ios - 从 iphone 流媒体

Question

我需要将音频从麦克风流式传输到 http 服务器。
这些录音设置是我需要的:

NSDictionary *audioOutputSettings = [NSDictionary dictionaryWithObjectsAndKeys:
                                             [NSNumber numberWithInt: kAudioFormatULaw],AVFormatIDKey,        
                                             [NSNumber numberWithFloat:8000.0],AVSampleRateKey,//was 44100.0
                                             [NSData dataWithBytes: &acl length: sizeof( AudioChannelLayout ) ], AVChannelLayoutKey,
                                             [NSNumber numberWithInt:1],AVNumberOfChannelsKey,
                                             [NSNumber numberWithInt:64000],AVEncoderBitRateKey,
                                             nil];

API 即时编码状态:

Send a continuous stream of audio to the currently viewed camera. Audio needs to be encoded at G711 mu-law at 64 kbit/s for transfer to the Axis camera at the bedside. send (this should be a POST URL in SSL to connected server): POST /transmitaudio?id= Content-type: audio/basic Content-Length: 99999 (length is ignored)

以下是我尝试使用的链接列表。

LINK -(SO)基本解释，只有音频单元和音频队列在通过麦克风录制时允许 nsdata 作为输出 | 不是示例，而是对所需内容(音频队列或音频单元)的良好定义

LINK - (SO) 音频回调示例 | 只包含回调

LINK - (SO)REMOTE IO 示例 | 没有启动/停止功能，用于保存到文件

LINK - (SO)REMOTE IO 示例 | 未答复无效

LINK - (SO)基本录音示例 | 很好的例子，但要记录在案

LINK - (SO)引导我进入 InMemoryAudioFile 类的问题(无法开始工作)| 访问了 inMemoryFile(或类似文件)的链接，但无法正常工作。

LINK - (SO) 更多音频单元和远程 io 示例/问题 | 让这个工作正常，但再次没有停止功能，即使我试图弄清楚调用是什么并让它停止，它似乎仍然没有将音频传输到服务器。

LINK - 不错的remoteIO 和音频队列示例但是| 另一个很好的例子，它几乎可以正常工作，但是代码有一些问题(编译器认为它不是 obj-c++)并且再次不知道如何从它而不是文件中获取音频“数据”。

LINK - 音频队列的 Apple 文档 | 框架有问题。完成了它(请参阅下面的问题)但最终无法使其正常工作，但可能没有像其他人那样给这个时间，也许应该有。

LINK - 我在尝试实现音频队列/单元时遇到的(SO)问题 | 不是例子

LINK - (SO) 另一个 remoteIO 例子 | 另一个很好的例子，但无法弄清楚如何将其获取到数据而不是文件。

LINK - 看起来也很有趣，循环缓冲区 | 无法弄清楚如何将其与音频回调相结合

这是我目前正在尝试直播的类(class)。这似乎有效，尽管接收器端(连接到服务器)的扬声器发出静电。这似乎表明音频数据格式有问题。

IOS 版本(减去 GCD 套接字的委托(delegate)方法):

@implementation MicCommunicator {
AVAssetWriter * assetWriter;
AVAssetWriterInput * assetWriterInput;
}

@synthesize captureSession = _captureSession;
@synthesize output = _output;
@synthesize restClient = _restClient;
@synthesize uploadAudio = _uploadAudio;
@synthesize outputPath = _outputPath;
@synthesize sendStream = _sendStream;
@synthesize receiveStream = _receiveStream;

@synthesize socket = _socket;
@synthesize isSocketConnected = _isSocketConnected;

-(id)init {
    if ((self = [super init])) {

        _receiveStream = [[NSStream alloc]init];
        _sendStream = [[NSStream alloc]init];
        _socket = [[GCDAsyncSocket alloc] initWithDelegate:self delegateQueue:dispatch_get_main_queue()];
        _isSocketConnected = FALSE;

        _restClient = [RestClient sharedManager];
        _uploadAudio = false;

        NSArray *searchPaths = NSSearchPathForDirectoriesInDomains(NSDocumentDirectory, NSUserDomainMask, YES);
        _outputPath = [NSURL fileURLWithPath:[[searchPaths objectAtIndex:0] stringByAppendingPathComponent:@"micOutput.output"]];

        NSError * assetError;

        AudioChannelLayout acl;
        bzero(&acl, sizeof(acl));
        acl.mChannelLayoutTag = kAudioChannelLayoutTag_Mono; //kAudioChannelLayoutTag_Stereo;
        NSDictionary *audioOutputSettings = [NSDictionary dictionaryWithObjectsAndKeys:
                                             [NSNumber numberWithInt: kAudioFormatULaw],AVFormatIDKey,        
                                             [NSNumber numberWithFloat:8000.0],AVSampleRateKey,//was 44100.0
                                             [NSData dataWithBytes: &acl length: sizeof( AudioChannelLayout ) ], AVChannelLayoutKey,
                                             [NSNumber numberWithInt:1],AVNumberOfChannelsKey,
                                             [NSNumber numberWithInt:64000],AVEncoderBitRateKey,
                                             nil];

        assetWriterInput = [[AVAssetWriterInput assetWriterInputWithMediaType:AVMediaTypeAudio outputSettings:audioOutputSettings]retain];
        [assetWriterInput setExpectsMediaDataInRealTime:YES];

        assetWriter = [[AVAssetWriter assetWriterWithURL:_outputPath fileType:AVFileTypeWAVE error:&assetError]retain]; //AVFileTypeAppleM4A

        if (assetError) {
            NSLog (@"error initing mic: %@", assetError);
            return nil;
        }
        if ([assetWriter canAddInput:assetWriterInput]) {
            [assetWriter addInput:assetWriterInput];
        } else {
            NSLog (@"can't add asset writer input...!");
            return nil;
        }

