\tvar ok=function(blob,mime,duration){\n\t\t\tvar tBlob=\"blob\",tABuf=\"arraybuffer\",tDT=\"dataType\",tDDT=\"DefaultDataType\";\n\t\t\tvar dType=This[tDT]||Recorder[tDDT]||tBlob,dTag=tDT+\"=\"+dType;\n\t\t\tvar isAB=blob instanceof ArrayBuffer,dErr=0;\n\t\t\tvar dLen=isAB?blob.byteLength:blob.size;\n\t\t\tif(dType==tABuf){\n\t\t\t\tif(!isAB) dErr=1;\n\t\t\t}else if(dType==tBlob){\n\t\t\t\tif(typeof Blob!=\"function\"){\n\t\t\t\t\tdErr=$T.G(\"NonBrowser-1\",[dTag])+$T(\"1skY::，请设置{1}\",0,RecTxt+\".\"+tDDT+'=\"'+tABuf+'\"');\n\t\t\t\t}else{\n\t\t\t\t\tif(isAB) blob=new Blob([blob],{type:mime});\n\t\t\t\t\tif(!(blob instanceof Blob)) dErr=1;\n\t\t\t\t\tmime=blob.type||mime;\n\t\t\t\t}\n\t\t\t}else{\n\t\t\t\tdErr=$T.G(\"NotSupport-1\",[dTag]);\n\t\t\t};\n\t\t\t\n\t\t\tThis.CLog($T(\"Wv7l::结束录音 编码花{1}ms 音频时长{2}ms 文件大小{3}b\",0,Date.now()-t1,duration,dLen)+\" \"+dTag+\",\"+mime);\n\t\t\tif(dErr){\n\t\t\t\terr(dErr!=1?dErr:$T(\"Vkbd::{1}编码器返回的不是{2}\",0,set.type,dType)+\", \"+dTag);\n\t\t\t\treturn;\n\t\t\t};\n\t\t\tif(set.takeoffEncodeChunk){//接管了输出，此时blob长度为0\n\t\t\t\tThis.CLog($T(\"QWnr::启用takeoffEncodeChunk后stop返回的blob长度为0不提供音频数据\"),3);\n\t\t\t}else if(dLen<Math.max(50,duration/5)){//1秒小于0.2k？\n\t\t\t\terr($T(\"Sz2H::生成的{1}无效\",0,set.type));\n\t\t\t\treturn;\n\t\t\t};\n\t\t\t\n\t\t\tTrue&&True(blob,duration,mime);\n\t\t\tend();\n\t\t};\n\t\tif(!This.isMock){\n\t\t\tvar isCtxWait=This.state==3;\n\t\t\tif(!This.state || isCtxWait){\n\t\t\t\terr($T(\"wf9t::未开始录音\")+(isCtxWait?$T(\"Dl2c::，开始录音前无用户交互导致AudioContext未运行\"):\"\"));\n\t\t\t\treturn;\n\t\t\t};\n\t\t};\n\t\tThis._stop(true); //将录音状态改成未录音\n\t\tvar size=This.recSize;\n\t\tif(!size){\n\t\t\terr($T(\"Ltz3::未采集到录音\"));\n\t\t\treturn;\n\t\t};\n\t\tif(!This[set.type]){\n\t\t\terr($T(\"xGuI::未加载{1}编码器，请尝试到{2}的src/engine内找到{1}的编码器并加载\",0,set.type,RecTxt));\n\t\t\treturn;\n\t\t};\n\t\t\n\t\t//环境配置检查，此处仅针对mock调用，因为open已经检查过了\n\t\tif(This.isMock){\n\t\t\tvar checkMsg=This.envCheck(This.mockEnvInfo||{envName:\"mock\",canProcess:false});//没有提供环境信息的mock时没有onProcess回调\n\t\t\tif(checkMsg){\n\t\t\t\terr($T(\"AxOH::录音错误：\")+checkMsg);\n\t\t\t\treturn;\n\t\t\t};\n\t\t};\n\t\t\n\t\t//此类型有边录边转码(Worker)支持\n\t\tvar engineCtx=This.engineCtx;\n\t\tif(This[set.type+\"_complete\"]&&engineCtx){\n\t\t\tvar