npm - @easyv/biz-components - Versions diffs - 1.0.28 → 1.0.30 - Mend

@easyv/biz-components 1.0.28 → 1.0.30

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package/dist/lib/utils/fun-ASR/recorder.es.js.map ADDED Viewed

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+ {"version":3,"file":"recorder.es.js","sources":["../../../../src/utils/fun-ASR/recorder.js"],"sourcesContent":["var NOOP = function () {};\n/**\n * @type {(set: any) => any}\n */\nexport const Recorder = function (set) {\n return new initFn(set);\n};\n\nRecorder.LM = '2023-02-01 18:05';\nvar RecTxt = 'Recorder';\nvar getUserMediaTxt = 'getUserMedia';\nvar srcSampleRateTxt = 'srcSampleRate';\nvar sampleRateTxt = 'sampleRate';\nvar CatchTxt = 'catch';\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 var stream = Recorder.Stream;\n if (stream) {\n var tracks = (stream.getTracks && stream.getTracks()) || stream.audioTracks || [];\n var track = tracks[0];\n if (track) {\n var state = track.readyState;\n return state == 'live' || state == track.LIVE;\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取值256, 512, 1024, 2048, 4096, 8192, or 16384\n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 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};\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 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。\nRecorder.Support = function () {\n var scope = navigator.mediaDevices || {};\n if (!scope[getUserMediaTxt]) {\n scope = navigator;\n scope[getUserMediaTxt] ||\n (scope[getUserMediaTxt] =\n scope.webkitGetUserMedia || scope.mozGetUserMedia || scope.msGetUserMedia);\n }\n if (!scope[getUserMediaTxt]) {\n return false;\n }\n Recorder.Scope = scope;\n\n if (!Recorder.GetContext()) {\n return false;\n }\n return true;\n};\n//获取全局的AudioContext对象，如果浏览器不支持将返回null\nRecorder.GetContext = function () {\n var AC = window.AudioContext;\n if (!AC) {\n AC = window.webkitAudioContext;\n }\n if (!AC) {\n return null;\n }\n\n if (!Recorder.Ctx || Recorder.Ctx.state == 'closed') {\n //不能反复构造，低版本number of hardware contexts reached maximum (6)\n Recorder.Ctx = new AC();\n\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 ctx.close();\n Recorder.Ctx = 0;\n }\n });\n }\n return Recorder.Ctx;\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, isUserMedia) {\n var bufferSize = streamStore.BufferSize || Recorder.BufferSize;\n\n var ctx = Recorder.Ctx,\n stream = streamStore.Stream;\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 = ctx[cmsdTxt]();\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) {\n for (var k0 in calls) {\n //has item\n var size = float32Arr.length;\n\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 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 =\n '。由于' +\n audioWorklet +\n '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' +\n audioWorklet +\n '。';\n var oldScript = function () {\n isWorklet = stream.isWorklet = false;\n _Disconn_n(stream);\n CLog(\n 'Connect采用老的' +\n scriptProcessor +\n '，' +\n (Recorder[ConnectEnableWorklet] ? '但已' : '可') +\n '设置' +\n RecTxt +\n '.' +\n ConnectEnableWorklet +\n '=true尝试启用' +\n audioWorklet +\n webMTips +\n oldIsBest,\n 3,\n );\n\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\n var _DsetTxt = '_D220626',\n _Dset = Recorder[_DsetTxt];\n if (_Dset) CLog('Use ' + RecTxt + '.' + _DsetTxt, 3);\n process.onaudioprocess = function (e) {\n var arr = e.inputBuffer.getChannelData(0);\n if (_Dset) {\n //临时调试用的参数，未来会被删除\n arr = new Float32Array(arr); //块是共享的，必须复制出来\n setTimeout(function () {\n onReceive(arr);\n }); //立即退出回调，试图减少对浏览器录音的影响\n } else {\n onReceive(arr);\n }\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\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 var clazzUrl = function () {\n var xf = function (f) {\n return f\n .toString()\n .replace(/^function|DEL_/g, '')\n .replace(/\\$RA/g, recAudioWorklet);\n };\n var clazz = 'class ' + RecProc + ' extends AudioWorkletProcessor{';\n clazz +=\n 'constructor ' +\n 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 console.log('$RA kill call');\n }\n };\n console.log('$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 +=\n 'process ' +\n 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\n var len = ~~(pos / bufferSize) * bufferSize;\n if (len) {\n this.port.postMessage({ 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 +=\n '}' +\n 'try{' +\n 'registerProcessor(\"' +\n RecProc +\n '\", ' +\n RecProc +\n ')' +\n '}catch(e){' +\n 'console.error(\"' +\n recAudioWorklet +\n '注册失败\",e)' +\n '}';\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\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(audioWorklet + '未返回任何音频，恢复使用' + scriptProcessor, 3);\n oldFn && oldScript(); //未来没有老的，可能是误判\n }\n }, 500);\n }\n });\n var createNode = function () {\n if (!awNext()) return;\n var node = (stream._n = new AwNode(ctx, RecProc, {\n processorOptions: { bufferSize: bufferSize },\n }));\n mediaConn(node);\n node.port.onmessage = function (e) {\n if (badInt) {\n clearTimeout(badInt);\n badInt = '';\n }\n if (awNext()) {\n onReceive(e.data.val);\n } else if (!isWorklet) {\n CLog(audioWorklet + '多余回调', 3);\n }\n };\n CLog(\n 'Connect采用' +\n audioWorklet +\n '，设置' +\n RecTxt +\n '.' +\n ConnectEnableWorklet +\n '=false可恢复老式' +\n scriptProcessor +\n webMTips +\n oldIsBest,\n 3,\n );\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，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 ctx.resume()[calls && 'finally'](function () {\n //注释掉这行观摩浏览器崩溃 STATUS_ACCESS_VIOLATION\n if (!awNext()) return;\n if (ctx[RecProc]) {\n createNode();\n return;\n }\n var url = clazzUrl();\n ctx[audioWorklet]\n .addModule(url)\n .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失败', 1, e);\n awNext() && oldScript();\n });\n });\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\n var supportMR = MR && onData in MR.prototype && MR.isTypeSupported(webmType);\n webMTips = supportMR ? '' : '（此浏览器不支持' + MRWebMPCM + '）';\n if (!isUserMedia || !isWebM || !supportMR) {\n connWorklet(); //非麦克风录音（MediaRecorder采样率不可控）或被禁用或不支持MediaRecorder 或不支持webm+pcm\n return;\n }\n\n var mrNext = function () {\n //可以继续，没有调用断开\n return isWebM && stream._ra;\n };\n var mrAlive = (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(MediaRecorderTxt + '未返回任何音频，降级使用' + audioWorklet, 3);\n connWorklet();\n }\n }, 500);\n }\n });\n\n var mrSet = Object.assign({ mimeType: webmType }, Recorder.ConnectWebMOptions);\n var mr = (stream._r = new MR(stream, mrSet));\n var webmData = (stream._rd = { sampleRate: ctx[sampleRateTxt] });\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 onReceive(f32arr);\n } else if (!isWebM) {\n CLog(MediaRecorderTxt + '多余回调', 3);\n }\n };\n reader.readAsArrayBuffer(e.data);\n };\n mr.start(~~(bufferSize / 48)); //按48k时的回调间隔\n CLog(\n 'Connect采用' +\n MRWebMPCM +\n '，设置' +\n RecTxt +\n '.' +\n ConnectEnableWebM +\n '=false可恢复使用' +\n audioWorklet +\n '或老式' +\n scriptProcessor,\n );\n };\n\n connWebM();\n};\nvar 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};\nvar _Disconn_n = function (stream) {\n stream._na = null;\n if (stream._n) {\n stream._n.port.postMessage({ kill: true });\n stream._n.disconnect();\n stream._n = null;\n }\n};\nvar _Disconn_r = function (stream) {\n stream._ra = null;\n if (stream._r) {\n stream._r.stop();\n stream._r = null;\n }\n};\nvar Disconnect = function (streamStore) {\n streamStore = streamStore || Recorder;\n var isGlobal = streamStore == Recorder;\n\n var stream = streamStore.Stream;\n if (stream) {\n if (stream._m) {\n stream._m.disconnect();\n stream._m = null;\n }\n if (stream._p) {\n stream._p.disconnect();\n stream._p.onaudioprocess = stream._p = null;\n }\n _Disconn_n(stream);\n _Disconn_r(stream);\n\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 var tracks = (stream.getTracks && stream.getTracks()) || stream.audioTracks || [];\n for (var i = 0; i < tracks.length; i++) {\n var track = tracks[i];\n track.stop && track.stop();\n }\n stream.stop && stream.stop();\n }\n }\n streamStore.Stream = 0;\n};\n\n/*对pcm数据的采样率进行转换\npcmDatas: [[Int16,...]] pcm片段列表\npcmSampleRate:48000 pcm数据的采样率\nnewSampleRate:16000 需要转换成的采样率，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 frameSize: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 frameType:\"\" 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 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}\n\n返回ChunkInfo:{\n//可定义，从指定位置开始转换到结尾\nindex:0 pcmDatas已处理到的索引\noffset:0.0 已处理到的index对应的pcm中的偏移的下一个位置\n\n//仅作为返回值\nframeNext:null||[Int16,...] 下一帧的部分数据，frameSize设置了的时候才可能会有\nsampleRate:16000 结果的采样率，<=newSampleRate\ndata:[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 prevChunkInfo || (prevChunkInfo = {});\n var index = prevChunkInfo.index || 0;\n var offset = prevChunkInfo.offset || 0;\n\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 nLen = pcmDatas.length;\n if (index > nLen + 1) {\n CLog('SampleData似乎传入了未重置chunk ' + index + '>' + nLen, 3);\n }\n var size = 0;\n for (var i = index; i < nLen; i++) {\n size += pcmDatas[i].length;\n }\n size = Math.max(0, size - Math.floor(offset));\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.floor(size / step);\n } else {\n //新采样高于录音采样不处理，省去了插值处理\n step = 1;\n newSampleRate = pcmSampleRate;\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 //处理数据\n for (; index < nLen; index++) {\n var o = pcmDatas[index];\n var i = offset,\n il = o.length;\n while (i < il) {\n //res[idx]=o[Math.round(i)]; 直接简单抽样\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 var after = Math.ceil(i);\n var atPoint = i - before;\n\n var beforeVal = o[before];\n var afterVal =\n after < il\n ? o[after]\n : //后个点越界了，查找下一个数组\n (pcmDatas[index + 1] || [beforeVal])[0] || 0;\n res[idx] = beforeVal + (afterVal - beforeVal) * atPoint;\n\n idx++;\n i += step; //抽样\n }\n offset = i - il;\n }\n //帧处理\n frameNext = null;\n var frameNextSize = res.length % frameSize;\n if (frameNextSize > 0) {\n var u8Pos = (res.length - frameNextSize) * 2;\n frameNext = new Int16Array(res.buffer.slice(u8Pos));\n res = new Int16Array(res.buffer.slice(0, u8Pos));\n }\n\n return {\n index: index,\n offset: offset,\n\n frameNext: frameNext,\n sampleRate: newSampleRate,\n data: res,\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 /*计算音量 https://blog.csdn.net/jody1989/article/details/73480259\n更高灵敏度算法:\n 限定最大感应值10000\n 线性曲线：低音量不友好\n power/10000*100 \n 对数曲线：低音量友好，但需限定最低感应值\n (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(\n 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）\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 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\nRecorder.CLog = function (msg, err) {\n var now = new Date();\n var t =\n ('0' + now.getMinutes()).substr(-2) +\n ':' +\n ('0' + now.getSeconds()).substr(-2) +\n '.' +\n ('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 console = window.console || {};\n var i = 2,\n fn = console.log;\n if (typeof err == 'number') {\n fn = err == 1 ? console.error : err == 3 ? console.warn : fn;\n } else {\n i = 1;\n }\n for (; i < a.length; i++) {\n arr.push(a[i]);\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};\nvar CLog = function () {\n Recorder.CLog.apply(this, arguments);\n};\nvar IsLoser = true;\ntry {\n IsLoser = !console.log.apply;\n} catch (e) {}\n\nvar ID = 0;\nfunction initFn(set) {\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\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 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 //,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 //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；环境要求比较苛刻：如果当前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 //目前只有mp3格式实现了实时编码，在支持实时处理的环境中将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 //目前除mp3外其他格式不可以提供此回调，提供了将在open时直接走fail逻辑\n };\n\n for (var k in set) {\n o[k] = set[k];\n }\n this.set = o;\n\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 = { 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 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 _streamStore: function () {\n if (this.set.sourceStream) {\n return this;\n } else {\n return Recorder;\n }\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\n open: function (True, False) {\n var This = this,\n streamStore = This._streamStore();\n True = True || NOOP;\n var failCall = function (errMsg, isUserNotAllow) {\n isUserNotAllow = !!isUserNotAllow;\n This.CLog('录音open失败：' + errMsg + ',isUserNotAllow:' + isUserNotAllow, 1);\n False && False(errMsg, isUserNotAllow);\n };\n\n var ok = function () {\n This.CLog('open ok id:' + This.id);\n True();\n\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 = '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 = 'open被中断';\n }\n failCall(err);\n return true;\n }\n };\n\n //环境配置检查\n var checkMsg = This.envCheck({ envName: 'H5', canProcess: true });\n if (checkMsg) {\n failCall('不能录音：' + checkMsg);\n return;\n }\n\n //***********已直接提供了音频流************\n if (This.set.sourceStream) {\n if (!Recorder.GetContext()) {\n failCall('不支持此浏览器从流中获取录音');\n return;\n }\n\n Disconnect(streamStore); //可能已open过，直接先尝试断开\n This.Stream = This.set.sourceStream;\n This.Stream._call = {};\n\n try {\n Connect(streamStore);\n } catch (e) {\n failCall('从流中打开录音失败：' + e.message);\n return;\n }\n ok();\n return;\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('无权录音(跨域，请尝试给iframe添加麦克风访问策略，如allow=\"camera;microphone\")');\n return;\n }\n\n if (/Permission|Allow/i.test(code)) {\n failCall('用户拒绝了录音权限', true);\n } else if (window.isSecureContext === false) {\n failCall('浏览器禁止不安全页面录音，可开启https解决');\n } else if (/Found/i.test(code)) {\n //可能是非安全环境导致的没有设备\n failCall(msg + '，无可用麦克风');\n } else {\n failCall(msg);\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 if (!Recorder.Support()) {\n codeFail('', '此浏览器不支持录音');\n return;\n }\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 //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 Recorder.Stream = stream;\n if (lockFail()) return;\n\n if (Recorder.IsOpen()) {\n if (oldStream) This.CLog('发现同时多次调用open', 1);\n\n Connect(streamStore, 1);\n ok();\n } else {\n failCall('录音功能无效：无音频流');\n }\n }, 100);\n };\n var f2 = function (e) {\n var code = e.name || e.message || e.code + ':' + e;\n This.CLog('请求录音权限错误', 1, e);\n\n codeFail(code, '无法录音：' + code);\n };\n\n var trackSet = {\n noiseSuppression: false, //默认禁用降噪，原声录制，免得移动端表现怪异（包括系统播放声音变小）\n echoCancellation: false, //回声消除\n };\n var trackSet2 = This.set.audioTrackSet;\n for (var k in trackSet2) trackSet[k] = trackSet2[k];\n trackSet.sampleRate = Recorder.Ctx.sampleRate; //必须指明采样率，不然手机上MediaRecorder采样率16k\n\n try {\n var pro = Recorder.Scope[getUserMediaTxt]({ audio: trackSet }, f1, f2);\n } catch (e) {\n //不能设置trackSet就算了\n This.CLog(getUserMediaTxt, 