播放来自NSStream的原始pcmaudio数据
我想从NSInputStream播放pcm数据。 任何人都可以提供正确的方法或代码来做到这一点。
我用下面的代码得到了StreamHasData事件中的audio。
uint8_t bytes[self.audioStreamReadMaxLength]; UInt32 length = [audioStream readData:bytes maxLength:self.audioStreamReadMaxLength]; 现在我怎么能在iPhone中播放字节audio数据？
我曾经遇到类似的问题，最后我解决了这个问题。
这是我所做的基本工作。 我正在使用套接字库
下面的课程负责获取audio并使其可用于连接的客户端。
#import <Foundation/Foundation.h> #import "GCDAsyncSocket.h" #import <AudioToolbox/AudioToolbox.h> @interface AudioServer : NSObject <GCDAsyncSocketDelegate> @property (nonatomic, strong)GCDAsyncSocket * serverSocket; @property (nonatomic, strong)NSMutableArray *connectedClients; @property (nonatomic) AudioComponentInstance audioUnit; -(void) start; -(void) stop; -(void) writeDataToClients:(NSData*)data; @end #define kOutputBus 0 #define kInputBus 1 #import "AudioServer.h" #import "SM_Utils.h" static OSStatus recordingCallback(void *inRefCon, AudioUnitRenderActionFlags *ioActionFlags, const AudioTimeStamp *inTimeStamp, UInt32 inBusNumber, UInt32 inNumberFrames, AudioBufferList *ioData) { // TODO: Use inRefCon to access our interface object to do stuff // Then, use inNumberFrames to figure out how much data is available, and make // that much space available in buffers in an AudioBufferList. AudioServer *server = (__bridge AudioServer*)inRefCon; AudioBufferList bufferList; SInt16 samples[inNumberFrames]; // A large enough size to not have to worry about buffer overrun memset (&samples, 0, sizeof (samples)); bufferList.mNumberBuffers = 1; bufferList.mBuffers[0].mData = samples; bufferList.mBuffers[0].mNumberChannels = 1; bufferList.mBuffers[0].mDataByteSize = inNumberFrames*sizeof(SInt16); // Then: // Obtain recorded samples OSStatus status; status = AudioUnitRender(server.audioUnit, ioActionFlags, inTimeStamp, inBusNumber, inNumberFrames, &bufferList); NSData *dataToSend = [NSData dataWithBytes:bufferList.mBuffers[0].mData length:bufferList.mBuffers[0].mDataByteSize]; [server writeDataToClients:dataToSend]; return noErr; } @implementation AudioServer -(id) init { return [super init]; } -(void) start { [UIApplication sharedApplication].idleTimerDisabled = YES; // Create a new instance of AURemoteIO AudioComponentDescription desc; desc.componentType = kAudioUnitType_Output; desc.componentSubType = kAudioUnitSubType_VoiceProcessingIO; desc.componentManufacturer = kAudioUnitManufacturer_Apple; desc.componentFlags = 0; desc.componentFlagsMask = 0; AudioComponent comp = AudioComponentFindNext(NULL, &desc); AudioComponentInstanceNew(comp, &_audioUnit); // Enable input and output on AURemoteIO // Input is enabled on the input scope of the input element // Output is enabled on the output scope of the output element UInt32 one = 1; AudioUnitSetProperty(_audioUnit, kAudioOutputUnitProperty_EnableIO, kAudioUnitScope_Input, 1, &one, sizeof(one)); AudioUnitSetProperty(_audioUnit, kAudioOutputUnitProperty_EnableIO, kAudioUnitScope_Output, 0, &one, sizeof(one)); // Explicitly set the input and output client formats // sample rate = 44100, num channels = 1, format = 32 bit floating point AudioStreamBasicDescription audioFormat = [self getAudioDescription]; AudioUnitSetProperty(_audioUnit, kAudioUnitProperty_StreamFormat, kAudioUnitScope_Output, 1, &audioFormat, sizeof(audioFormat)); AudioUnitSetProperty(_audioUnit, kAudioUnitProperty_StreamFormat, kAudioUnitScope_Input, 0, &audioFormat, sizeof(audioFormat)); // Set the MaximumFramesPerSlice