duration=Math.round(engineCtx.pcmSize/set[sampleRateTxt]*1000);//采用后的数据长度和buffers的长度可能微小的不一致，是采样率连续转换的精度问题\n\t\t\t\n\t\t\tt1=Date.now();\n\t\t\tThis[set.type+\"_complete\"](engineCtx,function(blob,mime){\n\t\t\t\tok(blob,mime,duration);\n\t\t\t},err);\n\t\t\treturn;\n\t\t};\n\t\t\n\t\t//标准UI线程转码，调整采样率\n\t\tt1=Date.now();\n\t\tif(!This.buffers[0]){\n\t\t\terr($T(\"xkKd::音频buffers被释放\"));\n\t\t\treturn;\n\t\t};\n\t\tvar chunk=Recorder.SampleData(This.buffers,This[srcSampleRateTxt],set[sampleRateTxt]);\n\t\t\n\t\tset[sampleRateTxt]=chunk[sampleRateTxt];\n\t\tvar res=chunk.data;\n\t\tvar duration=Math.round(res.length/set[sampleRateTxt]*1000);\n\t\t\n\t\tThis.CLog($T(\"CxeT::采样:{1} 花:{2}ms\",0,size+\"->\"+res.length,Date.now()-t1));\n\t\t\n\t\tsetTimeout(function(){\n\t\t\tt1=Date.now();\n\t\t\tThis[set.type](res,function(blob,mime){\n\t\t\t\tok(blob,mime,duration);\n\t\t\t},function(msg){\n\t\t\t\terr(msg);\n\t\t\t});\n\t\t});\n\t}\n\n};\n\n\n\n\n\n//=======从WebM字节流中提取pcm数据，提取成功返回Float32Array，失败返回null||-1=====\nvar WebM_Extract=function(inBytes, scope){\n\tif(!scope.pos){\n\t\tscope.pos=[0]; scope.tracks={}; scope.bytes=[];\n\t};\n\tvar tracks=scope.tracks, position=[scope.pos[0]];\n\tvar endPos=function(){ scope.pos[0]=position[0] };\n\t\n\tvar sBL=scope.bytes.length;\n\tvar bytes=new Uint8Array(sBL+inBytes.length);\n\tbytes.set(scope.bytes); bytes.set(inBytes,sBL);\n\tscope.bytes=bytes;\n\t\n\t//先读取文件头和Track信息\n\tif(!scope._ht){\n\t\treadMatroskaVInt(bytes, position);//EBML Header\n\t\treadMatroskaBlock(bytes, position);//跳过EBML Header内容\n\t\tif(!BytesEq(readMatroskaVInt(bytes, position), [0x18,0x53,0x80,0x67])){\n\t\t\treturn;//未识别到Segment\n\t\t}\n\t\treadMatroskaVInt(bytes, position);//跳过Segment长度值\n\t\twhile(position[0]<bytes.length){\n\t\t\tvar eid0=readMatroskaVInt(bytes, position);\n\t\t\tvar bytes0=readMatroskaBlock(bytes, position);\n\t\t\tvar pos0=[0],audioIdx=0;\n\t\t\tif(!bytes0)return;//数据不全，等待缓冲\n\t\t\t//Track完整数据，循环读取TrackEntry\n\t\t\tif(BytesEq(eid0, [0x16,0x54,0xAE,0x6B])){\n\t\t\t\twhile(pos0[0]<bytes0.length){\n\t\t\t\t\tvar eid1=readMatroskaVInt(bytes0, pos0);\n\t\t\t\t\tvar bytes1=readMatroskaBlock(bytes0, pos0);\n\t\t\t\t\tvar pos1=[0],track={channels:0,sampleRate:0};\n\t\t\t\t\tif(BytesEq(eid1, [0xAE])){//TrackEntry\n\t\t\t\t\t\twhile(pos1[0]<bytes1.length){\n\t\t\t\t\t\t\tvar eid2=readMatroskaVInt(bytes1, pos1);\n\t\t\t\t\t\t\tvar