3, e);\n pro = Recorder.Scope[getUserMediaTxt]({ audio: true }, f1, f2);\n }\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 //关闭释放录音资源\n close: function (call) {\n call = call || NOOP;\n\n var This = this,\n streamStore = This._streamStore();\n This._stop();\n\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('close被忽略（因为同时open了多个rec，只有最后一个会真正close）', 3);\n call();\n return;\n }\n Lock.C++; //获得控制权\n\n Disconnect(streamStore);\n\n This.CLog('close');\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[srcSampleRateTxt] = pcmSampleRate;\n return This;\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 (\n !errMsg &&\n !Recorder[tag] &&\n window.Int8Array &&\n !new Int8Array(new Int32Array([1]).buffer)[0]\n ) {\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 = '不支持' + tag + '架构';\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 if (This[type + '_envCheck']) {\n //编码器已实现环境检查\n errMsg = This[type + '_envCheck'](envInfo, set);\n } else {\n //未实现检查的手动检查配置是否有效\n if (set.takeoffEncodeChunk) {\n errMsg =\n type + '类型' + (This[type] ? '' : '(未加载编码器)') + '不支持设置takeoffEncodeChunk';\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\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 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(\n srcSampleRateTxt +\n ': ' +\n sampleRate +\n ' set.' +\n sampleRateTxt +\n ': ' +\n set[sampleRateTxt] +\n (setSr ? ' 忽略' + setSr : ''),\n setSr ? 3 : 0,\n );\n\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 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 var bufferSampleRate = This[srcSampleRateTxt];\n var size = pcm.length;\n var powerLevel = Recorder.PowerLevel(sum, size);\n\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 var envInFixTs = This.envInFixTs;\n envInFixTs.splice(0, 0, { t: now, d: pcmTime });\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 tsInStart = o.t;\n tsPcm += o.d;\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 + ']' + (fixOpen ? '' : '未') + '补偿' + addTime + 'ms', 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 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(\n buffers,\n bufferSampleRate,\n set[sampleRateTxt],\n engineCtx.chunkInfo,\n );\n engineCtx.chunkInfo = chunkInfo;\n\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 //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 for (; i < bufferNextIdxThis; i++) {\n buffersThis[i] = null;\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\n buffers[0] = null; //彻底被清理\n }\n if (engineCtx) {\n engineCtx.pcmSize += num;\n } else {\n This.recSize += num;\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(\n buffers,\n powerLevel,\n duration,\n bufferSampleRate,\n bufferFirstIdx,\n asyncEnd,\n );\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 + '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 + '低性能，耗时' + slowT + 'ms', 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 if (hasClear) {\n This.CLog('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 } else {\n asyncEnd();\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\n start: function () {\n var This = this,\n ctx = Recorder.Ctx;\n\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 if (!isOpen) {\n This.CLog('未open', 1);\n return;\n }\n This.CLog('开始录音');\n\n This._stop();\n This.state = 3; //0未录音 1录音中 2暂停 3等待ctx激活\n This.envStart(null, ctx[sampleRateTxt]);\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('start被中断', 3);\n return;\n }\n This._SO = 0;\n\n var end = function () {\n if (This.state == 3) {\n This.state = 1;\n This.resume();\n }\n };\n if (ctx.state == 'suspended') {\n var tag = 'AudioContext resume: ';\n This.CLog(tag + 'wait...');\n ctx\n .resume()\n .then(function () {\n This.CLog(tag + ctx.state);\n end();\n })\n [CatchTxt](function (e) {\n //比较少见，可能对录音没有影响\n This.CLog(tag + ctx.state + ' 可能无法录音：' + e.message, 1, e);\n end();\n });\n } else {\n end();\n }\n },\n /*暂停录音*/\n pause: function () {\n var This = this;\n if (This.state) {\n This.state = 2;\n This.CLog('pause');\n delete This._streamStore().Stream._call[This.id];\n }\n },\n /*恢复录音*/\n resume: function () {\n var This = this;\n if (This.state) {\n This.state = 1;\n This.CLog('resume');\n This.envResume();\n\n var stream = This._streamStore().Stream;\n stream._call[This.id] = function (pcm, sum) {\n if (This.state == 1) {\n This.envIn(pcm, sum);\n }\n };\n ConnAlive(stream); //AudioWorklet只会在ctx激活后运行\n }\n },\n\n _stop: function (keepEngine) {\n var This = this,\n set = This.set;\n if (!This.isMock) {\n This._S++;\n }\n if (This.state) {\n This.pause();\n This.state = 0;\n }\n if (!keepEngine && This[set.type + '_stop']) {\n This[set.type + '_stop'](This.engineCtx);\n This.engineCtx = 0;\n }\n },\n /*\n 结束录音并返回录音数据blob对象\n True(blob,duration) \n blob：录音数据audio/mp3|wav格式\n duration：录音时长，单位毫秒\n False(msg)\n autoClose: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(\n 'stop 和start时差' +\n (envInMS\n ? envInMS +\n 'ms 补偿' +\n This.envInFix +\n 'ms' +\n ' envIn:' +\n envInLen +\n ' fps:' +\n ((envInLen / envInMS) * 1000).toFixed(1)\n : '-'),\n );\n\n var end = function () {\n This._stop(); //彻底关掉engineCtx\n if (autoClose) {\n This.close();\n }\n };\n var err = function (msg) {\n This.CLog('结束录音失败：' + msg, 1);\n False && False(msg);\n end();\n };\n var ok = function (blob, duration) {\n This.CLog(\n '结束录音编码花' +\n (Date.now() - t1) +\n 'ms 音频时长' +\n duration +\n 'ms 文件大小' +\n blob.size +\n 'b',\n );\n if (set.takeoffEncodeChunk) {\n //接管了输出，此时blob长度为0\n This.CLog('启用takeoffEncodeChunk后stop返回的blob长度为0不提供音频数据', 3);\n } else if (blob.size < Math.max(100, duration / 2)) {\n //1秒小于0.5k？\n err('生成的' + set.type + '无效');\n return;\n }\n True && True(blob, duration);\n end();\n };\n if (!This.isMock) {\n var isCtxWait = This.state == 3;\n\n if (!This.state || isCtxWait) {\n console.log(\n '未开始录音' + (isCtxWait ? '，开始录音前无用户交互导致AudioContext未运行' : ''),\n );\n return;\n }\n This._stop(true);\n }\n var size = This.recSize;\n if (!size) {\n err('未采集到录音');\n return;\n }\n if (!This.buffers[0]) {\n err('音频buffers被释放');\n return;\n }\n if (!This[set.type]) {\n err('未加载' + set.type + '编码器');\n return;\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 || { envName: 'mock', canProcess: false }); //没有提供环境信息的mock时没有onProcess回调\n if (checkMsg) {\n err('录音错误：' + checkMsg);\n return;\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'](\n engineCtx,\n function (blob) {\n ok(blob, duration);\n },\n err,\n );\n return;\n }\n\n //标准UI线程转码，调整采样率\n t1 = Date.now();\n var chunk = Recorder.SampleData(This.buffers, This[srcSampleRateTxt], set[sampleRateTxt]);\n\n set[sampleRateTxt] = chunk[sampleRateTxt];\n var res = chunk.data;\n var duration = Math.round((res.length / set[sampleRateTxt]) * 1000);\n\n This.CLog('采样' + size + '->' + res.length + ' 花:' + (Date.now() - t1) + 'ms');\n\n setTimeout(function () {\n t1 = Date.now();\n This[set.type](\n res,\n function (blob) {\n ok(blob, duration);\n },\n function (msg) {\n err(msg);\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 if (!scope.pos) {\n scope.pos = [0];\n scope.tracks = {};\n scope.bytes = [];\n }\n var tracks = scope.tracks,\n position = [scope.pos[0]];\n var endPos = function () {\n scope.pos[0] = position[0];\n };\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); 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package/dist/lib/utils/fun-ASR/wav.es.js ADDED Viewed

