IOS播放PCM数据

关于PCM播放器源码学习是本文要介绍的内容，由于原始PCM文件不能容纳任何关于其自身频率或者帧大小之类的信息，本例将不得不对此自行设置。

我们会使用一种为经过压缩的PCM数据格式，具体参数喂16位、44KHz、单声道。这些信息通过程序顶部的三个预定义值指定：

1	#define BYTES_PER_SAMPLE 2

16位等于两个字节

1	#define SAMPLE_PATE  44100

每秒采样率44100次等于44KHz

#import<UIKit/UIKit.h>

#import"AppDelegate.h"

#include<AudioToolbox/AudioQueue.h>

#include"AKLib.h"

#define BYTES_PER_SAMPLE2

#define SAMPLE_PATE 44100

typedefshort sampleFrame;

#define FRAME_COUNT735

#define AUDIO_BUFFERS3

typedefstruct AQCallbackStruct{

   AudioQueueRef queue;//播放队列

    UInt32 FrameCount;

   AudioQueueBufferRef mBuffers[AUDIO_BUFFERS];

   AudioStreamBasicDescription mDataFormat;

    UInt32 sampleLen;

    UInt32 playPtr;

    sampleFrame *pcmBuffer;

}AQCallbackStruct;

int playbuffer(void *pcm,unsignedlong len);

void AQBufferCallback(void *in,AudioQueueRef inQ,AudioQueueBufferRef outQB);

int main(int argc,char *argv[])

{

    unsigned char  *pcmBuffer;

    

   unsigned char source[11] ={0x01,'9','0','0','6','0','2','3','4','1','6'};

    printf("要编码数据的长度 %ld\n",sizeof(source));

    int outLen;

    pcmBuffer = CreatePacket(source,sizeof(source),&outLen);

//    FILE *stream;

//    if ( (stream = fopen("/Users/weiwen/Desktop/encode.pcm", "wb+")) == NULL) {

//        fprintf(stderr,"Cannot open output file.\n");

//        return 1;

//    }

//    fwrite(pcmBuffer, 1, outLen, stream);

//    fclose(stream);

   printf("编码之后的数据长度 %d\n",outLen);

    int i;

    for (i=0; i<1000; i++) {

        printf("%d\n",i);

        playbuffer(pcmBuffer, outLen);

    }

   @autoreleasepool {

       return UIApplicationMain(argc, argv, nil,NSStringFromClass([AppDelegateclass]));

    }

}

int playbuffer(void *pcmBuffer,unsignedlong len)

{

   AQCallbackStruct agc;

    UInt32 err,bufferSize;

    int i;

    agc.mDataFormat.mSampleRate =SAMPLE_PATE;

    agc.mDataFormat.mFormatID = kAudioFormatLinearPCM;

    agc.mDataFormat.mFormatFlags = kLinearPCMFormatFlagIsSignedInteger | kAudioFormatFlagIsPacked;

    agc.mDataFormat.mBytesPerPacket =4;

    agc.mDataFormat.mFramesPerPacket =1;

    agc.mDataFormat.mBytesPerFrame =4;

    agc.mDataFormat.mChannelsPerFrame =2;

    agc.mDataFormat.mBitsPerChannel =16;

    agc.FrameCount =FRAME_COUNT;

    agc.sampleLen = len/BYTES_PER_SAMPLE;

    agc.playPtr =0;

    agc.pcmBuffer = (short *)pcmBuffer;

    err = AudioQueueNewOutput(&agc.mDataFormat,AQBufferCallback,&agc,NULL,

                             kCFRunLoopCommonModes,0,&agc.queue);

    if(err) return err;

    agc.FrameCount =FRAME_COUNT;

    bufferSize = agc.FrameCount * agc.mDataFormat.mBytesPerFrame;

    for (i=0; i<AUDIO_BUFFERS; i++)

    {

        err = AudioQueueAllocateBuffer(agc.queue,bufferSize,&agc.mBuffers[i]);

        if(err) return err;

        AQBufferCallback(&agc,agc.queue,agc.mBuffers[i]);

    }

    err = AudioQueueStart(agc.queue,NULL);

    if(err) return err;

    int v=0;

    while (agc.playPtr<agc.sampleLen)

    {

        struct timeval tv={1,0};

        select(v,NULL,NULL,NULL,&tv);

    }

   //AudioQueueReset(agc.queue);

   AudioQueueStop(agc.queue,true);

    sleep(1);

    return 0;

}

void AQBufferCallback(void *pIn, AudioQueueRef inQ,AudioQueueBufferRef outQB)

{

   AQCallbackStruct *agc;

    short *audioBuf=NULL;

    short sample;

    int i;

    agc=(AQCallbackStruct *)pIn;

    audioBuf=(short*)(outQB->mAudioData);

    

   //printf("Sync:%lu / %lu \n",agc->playPtr,agc->sampleLen);

    if (agc->FrameCount >0)

    {

        outQB->mAudioDataByteSize =4*agc->FrameCount;

        for (i=0; i<agc->FrameCount*2; i++)

        {

            if(agc->playPtr > agc->sampleLen)

            {

                sample =0;

            }

            else

            {

                sample = (agc->pcmBuffer[agc->playPtr]);

            }

            audioBuf[i] = sample;

            audioBuf[i+1] = sample;

            agc->playPtr++;

        }

       AudioQueueEnqueueBuffer(inQ,outQB,0,NULL);

    }

}