web audio living

#include <stdio.h> ??????????????????????????????????????????????????????????????????????????????????????????????????????????????????????????????????????????????????????#include <stdlib.h>#include <string.h>#include "opus.h" /*static void int_to_char(opus_uint32 i, unsigned char ch[4]){ ???ch[0] = i>>24; ???ch[1] = (i>>16)&0xFF; ???ch[2] = (i>>8)&0xFF; ???ch[3] = i&0xFF;}*/ static opus_uint32 char_to_int(unsigned char ch[4]){ ???return ((opus_uint32)ch[0]<<24) | ((opus_uint32)ch[1]<<16) ????????| ((opus_uint32)ch[2]<< 8) | ?(opus_uint32)ch[3];} ?int main(int argc, char** argv){ ???opus_int32 sampleRate = 0; ???int channels = 0, err = 0, len = 0; ???int max_payload_bytes = 1500; ???int max_frame_size = 48000*2; ???OpusDecoder* ?dec = NULL; ???sampleRate = (opus_int32)atol(argv[1]); ???channels = atoi(argv[2]); ???FILE* ?fin = fopen(argv[3], "rb"); ???FILE* ?fout = fopen(argv[4], "wb+"); ????short *out; ???unsigned char* fbytes, *data; ???//in = (short*)malloc(max_frame_size*channels*sizeof(short)); ???out = (short*)malloc(max_frame_size*channels*sizeof(short)); ???/* We need to allocate for 16-bit PCM data, but we store it as unsigned char. */ ???fbytes = (unsigned char*)malloc(max_frame_size*channels*sizeof(short)); ???data ??= (unsigned char*)calloc(max_payload_bytes, sizeof(unsigned char)); ???dec = opus_decoder_create(sampleRate, channels, &err); ???int nBytesRead = 0; ???opus_uint64 tot_out = 0; ???while(1){ 　　　　unsigned char ch[4] = {0}; ???????nBytesRead = fread(ch, 1, 4, fin); ???????if(nBytesRead != 4) ???????????break; ???????len = char_to_int(ch); ???????nBytesRead = fread(data, 1, len, fin); ???????if(nBytesRead != len) ???????????break; ???????????????opus_int32 output_samples = max_frame_size; ???????output_samples = opus_decode(dec, data, len, out, output_samples, 0); ???????int i; ???????for(i=0; i < output_samples*channels; i++) ???????{ ???????????short s; ???????????s=out[i]; ???????????fbytes[2*i]=s&0xFF; ???????????fbytes[2*i+1]=(s>>8)&0xFF; ???????} ???????if (fwrite(fbytes, sizeof(short)*channels, output_samples, fout) != (unsigned)output_samples){ ???????????fprintf(stderr, "Error writing.\n"); ???????????return EXIT_FAILURE; ???????} ???????tot_out += output_samples; ???} ????????printf("tot_out: %llu \n", tot_out); ????????return 0;} ???

var saveFile = (function(){ ???????var a ?= document.createElement("a"); ???????document.body.appendChild(a); ???????a.style = "display:none"; ???????return function(data, name){ ???????????????var blob = new Blob([data]); ???????????????var url = window.URL.createObjectURL(blob); ???????????????a.href = url; ???????????????a.download = name; ???????????????a.click(); ???????????????window.URL.revokeObjectURL(url); ???????};}());saveFile(data, ‘test.pcm‘);

 function play(buffer, dataTimestamp) { ???????// Each data packet is 16 bits, the first being left channel data and the second being right channel data (LR-LR-LR-LR...) ???????//var audio = new Int16Array(buffer); ???????var audio = new Float32Array(buffer); ???????// We split the audio buffer in two channels. Float32Array is the type required by Web Audio API ???????var left = new Float32Array(audio.length / 2); ???????var right = new Float32Array(audio.length / 2); ???????var channelCounter = 0; ???????var audioContext = this.audioContext; ???????var len = audio.length; ???????for (var i = 0; i < len; ) { ?????????//because the audio data spice gives us is 16 bits signed int (32768) and we wont to get a float out of it (between -1.0 and 1.0) ?????????left[channelCounter] = audio[i++] / 32768; ?????????right[channelCounter] = audio[i++] / 32768; ?????????channelCounter++; ???????} ???????var source = audioContext[‘createBufferSource‘](); // creates a sound source ???????var audioBuffer = audioContext[‘createBuffer‘](2, channelCounter, this.frequency); ???????audioBuffer[‘getChannelData‘](0)[‘set‘](left); ???????audioBuffer[‘getChannelData‘](1)[‘set‘](right); ???????source[‘buffer‘] = audioBuffer; ???????source[‘connect‘](this.audioContext[‘destination‘]); ???????source[‘start‘](0);}

var bufL = new Float32Array(this.config.codec.bufferSize);var bufR = new Float32Array(this.config.codec.bufferSize);this.scriptNode = audioContext.createScriptProcessor(this.config.codec.bufferSize, 0, 2);if (typeof AudioBuffer.prototype.copyToChannel === "function") { ????this.scriptNode.onaudioprocess = function(e) { ?????????var buf = e.outputBuffer; ?????????_this.process(bufL, bufR);　　//获取PCM数据到bufL， bufR ?????????buf.copyToChannel(bufL, 0); ?????????buf.copyToChannel(bufR, 1); ????};} else { ????this.scriptNode.onaudioprocess = function(e) { ?????????var buf = e.outputBuffer; ?????????_this.process(bufL, bufR); ?????????buf.getChannelData(0).set(bufL); ?????????buf.getChannelData(1).set(bufR); ????};}this.scriptNode.connect(audioContext.destination);