stream媒体从iphone

我需要将audio从麦克风传输到http服务器。
这些录音设置是我所需要的：

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：

发送连续的audiostream到当前浏览的摄像机。 audio需要以64 kbit / s的G711 mu-law编码传输到床边的Axis相机。 发送（这应该是SSL连接服务器中的POST URL）：POST / transmitaudio？id = Content-type：audio / basic Content-Length：99999（忽略长度）

以下是我尝试使用的链接列表。

LINK – （SO）基本解释：只有audio单元和audio队列允许通过麦克风录制时的nsdata输出。 不是一个例子，而是一个很好的定义需要什么（audio队列，或audio单元）

LINK – （SO）audiocallback示例| 只包括callback

LINK – （SO）REMOTE IO示例| 没有启动/停止，并保存到文件

LINK – （SO）REMOTE IO示例| 没有回答没有工作

LINK – （SO）基本录音示例| 很好的例子，但logging档案

LINK – （SO）这个问题引导我到InMemoryAudioFile类（无法工作）| 其次是链接到inMemoryFile（或类似的东西），但无法得到它的工作。

链接 – （SO）更多的audio单元和远程IO示例/问题| 得到这个工作，但再次没有停止function，即使当我试图找出是什么呼叫，并使其停止，似乎仍然没有传输到服务器的audio。

LINK – 体面的remoteIO和audio队列示例，但是| 另一个很好的例子，几乎得到它的工作，但有代码（编译器认为它不是obj-c + +）的一些问题，并再次不知道如何获得audio“数据”，而不是一个文件。

LINK – 用于audio队列的Apple文档 有框架问题。 （见下面的问题），但最终无法得到它的工作，但可能没有给这个时间像其他人一样，也许应该有。

LINK – （SO）在尝试实现audio队列/单元时遇到的问题 不是一个例子

LINK – （SO）另一个remoteIO例子| 另一个很好的例子，但不能弄清楚如何得到它的数据，而不是文件。

链接 – 也看起来很有趣，循环缓冲区| 无法弄清楚如何将这与audiocallback

这是我目前正在尝试stream式传输的类。 这似乎工作，虽然在接收端（连接到服务器）的扬声器有静态的。 这似乎表明audio数据格式有问题。

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; } } 

Answer 1

我在这个话题上的工作是惊人的，漫长的。 我终于得到了这个工作，但它可能是黑客入侵。 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，我得到了警告，所以在这里出现了一些问题（但是从我的研究中，使用一个池的方法与释放相同，所以我不认为这很重要）：

   对象0x13cd20类__NSCFString autoreleased没有到位的地方 – 只是泄漏 – 打破objc_autoreleaseNoPool（）debugging

3. 为了得到这个工作，我不得不从SpeakHereController（见下文）注释掉所有的AQPlayer引用，因为我无法修复任何其他的错误。 这对我来说并不重要，因为我只是录音

所以上面的主要答案是在AVAssetWriter中有一个错误，它阻止了它追加字节和写入audio数据。 联系苹果支持后，我终于find了这个，让他们通知我这件事。 据我所知该错误是特定于ulaw和AVAssetWriter，虽然我havnt尝试了许多其他格式来validation。
为了响应这个唯一的其他select是/是使用AudioQueues。 之前我曾尝试的一些东西，但带来了一堆问题。 最大的问题是我在obj-c ++中缺乏知识。 下面这个让事情起作用的类是从speakHere例子中稍作改动，这样audio就被ulaw格式化了。 其他问题是为了让所有文件都能很好地播放。 但是，这很容易通过将链中的所有文件名更改为.mm。 接下来的问题是试图使用和谐的课程。 这仍然是一个在制品，并与2号警告联系在一起。但是我的基本解决scheme是使用SpeakHereController（也包含在说明中）而不是直接访问AQRecorder。

反正这里是代码：

从obj-c类使用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); } 

Please feel free to comment or refine my answer, I will accept it as the answer if its a better solution. Please note this was my first attempt and Im sure it is not the most elegant or proper solution.

Answer 2

You could use the gamekit Framework? Then send the audio over bluetooth. There are examples in the ios developer library