@@ -0,0 +1,54 @@
+function wavEnhanceRecorder(Recorder) {
+  Recorder.prototype.enc_wav = {
+    stable: true,
+    testmsg: "支持位数8位、16位（填在比特率里面），采样率取值无限制"
+  };
+  Recorder.prototype.wav = function(res, True, False) {
+    var This = this, set = This.set, size = res.length, sampleRate = set.sampleRate, bitRate = set.bitRate == 8 ? 8 : 16;
+    var dataLength = size * (bitRate / 8);
+    var buffer = new ArrayBuffer(44 + dataLength);
+    var data = new DataView(buffer);
+    var offset = 0;
+    var writeString = function(str) {
+      for (var i2 = 0; i2 < str.length; i2++, offset++) {
+        data.setUint8(offset, str.charCodeAt(i2));
+      }
+    };
+    var write16 = function(v) {
+      data.setUint16(offset, v, true);
+      offset += 2;
+    };
+    var write32 = function(v) {
+      data.setUint32(offset, v, true);
+      offset += 4;
+    };
+    writeString("RIFF");
+    write32(36 + dataLength);
+    writeString("WAVE");
+    writeString("fmt ");
+    write32(16);
+    write16(1);
+    write16(1);
+    write32(sampleRate);
+    write32(sampleRate * (bitRate / 8));
+    write16(bitRate / 8);
+    write16(bitRate);
+    writeString("data");
+    write32(dataLength);
+    if (bitRate == 8) {
+      for (var i = 0; i < size; i++, offset++) {
+        var val = (res[i] >> 8) + 128;
+        data.setInt8(offset, val, true);
+      }
+    } else {
+      for (var i = 0; i < size; i++, offset += 2) {
+        data.setInt16(offset, res[i], true);
+      }
+    }
+    True(new Blob([data.buffer], { type: "audio/wav" }));
+  };
+}
+export {
+  wavEnhanceRecorder
+};
+//# sourceMappingURL=wav.es.js.map