property. This property is used to describe to an audio unit the maximum number // of samples it will be asked to produce on any single given call to AudioUnitRender UInt32 maxFramesPerSlice = 4096; AudioUnitSetProperty(_audioUnit, kAudioUnitProperty_MaximumFramesPerSlice, kAudioUnitScope_Global, 0, &maxFramesPerSlice, sizeof(UInt32)); // Get the property value back from AURemoteIO. We are going to use this value to allocate buffers accordingly UInt32 propSize = sizeof(UInt32); AudioUnitGetProperty(_audioUnit, kAudioUnitProperty_MaximumFramesPerSlice, kAudioUnitScope_Global, 0, &maxFramesPerSlice, &propSize); AURenderCallbackStruct renderCallback; renderCallback.inputProc = recordingCallback; renderCallback.inputProcRefCon = (__bridge void *)(self); AudioUnitSetProperty(_audioUnit, kAudioOutputUnitProperty_SetInputCallback, kAudioUnitScope_Global, 0, &renderCallback, sizeof(renderCallback)); // Initialize the AURemoteIO instance AudioUnitInitialize(_audioUnit); AudioOutputUnitStart(_audioUnit); _connectedClients = [[NSMutableArray alloc] init]; _serverSocket = [[GCDAsyncSocket alloc] initWithDelegate:self delegateQueue:dispatch_get_main_queue()]; [self startAcceptingConnections]; } - (AudioStreamBasicDescription)getAudioDescription { AudioStreamBasicDescription audioDescription = {0}; audioDescription.mFormatID = kAudioFormatLinearPCM; audioDescription.mFormatFlags = kAudioFormatFlagIsSignedInteger | kAudioFormatFlagIsPacked | kAudioFormatFlagsNativeEndian; audioDescription.mChannelsPerFrame = 1; audioDescription.mBytesPerPacket = sizeof(SInt16)*audioDescription.mChannelsPerFrame; audioDescription.mFramesPerPacket = 1; audioDescription.mBytesPerFrame = sizeof(SInt16)*audioDescription.mChannelsPerFrame; audioDescription.mBitsPerChannel = 8 * sizeof(SInt16); audioDescription.mSampleRate = 44100.0; return audioDescription; } -(void) startAcceptingConnections { NSError *error = nil; if(_serverSocket) [_serverSocket acceptOnPort:[SM_Utils serverPort] error:&error]; } -(void)socketDidDisconnect:(GCDAsyncSocket *)sock withError:(NSError *)err { if(_connectedClients) [_connectedClients removeObject:sock]; } - (void)socket:(GCDAsyncSocket *)socket didAcceptNewSocket:(GCDAsyncSocket *)newSocket { NSLog(@"Accepted New Socket from %@:%hu", [newSocket connectedHost], [newSocket connectedPort]); @synchronized(_connectedClients) { dispatch_async(dispatch_get_main_queue(), ^{ if(_connectedClients) [_connectedClients addObject:newSocket]; }); } NSError *error = nil; if(_serverSocket) [_serverSocket acceptOnPort:[SM_Utils serverPort] error:&error]; } -(void) writeDataToClients:(NSData *)data { if(_connectedClients) { for (GCDAsyncSocket *socket in _connectedClients) { if([socket isConnected]) { [socket writeData:data withTimeout:-1 tag:0]; } else{ if([_connectedClients containsObject:socket]) [_connectedClients removeObject:socket]; } } } } -(void) stop { if(_serverSocket) { _serverSocket = nil; } [UIApplication sharedApplication].idleTimerDisabled = NO; AudioOutputUnitStop(_audioUnit); } -(void) dealloc { if(_serverSocket) { _serverSocket = nil; } [UIApplication sharedApplication].idleTimerDisabled = NO; AudioOutputUnitStop(_audioUnit); } @end
然后下面的类负责从服务器检索audio并播放它