    }
    return self;
}

-(void)dealloc {
    [_output release];
    [_captureSession release];
    [_captureSession release];
    [assetWriter release];
    [assetWriterInput release];
    [super dealloc];
}


-(void)beginStreaming {

    NSLog(@"avassetwrter class is %@",NSStringFromClass([assetWriter class]));

    self.captureSession = [[AVCaptureSession alloc] init];
    AVCaptureDevice *audioCaptureDevice = [AVCaptureDevice defaultDeviceWithMediaType:AVMediaTypeAudio];
    NSError *error = nil;
    AVCaptureDeviceInput *audioInput = [AVCaptureDeviceInput deviceInputWithDevice:audioCaptureDevice error:&error];
    if (audioInput)
        [self.captureSession addInput:audioInput];
    else {
        NSLog(@"No audio input found.");
        return;
    }

    self.output = [[AVCaptureAudioDataOutput alloc] init];

    dispatch_queue_t outputQueue = dispatch_queue_create("micOutputDispatchQueue", NULL);
    [self.output setSampleBufferDelegate:self queue:outputQueue];
    dispatch_release(outputQueue);

    self.uploadAudio = FALSE;

    [self.captureSession addOutput:self.output];
    [assetWriter startWriting];
    [self.captureSession startRunning];
}

-(void)pauseStreaming
{
    self.uploadAudio = FALSE;
}

-(void)resumeStreaming
{
    self.uploadAudio = TRUE;
}

-(void)finishAudioWork
{
    [self dealloc];
}

-(void)captureOutput:(AVCaptureOutput *)captureOutput didOutputSampleBuffer:(CMSampleBufferRef)sampleBuffer fromConnection:(AVCaptureConnection *)connection {


    AudioBufferList audioBufferList;
    NSMutableData *data= [[NSMutableData alloc] init];
    CMBlockBufferRef blockBuffer;
    CMSampleBufferGetAudioBufferListWithRetainedBlockBuffer(sampleBuffer, NULL, &audioBufferList, sizeof(audioBufferList), NULL, NULL, 0, &blockBuffer);

    for (int y = 0; y < audioBufferList.mNumberBuffers; y++) {
        AudioBuffer audioBuffer = audioBufferList.mBuffers[y];
        Float32 *frame = (Float32*)audioBuffer.mData;

        [data appendBytes:frame length:audioBuffer.mDataByteSize];
    }

    // append [data bytes] to your NSOutputStream 

    // These two lines write to disk, you may not need this, just providing an example
    [assetWriter startSessionAtSourceTime:CMSampleBufferGetPresentationTimeStamp(sampleBuffer)];
    [assetWriterInput appendSampleBuffer:sampleBuffer];

    //start upload audio data
    if (self.uploadAudio) { 

        if (!self.isSocketConnected) {
            [self connect];
        }
            NSString *requestStr = [NSString stringWithFormat:@"POST /transmitaudio?id=%@ HTTP/1.0\r\n\r\n",self.restClient.sessionId];

            NSData *requestData = [requestStr dataUsingEncoding:NSUTF8StringEncoding];        
        [self.socket writeData:requestData withTimeout:5 tag:0];     
        [self.socket writeData:data withTimeout:5 tag:0]; 
    }
    //stop upload audio data

    CFRelease(blockBuffer);
    blockBuffer=NULL;
    [data release];
}

和JAVA版本:

import java.io.BufferedInputStream;
import java.io.BufferedOutputStream;
import java.io.BufferedReader;
import java.io.DataInputStream;
import java.io.DataOutputStream;
import java.io.IOException;
import java.io.InputStream;
import java.io.InputStreamReader;
import java.io.OutputStream;
import java.io.PrintWriter;
import java.nio.ByteBuffer;
import java.nio.ByteOrder;
import java.util.Arrays;

import javax.net.ssl.SSLContext;
import javax.net.ssl.SSLSocket;
import javax.net.ssl.SSLSocketFactory;
import javax.net.ssl.TrustManager;
import javax.net.ssl.X509TrustManager;

import android.media.AudioFormat;
import android.media.AudioManager;
import android.media.AudioRecord;
import android.media.AudioTrack;
import android.media.MediaRecorder.AudioSource;
import android.util.Log;

public class AudioWorker extends Thread
{ 
    private boolean stopped = false;

    private String host;
    private int port;
    private long id=0;
    boolean run=true;
    AudioRecord recorder;

    //ulaw encoder stuff
    private final static String TAG = "UlawEncoderInputStream";

    private final static int MAX_ULAW = 8192;
    private final static int SCALE_BITS = 16;

    private InputStream mIn;

    private int mMax = 0;

    private final byte[] mBuf = new byte[1024];
    private int mBufCount = 0; // should be 0 or 1

    private final byte[] mOneByte = new byte[1];
    ////
    /**
     * Give the thread high priority so that it's not canceled unexpectedly, and start it
     */
    public AudioWorker(String host, int port, long id)
    { 
        this.host = host;
        this.port = port;
        this.id = id;
        android.os.Process.setThreadPriority(android.os.Process.THREAD_PRIORITY_URGENT_AUDIO);
//        start();
    }

    @Override
    public void run()
    { 
        Log.i("AudioWorker", "Running AudioWorker Thread");
        recorder = null;
        AudioTrack track = null;
        short[][]   buffers  = new short[256][160];
        int ix = 0;