package/dist/lib/utils/fun-ASR/wav.es.js.map ADDED Viewed

@@ -0,0 +1 @@

+ {"version":3,"file":"wav.es.js","sources":["../../../../src/utils/fun-ASR/wav.js"],"sourcesContent":["/*\nwav编码器+编码引擎\nhttps://github.com/xiangyuecn/Recorder\n\n当然最佳推荐使用mp3、wav格式，代码也是优先照顾这两种格式\n浏览器支持情况\nhttps://developer.mozilla.org/en-US/docs/Web/HTML/Supported_media_formats\n\n编码原理：给pcm数据加上一个44直接的wav头即成wav文件；pcm数据就是Recorder中的buffers原始数据（重新采样），16位时为LE小端模式（Little Endian），实质上是未经过任何编码处理\n*/\nexport function wavEnhanceRecorder(Recorder) {\n 'use strict';\n\n Recorder.prototype.enc_wav = {\n stable: true,\n testmsg: '支持位数8位、16位（填在比特率里面），采样率取值无限制',\n };\n Recorder.prototype.wav = function (res, True, False) {\n var This = this,\n set = This.set,\n size = res.length,\n sampleRate = set.sampleRate,\n bitRate = set.bitRate == 8 ? 8 : 16;\n\n //编码数据 https://github.com/mattdiamond/Recorderjs https://www.cnblogs.com/blqw/p/3782420.html https://www.cnblogs.com/xiaoqi/p/6993912.html\n var dataLength = size * (bitRate / 8);\n var buffer = new ArrayBuffer(44 + dataLength);\n var data = new DataView(buffer);\n\n var offset = 0;\n var writeString = function (str) {\n for (var i = 0; i < str.length; i++, offset++) {\n data.setUint8(offset, str.charCodeAt(i));\n }\n };\n var write16 = function (v) {\n data.setUint16(offset, v, true);\n offset += 2;\n };\n var write32 = function (v) {\n data.setUint32(offset, v, true);\n offset += 4;\n };\n\n /* RIFF identifier */\n writeString('RIFF');\n /* RIFF chunk length */\n write32(36 + dataLength);\n /* RIFF type */\n writeString('WAVE');\n /* format chunk identifier */\n writeString('fmt ');\n /* format chunk length */\n write32(16);\n /* sample format (raw) */\n write16(1);\n /* channel count */\n write16(1);\n /* sample rate */\n write32(sampleRate);\n /* byte rate (sample rate * block align) */\n write32(sampleRate * (bitRate / 8)); // *1 声道\n /* block align (channel count * bytes per sample) */\n write16(bitRate / 8); // *1 声道\n /* bits per sample */\n write16(bitRate);\n /* data chunk identifier */\n writeString('data');\n /* data chunk length */\n write32(dataLength);\n // 写入采样数据\n if (bitRate == 8) {\n for (var i = 0; i < size; i++, offset++) {\n //16转8据说是雷霄骅的 https://blog.csdn.net/sevennight1989/article/details/85376149 细节比blqw的按比例的算法清晰点，虽然都有明显杂音\n var val = (res[i] >> 8) + 128;\n data.setInt8(offset, val, true);\n }\n } else {\n for (var i = 0; i < size; i++, offset += 2) {\n data.setInt16(offset, res[i], true);\n }\n }\n\n True(new Blob([data.buffer], { type: 'audio/wav' }));\n };\n}\n"],"names":["i"],"mappings":"AAUO,SAAS,mBAAmB,UAAU;AAG3C,WAAS,UAAU,UAAU;AAAA,IAC3B,QAAQ;AAAA,IACR,SAAS;AAAA,EACV;AACD,WAAS,UAAU,MAAM,SAAU,KAAK,MAAM,OAAO;AACnD,QAAI,OAAO,MACT,MAAM,KAAK,KACX,OAAO,IAAI,QACX,aAAa,IAAI,YACjB,UAAU,IAAI,WAAW,IAAI,IAAI;AAGnC,QAAI,aAAa,QAAQ,UAAU;AACnC,QAAI,SAAS,IAAI,YAAY,KAAK,UAAU;AAC5C,QAAI,OAAO,IAAI,SAAS,MAAM;AAE9B,QAAI,SAAS;AACb,QAAI,cAAc,SAAU,KAAK;AAC/B,eAASA,KAAI,GAAGA,KAAI,IAAI,QAAQA,MAAK,UAAU;AAC7C,aAAK,SAAS,QAAQ,IAAI,WAAWA,EAAC,CAAC;AAAA,MAC/C;AAAA,IACK;AACD,QAAI,UAAU,SAAU,GAAG;AACzB,WAAK,UAAU,QAAQ,GAAG,IAAI;AAC9B,gBAAU;AAAA,IACX;AACD,QAAI,UAAU,SAAU,GAAG;AACzB,WAAK,UAAU,QAAQ,GAAG,IAAI;AAC9B,gBAAU;AAAA,IACX;AAGD,gBAAY,MAAM;AAElB,YAAQ,KAAK,UAAU;AAEvB,gBAAY,MAAM;AAElB,gBAAY,MAAM;AAElB,YAAQ,EAAE;AAEV,YAAQ,CAAC;AAET,YAAQ,CAAC;AAET,YAAQ,UAAU;AAElB,YAAQ,cAAc,UAAU,EAAE;AAElC,YAAQ,UAAU,CAAC;AAEnB,YAAQ,OAAO;AAEf,gBAAY,MAAM;AAElB,YAAQ,UAAU;AAElB,QAAI,WAAW,GAAG;AAChB,eAAS,IAAI,GAAG,IAAI,MAAM,KAAK,UAAU;AAEvC,YAAI,OAAO,IAAI,CAAC,KAAK,KAAK;AAC1B,aAAK,QAAQ,QAAQ,KAAK,IAAI;AAAA,MACtC;AAAA,IACA,OAAW;AACL,eAAS,IAAI,GAAG,IAAI,MAAM,KAAK,UAAU,GAAG;AAC1C,aAAK,SAAS,QAAQ,IAAI,CAAC,GAAG,IAAI;AAAA,MAC1C;AAAA,IACA;AAEI,SAAK,IAAI,KAAK,CAAC,KAAK,MAAM,GAAG,EAAE,MAAM,YAAW,CAAE,CAAC;AAAA,EACpD;AACH;"}

package/dist/lib/utils/fun-ASR/ws-connecter.es.js ADDED Viewed

@@ -0,0 +1,98 @@
+function WebSocketConnectMethod(config) {
+  let speechSokt;
+  const msgHandle = config.msgHandle;
+  const stateHandle = config.stateHandle;
+  const propsOnError = config.onError;
+  this.getSocket = function() {
+    return speechSokt;
+  };
+  this.wsStart = function() {
+    const Uri = config.uri;
+    if (Uri.match(/wss:\S*|ws:\S*/)) {
+      console.log("Uri" + Uri);
+    } else {
+      alert("请检查wss地址正确性");
+      return 0;
+    }
+    if (speechSokt && speechSokt.readyState === WebSocket.OPEN) {
+      return 1;
+    }
+    if ("WebSocket" in window) {
+      try {
+        speechSokt = new WebSocket(Uri);
+        speechSokt.onopen = function(e) {
+          onOpen(e);
+        };
+        speechSokt.onclose = function(e) {
+          console.log("onclose ws!");
+          onClose(e);
+        };
+        speechSokt.onmessage = function(e) {
+          onMessage(e);
+        };
+        speechSokt.onerror = function(e) {
+          onError(e);
+        };
+      } catch (error) {
+        propsOnError(error.message);
+      }
+      return 1;
+    } else {
+      alert("当前浏览器不支持 WebSocket");
+      return 0;
+    }
+  };
+  this.wsStop = function() {
+    if (speechSokt != void 0) {
+      console.log("stop ws!");
+      speechSokt.close();
+    }
+  };
+  this.wsSend = function(oneData) {
+    if (speechSokt == void 0) return;
+    if (speechSokt.readyState === 1) {
+      return speechSokt.send(oneData);
+    }
+  };
+  function onOpen(e) {
+    const chunk_size = [5, 10, 5];
+    const request = {
+      chunk_size,
+      wav_name: "h5",
+      is_speaking: true,
+      chunk_interval: 10,
+      itn: config.itn || true,
+      mode: config.mode || "2pass"
+    };
+    if (config.isFileMode) {
+      request.wav_format = file_ext;
+      if (file_ext == "wav") {
+        request.wav_format = "PCM";
+        request.audio_fs = file_sample_rate;
+      }
+    }
+    const hotwords = config.hotWords ? JSON.stringify(config.hotWords) : "";
+    if (hotwords != null) {
+      request.hotwords = hotwords;
+    }
+    console.log(JSON.stringify(request));
+    speechSokt.send(JSON.stringify(request));
+    console.log("连接成功");
+    stateHandle(0);
+  }
+  function onClose(e) {
+    stateHandle(1);
+  }
+  function onMessage(e) {
+    msgHandle(e);
+  }
+  function onError(e) {
+    console.error("socket error:", e);
+    stateHandle(2);
+    propsOnError("websocket connect error");
+  }
+}
+export {
+  WebSocketConnectMethod
+};
+//# sourceMappingURL=ws-connecter.es.js.map