#import <Foundation/Foundation.h> #import "GCDAsyncSocket.h" #import <AudioToolbox/AudioToolbox.h> #import "TPCircularBuffer.h" @protocol AudioClientDelegate <NSObject> -(void) connected; -(void) animateSoundIndicator:(float) rms; @end @interface AudioClient : NSObject<GCDAsyncSocketDelegate> { NSString *ipAddress; BOOL stopped; } @property (nonatomic) TPCircularBuffer circularBuffer; @property (nonatomic) AudioComponentInstance audioUnit; @property (nonatomic, strong) GCDAsyncSocket *socket; @property (nonatomic, strong) id<AudioClientDelegate> delegate; -(id) initWithDelegate:(id)delegate; -(void) start:(NSString *)ip; -(void) stop; -(TPCircularBuffer *) outputShouldUseCircularBuffer; @end static OSStatus OutputRenderCallback(void *inRefCon, AudioUnitRenderActionFlags *ioActionFlags, const AudioTimeStamp *inTimeStamp, UInt32 inBusNumber, UInt32 inNumberFrames, AudioBufferList *ioData){ AudioClient *output = (__bridge AudioClient*)inRefCon; TPCircularBuffer *circularBuffer = [output outputShouldUseCircularBuffer]; if( !circularBuffer ){ AudioUnitSampleType *left = (AudioUnitSampleType*)ioData->mBuffers[0].mData; for(int i = 0; i < inNumberFrames; i++ ){ left[ i ] = 0.0f; } return noErr; }; int32_t bytesToCopy = ioData->mBuffers[0].mDataByteSize; SInt16* outputBuffer = ioData->mBuffers[0].mData; int32_t availableBytes; SInt16 *sourceBuffer = TPCircularBufferTail(circularBuffer, &availableBytes); int32_t amount = MIN(bytesToCopy,availableBytes); memcpy(outputBuffer, sourceBuffer, amount); TPCircularBufferConsume(circularBuffer,amount); return noErr; } -(id) initWithDelegate:(id)delegate { if(!self) { self = [super init]; } [self circularBuffer:&_circularBuffer withSize:24576*5]; _delegate = delegate; stopped = NO; return self; } -(void) start:(NSString *)ip { _socket = [[GCDAsyncSocket alloc] initWithDelegate:self delegateQueue: dispatch_get_main_queue()]; NSError *err; ipAddress = ip; [UIApplication sharedApplication].idleTimerDisabled = YES; if(![_socket connectToHost:ipAddress onPort:[SM_Utils serverPort] error:&err]) { } [self setupAudioUnit]; } -(void) setupAudioUnit { AudioComponentDescription desc; desc.componentType = kAudioUnitType_Output; desc.componentSubType = kAudioUnitSubType_VoiceProcessingIO; desc.componentManufacturer = kAudioUnitManufacturer_Apple; desc.componentFlags = 0; desc.componentFlagsMask = 0; AudioComponent comp = AudioComponentFindNext(NULL, &desc); OSStatus status; status = AudioComponentInstanceNew(comp, &_audioUnit); if(status != noErr) { NSLog(@"Error creating AudioUnit instance"); } // Enable input and output on AURemoteIO // Input is enabled on the input scope of the input element // Output is enabled on the output scope of the output element UInt32 one = 1; status = AudioUnitSetProperty(_audioUnit, kAudioOutputUnitProperty_EnableIO, kAudioUnitScope_Output, kOutputBus, &one, sizeof(one)); if(status != noErr) { NSLog(@"Error enableling AudioUnit output bus"); } // Explicitly set the input and output client formats // sample rate = 44100, num channels = 1, format = 16 bit int point AudioStreamBasicDescription audioFormat = [self getAudioDescription]; status = AudioUnitSetProperty(_audioUnit, kAudioUnitProperty_StreamFormat, kAudioUnitScope_Input, kOutputBus, &audioFormat, sizeof(audioFormat)); if(status != noErr) { NSLog(@"Error setting audio format"); } AURenderCallbackStruct renderCallback; renderCallback.inputProc = OutputRenderCallback; renderCallback.inputProcRefCon = (__bridge void *)(self); status = AudioUnitSetProperty(_audioUnit, kAudioUnitProperty_SetRenderCallback, kAudioUnitScope_Global, kOutputBus, &renderCallback, sizeof(renderCallback)); if(status != noErr) { NSLog(@"Error setting rendering callback"); } // Initialize the AURemoteIO instance status = AudioUnitInitialize(_audioUnit); if(status != noErr) { NSLog(@"Error initializing audio unit"); } } - (AudioStreamBasicDescription)getAudioDescription { AudioStreamBasicDescription audioDescription = {0}; audioDescription.mFormatID = kAudioFormatLinearPCM; audioDescription.mFormatFlags = kAudioFormatFlagIsSignedInteger | kAudioFormatFlagIsPacked | kAudioFormatFlagsNativeEndian; audioDescription.mChannelsPerFrame = 1; audioDescription.mBytesPerPacket = sizeof(SInt16)*audioDescription.mChannelsPerFrame; audioDescription.mFramesPerPacket = 1; audioDescription.mBytesPerFrame = sizeof(SInt16)*audioDescription.mChannelsPerFrame; audioDescription.mBitsPerChannel = 8 * sizeof(SInt16); audioDescription.mSampleRate = 44100.0; return audioDescription; } -(void) socketDidDisconnect:(GCDAsyncSocket *)sock withError:(NSError *)err { if(!stopped) if(![_socket connectToHost:ipAddress onPort:[SM_Utils serverPort] error:&err]) { } } -(void) socket:(GCDAsyncSocket *)socket didReadData:(NSData *)data withTag:(long)tag { if(data.length > 0) { unsigned long len = [data length]; SInt16* byteData = (SInt16*)malloc(len); memcpy(byteData, [data bytes], len); double sum = 0.0; for(int i = 0; i < len/2; i++) { sum += byteData[i] * byteData[i]; } double average = sum / len; double rms = sqrt(average); [_delegate animateSoundIndicator:rms]; Byte* soundData = (Byte*)malloc(len); memcpy(soundData, [data bytes], len); if(soundData) { AudioBufferList *theDataBuffer = (AudioBufferList*) malloc(sizeof(AudioBufferList) *1); theDataBuffer->mNumberBuffers = 1; theDataBuffer->mBuffers[0].mDataByteSize = (UInt32)len; theDataBuffer->mBuffers[0].mNumberChannels = 1; theDataBuffer->mBuffers[0].mData = (SInt16*)soundData; [self appendDataToCircularBuffer:&_circularBuffer fromAudioBufferList:theDataBuffer]; } } [socket readDataToLength:18432 withTimeout:-1 tag:0]; } -(void)circularBuffer:(TPCircularBuffer *)circularBuffer withSize:(int)size { TPCircularBufferInit(circularBuffer,size); } -(void)appendDataToCircularBuffer:(TPCircularBuffer*)circularBuffer fromAudioBufferList:(AudioBufferList*)audioBufferList { TPCircularBufferProduceBytes(circularBuffer, audioBufferList->mBuffers[0].mData, audioBufferList->mBuffers[0].mDataByteSize); } -(void)freeCircularBuffer:(TPCircularBuffer *)circularBuffer { TPCircularBufferClear(circularBuffer); TPCircularBufferCleanup(circularBuffer); } -(void) socket:(GCDAsyncSocket *)socket didConnectToHost:(NSString *)host port:(uint16_t)port { OSStatus status = AudioOutputUnitStart(_audioUnit); if(status != noErr) { NSLog(@"Error starting audio unit"); } [socket readDataToLength:18432 withTimeout:-1 tag:0]; [_delegate connected]; } -(TPCircularBuffer *) outputShouldUseCircularBuffer { return &_circularBuffer; } -(void) stop { OSStatus status = AudioOutputUnitStop(_audioUnit); if(status != noErr) { NSLog(@"Error stopping audio unit"); } [UIApplication sharedApplication].idleTimerDisabled = NO; TPCircularBufferClear(&_circularBuffer); _audioUnit = nil; stopped = YES; } -(void) dealloc { OSStatus status = AudioOutputUnitStop(_audioUnit); if(status != noErr) { NSLog(@"Error stopping audio unit"); } [UIApplication sharedApplication].idleTimerDisabled = NO; TPCircularBufferClear(&_circularBuffer); _audioUnit = nil; stopped = YES; } @end
一些代码是独特的，我的要求，但其中大部分可以重新使用，我希望这有助于。
