使用WebRTC来设置立体声以改变音频的通道

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本文介绍了如何使用WebRTC和JavaScript来设置立体声,改变音频的通道。通过获取用户媒体设备,创建MediaStream对象,然后处理左右声道,实现了在Web浏览器中自定义音频通道,提供更丰富的音频体验。

WebRTC(Web Real-Time Communication)是一种支持实时音视频通信的开放标准。它使得在Web浏览器中实现音频和视频通信变得更加便捷。在WebRTC中,我们可以通过设置立体声来改变音频的通道。

下面我们将通过源代码示例来介绍如何使用WebRTC来设置立体声以改变音频的通道。请注意,以下代码是使用JavaScript编写的。

首先,我们需要在HTML文件中添加一个用于渲染音频的元素。这可以是一个<audio>标签,例如:

<audio id="audioPlayer" controls>
Record sounds / noises around you and turn them into music. It’s a work in progress, at the moment it enables you to record live audio straight from your browser, edit it and save these sounds as a WAV file. There's also a sequencer part where you can create small loops using these sounds with a drone synth overlaid on them. See it working: http://daaain.github.com/JSSoundRecorder Technology ---------- No servers involved, only Web Audio API with binary sound Blobs passed around! ### Web Audio API #### GetUserMedia audio for live recording Experimental API to record any system audio input (including USB soundcards, musical instruments, etc). ```javascript // shim and create AudioContext window.AudioContext = window.AudioContext || window.webkitAudioContext || window.mozAudioContext; var audio_context = new AudioContext(); // shim and start GetUserMedia audio stream navigator.getUserMedia = navigator.getUserMedia || navigator.webkitGetUserMedia || navigator.mozGetUserMedia; navigator.getUserMedia({audio: true}, startUserMedia, function(e) { console.log('No live audio input: ' + e); }); ``` #### Audio nodes for routing You can route audio stream around, with input nodes (microphone, synths, etc), filters (volume / gain, equaliser, low pass, etc) and outputs (speakers, binary streams, etc). ```javascript function startUserMedia(stream) { // create MediaStreamSource and GainNode var input = audio_context.createMediaStreamSource(stream); var volume = audio_context.createGain(); volume.gain.value = 0.7; // connect them and pipe output input.connect(volume); volume.connect(audio_context.destination); // connect recorder as well - see below var recorder = new Recorder(input); } ``` ### WebWorker Processing (interleaving) record buffer is done in the background to not block the main thread and the UI. Also WAV conversion for export is also quite heavy for longer recordings, so best left to run in the background. ```javascript this.context = input.context; this.node = this.context.createScriptProcessor(4096, 2, 2); this.node.onaudioprocess = function(e){ worker.postMessage({ command: 'record', buffer: [ e.inputBuffer.getChannelData(0), e.inputBuffer.getChannelData(1) ] }); } ``` ```javascript function record(inputBuffer){ var bufferL = inputBuffer[0]; var bufferR = inputBuffer[1]; var interleaved = interleave(bufferL, bufferR); recBuffers.push(interleaved); recLength += interleaved.length; } function interleave(inputL, inputR){ var length = inputL.length + inputR.length; var result = new Float32Array(length); var index = 0, inputIndex = 0; while (index < length){ result[index++] = inputL[inputIndex]; result[index++] = inputR[inputIndex]; inputIndex++; } return result; } ``` ```javascript function encodeWAV(samples){ var buffer = new ArrayBuffer(44 + samples.length * 2); var view = new DataView(buffer); /* RIFF identifier */ writeString(view, 0, 'RIFF'); /* file length */ view.setUint32(4, 32 + samples.length * 2, true); /* RIFF type */ writeString(view, 8, 'WAVE'); /* format chunk identifier */ writeString(view, 12, 'fmt '); /* format chunk length */ view.setUint32(16, 16, true); /* sample format (raw) */ view.setUint16(20, 1, true); /* channel count */ view.setUint16(22, 2, true); /* sample rate */ view.setUint32(24, sampleRate, true); /* byte rate (sample rate * block align) */ view.setUint32(28, sampleRate * 4, true); /* block align (channel count * bytes per sample) */ view.setUint16(32, 4, true); /* bits per sample */ view.setUint16(34, 16, true); /* data chunk identifier */ writeString(view, 36, 'data'); /* data chunk length */ view.setUint32(40, samples.length * 2, true); floatTo16BitPCM(view, 44, samples); return view; } ``` ### Binary Blob Instead of file drag and drop interface this binary blob is passed to editor. Note: BlobBuilder deprecated (but a lot of examples use it), you should use Blob constructor instead! ```javascript var f = new FileReader(); f. { audio_context.decodeAudioData(e.target.result, function(buffer) { $('#audioLayerControl')[0].handleAudio(buffer); }, function(e) { console.warn(e); }); }; f.readAsArrayBuffer(blob); ``` ```javascript function exportWAV(type){ var buffer = mergeBuffers(recBuffers, recLength); var dataview = encodeWAV(buffer); var audioBlob = new Blob([dataview], { type: type }); this.postMessage(audioBlob); } ``` ### Virtual File – URL.createObjectURL You can create file download link pointing to WAV blob, but also set it as the source of an Audio element. ```javascript var url = URL.createObjectURL(blob); var audioElement = document.createElement('audio'); var downloadAnchor = document.createElement('a'); audioElement.controls = true; audioElement.src = url; downloadAnchor.href = url; ``` TODO ---- * Sequencer top / status row should be radio buttons :) * Code cleanup / restructuring * Enable open / drag and drop files for editing * Visual feedback (levels) for live recording * Sequencer UI (and separation to a different module) Credits / license ----------------- Live recording code adapted from: http://www.phpied.com/files/webaudio/record.html Editor code adapted from: https://github.com/plucked/html5-audio-editor Copyright (c) 2012 Daniel Demmel MIT License
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