connection_socket, buf, bytesRead, 0); // 关闭socket close(connection_socket); 2、AudioQueue播放音频 AudioQueue的播放时,需要先给audioBuffer 填充数据,并把audioBuffer放入AudioQueue,然后通知AudioQueue开始播放; AudioQueue从已经填充的audioBuffer里面开始播放数据,实时把播放完毕的audioBuffer 回调给业务层,业务继续填充播放完毕的audioBuffer,重复流程直到音频播放完毕。 是播完结束的回调 AudioQueueNewOutput(&asbd, MyAudioQueueOutputCallback, myData, NULL, NULL, 0, &audioQueue); // AudioBuffer
的获取 流程我们了解到了,如何取到 AudioBuffer 呢? 一般的做法是通过 FileReader 来读取到 File 类型,转为 ArrayBuffer 类型,最后再转 AudioBuffer 类型。有没有更方便的做法呢?当然有了!白嫖啊! 我们打印引擎封装好的 cc.AudioClip 资源,可以清晰的看到 _audio 就是我们想要的 AudioBuffer 类型,完美! ? 数据处理 好了,万事俱备,就差如何分析音频数据了。 // 也就是将 AudioBuffer 数据传递进去。 // 以下就是创建音频资源节点管理者。 this.audioBufferSourceNode = audioContext.createBufferSource(); // 将 AudioBuffer 传递进去。
include <QFile> class MyAudioInput { private: QAudioOutput* audio; // QAudioOutput对象 QBuffer audioBuffer sourceFile.setFileName("/tmp/test.raw"); sourceFile.open(QIODevice::ReadOnly); audioBuffer.setData (sourceFile.readAll()); // 读取文件内容到缓冲区 audioBuffer.open(QIODevice::ReadOnly); // 打开缓冲区 audio->start(&audioBuffer); // 开始播放 } void stop() { audio->stop(); audioBuffer.close
; mCbf(EVENT_MORE_DATA, mUserData, &audioBuffer); // 利用有数据了 size_t readSize = audioBuffer.size status_t err = obtainBuffer(&audioBuffer, blocking ? = TRANSFER_OBTAIN) { audioBuffer->frameCount = 0; audioBuffer->size = 0; audioBuffer ->frameCount = buffer.mFrameCount; audioBuffer->size = buffer.mFrameCount * mFrameSize; audioBuffer ->raw = buffer.mRaw; audioBuffer->sequence = oldSequence; if (nonContig !
, size=%zu (%zd)", buffer, userSize, userSize); return BAD_VALUE; } size_t written = 0; Buffer audioBuffer ; while (userSize >= mFrameSize) { audioBuffer.frameCount = userSize / mFrameSize; status_t err = obtainBuffer(&audioBuffer, blocking ? ; memcpy(audioBuffer.i8, buffer, toWrite); mBuffer = malloc(toWrite); memcpy(mBuffer size_t toWrite = audioBuffer.size; memcpy(audioBuffer.i8, buffer, toWrite); mBuffer
是回调的结构体; AudioBufferList是音频的缓存数据结构,具体如下: struct AudioBufferList { UInt32 mNumberBuffers; AudioBuffer mBuffers[1]; // this is a variable length array of mNumberBuffers elements }; struct AudioBuffer { mNumberChannels; UInt32 mDataByteSize; void* __nullable mData; }; mNumberBuffers: AudioBuffer 的数量 mBuffers:AudioBuffer的指针数组,数组长度等于mNumberBuffers AudioBuffer:mNumberChannels是声道数,mDataByteSize是buffer
window.AudioContext = window.AudioContext || window.webkitAudioContext; var context = new window.AudioContext(); 创建audioBuffer var audioBuffers = [] var audioBuffer = context.createBuffer(channels, frameCount, samplerate); 播放音频 for (var channel = 0; channel < channels; channel++) { var nowBuffering = audioBuffer.getChannelData } } : function(fromBuffer) { var nowBuffering = audioBuffer.getChannelData var source = context.createBufferSource(); source.buffer = audioBuffer
mAudioRecord.startRecording(); 4、读取采集到的声音数据 /* * 不断的读取采集到的声音数据,放进编码器的输入缓存inputBuffers中进行编码 * audioBuffer 存储写入音频录制数据的缓冲区 * sizeInBytes 请求的最大字节数 * public int read (ByteBuffer audioBuffer, int sizeInBytes
window.mozAudioContext || window.msAudioContext; try { var context = new contextClass(); var source = null; var audioBuffer 立即停止 } } function playSound() { source = context.createBufferSource(); source.buffer = audioBuffer initSound(arrayBuffer) { context.decodeAudioData(arrayBuffer, function(buffer) { //解码成功时的回调函数 audioBuffer
需要一个AudioBufferSourceNode对象,它代表一个音频源,通过AudioContext的createBufferSourceAPI来创建,它还需要一个AudioBuffer类型的音频资源 把音频放入AudioBuffer后,可以传入到一个 AudioBufferSourceNode进行播放。 来源 MDN 简单来说就是把我们网上的或者本地的又或者通过设备麦克风等获取的一段音频转换成AudioBuffer然后“喂”给AudioBufferSourceNode对象,这样我们这个音频源就可以播放啦
]; CFRelease(videoBuffer); } CMSampleBufferRef audioBuffer = [self.demuxer copyNextAudioSampleBuffer]; if (audioBuffer) { [self.muxer appendSampleBuffer:audioBuffer]; CFRelease(audioBuffer); } }
status_t err = obtainBuffer(&audioBuffer, blocking ? ; memcpy(audioBuffer.i8, buffer, toWrite); buffer = ((const char *) buffer) + toWrite ; userSize -= toWrite; written += toWrite; releaseBuffer(&audioBuffer); ->frameCount = buffer.mFrameCount; audioBuffer->size = buffer.mFrameCount * mFrameSize; audioBuffer ->raw = buffer.mRaw; audioBuffer->sequence = oldSequence; if (nonContig !