        /*
         * Initialize buffer to hold continuously recorded AudioWorker data, start recording, and start
         * playback.
         */
        try
        {
            int N = AudioRecord.getMinBufferSize(8000,AudioFormat.CHANNEL_IN_MONO,AudioFormat.ENCODING_PCM_16BIT);
            recorder = new AudioRecord(AudioSource.MIC, 8000, AudioFormat.CHANNEL_IN_MONO, AudioFormat.ENCODING_PCM_16BIT, N*10);
            track = new AudioTrack(AudioManager.STREAM_MUSIC, 8000,   AudioFormat.CHANNEL_OUT_MONO, AudioFormat.ENCODING_PCM_16BIT, N*10, AudioTrack.MODE_STREAM);
            recorder.startRecording();
//            track.play();
            /*
             * Loops until something outside of this thread stops it.
             * Reads the data from the recorder and writes it to the AudioWorker track for playback.
             */


            SSLContext sc = SSLContext.getInstance("SSL");
            sc.init(null, trustAllCerts, new java.security.SecureRandom());
            SSLSocketFactory sslFact = sc.getSocketFactory();
            SSLSocket socket = (SSLSocket)sslFact.createSocket(host, port);

            socket.setSoTimeout(10000);
            InputStream inputStream = socket.getInputStream();
            DataInputStream in = new DataInputStream(new BufferedInputStream(inputStream));
            OutputStream outputStream = socket.getOutputStream();
            DataOutputStream os = new DataOutputStream(new BufferedOutputStream(outputStream));
            PrintWriter socketPrinter = new PrintWriter(os);
            BufferedReader br = new BufferedReader(new InputStreamReader(in));

//          socketPrinter.println("POST /transmitaudio?patient=1333369798370 HTTP/1.0");
            socketPrinter.println("POST /transmitaudio?id="+id+" HTTP/1.0");
            socketPrinter.println("Content-Type: audio/basic");
            socketPrinter.println("Content-Length: 99999");
            socketPrinter.println("Connection: Keep-Alive");
            socketPrinter.println("Cache-Control: no-cache");
            socketPrinter.println();
            socketPrinter.flush();


            while(!stopped)
            { 
                Log.i("Map", "Writing new data to buffer");
                short[] buffer = buffers[ix++ % buffers.length];

                N = recorder.read(buffer,0,buffer.length);
                track.write(buffer, 0, buffer.length);

                byte[] bytes2 = new byte[buffer.length * 2];
                ByteBuffer.wrap(bytes2).order(ByteOrder.LITTLE_ENDIAN).asShortBuffer().put(buffer);

                read(bytes2, 0, bytes2.length);
                os.write(bytes2,0,bytes2.length);

//
//                ByteBuffer byteBuf = ByteBuffer.allocate(2*N);
//              System.out.println("byteBuf length "+2*N);
//                int i = 0;
//                while (buffer.length > i) {
//                    byteBuf.putShort(buffer[i]);
//                    i++;
//                }         
//                byte[] b = new byte[byteBuf.remaining()];
            }
            os.close();
        }
        catch(Throwable x)
        { 
            Log.w("AudioWorker", "Error reading voice AudioWorker", x);
        }
        /*
         * Frees the thread's resources after the loop completes so that it can be run again
         */
        finally
        { 
            recorder.stop();
            recorder.release();
            track.stop();
            track.release();
        }
    }

    /**
     * Called from outside of the thread in order to stop the recording/playback loop
     */
    public void close()
    { 
         stopped = true;
    }
    public void resumeThread()
    { 
         stopped = false;
         run();
    }

    TrustManager[] trustAllCerts = new TrustManager[]{
            new X509TrustManager() {
                public java.security.cert.X509Certificate[] getAcceptedIssuers() {
                    return null;
                }
                public void checkClientTrusted(
                        java.security.cert.X509Certificate[] certs, String authType) {
                }
                public void checkServerTrusted(
                        java.security.cert.X509Certificate[] chain, String authType) {
                    for (int j=0; j<chain.length; j++)
                    {
                        System.out.println("Client certificate information:");
                        System.out.println("  Subject DN: " + chain[j].getSubjectDN());
                        System.out.println("  Issuer DN: " + chain[j].getIssuerDN());
                        System.out.println("  Serial number: " + chain[j].getSerialNumber());
                        System.out.println("");
                    }
                }
            }
    };


    public static void encode(byte[] pcmBuf, int pcmOffset,
            byte[] ulawBuf, int ulawOffset, int length, int max) {

        // from  'ulaw' in wikipedia
        // +8191 to +8159                          0x80
        // +8158 to +4063 in 16 intervals of 256   0x80 + interval number
        // +4062 to +2015 in 16 intervals of 128   0x90 + interval number
        // +2014 to  +991 in 16 intervals of  64   0xA0 + interval number
        //  +990 to  +479 in 16 intervals of  32   0xB0 + interval number
        //  +478 to  +223 in 16 intervals of  16   0xC0 + interval number
        //  +222 to   +95 in 16 intervals of   8   0xD0 + interval number
        //   +94 to   +31 in 16 intervals of   4   0xE0 + interval number
        //   +30 to    +1 in 15 intervals of   2   0xF0 + interval number
        //     0                                   0xFF

        //    -1                                   0x7F
        //   -31 to    -2 in 15 intervals of   2   0x70 + interval number
        //   -95 to   -32 in 16 intervals of   4   0x60 + interval number
        //  -223 to   -96 in 16 intervals of   8   0x50 + interval number
        //  -479 to  -224 in 16 intervals of  16   0x40 + interval number
        //  -991 to  -480 in 16 intervals of  32   0x30 + interval number
        // -2015 to  -992 in 16 intervals of  64   0x20 + interval number
        // -4063 to -2016 in 16 intervals of 128   0x10 + interval number
        // -8159 to -4064 in 16 intervals of 256   0x00 + interval number
        // -8192 to -8160                          0x00