bytes2=readMatroskaBlock(bytes1, pos1);\n\t\t\t\t\t\t\tvar pos2=[0];\n\t\t\t\t\t\t\tif(BytesEq(eid2, [0xD7])){//Track Number\n\t\t\t\t\t\t\t\tvar val=BytesInt(bytes2);\n\t\t\t\t\t\t\t\ttrack.number=val;\n\t\t\t\t\t\t\t\ttracks[val]=track;\n\t\t\t\t\t\t\t}else if(BytesEq(eid2, [0x83])){//Track Type\n\t\t\t\t\t\t\t\tvar val=BytesInt(bytes2);\n\t\t\t\t\t\t\t\tif(val==1) track.type=\"video\";\n\t\t\t\t\t\t\t\telse if(val==2) {\n\t\t\t\t\t\t\t\t\ttrack.type=\"audio\";\n\t\t\t\t\t\t\t\t\tif(!audioIdx) scope.track0=track;\n\t\t\t\t\t\t\t\t\ttrack.idx=audioIdx++;\n\t\t\t\t\t\t\t\t}else track.type=\"Type-\"+val;\n\t\t\t\t\t\t\t}else if(BytesEq(eid2, [0x86])){//Track Codec\n\t\t\t\t\t\t\t\tvar str=\"\";\n\t\t\t\t\t\t\t\tfor(var i=0;i<bytes2.length;i++){\n\t\t\t\t\t\t\t\t\tstr+=String.fromCharCode(bytes2[i]);\n\t\t\t\t\t\t\t\t}\n\t\t\t\t\t\t\t\ttrack.codec=str;\n\t\t\t\t\t\t\t}else if(BytesEq(eid2, [0xE1])){\n\t\t\t\t\t\t\t\twhile(pos2[0]<bytes2.length){//循环读取 Audio 属性\n\t\t\t\t\t\t\t\t\tvar eid3=readMatroskaVInt(bytes2, pos2);\n\t\t\t\t\t\t\t\t\tvar bytes3=readMatroskaBlock(bytes2, pos2);\n\t\t\t\t\t\t\t\t\t//采样率、位数、声道数\n\t\t\t\t\t\t\t\t\tif(BytesEq(eid3, [0xB5])){\n\t\t\t\t\t\t\t\t\t\tvar val=0,arr=new Uint8Array(bytes3.reverse()).buffer;\n\t\t\t\t\t\t\t\t\t\tif(bytes3.length==4) val=new Float32Array(arr)[0];\n\t\t\t\t\t\t\t\t\t\telse if(bytes3.length==8) val=new Float64Array(arr)[0];\n\t\t\t\t\t\t\t\t\t\telse CLog(\"WebM Track !Float\",1,bytes3);\n\t\t\t\t\t\t\t\t\t\ttrack[sampleRateTxt]=Math.round(val);\n\t\t\t\t\t\t\t\t\t}else if(BytesEq(eid3, [0x62,0x64])) track.bitDepth=BytesInt(bytes3);\n\t\t\t\t\t\t\t\t\telse if(BytesEq(eid3, [0x9F])) track.channels=BytesInt(bytes3);\n\t\t\t\t\t\t\t\t}\n\t\t\t\t\t\t\t}\n\t\t\t\t\t\t}\n\t\t\t\t\t}\n\t\t\t\t};\n\t\t\t\tscope._ht=1;\n\t\t\t\tCLog(\"WebM Tracks\",tracks);\n\t\t\t\tendPos();\n\t\t\t\tbreak;\n\t\t\t}\n\t\t}\n\t}\n\t\n\t//校验音频参数信息，如果不符合代码要求，统统拒绝处理\n\tvar track0=scope.track0;\n\tif(!track0)return;\n\tvar trackSR=track0[sampleRateTxt]; scope.webmSR=trackSR;\n\tif(track0.bitDepth==16 && /FLOAT/i.test(track0.codec)){\n\t\ttrack0.bitDepth=32; //chrome v66 实际为浮点数\n\t\tCLog(\"WebM 16->32 bit\",3);\n\t}\n\tif(trackSR<8000 || track0.bitDepth!=32 || track0.channels<1 || !