package/dist/lib/utils/fun-ASR/ws-connecter.es.js.map ADDED Viewed

@@ -0,0 +1 @@

+ {"version":3,"file":"ws-connecter.es.js","sources":["../../../../src/utils/fun-ASR/ws-connecter.js"],"sourcesContent":["/**\n * Copyright FunASR (https://github.com/alibaba-damo-academy/FunASR). All Rights\n * Reserved. MIT License (https://opensource.org/licenses/MIT)\n */\n/* 2021-2023 by zhaoming,mali aihealthx.com */\n\nexport function WebSocketConnectMethod(config) {\n //定义socket连接方法类\n\n let speechSokt;\n let connKeeperID;\n\n const msgHandle = config.msgHandle;\n const stateHandle = config.stateHandle;\n const propsOnError = config.onError;\n\n this.getSocket = function () {\n return speechSokt;\n };\n\n this.wsStart = function () {\n const Uri = config.uri; //\"wss://111.205.137.58:5821/wss/\" //设置wss asr online接口地址如 wss://X.X.X.X:port/wss/\n if (Uri.match(/wss:\\S*|ws:\\S*/)) {\n console.log('Uri' + Uri);\n } else {\n alert('请检查wss地址正确性');\n return 0;\n }\n\n if (speechSokt && speechSokt.readyState === WebSocket.OPEN) {\n return 1;\n }\n\n if ('WebSocket' in window) {\n try {\n speechSokt = new WebSocket(Uri); // 定义socket连接对象\n speechSokt.onopen = function (e) {\n onOpen(e);\n }; // 定义响应函数\n speechSokt.onclose = function (e) {\n console.log('onclose ws!');\n onClose(e);\n };\n speechSokt.onmessage = function (e) {\n onMessage(e);\n };\n speechSokt.onerror = function (e) {\n onError(e);\n };\n } catch (error) {\n propsOnError(error.message);\n }\n return 1;\n } else {\n alert('当前浏览器不支持 WebSocket');\n return 0;\n }\n };\n\n // 定义停止与发送函数\n this.wsStop = function () {\n if (speechSokt != undefined) {\n console.log('stop ws!');\n speechSokt.close();\n }\n };\n\n this.wsSend = function (oneData) {\n if (speechSokt == undefined) return;\n if (speechSokt.readyState === 1) {\n // 0:CONNECTING, 1:OPEN, 2:CLOSING, 3:CLOSED\n\n return speechSokt.send(oneData);\n }\n };\n\n // SOCEKT连接中的消息与状态响应\n function onOpen(e) {\n // 发送json\n const chunk_size = [5, 10, 5];\n const request = {\n chunk_size: chunk_size,\n wav_name: 'h5',\n is_speaking: true,\n chunk_interval: 10,\n itn: config.itn || true,\n mode: config.mode || '2pass',\n };\n if (config.isFileMode) {\n request.wav_format = file_ext;\n if (file_ext == 'wav') {\n request.wav_format = 'PCM';\n request.audio_fs = file_sample_rate;\n }\n }\n\n const hotwords = config.hotWords ? JSON.stringify(config.hotWords) : '';\n\n if (hotwords != null) {\n request.hotwords = hotwords;\n }\n console.log(JSON.stringify(request));\n speechSokt.send(JSON.stringify(request));\n console.log('连接成功');\n stateHandle(0);\n }\n\n function onClose(e) {\n stateHandle(1);\n }\n\n function onMessage(e) {\n msgHandle(e);\n }\n\n function onError(e) {\n console.error('socket error:', e);\n stateHandle(2);\n propsOnError('websocket connect error');\n }\n}\n"],"names":[],"mappings":"AAMO,SAAS,uBAAuB,QAAQ;AAG7C,MAAI;AAGJ,QAAM,YAAY,OAAO;AACzB,QAAM,cAAc,OAAO;AAC3B,QAAM,eAAe,OAAO;AAE5B,OAAK,YAAY,WAAY;AAC3B,WAAO;AAAA,EACR;AAED,OAAK,UAAU,WAAY;AACzB,UAAM,MAAM,OAAO;AACnB,QAAI,IAAI,MAAM,gBAAgB,GAAG;AAC/B,cAAQ,IAAI,QAAQ,GAAG;AAAA,IAC7B,OAAW;AACL,YAAM,aAAa;AACnB,aAAO;AAAA,IACb;AAEI,QAAI,cAAc,WAAW,eAAe,UAAU,MAAM;AAC1D,aAAO;AAAA,IACb;AAEI,QAAI,eAAe,QAAQ;AACzB,UAAI;AACF,qBAAa,IAAI,UAAU,GAAG;AAC9B,mBAAW,SAAS,SAAU,GAAG;AAC/B,iBAAO,CAAC;AAAA,QAClB;AACQ,mBAAW,UAAU,SAAU,GAAG;AAChC,kBAAQ,IAAI,aAAa;AACzB,kBAAQ,CAAC;AAAA,QACV;AACD,mBAAW,YAAY,SAAU,GAAG;AAClC,oBAAU,CAAC;AAAA,QACZ;AACD,mBAAW,UAAU,SAAU,GAAG;AAChC,kBAAQ,CAAC;AAAA,QACV;AAAA,MACF,SAAQ,OAAO;AACd,qBAAa,MAAM,OAAO;AAAA,MAClC;AACM,aAAO;AAAA,IACb,OAAW;AACL,YAAM,oBAAoB;AAC1B,aAAO;AAAA,IACb;AAAA,EACG;AAGD,OAAK,SAAS,WAAY;AACxB,QAAI,cAAc,QAAW;AAC3B,cAAQ,IAAI,UAAU;AACtB,iBAAW,MAAO;AAAA,IACxB;AAAA,EACG;AAED,OAAK,SAAS,SAAU,SAAS;AAC/B,QAAI,cAAc,OAAW;AAC7B,QAAI,WAAW,eAAe,GAAG;AAG/B,aAAO,WAAW,KAAK,OAAO;AAAA,IACpC;AAAA,EACG;AAGD,WAAS,OAAO,GAAG;AAEjB,UAAM,aAAa,CAAC,GAAG,IAAI,CAAC;AAC5B,UAAM,UAAU;AAAA,MACd;AAAA,MACA,UAAU;AAAA,MACV,aAAa;AAAA,MACb,gBAAgB;AAAA,MAChB,KAAK,OAAO,OAAO;AAAA,MACnB,MAAM,OAAO,QAAQ;AAAA,IACtB;AACD,QAAI,OAAO,YAAY;AACrB,cAAQ,aAAa;AACrB,UAAI,YAAY,OAAO;AACrB,gBAAQ,aAAa;AACrB,gBAAQ,WAAW;AAAA,MAC3B;AAAA,IACA;AAEI,UAAM,WAAW,OAAO,WAAW,KAAK,UAAU,OAAO,QAAQ,IAAI;AAErE,QAAI,YAAY,MAAM;AACpB,cAAQ,WAAW;AAAA,IACzB;AACI,YAAQ,IAAI,KAAK,UAAU,OAAO,CAAC;AACnC,eAAW,KAAK,KAAK,UAAU,OAAO,CAAC;AACvC,YAAQ,IAAI,MAAM;AAClB,gBAAY,CAAC;AAAA,EACjB;AAEE,WAAS,QAAQ,GAAG;AAClB,gBAAY,CAAC;AAAA,EACjB;AAEE,WAAS,UAAU,GAAG;AACpB,cAAU,CAAC;AAAA,EACf;AAEE,WAAS,QAAQ,GAAG;AAClB,YAAQ,MAAM,iBAAiB,CAAC;AAChC,gBAAY,CAAC;AACb,iBAAa,yBAAyB;AAAA,EAC1C;AACA;"}

package/dist/lib/utils/vosk-browser-manager/audio-manager/audio-manager.es.js ADDED Viewed