audioContext 连接的是 AudioBuffer 和音频播放器。中间可以和一些 Node 连接,这些 Node 是用来对流进行处理的。 analyserNode: 频谱分析 offlineContext : 快速音视频处理 下面是直接解码一个 Buffer 为音频处理格式流: context.decodeAudioData(buffer,audioBuffer =>{ // audioBuffer 对象 }); 这意味着,我们可以直接拿到音频流里面的具体数据,而剩下的就是我们的想象空间了。
解码为AudioBuffer(解压缩) const audioBuffer = awaitthis.audioCtx.decodeAudioData(arrayBuffer); 创建音源节点 this.source = this.audioCtx.createBufferSource(); this.source.buffer = audioBuffer; bgm.mp3'); // 游戏逻辑里调用 player.enterUnderwater(); // 角色跳入水中 流程图 [网络] ─fetch→ [ArrayBuffer] ─decode→ [AudioBuffer this.listener.upX.value = 0; this.listener.upY.value = 1; this.listener.upZ.value = 0; } createSpatialSound(audioBuffer , x, y, z) { const source = this.audioCtx.createBufferSource(); source.buffer = audioBuffer;
new AudioContext(); var source = context.createBufferSource() context.decodeAudioData(arrayBuffer, (audioBuffer ) => { source.buffer = audioBuffer var gain = context.createGain() gain.gain.value = 0.6 source.connect (gain) source.start(0) gain.connect(context.destination) }) 首先需要一个音频源,在这里我们使用 audioBuffer通过 Ajax或者 Fetch拿到 ArrayBuffer,然后 decodeAudioData 获得 audioBuffer,这就是 input,然后添加声音音量(Effect),最后连接到 destination,默认就是扬声器
我找到了免费的ASR接口,通过getUserMedia获取麦克风输入,通过MediaRecorder获得audioBuffer,通过AudioContext和audiobuffer-to-wav进行转码
样式采用的是预处理sass,感兴趣的可以去看一下代码 2.3 创建音频 /** * 创建音频 * @param AudioBuffer buffer AudioBuffer对象 * @return ac.state === 'closed') { ac = new (window.AudioContext || window.webkitAudioContext)(); } audioBuffer gainNode.disconnect(ac.destination); bufferSource = ac.createBufferSource(); bufferSource.buffer = audioBuffer
response.arrayBuffer()) .then((arrayBuffer) => audioContext.decodeAudioData(arrayBuffer)) .then((audioBuffer ) => { const source = audioContext.createBufferSource(); source.buffer = audioBuffer; source.connect
方式二:createBufferSource AudioContext.createBufferSource() 创建一个 AudioBufferSourceNode 对象, 他可以通过 AudioBuffer AudioBuffer可以用AudioContext 接口的 decodeAudioData() 方法异步解码音频文件中的 ArrayBuffer。 addFlvHeader (chunk) { let audioBuffer = null if (this.flvHeader == null) { // copy first 9 bytes (flv header) this.flvHeader = chunk.slice(0, 9) audioBuffer = chunk } else { audioBuffer = this.appendBuffer(this.flvHeader, chunk) } return audioBuffer } 五.
PacketTable AudioFilePacketTableInfo 读写 struct AudioBufferList { UInt32 mNumberBuffers; AudioBuffer operator=(const AudioBufferList&); #endif }; typedef struct AudioBufferList AudioBufferList; struct AudioBuffer mNumberChannels; UInt32 mDataByteSize; void* __nullable mData; }; typedef struct AudioBuffer AudioBuffer; 写入文件内容 写入和读取类似,只是要预先填好BufferList的内容: OSStatus ExtAudioFileWrite ( ExtAudioFileRef inExtAudioFile