/(\\b|_)PCM\\b/i.test(track0.codec)){\n\t\tscope.bytes=[];//格式非预期 无法处理，清空缓冲数据\n\t\tif(!scope.bad)CLog(\"WebM Track Unexpected\",3,scope);\n\t\tscope.bad=1;\n\t\treturn -1;\n\t}\n\t\n\t//循环读取Cluster内的SimpleBlock\n\tvar datas=[],dataLen=0;\n\twhile(position[0]<bytes.length){\n\t\tvar eid1=readMatroskaVInt(bytes, position);\n\t\tvar bytes1=readMatroskaBlock(bytes, position);\n\t\tif(!bytes1)break;//数据不全，等待缓冲\n\t\tif(BytesEq(eid1, [0xA3])){//SimpleBlock完整数据\n\t\t\tvar trackNo=bytes1[0]&0xf;\n\t\t\tvar track=tracks[trackNo];\n\t\t\tif(!track){//不可能没有，数据出错？\n\t\t\t\tCLog(\"WebM !Track\"+trackNo,1,tracks);\n\t\t\t}else if(track.idx===0){\n\t\t\t\tvar u8arr=new Uint8Array(bytes1.length-4);\n\t\t\t\tfor(var i=4;i<bytes1.length;i++){\n\t\t\t\t\tu8arr[i-4]=bytes1[i];\n\t\t\t\t}\n\t\t\t\tdatas.push(u8arr); dataLen+=u8arr.length;\n\t\t\t}\n\t\t}\n\t\tendPos();\n\t}\n\t\n\tif(dataLen){\n\t\tvar more=new Uint8Array(bytes.length-scope.pos[0]);\n\t\tmore.set(bytes.subarray(scope.pos[0]));\n\t\tscope.bytes=more; //清理已读取了的缓冲数据\n\t\tscope.pos[0]=0;\n\t\t\n\t\tvar u8arr=new Uint8Array(dataLen); //已获取的音频数据\n\t\tfor(var i=0,i2=0;i<datas.length;i++){\n\t\t\tu8arr.set(datas[i],i2);\n\t\t\ti2+=datas[i].length;\n\t\t}\n\t\tvar arr=new Float32Array(u8arr.buffer);\n\t\t\n\t\tif(track0.channels>1){//多声道，提取一个声道\n\t\t\tvar arr2=[];\n\t\t\tfor(var i=0;i<arr.length;){\n\t\t\t\tarr2.push(arr[i]);\n\t\t\t\ti+=track0.channels;\n\t\t\t}\n\t\t\tarr=new Float32Array(arr2);\n\t\t};\n\t\treturn arr;\n\t}\n};\n//两个字节数组内容是否相同\nvar BytesEq=function(bytes1,bytes2){\n\tif(!bytes1 || bytes1.length!=bytes2.length) return false;\n\tif(bytes1.length==1) return bytes1[0]==bytes2[0];\n\tfor(var i=0;i<bytes1.length;i++){\n\t\tif(bytes1[i]!=bytes2[i]) return false;\n\t}\n\treturn true;\n};\n//字节数组BE转成int数字\nvar BytesInt=function(bytes){\n\tvar s=\"\";//0-8字节，js位运算只支持4字节\n\tfor(var i=0;i<bytes.length;i++){var n=bytes[i];s+=(n<16?\"0\":\"\")+n.toString(16)};\n\treturn parseInt(s,16)||0;\n};\n//读取一个可变长数值字节数组\nvar readMatroskaVInt=function(arr,pos,trim){\n\tvar i=pos[0];\n\tif(i>=arr.length)return;\n\tvar b0=arr[i],b2=(\"0000000\"+b0.toString(2)).substr(-8);\n\tvar m=/^(0*1)(\\d*)$/.exec(b2);\n\tif(!m)return;\n\tvar len=m[1].length, val=[];\n\tif(i+len>arr.length)return;\n\tfor(var i2=0;i2<len;i2++){ val[i2]=arr[i]; i++; }\n\tif(trim) val[0]=parseInt(m[2]||'0',2);\n\tpos[0]=i;\n\treturn val;\n};\n//读取一个自带长度的内容字节数组\nvar readMatroskaBlock=function(arr,pos){\n\tvar