        // set scale factors
        if (max <= 0) max = MAX_ULAW;

        int coef = MAX_ULAW * (1 << SCALE_BITS) / max;

        for (int i = 0; i < length; i++) {
            int pcm = (0xff & pcmBuf[pcmOffset++]) + (pcmBuf[pcmOffset++] << 8);
            pcm = (pcm * coef) >> SCALE_BITS;

            int ulaw;
            if (pcm >= 0) {
                ulaw = pcm <= 0 ? 0xff :
                        pcm <=   30 ? 0xf0 + ((  30 - pcm) >> 1) :
                        pcm <=   94 ? 0xe0 + ((  94 - pcm) >> 2) :
                        pcm <=  222 ? 0xd0 + (( 222 - pcm) >> 3) :
                        pcm <=  478 ? 0xc0 + (( 478 - pcm) >> 4) :
                        pcm <=  990 ? 0xb0 + (( 990 - pcm) >> 5) :
                        pcm <= 2014 ? 0xa0 + ((2014 - pcm) >> 6) :
                        pcm <= 4062 ? 0x90 + ((4062 - pcm) >> 7) :
                        pcm <= 8158 ? 0x80 + ((8158 - pcm) >> 8) :
                        0x80;
            } else {
                ulaw = -1 <= pcm ? 0x7f :
                          -31 <= pcm ? 0x70 + ((pcm -   -31) >> 1) :
                          -95 <= pcm ? 0x60 + ((pcm -   -95) >> 2) :
                         -223 <= pcm ? 0x50 + ((pcm -  -223) >> 3) :
                         -479 <= pcm ? 0x40 + ((pcm -  -479) >> 4) :
                         -991 <= pcm ? 0x30 + ((pcm -  -991) >> 5) :
                        -2015 <= pcm ? 0x20 + ((pcm - -2015) >> 6) :
                        -4063 <= pcm ? 0x10 + ((pcm - -4063) >> 7) :
                        -8159 <= pcm ? 0x00 + ((pcm - -8159) >> 8) :
                        0x00;
            }
            ulawBuf[ulawOffset++] = (byte)ulaw;
        }
    }
    public static int maxAbsPcm(byte[] pcmBuf, int offset, int length) {
        int max = 0;
        for (int i = 0; i < length; i++) {
            int pcm = (0xff & pcmBuf[offset++]) + (pcmBuf[offset++] << 8);
            if (pcm < 0) pcm = -pcm;
            if (pcm > max) max = pcm;
        }
        return max;
    }

    public int read(byte[] buf, int offset, int length) throws IOException {
        if (recorder == null) throw new IllegalStateException("not open");

        // return at least one byte, but try to fill 'length'
        while (mBufCount < 2) {
            int n = recorder.read(mBuf, mBufCount, Math.min(length * 2, mBuf.length - mBufCount));
            if (n == -1) return -1;
            mBufCount += n;
        }

        // compand data
        int n = Math.min(mBufCount / 2, length);
        encode(mBuf, 0, buf, offset, n, mMax);

        // move data to bottom of mBuf
        mBufCount -= n * 2;
        for (int i = 0; i < mBufCount; i++) mBuf[i] = mBuf[i + n * 2];

        return n;
    }

}

Answer 1

我在这个主题上的工作是惊人的和漫长的。我终于让这个工作了，但它可能被黑了。因此，我会在发布答案之前列出一些警告:

1. 缓冲区之间仍然有咔哒声

2. 由于我在 obj-c++ 类中使用我的 obj-c 类的方式，我收到警告，所以那里有问题(但是根据我的研究，使用池与发布一样，所以我不相信这很重要):

   Object 0x13cd20 of class __NSCFString autoreleased with no pool in place - just leaking - break on objc_autoreleaseNoPool() to debug

3. 为了使这项工作正常进行，我不得不从 SpeakHereController(见下文)中注释掉所有 AQPlayer 引用，因为我无法通过任何其他方式修复错误。不过这对我来说并不重要，因为我只是在录音

所以上面的主要答案是 AVAssetWriter 中有一个错误阻止它附加字节和写入音频数据。在联系苹果支持并让他们通知我后，我终于发现了这一点。据我所知，这个错误是 ulaw 和 AVAssetWriter 特有的，尽管我还没有尝试过许多其他格式来验证。
对此，唯一的其他选择是/曾经是使用 AudioQueues。我以前尝试过但带来了很多问题。最大的问题是我对 obj-c++ 缺乏了解。下面使事情正常运行的类来自 speakHere 示例，稍作更改，以便音频采用 ulaw 格式。其他问题来自试图让所有文件都能很好地播放。然而，这很容易通过将链中的所有文件名更改为 .mm 来解决。下一个问题是尝试和谐地使用这些类。这仍然是一个 WIP，并且与警告号 2 相关。但我对此的基本解决方案是使用 SpeakHereController(也包含在 speakhere 示例中)而不是直接访问 AQRecorder。

无论如何这里是代码:

使用 obj-c 类中的 SpeakHereController

.h

@property(nonatomic,strong) SpeakHereController * recorder;

.毫米

[init method]
        //AQRecorder wrapper (SpeakHereController) allocation
        _recorder = [[SpeakHereController alloc]init];
        //AQRecorder wrapper (SpeakHereController) initialization
        //technically this class is a controller and thats why its init method is awakeFromNib
        [_recorder awakeFromNib];

