来自iphone的流媒体

时间:2012-05-08 15:03:01

标签: ios audio-streaming

我需要将音频从麦克风传输到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 im编码状态:

  

向当前查看的相机发送连续的音频流。   音频需要以64 kbit / s的G711 mu-law进行编码以便传输到   床边的Axis相机。发送(这应该是SSL中的POST URL   连接服务器):POST / transmitaudio?id =   内容类型:audio / basic内容长度:99999(长度被忽略)   

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音时才允许nsdata作为输出不是一个例子,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| 仅包含回调

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

LINK - (SO)REMOTE IO示例| 无法回答

LINK - (SO)基本录音示例| 好的例子,但记录到文件

LINK - (SO)引导我进入InMemoryAudioFile类的问题(无法开始工作) 跟随到inMemoryFile(或类似的东西)的链接,但无法使其正常工作。

LINK - (SO)更多音频单元和远程io示例/问题| 让这一个工作,但再一次没有停止功能,即使我试图弄清楚呼叫是什么并使其停止,它仍然似乎没有将音频传输到服务器。

LINK - 体面的remoteIO和音频队列示例,但是| 另一个很好的例子,几乎让它工作但是代码有些问题(编译器认为它不是obj-c ++)而且再一次不知道如何从它而不是文件中获取音频“数据”。

LINK - Apple docs for audio queue | 有框架问题。通过它(见下面的问题),但最终无法使它工作,但可能没有给这个与其他人一样多的时间,也许应该有。

LINK - (SO)在尝试实现音频队列/单元时遇到的问题不是示例

LINK - (SO)另一个remoteIO示例| 另一个很好的例子,但无法弄清楚如何将其转换为数据而不是文件。

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

这是我当前正在尝试传输的课程。这似乎有效,尽管接收器端(连接到服务器)的扬声器有静电。这似乎表明音频数据格式存在问题。

IOS VERSION(减去GCD套接字的委托方法):

@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;
    }

}

2 个答案:

答案 0 :(得分:3)

我在这个主题上的工作是惊人的,漫长的。我终于得到了这个工作,但它可能是黑客攻击。因此,我会在发布答案之前列出一些警告:

  1. 缓冲区之间仍然存在点击噪音

  2. 由于我在obj-c ++类中使用obj-c类的方式,我收到警告,所以有一些错误(但是我使用池的研究与释放相同,所以我不相信这很重要):

      

    类__NSCFString的对象0x13cd20自动释放,没有池   地方 - 只是泄漏 - 在objc_autoreleaseNoPool()上打破调试

  3. 为了使这个工作,我必须注释掉SpeakHereController的所有AQPlayer引用(见下文),因为我无法通过任何其他方式修复错误。然而,这对我来说无关紧要,因为我只记录

  4. 所以上面的主要答案是AVAssetWriter中存在一个错误,它阻止它附加字节并写入音频数据。我终于在联系苹果支持后发现了这一点,让他们通知我这件事。据我所知,这个bug是特定于ulaw和AVAssetWriter的,虽然我已经尝试了许多其他格式来验证。
    为此,唯一的另一个选择是使用AudioQueues。我曾经尝试过的东西但带来了一堆问题。最大的问题是我对obj-c ++缺乏了解。让下面的东西工作的是来自speakHere的例子,稍有变化,因此音频是ulaw格式的。其他问题是试图让所有文件都能很好地发挥作用。但是,通过将链中的所有文件名更改为.mm,可以轻松解决此问题。下一个问题是尝试和谐地使用这些类。这仍然是WIP,并且与警告编号2相关。但我的基本解决方案是使用SpeakHereController(也包括在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类的SpeakHereController

    ·H

    @property(nonatomic,strong) SpeakHereController * recorder;
    

    .mm

    [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];
    }
    

    SpeakHereController

    #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
    

    AQRecorder (.h有2行重要性

    #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);
    }
    

    请随时评论或改进我的答案,我会接受它作为答案,如果它是一个更好的解决方案。请注意这是我的第一次尝试,我相信它不是最优雅或最合适的解决方案。

答案 1 :(得分:0)

你可以使用gamekit框架吗?然后通过蓝牙发送音频。 ios开发人员库中有一些示例