lenVal=readMatroskaVInt(arr,pos,1);\n\tif(!lenVal)return;\n\tvar len=BytesInt(lenVal);\n\tvar i=pos[0], val=[];\n\tif(len<0x7FFFFFFF){ //超大值代表没有长度\n\t\tif(i+len>arr.length)return;\n\t\tfor(var i2=0;i2<len;i2++){ val[i2]=arr[i]; i++; }\n\t}\n\tpos[0]=i;\n\treturn val;\n};\n//=====End WebM读取=====\n\n\n//=====i18n 简版国际化多语言支持=====\nvar i18n=Recorder.i18n={\n\tlang: \"zh-CN\" //默认中文\n\t,alias:{\"zh-CN\":\"zh\",\"en-US\":\"en\"} //别名配置，映射到一个语言实例\n\t,locales:{} //语言实例：zh:{key:\"text\"}\n\t,data:{} //各种数据，desc$xx：语言描述，rtl$xx：文本方向是否从右到左 rtl$zh:false rtl$ar:true\n\t//添加语言文本内容，set={lang:\"\",overwrite:true}，texts=[\"key:text{1}\",...]\n\t,put:function(set,texts){\n\t\tvar tag=RecTxt+\".i18n.put: \";\n\t\tvar overwrite=set.overwrite; overwrite=overwrite==null||overwrite;\n\t\tvar lang=set.lang; lang=i18n.alias[lang]||lang;\n\t\tif(!lang)throw new Error(tag+\"set.lang?\");\n\t\tvar locale=i18n.locales[lang];\n\t\tif(!locale){ locale={}; i18n.locales[lang]=locale };\n\t\tvar exp=/^([\\w\\-]+):/,m;\n\t\tfor(var i=0;i<texts.length;i++){\n\t\t\tvar v=texts[i]; m=exp.exec(v);\n\t\t\tif(!m){ CLog(tag+\"'key:'? \"+v,3,set); continue }\n\t\t\tvar key=m[1],v=v.substr(key.length+1);\n\t\t\tif(!overwrite && locale[key]) continue;\n\t\t\tlocale[key]=v;\n\t\t}\n\t}\n\t//获取key对应的文本，如果没有对应文本，将返回en的，en的也没有将返回zh的\n\t,get:function(/*key,args,lang*/){\n\t\treturn i18n.v_G.apply(null,arguments);\n\t}, v_G:function(key,args,lang){ //全局可重写get\n\t\targs=args||[];\n\t\tlang=lang||i18n.lang; lang=i18n.alias[lang]||lang;\n\t\tvar locale=i18n.locales[lang];\n\t\tvar val=locale&&locale[key]||\"\";\n\t\tif(!val&&lang!=\"zh\"){\n\t\t\tif(lang==\"en\")return i18n.v_G(key,args,\"zh\");\n\t\t\treturn i18n.v_G(key,args,\"en\");\n\t\t}\n\t\ti18n.lastLang=lang;\n\t\tif(val==\"=Empty\")return \"\";\n\t\treturn val.replace(/\\{(\\d+)(\\!?)\\}/g,function(v,a,b){\n\t\t\ta=+a||0; v=args[a-1];\n\t\t\tif(a<1 || a>args.length){ v=\"{?}\"; CLog(\"i18n[\"+key+\"] no {\"+a+\"}: \"+val,3) }\n\t\t\treturn b?\"\":v;\n\t\t});\n\t}\n\t/**返回一个国际化文本，返回的文本取决于i18n.lang。