[recording]
     bool buttonState = self.audioRecord.isSelected;
[self.audioRecord setSelected:!buttonState];

if ([self.audioRecord isSelected]) {

    [self.recorder startRecord];
}else {
    [self.recorder stopRecord];
}

SpeakHere Controller

#import "SpeakHereController.h"

@implementation SpeakHereController

@synthesize player;
@synthesize recorder;

@synthesize btn_record;
@synthesize btn_play;
@synthesize fileDescription;
@synthesize lvlMeter_in;
@synthesize playbackWasInterrupted;

char *OSTypeToStr(char *buf, OSType t)
{
    char *p = buf;
    char str[4], *q = str;
    *(UInt32 *)str = CFSwapInt32(t);
    for (int i = 0; i < 4; ++i) {
        if (isprint(*q) && *q != '\\')
            *p++ = *q++;
        else {
            sprintf(p, "\\x%02x", *q++);
            p += 4;
        }
    }
    *p = '\0';
    return buf;
}

-(void)setFileDescriptionForFormat: (CAStreamBasicDescription)format withName:(NSString*)name
{
    char buf[5];
    const char *dataFormat = OSTypeToStr(buf, format.mFormatID);
    NSString* description = [[NSString alloc] initWithFormat:@"(%d ch. %s @ %g Hz)", format.NumberChannels(), dataFormat, format.mSampleRate, nil];
    fileDescription.text = description;
    [description release];  
}

#pragma mark Playback routines

-(void)stopPlayQueue
{
//  player->StopQueue();
    [lvlMeter_in setAq: nil];
    btn_record.enabled = YES;
}

-(void)pausePlayQueue
{
//  player->PauseQueue();
    playbackWasPaused = YES;
}


-(void)startRecord
{
    //    recorder = new AQRecorder();

    if (recorder->IsRunning()) // If we are currently recording, stop and save the file.
    {
        [self stopRecord];
    }
    else // If we're not recording, start.
    {
        //      btn_play.enabled = NO;  

        // Set the button's state to "stop"
        //      btn_record.title = @"Stop";

        // Start the recorder
        recorder->StartRecord(CFSTR("recordedFile.caf"));

        [self setFileDescriptionForFormat:recorder->DataFormat() withName:@"Recorded File"];

        // Hook the level meter up to the Audio Queue for the recorder
        //      [lvlMeter_in setAq: recorder->Queue()];
    } 
}

- (void)stopRecord
{
    // Disconnect our level meter from the audio queue
//  [lvlMeter_in setAq: nil];

    recorder->StopRecord();

    // dispose the previous playback queue
//  player->DisposeQueue(true);

    // now create a new queue for the recorded file
    recordFilePath = (CFStringRef)[NSTemporaryDirectory() stringByAppendingPathComponent: @"recordedFile.caf"];
//  player->CreateQueueForFile(recordFilePath);

    // Set the button's state back to "record"
//  btn_record.title = @"Record";
//  btn_play.enabled = YES;
}

- (IBAction)play:(id)sender
{
    if (player->IsRunning())
    {
        if (playbackWasPaused) {
//          OSStatus result = player->StartQueue(true);
//          if (result == noErr)
//              [[NSNotificationCenter defaultCenter] postNotificationName:@"playbackQueueResumed" object:self];
        }
        else
//          [self stopPlayQueue];
            nil;
    }
    else
    {       
//      OSStatus result = player->StartQueue(false);
//      if (result == noErr)
//          [[NSNotificationCenter defaultCenter] postNotificationName:@"playbackQueueResumed" object:self];
    }
}

- (IBAction)record:(id)sender
{
    if (recorder->IsRunning()) // If we are currently recording, stop and save the file.
    {
        [self stopRecord];
    }
    else // If we're not recording, start.
    {
//      btn_play.enabled = NO;  
//      
//      // Set the button's state to "stop"
//      btn_record.title = @"Stop";

        // Start the recorder
        recorder->StartRecord(CFSTR("recordedFile.caf"));

        [self setFileDescriptionForFormat:recorder->DataFormat() withName:@"Recorded File"];

        // Hook the level meter up to the Audio Queue for the recorder
        [lvlMeter_in setAq: recorder->Queue()];
    }   
}
#pragma mark AudioSession listeners
void interruptionListener(  void *  inClientData,
                            UInt32  inInterruptionState)
{
    SpeakHereController *THIS = (SpeakHereController*)inClientData;
    if (inInterruptionState == kAudioSessionBeginInterruption)
    {
        if (THIS->recorder->IsRunning()) {
            [THIS stopRecord];
        }
        else if (THIS->player->IsRunning()) {
            //the queue will stop itself on an interruption, we just need to update the UI
            [[NSNotificationCenter defaultCenter] postNotificationName:@"playbackQueueStopped" object:THIS];
            THIS->playbackWasInterrupted = YES;
        }
    }
    else if ((inInterruptionState == kAudioSessionEndInterruption) && THIS->playbackWasInterrupted)
    {
        // we were playing back when we were interrupted, so reset and resume now
//      THIS->player->StartQueue(true);
        [[NSNotificationCenter defaultCenter] postNotificationName:@"playbackQueueResumed" object:THIS];
        THIS->playbackWasInterrupted = NO;
    }
}