@@ -0,0 +1,92 @@
+var __defProp = Object.defineProperty;
+var __defNormalProp = (obj, key, value) => key in obj ? __defProp(obj, key, { enumerable: true, configurable: true, writable: true, value }) : obj[key] = value;
+var __publicField = (obj, key, value) => __defNormalProp(obj, typeof key !== "symbol" ? key + "" : key, value);
+import { requestMicrophonePermission } from "../../xun-fei-voice/utils.es.js";
+import defaultWorkletProcessorStr from "./workletProcessor.es.js";
+const defaultSampleRate = 48e3;
+const generateWorkletUrl = (jsStr) => {
+  const blob = new Blob([jsStr], { type: "application/javascript" });
+  const scriptUrl = URL.createObjectURL(blob);
+  return scriptUrl;
+};
+class AudioManager {
+  constructor(config) {
+    __publicField(this, "workletUrl");
+    __publicField(this, "sampleRate", defaultSampleRate);
+    __publicField(this, "audioContext");
+    __publicField(this, "mediaSource");
+    __publicField(this, "mediaStream");
+    __publicField(this, "config", {});
+    const processorStr = config.workletProcessor || defaultWorkletProcessorStr;
+    this.workletUrl = generateWorkletUrl(processorStr);
+    this.sampleRate = config.sampleRate;
+    this.config = config;
+  }
+  async init() {
+    if (this.audioContext) {
+      await this.destroy();
+    }
+    const audioContext = new window.AudioContext({
+      sampleRate: this.sampleRate
+    });
+    if (!this.workletUrl) {
+      throw Error("加载 workletProcessor 失败！");
+    }
+    await audioContext.audioWorklet.addModule(this.workletUrl);
+    const audioWorkletNode = new AudioWorkletNode(audioContext, "audio-worklet-processor");
+    const mediaStream = await requestMicrophonePermission();
+    const mediaSource = audioContext.createMediaStreamSource(mediaStream);
+    audioWorkletNode.port.postMessage({
+      type: "init",
+      frameSize: this.config.frameSize
+    });
+    audioWorkletNode.port.onmessage = (e) => {
+      var _a, _b;
+      (_b = (_a = this.config).onFrameRecorded) == null ? void 0 : _b.call(_a, e.data);
+    };
+    mediaSource.connect(audioWorkletNode);
+    this.audioContext = audioContext;
+    this.mediaSource = mediaSource;
+    this.mediaStream = mediaStream;
+  }
+  async start() {
+    var _a, _b, _c, _d, _e;
+    try {
+      if (((_a = this.audioContext) == null ? void 0 : _a.state) === "suspended") {
+        this.audioContext.resume();
+      }
+      if (((_b = this.audioContext) == null ? void 0 : _b.state) === "closed" || !this.audioContext) {
+        await this.init();
+        await ((_c = this.audioContext) == null ? void 0 : _c.resume());
+      }
+      (_e = (_d = this.config).onStart) == null ? void 0 : _e.call(_d);
+    } catch (error) {
+      console.error("Failed to start recording:", error);
+      throw error;
+    }
+  }
+  async stop() {
+    var _a, _b, _c;
+    if (((_a = this.audioContext) == null ? void 0 : _a.state) === "running") {
+      await this.audioContext.suspend();
+    }
+    (_c = (_b = this.config).onStop) == null ? void 0 : _c.call(_b);
+  }
+  close() {
+    var _a;
+    (_a = this.mediaSource) == null ? void 0 : _a.disconnect();
+    if (this.audioContext && this.audioContext.state !== "closed") {
+      return this.audioContext.close();
+    }
+  }
+  async destroy() {
+    var _a, _b;
+    await this.close();
+    (_b = (_a = this.mediaStream) == null ? void 0 : _a.getTracks()) == null ? void 0 : _b.forEach((track) => track.stop());
+  }
+}
+export {
+  AudioManager,
+  defaultSampleRate
+};
+//# sourceMappingURL=audio-manager.es.js.map

package/dist/lib/utils/vosk-browser-manager/audio-manager/audio-manager.es.js.map ADDED Viewed

@@ -0,0 +1 @@

+ {"version":3,"file":"audio-manager.es.js","sources":["../../../../../src/utils/vosk-browser-manager/audio-manager/audio-manager.ts"],"sourcesContent":["import { requestMicrophonePermission } from '../../xun-fei-voice/utils';\nimport { SampleRate } from './types';\nimport defaultWorkletProcessorStr from './workletProcessor.js?raw';\n\nexport const defaultSampleRate = 48000;\n\nexport interface AudiManagerConfig {\n /** 一个转换成 string js 文件。其内容要符合 AudioWorkletProcessor 的要求。\n * - 需使用 registerProcessor 注册，第一个参数为 “audio-worklet-processor”，如： registerProcessor('audio-worklet-processor', audioToFloat32Processor);\n * - 参考：https://developer.mozilla.org/en-US/docs/Web/API/AudioWorkletProcessor\n */\n workletProcessor?: string;\n /** 采样率（每秒采样数）的浮点数，常用值：16000, 32000, 44100, 48000 */\n sampleRate?: SampleRate;\n onFrameRecorded?: (data: Float32Array) => void;\n onStart?: () => void;\n onStop?: () => void;\n frameSize?: number;\n}\n\nconst generateWorkletUrl = (jsStr: string) => {\n const blob = 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package/dist/lib/utils/vosk-browser-manager/audio-manager/workletProcessor.es.js ADDED Viewed

@@ -0,0 +1,5 @@
+const defaultWorkletProcessorStr = "/** 录音数据处理器，把 录音数据转换成 Float32 的数据 */\nclass audioToFloat32Processor extends AudioWorkletProcessor {\n  /** 每一帧的数据大小 */\n  frameSize = 4096;\n  float32Data = new Float32Array(this.frameSize);\n  _pos = 0;\n\n  constructor() {\n    super();\n    this.port.onMessage = (e) => {\n      console.log('%c  🐱 cat  ==== set worklet', 'color: orange; font-size: 16px;');\n      if (e?.type === 'init' && typeof e?.frameSize === 'number') {\n        this.frameSize = e.frameSize;\n        this.float32Data = new Float32Array(this.frameSize);\n        this._pos = 0;\n      }\n    };\n  }\n\n  process(inputs) {\n    // 获取第一个输入通道的音频数据\n    const input = inputs[0][0];\n    if (input && input.length > 0) {\n      for (let i = 0; i < input.length; i++) {\n        this.float32Data[this._pos++] = input[i];\n        if (this._pos === this.frameSize) {\n          this.port.postMessage(this.float32Data);\n          this._pos = 0;\n        }\n      }\n    }\n    return true;\n  }\n}\n\nregisterProcessor('audio-worklet-processor', audioToFloat32Processor);\n";
+export {
+  defaultWorkletProcessorStr as default
+};
+//# sourceMappingURL=workletProcessor.es.js.map

package/dist/lib/utils/vosk-browser-manager/audio-manager/workletProcessor.es.js.map ADDED Viewed