\n\t\t调用参数：T(\"key:[lang]:中文{1}\",\"[lang]:英文{1}\",...,0,\"args1\",\"args2\")，除了key:，其他都是可选的，文本如果在语言实例中不存在会push进去，第一个文本无lang时默认zh，第二个无lang时默认en，文本中用{1-n}来插入args\n\t\t第一个args前面这个参数必须是0；也可以不提供args，这个参数填args的数量，此时不返回文本，只返回key，再用i18n.get提供参数获取文本\n\t\t本函数调用，第一个文本需中文，调用尽量简单，不要换行，方便后续自动提取翻译列表\n\t\targs如果旧的参数位置发生了变更，应当使用新的key，让旧的翻译失效，增加args无需更换key\n\t\tkey的生成使用随机字符串，不同源码里面可以搞个不同前缀:\n\t\t\ts=\"\";L=4; for(var i=0,n;i<L;i++){ n=~~(Math.random()*62);s+=n<10?n:String.fromCharCode(n<36?55+n:61+n); }; s\n\t**/\n\t,$T:function(){\n\t\treturn i18n.v_T.apply(null,arguments);\n\t} ,v_T:function(){ //全局可重写$T\n\t\tvar a=arguments,key=\"\",args=[],isArgs=0,tag=RecTxt+\".i18n.$T:\";\n\t\tvar exp=/^([\\w\\-]*):/,m;\n\t\tfor(var i=0;i<a.length;i++){\n\t\t\tvar v=a[i];\n\t\t\tif(i==0){\n\t\t\t\tm=exp.exec(v); key=m&&m[1];\n\t\t\t\tif(!key)throw new Error(tag+\"0 'key:'?\");\n\t\t\t\tv=v.substr(key.length+1);\n\t\t\t}\n\t\t\tif(isArgs===-1) args.push(v);\n\t\t\telse if(isArgs) throw new Error(tag+\" bad args\");\n\t\t\telse if(v===0) isArgs=-1;\n\t\t\telse if(IsNum(v)){\n\t\t\t\tif(v<1) throw new Error(tag+\" bad args\");\n\t\t\t\tisArgs=v;\n\t\t\t}else{\n\t\t\t\tvar lang=i==1?\"en\":i?\"\":\"zh\";\n\t\t\t\tm=exp.exec(v); if(m){ lang=m[1]||lang; v=v.substr(m[1].length+1); }\n\t\t\t\tif(!m || !lang)throw new Error(tag+i+\" 'lang:'?\");\n\t\t\t\ti18n.put({lang:lang,overwrite:false},[key+\":\"+v]);\n\t\t\t}\n\t\t}\n\t\tif(!key)return \"\";\n\t\tif(isArgs>0)return key;\n\t\treturn i18n.v_G(key,args);\n\t}\n};\nvar $T=i18n.$T; $T.G=i18n.get;\n//预定义文本\n$T(\"NonBrowser-1::非浏览器环境，不支持{1}\",1);\n$T(\"IllegalArgs-1::参数错误：{1}\",1);\n$T(\"NeedImport-2::调用{1}需要先导入{2}\",2);\n$T(\"NotSupport-1::不支持：{1}\",1);\n//=====End i18n=====\n\n\n\n//流量统计用1像素图片地址，设置为空将不参与统计\nRecorder.TrafficImgUrl=\"//ia.51.la/go1?id=20469973&pvFlag=1\";\nvar Traffic=Recorder.Traffic=function(report){\n\tif(!isBrowser)return;\n\treport=report?\"/\"+RecTxt+\"/Report/\"+report:\"\";\n\tvar imgUrl=Recorder.TrafficImgUrl;\n\tif(imgUrl){\n\t\tvar data=Recorder.Traffic;\n\t\tvar m=/^(https?:..[^\\/#]*\\/?)[^#]*/i.exec(location.href)||[];\n\t\tvar host=(m[1]||\"http://file/\");\n\t\tvar idf=(m[0]||host)+report;\n\t\t\n\t\tif(imgUrl.indexOf(\"//\")==0){\n\t\t\t//给url加上http前缀，如果是file协议下，不加前缀没法用\n\t\t\tif(/^https:/i.test(idf)){\n\t\t\t\timgUrl=\"https:\"+imgUrl;\n\t\t\t}else{\n\t\t\t\timgUrl=\"http:\"+imgUrl;\n\t\t\t};\n\t\t};\n\t\tif(report){\n\t\t\timgUrl=imgUrl+\"&cu=\"+encodeURIComponent(host+report);\n\t\t};\n\t\t\n\t\tif(!data[idf]){\n\t\t\tdata[idf]=1;\n\t\t\t\n\t\t\tvar img=new Image();\n\t\t\timg.src=imgUrl;\n\t\t\tCLog(\"Traffic Analysis Image: 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