void propListener(  void *                  inClientData,
                    AudioSessionPropertyID  inID,
                    UInt32                  inDataSize,
                    const void *            inData)
{
    SpeakHereController *THIS = (SpeakHereController*)inClientData;
    if (inID == kAudioSessionProperty_AudioRouteChange)
    {
        CFDictionaryRef routeDictionary = (CFDictionaryRef)inData;          
        //CFShow(routeDictionary);
        CFNumberRef reason = (CFNumberRef)CFDictionaryGetValue(routeDictionary, CFSTR(kAudioSession_AudioRouteChangeKey_Reason));
        SInt32 reasonVal;
        CFNumberGetValue(reason, kCFNumberSInt32Type, &reasonVal);
        if (reasonVal != kAudioSessionRouteChangeReason_CategoryChange)
        {
            /*CFStringRef oldRoute = (CFStringRef)CFDictionaryGetValue(routeDictionary, CFSTR(kAudioSession_AudioRouteChangeKey_OldRoute));
            if (oldRoute)   
            {
                printf("old route:\n");
                CFShow(oldRoute);
            }
            else 
                printf("ERROR GETTING OLD AUDIO ROUTE!\n");

            CFStringRef newRoute;
            UInt32 size; size = sizeof(CFStringRef);
            OSStatus error = AudioSessionGetProperty(kAudioSessionProperty_AudioRoute, &size, &newRoute);
            if (error) printf("ERROR GETTING NEW AUDIO ROUTE! %d\n", error);
            else
            {
                printf("new route:\n");
                CFShow(newRoute);
            }*/

            if (reasonVal == kAudioSessionRouteChangeReason_OldDeviceUnavailable)
            {           
                if (THIS->player->IsRunning()) {
                    [THIS pausePlayQueue];
                    [[NSNotificationCenter defaultCenter] postNotificationName:@"playbackQueueStopped" object:THIS];
                }       
            }

            // stop the queue if we had a non-policy route change
            if (THIS->recorder->IsRunning()) {
                [THIS stopRecord];
            }
        }   
    }
    else if (inID == kAudioSessionProperty_AudioInputAvailable)
    {
        if (inDataSize == sizeof(UInt32)) {
            UInt32 isAvailable = *(UInt32*)inData;
            // disable recording if input is not available
            THIS->btn_record.enabled = (isAvailable > 0) ? YES : NO;
        }
    }
}

#pragma mark Initialization routines
- (void)awakeFromNib
{       
    // Allocate our singleton instance for the recorder & player object
    recorder = new AQRecorder();
    player = nil;//new AQPlayer();

    OSStatus error = AudioSessionInitialize(NULL, NULL, interruptionListener, self);
    if (error) printf("ERROR INITIALIZING AUDIO SESSION! %d\n", error);
    else 
    {
        UInt32 category = kAudioSessionCategory_PlayAndRecord;  
        error = AudioSessionSetProperty(kAudioSessionProperty_AudioCategory, sizeof(category), &category);
        if (error) printf("couldn't set audio category!");

        error = AudioSessionAddPropertyListener(kAudioSessionProperty_AudioRouteChange, propListener, self);
        if (error) printf("ERROR ADDING AUDIO SESSION PROP LISTENER! %d\n", error);
        UInt32 inputAvailable = 0;
        UInt32 size = sizeof(inputAvailable);

        // we do not want to allow recording if input is not available
        error = AudioSessionGetProperty(kAudioSessionProperty_AudioInputAvailable, &size, &inputAvailable);
        if (error) printf("ERROR GETTING INPUT AVAILABILITY! %d\n", error);
//      btn_record.enabled = (inputAvailable) ? YES : NO;

        // we also need to listen to see if input availability changes
        error = AudioSessionAddPropertyListener(kAudioSessionProperty_AudioInputAvailable, propListener, self);
        if (error) printf("ERROR ADDING AUDIO SESSION PROP LISTENER! %d\n", error);

        error = AudioSessionSetActive(true); 
        if (error) printf("AudioSessionSetActive (true) failed");
    }

//  [[NSNotificationCenter defaultCenter] addObserver:self selector:@selector(playbackQueueStopped:) name:@"playbackQueueStopped" object:nil];
//  [[NSNotificationCenter defaultCenter] addObserver:self selector:@selector(playbackQueueResumed:) name:@"playbackQueueResumed" object:nil];

//  UIColor *bgColor = [[UIColor alloc] initWithRed:.39 green:.44 blue:.57 alpha:.5];
//  [lvlMeter_in setBackgroundColor:bgColor];
//  [lvlMeter_in setBorderColor:bgColor];
//  [bgColor release];

    // disable the play button since we have no recording to play yet
//  btn_play.enabled = NO;
//  playbackWasInterrupted = NO;
//  playbackWasPaused = NO;
}

# pragma mark Notification routines
- (void)playbackQueueStopped:(NSNotification *)note
{
    btn_play.title = @"Play";
    [lvlMeter_in setAq: nil];
    btn_record.enabled = YES;
}

- (void)playbackQueueResumed:(NSNotification *)note
{
    btn_play.title = @"Stop";
    btn_record.enabled = NO;
    [lvlMeter_in setAq: player->Queue()];
}

#pragma mark Cleanup
- (void)dealloc
{
    [btn_record release];
    [btn_play release];
    [fileDescription release];
    [lvlMeter_in release];

//  delete player;
    delete recorder;

    [super dealloc];
}

@end

AQ记录仪(.h有2行重要

#define kNumberRecordBuffers    3
#define kBufferDurationSeconds 5.0

)

#include "AQRecorder.h"
//#include "UploadAudioWrapperInterface.h"
//#include "RestClient.h"

RestClient * restClient;
NSData* data;