苹果有一个例子，做同样的东西： –
void* MyGetOpenALAudioData(CFURLRef inFileURL, ALsizei *outDataSize, ALenum *outDataFormat, ALsizei* outSampleRate) { OSStatus err = noErr; SInt64 theFileLengthInFrames = 0; AudioStreamBasicDescription theFileFormat; UInt32 thePropertySize = sizeof(theFileFormat); ExtAudioFileRef extRef = NULL; void* theData = NULL; AudioStreamBasicDescription theOutputFormat; // Open a file with ExtAudioFileOpen() err = ExtAudioFileOpenURL(inFileURL, &extRef); if(err) { printf("MyGetOpenALAudioData: ExtAudioFileOpenURL FAILED, Error = %ld\n", err); goto Exit; } // Get the audio data format err = ExtAudioFileGetProperty(extRef, kExtAudioFileProperty_FileDataFormat, &thePropertySize, &theFileFormat); if(err) { printf("MyGetOpenALAudioData: ExtAudioFileGetProperty(kExtAudioFileProperty_FileDataFormat) FAILED, Error = %ld\n", err); goto Exit; } if (theFileFormat.mChannelsPerFrame > 2) { printf("MyGetOpenALAudioData - Unsupported Format, channel count is greater than stereo\n"); goto Exit;} // Set the client format to 16 bit signed integer (native-endian) data // Maintain the channel count and sample rate of the original source format theOutputFormat.mSampleRate = theFileFormat.mSampleRate; theOutputFormat.mChannelsPerFrame = theFileFormat.mChannelsPerFrame; theOutputFormat.mFormatID = kAudioFormatLinearPCM; theOutputFormat.mBytesPerPacket = 2 * theOutputFormat.mChannelsPerFrame; theOutputFormat.mFramesPerPacket = 1; theOutputFormat.mBytesPerFrame = 2 * theOutputFormat.mChannelsPerFrame; theOutputFormat.mBitsPerChannel = 16; theOutputFormat.mFormatFlags = kAudioFormatFlagsNativeEndian | kAudioFormatFlagIsPacked | kAudioFormatFlagIsSignedInteger; // Set the desired client (output) data format err = ExtAudioFileSetProperty(extRef, kExtAudioFileProperty_ClientDataFormat, sizeof(theOutputFormat), &theOutputFormat); if(err) { printf("MyGetOpenALAudioData: ExtAudioFileSetProperty(kExtAudioFileProperty_ClientDataFormat) FAILED, Error = %ld\n", err); goto Exit; } // Get the total frame count thePropertySize = sizeof(theFileLengthInFrames); err = ExtAudioFileGetProperty(extRef, kExtAudioFileProperty_FileLengthFrames, &thePropertySize, &theFileLengthInFrames); if(err) { printf("MyGetOpenALAudioData: ExtAudioFileGetProperty(kExtAudioFileProperty_FileLengthFrames) FAILED, Error = %ld\n", err); goto Exit; } // Read all the data into memory UInt32 theFramesToRead = (UInt32)theFileLengthInFrames; UInt32 dataSize = theFramesToRead * theOutputFormat.mBytesPerFrame;; theData = malloc(dataSize); if (theData) { AudioBufferList theDataBuffer; theDataBuffer.mNumberBuffers = 1; theDataBuffer.mBuffers[0].mDataByteSize = dataSize; theDataBuffer.mBuffers[0].mNumberChannels = theOutputFormat.mChannelsPerFrame; theDataBuffer.mBuffers[0].mData = theData; // Read the data into an AudioBufferList err = ExtAudioFileRead(extRef, &theFramesToRead, &theDataBuffer); if(err == noErr) { // success *outDataSize = (ALsizei)dataSize; *outDataFormat = (theOutputFormat.mChannelsPerFrame > 1) ? AL_FORMAT_STEREO16 : AL_FORMAT_MONO16; *outSampleRate = (ALsizei)theOutputFormat.mSampleRate; } else { // failure free (theData); theData = NULL; // make sure to return NULL printf("MyGetOpenALAudioData: ExtAudioFileRead FAILED, Error = %ld\n", err); goto Exit; } } Exit: // Dispose the ExtAudioFileRef, it is no longer needed if (extRef) ExtAudioFileDispose(extRef); return theData; }
find示例代码在这里 ，希望这可以帮助。