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都说iOS最恶心的部分是流媒体，其中恶心的恶心之处更在即时语音。
所以我们先不谈即时语音，研究一下，iOS中声音采集与播放的实现。
要在iOS设备上实现录音和播放功能，苹果提供了简单的做法，那就是利用AVAudioRecorder和AVAudioPlayer。度娘大多数 也是如此。但是这种方法有很大的局限性。单说说这种做法：录音，首先得设置录音文件路径，然后录音数据直接写入了文件。播放也是首先给出文件路径，等到音 频整个加载完成了，才能开始播放。这相当不灵活。
我的做法是利用音频队列AudioQueue，将声音暂存至缓冲区，然后从缓冲区取出音频数据，进行播放。
声音采集：
使用AudioQueue框架以队列的形式处理音频数据。因此使用时需要给队列分配缓存空间，由回调（Callback）函数完成向队列缓存读写音频数据的功能。
一个Recording Audio Queue，包括Buffer(缓冲器)组成的Buffer Queue(缓冲队列)，以及一个Callback(回调)。实现主要步骤为：
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Record.h#import <Foundation/Foundation.h>#import <AudioToolbox/AudioToolbox.h>#import <CoreAudio/CoreAudioTypes.h>#import "AudioConstant.h"// use Audio Queuetypedef struct AQCallbackStruct{ AudioStreamBasicDescription mDataFormat; AudioQueueRef queue; AudioQueueBufferRef mBuffers[kNumberBuffers]; AudioFileID outputFile; unsigned long frameSize; long long recPtr; int run; } AQCallbackStruct;@interface Record : NSObject{ AQCallbackStruct aqc; AudioFileTypeID fileFormat; long audioDataLength; Byte audioByte[999999]; long audioDataIndex;}- (id) init;- (void) start;- (void) stop;- (void) pause;- (Byte *) getBytes;- (void) processAudioBuffer:(AudioQueueBufferRef) buffer withQueue:(AudioQueueRef) queue;@property (nonatomic, assign) AQCallbackStruct aqc;@property (nonatomic, assign) long audioDataLength;@end |
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Record.mm#import "Record.h"@implementation Record@synthesize aqc;@synthesize audioDataLength;static void AQInputCallback (void * inUserData, AudioQueueRef inAudioQueue, AudioQueueBufferRef inBuffer, const AudioTimeStamp * inStartTime, unsigned long inNumPackets, const AudioStreamPacketDescription * inPacketDesc){ Record * engine = (__bridge Record *) inUserData; if (inNumPackets > 0) { [engine processAudioBuffer:inBuffer withQueue:inAudioQueue]; } if (engine.aqc.run) { AudioQueueEnqueueBuffer(engine.aqc.queue, inBuffer, 0, NULL); }}- (id) init{ self = [super init]; if (self) { aqc.mDataFormat.mSampleRate = kSamplingRate; aqc.mDataFormat.mFormatID = kAudioFormatLinearPCM; aqc.mDataFormat.mFormatFlags = kLinearPCMFormatFlagIsSignedInteger |kLinearPCMFormatFlagIsPacked; aqc.mDataFormat.mFramesPerPacket = 1; aqc.mDataFormat.mChannelsPerFrame = kNumberChannels; aqc.mDataFormat.mBitsPerChannel = kBitsPerChannels; aqc.mDataFormat.mBytesPerPacket = kBytesPerFrame; aqc.mDataFormat.mBytesPerFrame = kBytesPerFrame; aqc.frameSize = kFrameSize; AudioQueueNewInput(&aqc.mDataFormat, AQInputCallback, (__bridge void *)(self), NULL, kCFRunLoopCommonModes,0, &aqc.queue); for (int i=0;i<kNumberBuffers;i++) { AudioQueueAllocateBuffer(aqc.queue, aqc.frameSize, &aqc.mBuffers[i]); AudioQueueEnqueueBuffer(aqc.queue, aqc.mBuffers[i], 0, NULL); } aqc.recPtr = 0; aqc.run = 1; } audioDataIndex = 0; return self;}- (void) dealloc{ AudioQueueStop(aqc.queue, true); aqc.run = 0; AudioQueueDispose(aqc.queue, true);}- (void) start{ AudioQueueStart(aqc.queue, NULL);}- (void) stop{ AudioQueueStop(aqc.queue, true);}- (void) pause{ AudioQueuePause(aqc.queue);}- (Byte *)getBytes{ return audioByte;}- (void) processAudioBuffer:(AudioQueueBufferRef) buffer withQueue:(AudioQueueRef) queue{ NSLog(@"processAudioData :%ld", buffer->mAudioDataByteSize); //处理data：忘记oc怎么copy内存了，于是采用的C++代码，记得把类后缀改为.mm。同Play memcpy(audioByte+audioDataIndex, buffer->mAudioData, buffer->mAudioDataByteSize); audioDataIndex +=buffer->mAudioDataByteSize; audioDataLength = audioDataIndex;}@end |