// ____________________________________________________________________________________
// Determine the size, in bytes, of a buffer necessary to represent the supplied number
// of seconds of audio data.
int AQRecorder::ComputeRecordBufferSize(const AudioStreamBasicDescription *format, float seconds)
{
    int packets, frames, bytes = 0;
    try {
        frames = (int)ceil(seconds * format->mSampleRate);

        if (format->mBytesPerFrame > 0)
            bytes = frames * format->mBytesPerFrame;
        else {
            UInt32 maxPacketSize;
            if (format->mBytesPerPacket > 0)
                maxPacketSize = format->mBytesPerPacket;    // constant packet size
            else {
                UInt32 propertySize = sizeof(maxPacketSize);
                XThrowIfError(AudioQueueGetProperty(mQueue, kAudioQueueProperty_MaximumOutputPacketSize, &maxPacketSize,
                                                 &propertySize), "couldn't get queue's maximum output packet size");
            }
            if (format->mFramesPerPacket > 0)
                packets = frames / format->mFramesPerPacket;
            else
                packets = frames;   // worst-case scenario: 1 frame in a packet
            if (packets == 0)       // sanity check
                packets = 1;
            bytes = packets * maxPacketSize;
        }
    } catch (CAXException e) {
        char buf[256];
        fprintf(stderr, "Error: %s (%s)\n", e.mOperation, e.FormatError(buf));
        return 0;
    }   
    return bytes;
}

// ____________________________________________________________________________________
// AudioQueue callback function, called when an input buffers has been filled.
void AQRecorder::MyInputBufferHandler(  void *                              inUserData,
                                        AudioQueueRef                       inAQ,
                                        AudioQueueBufferRef                 inBuffer,
                                        const AudioTimeStamp *              inStartTime,
                                        UInt32                              inNumPackets,
                                        const AudioStreamPacketDescription* inPacketDesc)
{
    AQRecorder *aqr = (AQRecorder *)inUserData;


    try {
        if (inNumPackets > 0) {
            // write packets to file
//          XThrowIfError(AudioFileWritePackets(aqr->mRecordFile, FALSE, inBuffer->mAudioDataByteSize,
//                                           inPacketDesc, aqr->mRecordPacket, &inNumPackets, inBuffer->mAudioData),
//                     "AudioFileWritePackets failed");
            aqr->mRecordPacket += inNumPackets;



//            int numBytes = inBuffer->mAudioDataByteSize;       
//            SInt8 *testBuffer = (SInt8*)inBuffer->mAudioData;
//            
//            for (int i=0; i < numBytes; i++)
//            {
//                SInt8 currentData = testBuffer[i];
//                printf("Current data in testbuffer is %d", currentData);
//                
//                NSData * temp = [NSData dataWithBytes:currentData length:sizeof(currentData)];
//            }


            data=[[NSData dataWithBytes:inBuffer->mAudioData length:inBuffer->mAudioDataByteSize]retain];

            [restClient uploadAudioData:data url:nil];

        }


        // if we're not stopping, re-enqueue the buffer so that it gets filled again
        if (aqr->IsRunning())
            XThrowIfError(AudioQueueEnqueueBuffer(inAQ, inBuffer, 0, NULL), "AudioQueueEnqueueBuffer failed");
    } catch (CAXException e) {
        char buf[256];
        fprintf(stderr, "Error: %s (%s)\n", e.mOperation, e.FormatError(buf));
    }

}

AQRecorder::AQRecorder()
{
    mIsRunning = false;
    mRecordPacket = 0;

    data = [[NSData alloc]init];
    restClient = [[RestClient sharedManager]retain];
}

AQRecorder::~AQRecorder()
{
    AudioQueueDispose(mQueue, TRUE);
    AudioFileClose(mRecordFile);

    if (mFileName){
     CFRelease(mFileName);   
    }

    [restClient release];
    [data release];
}

// ____________________________________________________________________________________
// Copy a queue's encoder's magic cookie to an audio file.
void AQRecorder::CopyEncoderCookieToFile()
{
    UInt32 propertySize;
    // get the magic cookie, if any, from the converter     
    OSStatus err = AudioQueueGetPropertySize(mQueue, kAudioQueueProperty_MagicCookie, &propertySize);

    // we can get a noErr result and also a propertySize == 0
    // -- if the file format does support magic cookies, but this file doesn't have one.
    if (err == noErr && propertySize > 0) {
        Byte *magicCookie = new Byte[propertySize];
        UInt32 magicCookieSize;
        XThrowIfError(AudioQueueGetProperty(mQueue, kAudioQueueProperty_MagicCookie, magicCookie, &propertySize), "get audio converter's magic cookie");
        magicCookieSize = propertySize; // the converter lies and tell us the wrong size

        // now set the magic cookie on the output file
        UInt32 willEatTheCookie = false;
        // the converter wants to give us one; will the file take it?
        err = AudioFileGetPropertyInfo(mRecordFile, kAudioFilePropertyMagicCookieData, NULL, &willEatTheCookie);
        if (err == noErr && willEatTheCookie) {
            err = AudioFileSetProperty(mRecordFile, kAudioFilePropertyMagicCookieData, magicCookieSize, magicCookie);
            XThrowIfError(err, "set audio file's magic cookie");
        }
        delete[] magicCookie;
    }
}

void AQRecorder::SetupAudioFormat(UInt32 inFormatID)
{
    memset(&mRecordFormat, 0, sizeof(mRecordFormat));

    UInt32 size = sizeof(mRecordFormat.mSampleRate);
    XThrowIfError(AudioSessionGetProperty(  kAudioSessionProperty_CurrentHardwareSampleRate,
                                        &size, 
                                        &mRecordFormat.mSampleRate), "couldn't get hardware sample rate");