@@ -0,0 +1 @@

+ {"version":3,"file":"workletProcessor.es.js","sources":["../../../../../src/utils/vosk-browser-manager/audio-manager/workletProcessor.js?raw"],"sourcesContent":["export default \"/** 录音数据处理器，把录音数据转换成 Float32 的数据 */\\nclass audioToFloat32Processor extends AudioWorkletProcessor {\\n /** 每一帧的数据大小 */\\n frameSize = 4096;\\n float32Data = new Float32Array(this.frameSize);\\n _pos = 0;\\n\\n constructor() {\\n super();\\n this.port.onMessage = (e) => {\\n console.log('%c 🐱 cat ==== set worklet', 'color: orange; font-size: 16px;');\\n if (e?.type === 'init' && typeof e?.frameSize === 'number') {\\n this.frameSize = e.frameSize;\\n this.float32Data = new Float32Array(this.frameSize);\\n this._pos = 0;\\n }\\n };\\n }\\n\\n process(inputs) {\\n // 获取第一个输入通道的音频数据\\n const input = inputs[0][0];\\n if (input && input.length > 0) {\\n for (let i = 0; i < input.length; i++) {\\n this.float32Data[this._pos++] = input[i];\\n if (this._pos === this.frameSize) {\\n this.port.postMessage(this.float32Data);\\n this._pos = 0;\\n }\\n }\\n }\\n return true;\\n }\\n}\\n\\nregisterProcessor('audio-worklet-processor', audioToFloat32Processor);\\n\""],"names":[],"mappings":"AAAA,MAAe,6BAAA;"}

package/dist/lib/utils/vosk-browser-manager/vosk-browser-manager.es.js ADDED Viewed

@@ -0,0 +1,116 @@
+var __defProp = Object.defineProperty;
+var __defNormalProp = (obj, key, value) => key in obj ? __defProp(obj, key, { enumerable: true, configurable: true, writable: true, value }) : obj[key] = value;
+var __publicField = (obj, key, value) => __defNormalProp(obj, typeof key !== "symbol" ? key + "" : key, value);
+import { waitSomeThingBeTrue, wait } from "../wait-something/index.es.js";
+import { defaultSampleRate, AudioManager } from "./audio-manager/audio-manager.es.js";
+import omit from "../../node_modules/.pnpm/lodash-es@4.17.21/node_modules/lodash-es/omit.es.js";
+const pickAudioManagerConfig = (config) => {
+  return omit(config, [
+    "vosk",
+    "modelUrl",
+    "autoStart",
+    "onResult",
+    "onPartialResult",
+    "onInitialized"
+  ]);
+};
+class VoskBrowserManager {
+  constructor(config) {
+    __publicField(this, "config");
+    /** 采样率（每秒采样数）的浮点数，常用值：16000, 32000, 44100, 48000 */
+    __publicField(this, "sampleRate", defaultSampleRate);
+    __publicField(this, "audioManager");
+    __publicField(this, "model");
+    __publicField(this, "autoStart", false);
+    __publicField(this, "startResolve");
+    __publicField(this, "status");
+    __publicField(this, "recognizer");
+    this.config = config;
+    this.sampleRate = config.sampleRate || defaultSampleRate;
+    this.autoStart = config.autoStart ?? false;
+    this.init(config.vosk);
+  }
+  setConfig(config) {
+    this.config = {
+      ...this.config,
+      ...config
+    };
+  }
+  async init(vosk) {
+    var _a, _b, _c;
+    if (this.model) {
+      this.model.terminate();
+    }
+    this.status = "initializing";
+    console.time("load model:");
+    this.model = await vosk.createModel(this.config.modelUrl);
+    console.timeEnd("load model:");
+    const hotWords = this.config.hotWords ? JSON.stringify(this.config.hotWords) : void 0;
+    const _recognizer = new this.model.KaldiRecognizer(this.sampleRate, hotWords);
+    _recognizer.on("result", (message) => {
+      var _a2, _b2;
+      const resultText = message.result.text;
+      (_b2 = (_a2 = this.config).onResult) == null ? void 0 : _b2.call(_a2, resultText);
+    });
+    _recognizer.on("partialresult", (message) => {
+      var _a2, _b2;
+      const partialText = message.result.partial;
+      (_b2 = (_a2 = this.config).onPartialResult) == null ? void 0 : _b2.call(_a2, partialText);
+    });
+    this.recognizer = _recognizer;
+    if (this.audioManager) {
+      this.audioManager.destroy();
+    }
+    this.audioManager = new AudioManager({
+      ...pickAudioManagerConfig(this.config),
+      sampleRate: this.sampleRate,
+      onFrameRecorded: (data) => {
+        var _a2, _b2;
+        _recognizer.acceptWaveformFloat(data, this.sampleRate);
+        (_b2 = (_a2 = this.config).onFrameRecorded) == null ? void 0 : _b2.call(_a2, data);
+      }
+    });
+    (_b = (_a = this.config).onInitialized) == null ? void 0 : _b.call(_a);
+    (_c = this.startResolve) == null ? void 0 : _c.call(this);
+    this.status = "initialized";
+    if (this.autoStart) {
+      this.start();
+    }
+  }
+  /** 设置热词会重启识别器
+   * - 唤醒词识别阈值非常大，容易误唤醒，慎用！！！
+   */
+  async setHotWords(hotWords) {
+    this.config.hotWords = hotWords;
+    if (this.status !== "initialized") {
+      await waitSomeThingBeTrue(() => this.status === "initialized", 300, 5e3);
+    }
+    this.destroy();
+    await wait(300);
+    await this.init(this.config.vosk);
+  }
+  async start() {
+    var _a, _b;
+    if (this.status === "initialized") {
+      await ((_a = this.audioManager) == null ? void 0 : _a.start());
+    } else {
+      await new Promise((resolve) => {
+        this.startResolve = resolve;
+      });
+      await ((_b = this.audioManager) == null ? void 0 : _b.start());
+    }
+  }
+  stop() {
+    var _a;
+    (_a = this.audioManager) == null ? void 0 : _a.stop();
+  }
+  destroy() {
+    var _a, _b;
+    (_a = this.audioManager) == null ? void 0 : _a.destroy();
+    (_b = this.model) == null ? void 0 : _b.terminate();
+  }
+}
+export {
+  VoskBrowserManager
+};
+//# sourceMappingURL=vosk-browser-manager.es.js.map