声音播放：
同采集一样，播放主要步骤如下：
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Play.h#import <Foundation/Foundation.h>#import <AudioToolbox/AudioToolbox.h>#import "AudioConstant.h"@interface Play : NSObject{ //音频参数 AudioStreamBasicDescription audioDescription; // 音频播放队列 AudioQueueRef audioQueue; // 音频缓存 AudioQueueBufferRef audioQueueBuffers[QUEUE_BUFFER_SIZE];}-(void)Play:(Byte *)audioByte Length:(long)len;@end |
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Play.mm#import "Play.h"@interface Play(){ Byte *audioByte; long audioDataIndex; long audioDataLength;}@end@implementation Play//回调函数(Callback)的实现static void BufferCallback(void *inUserData,AudioQueueRef inAQ,AudioQueueBufferRef buffer){ NSLog(@"processAudioData :%u", (unsigned int)buffer->mAudioDataByteSize); Play* player=(__bridge Play*)inUserData; [player FillBuffer:inAQ queueBuffer:buffer];}//缓存数据读取方法的实现-(void)FillBuffer:(AudioQueueRef)queue queueBuffer:(AudioQueueBufferRef)buffer{ if(audioDataIndex + EVERY_READ_LENGTH < audioDataLength) { memcpy(buffer->mAudioData, audioByte+audioDataIndex, EVERY_READ_LENGTH); audioDataIndex += EVERY_READ_LENGTH; buffer->mAudioDataByteSize =EVERY_READ_LENGTH; AudioQueueEnqueueBuffer(queue, buffer, 0, NULL); } }-(void)SetAudioFormat{ ///设置音频参数 audioDescription.mSampleRate = kSamplingRate;//采样率 audioDescription.mFormatID = kAudioFormatLinearPCM; audioDescription.mFormatFlags = kAudioFormatFlagIsSignedInteger;//|kAudioFormatFlagIsNonInterleaved; audioDescription.mChannelsPerFrame = kNumberChannels; audioDescription.mFramesPerPacket = 1;//每一个packet一侦数据 audioDescription.mBitsPerChannel = kBitsPerChannels;//av_get_bytes_per_sample(AV_SAMPLE_FMT_S16)*8;//每个采样点16bit量化 audioDescription.mBytesPerFrame = kBytesPerFrame; audioDescription.mBytesPerPacket = kBytesPerFrame; [self CreateAudioQueue];}-(void)CreateAudioQueue{ [self Cleanup]; //使用player的内部线程播 AudioQueueNewOutput(&audioDescription, BufferCallback, (__bridge void *)(self), nil, nil, 0, &audioQueue); if(audioQueue) { ////添加buffer区 for(int i=0;i<QUEUE_BUFFER_SIZE;i++) { int result = AudioQueueAllocateBuffer(audioQueue, EVERY_READ_LENGTH, &audioQueueBuffers[i]); ///创建buffer区，MIN_SIZE_PER_FRAME为每一侦所需要的最小的大小，该大小应该比每次往buffer里写的最大的一次还大 NSLog(@"AudioQueueAllocateBuffer i = %d,result = %d",i,result); } }}-(void)Cleanup{ if(audioQueue) { NSLog(@"Release AudioQueueNewOutput"); [self Stop]; for(int i=0; i < QUEUE_BUFFER_SIZE; i++) { AudioQueueFreeBuffer(audioQueue, audioQueueBuffers[i]); audioQueueBuffers[i] = nil; } audioQueue = nil; }}-(void)Stop{ NSLog(@"Audio Player Stop"); AudioQueueFlush(audioQueue); AudioQueueReset(audioQueue); AudioQueueStop(audioQueue,TRUE);}-(void)Play:(Byte *)byte Length:(long)len{ [self Stop]; audioByte = byte; audioDataLength = len; NSLog(@"Audio Play Start >>>>>"); [self SetAudioFormat]; AudioQueueReset(audioQueue); audioDataIndex = 0; for(int i=0; i<QUEUE_BUFFER_SIZE; i++) { [self FillBuffer:audioQueue queueBuffer:audioQueueBuffers[i]]; } AudioQueueStart(audioQueue, NULL);}@end |
以上，实现了通过内存缓存，声音的采集和播放，包括了声音采集，暂停，结束，播放等主要流程。
PS:由于本人水品有限加之这方面资料较少，只跑通了正常流程，暂时没做异常处理。采集的声音Buffer限定大小每次只有十来秒钟的样子，这个留给需要的人自己去优化了。
AudioToolbox－－利用AudioQueue音频队列，通过缓存对声音进行采集与播放
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原文地址：http://www.cnblogs.com/sunminmin/p/4476596.html