    //override samplearate to 8k from device sample rate

    mRecordFormat.mSampleRate = 8000.0;

    size = sizeof(mRecordFormat.mChannelsPerFrame);
    XThrowIfError(AudioSessionGetProperty(  kAudioSessionProperty_CurrentHardwareInputNumberChannels, 
                                        &size, 
                                        &mRecordFormat.mChannelsPerFrame), "couldn't get input channel count");


//    mRecordFormat.mChannelsPerFrame = 1;

    mRecordFormat.mFormatID = inFormatID;
    if (inFormatID == kAudioFormatLinearPCM)
    {
        // if we want pcm, default to signed 16-bit little-endian
        mRecordFormat.mFormatFlags = kLinearPCMFormatFlagIsSignedInteger | kLinearPCMFormatFlagIsPacked;
        mRecordFormat.mBitsPerChannel = 16;
        mRecordFormat.mBytesPerPacket = mRecordFormat.mBytesPerFrame = (mRecordFormat.mBitsPerChannel / 8) * mRecordFormat.mChannelsPerFrame;
        mRecordFormat.mFramesPerPacket = 1;
    }

    if (inFormatID == kAudioFormatULaw) {
//        NSLog(@"is ulaw");
        mRecordFormat.mFormatFlags = kLinearPCMFormatFlagIsSignedInteger;
        mRecordFormat.mSampleRate = 8000.0;
//        mRecordFormat.mFormatFlags = 0;
        mRecordFormat.mFramesPerPacket = 1;
        mRecordFormat.mChannelsPerFrame = 1;
        mRecordFormat.mBitsPerChannel = 16;//was 8
        mRecordFormat.mBytesPerPacket = 1;
        mRecordFormat.mBytesPerFrame = 1;
    }
}

NSString * GetDocumentDirectory(void)
{    
    NSArray *paths = NSSearchPathForDirectoriesInDomains(NSDocumentDirectory, NSUserDomainMask, YES);
    NSString *basePath = ([paths count] > 0) ? [paths objectAtIndex:0] : nil;
    return basePath;
}


void AQRecorder::StartRecord(CFStringRef inRecordFile)
{
    int i, bufferByteSize;
    UInt32 size;
    CFURLRef url;

    try {       
        mFileName = CFStringCreateCopy(kCFAllocatorDefault, inRecordFile);

        // specify the recording format
        SetupAudioFormat(kAudioFormatULaw /*kAudioFormatLinearPCM*/);

        // create the queue
        XThrowIfError(AudioQueueNewInput(
                                      &mRecordFormat,
                                      MyInputBufferHandler,
                                      this /* userData */,
                                      NULL /* run loop */, NULL /* run loop mode */,
                                      0 /* flags */, &mQueue), "AudioQueueNewInput failed");

        // get the record format back from the queue's audio converter --
        // the file may require a more specific stream description than was necessary to create the encoder.
        mRecordPacket = 0;

        size = sizeof(mRecordFormat);
        XThrowIfError(AudioQueueGetProperty(mQueue, kAudioQueueProperty_StreamDescription,  
                                         &mRecordFormat, &size), "couldn't get queue's format");

        NSString *basePath = GetDocumentDirectory();
        NSString *recordFile = [basePath /*NSTemporaryDirectory()*/ stringByAppendingPathComponent: (NSString*)inRecordFile];   

        url = CFURLCreateWithString(kCFAllocatorDefault, (CFStringRef)recordFile, NULL);

        // create the audio file
        XThrowIfError(AudioFileCreateWithURL(url, kAudioFileCAFType, &mRecordFormat, kAudioFileFlags_EraseFile,
                                          &mRecordFile), "AudioFileCreateWithURL failed");
        CFRelease(url);

        // copy the cookie first to give the file object as much info as we can about the data going in
        // not necessary for pcm, but required for some compressed audio
        CopyEncoderCookieToFile();


        // allocate and enqueue buffers
        bufferByteSize = ComputeRecordBufferSize(&mRecordFormat, kBufferDurationSeconds);   // enough bytes for half a second
        for (i = 0; i < kNumberRecordBuffers; ++i) {
            XThrowIfError(AudioQueueAllocateBuffer(mQueue, bufferByteSize, &mBuffers[i]),
                       "AudioQueueAllocateBuffer failed");
            XThrowIfError(AudioQueueEnqueueBuffer(mQueue, mBuffers[i], 0, NULL),
                       "AudioQueueEnqueueBuffer failed");
        }
        // start the queue
        mIsRunning = true;
        XThrowIfError(AudioQueueStart(mQueue, NULL), "AudioQueueStart failed");
    }
    catch (CAXException &e) {
        char buf[256];
        fprintf(stderr, "Error: %s (%s)\n", e.mOperation, e.FormatError(buf));
    }
    catch (...) {
        fprintf(stderr, "An unknown error occurred\n");
    }   

}

void AQRecorder::StopRecord()
{
    // end recording
    mIsRunning = false;
//    XThrowIfError(AudioQueueReset(mQueue), "AudioQueueStop failed");  
    XThrowIfError(AudioQueueStop(mQueue, true), "AudioQueueStop failed");   
    // a codec may update its cookie at the end of an encoding session, so reapply it to the file now
    CopyEncoderCookieToFile();
    if (mFileName)
    {
        CFRelease(mFileName);
        mFileName = NULL;
    }
    AudioQueueDispose(mQueue, true);
    AudioFileClose(mRecordFile);
}

请随时评论或完善我的答案，如果它是更好的解决方案，我会接受它作为答案。请注意，这是我的第一次尝试，我确信这不是最优雅或最合适的解决方案。