我正在尝试在 iOS 中构建某种 VoIP 应用程序。到目前为止，我已经能够成功地将麦克风数据作为缓冲区从麦克风发送到使用GCDAsyncSocket. 现在我需要回放我收到的数据，这让我很困惑。我在网上看过，但我看到的只是从远程播放音频文件或从 URL 播放音频流。我实际上NSData定期接收，需要弄清楚如何使用它们NSData来填充音频单元缓冲区列表。我是 C 的新手，发现很难通过它。这是我NSData从服务器获取的。
- (void)socket:(GCDAsyncSocket *)sender didReadData:(NSData *)data withTag:(long)tag
{
if (tag == 1 ){
//this is where I read password and stuff to authenticate
}
else{
[self setUpAQOutput:data];//this should somehow initialize AU and fill the buffer
}
在我的 中AudioUnitProcessor，这就是我AUnit使用Stefan Popp 的代码进行设置的方式：
//
// AudioProcessor.m
// MicInput
//
// Created by Stefan Popp on 21.09.11.
//
#import "AudioProcessor.h"
#import "PTTClient.h"
#pragma mark Recording callback
static OSStatus recordingCallback(void *inRefCon,
AudioUnitRenderActionFlags *ioActionFlags,
const AudioTimeStamp *inTimeStamp,
UInt32 inBusNumber,
UInt32 inNumberFrames,
AudioBufferList *ioData) {
// the data gets rendered here
AudioBuffer buffer;
// a variable where we check the status
OSStatus status;
/**
This is the reference to the object who owns the callback.
*/
AudioProcessor *audioProcessor = (AudioProcessor*) inRefCon;
/**
on this point we define the number of channels, which is mono
for the iphone. the number of frames is usally 512 or 1024.
*/
buffer.mDataByteSize = inNumberFrames * 2; // sample size
buffer.mNumberChannels = 1; // one channel
buffer.mData = malloc( inNumberFrames * 2 ); // buffer size
// we put our buffer into a bufferlist array for rendering
AudioBufferList bufferList;
bufferList.mNumberBuffers = 1;
bufferList.mBuffers[0] = buffer;
// render input and check for error
status = AudioUnitRender([audioProcessor audioUnit], ioActionFlags, inTimeStamp, inBusNumber, inNumberFrames, &bufferList);
// process the bufferlist in the audio processor
[audioProcessor processBuffer:&bufferList];
// clean up the buffer
free(bufferList.mBuffers[0].mData);
return noErr;
}
#pragma mark Playback callback
static OSStatus playbackCallback(void *inRefCon,
AudioUnitRenderActionFlags *ioActionFlags,
const AudioTimeStamp *inTimeStamp,
UInt32 inBusNumber,
UInt32 inNumberFrames,
AudioBufferList *ioData) {
//does nothing
return noErr;
}
#pragma mark objective-c class
@implementation AudioProcessor
@synthesize audioUnit, inAudioBuffer;
-(AudioProcessor*)init
{
self = [super init];
if (self) {
[self initializeAudio];
}
return self;
}
+ (OSStatus) playBytes:(NSArray*) byteArray {
/**
This is the reference to the object who owns the callback.
*/
// NSArray * byteArray = nil;
AudioProcessor *audioProcessor = [[AudioProcessor alloc] init];
// iterate over incoming stream an copy to output stream
for (int i=0; i < [byteArray count]; i++) {
// AudioBuffer buffer = ioData->mBuffers[i];
// find minimum size
UInt32 size = [audioProcessor inAudioBuffer].mDataByteSize;
// copy buffer to audio buffer which gets played after function return
memcpy(byteArray[i], [audioProcessor inAudioBuffer].mData, size);
// set data size
//buffer.mDataByteSize = size;
}
return noErr;
}
-(void)initializeAudio
{
OSStatus status;
// We define the audio component
AudioComponentDescription desc;
desc.componentType = kAudioUnitType_Output; // we want to ouput
desc.componentSubType = kAudioUnitSubType_RemoteIO; // we want in and ouput
desc.componentFlags = 0; // must be zero
desc.componentFlagsMask = 0; // must be zero
desc.componentManufacturer = kAudioUnitManufacturer_Apple; // select provider
// find the AU component by description
AudioComponent inputComponent = AudioComponentFindNext(NULL, &desc);
// create audio unit by component
status = AudioComponentInstanceNew(inputComponent, &audioUnit);
// define that we want record io on the input bus
UInt32 flag = 1;
status = AudioUnitSetProperty(audioUnit,
kAudioOutputUnitProperty_EnableIO, // use io
kAudioUnitScope_Input, // scope to input
kInputBus, // select input bus (1)
&flag, // set flag
sizeof(flag));
// define that we want play on io on the output bus
UInt32 stopFlag = 0;//stop flag 0 because we dont want to play audio back in device
status = AudioUnitSetProperty(audioUnit,
kAudioOutputUnitProperty_EnableIO, // use io
kAudioUnitScope_Output, // scope to output
kOutputBus, // select output bus (0)
&stopFlag, // set flag
sizeof(stopFlag));
/*
We need to specify our format on which we want to work.
We use Linear PCM cause its uncompressed and we work on raw data.
for more informations check.
We want 16 bits, 2 bytes per packet/frames at 44khz
*/
AudioStreamBasicDescription audioFormat;
audioFormat.mSampleRate = SAMPLE_RATE;
audioFormat.mFormatID = kAudioFormatLinearPCM;
audioFormat.mFormatFlags = kAudioFormatFlagIsPacked | kAudioFormatFlagIsSignedInteger;
audioFormat.mFramesPerPacket = 1;
audioFormat.mChannelsPerFrame = 1;
audioFormat.mBitsPerChannel = 16;
audioFormat.mBytesPerPacket = audioFormat.mChannelsPerFrame * sizeof( SInt16);
audioFormat.mBytesPerFrame = audioFormat.mChannelsPerFrame * sizeof( SInt16);
// set the format on the output stream
status = AudioUnitSetProperty(audioUnit,
kAudioUnitProperty_StreamFormat,
kAudioUnitScope_Output,
kInputBus,
&audioFormat,
sizeof(audioFormat));
// set the format on the input stream
status = AudioUnitSetProperty(audioUnit,
kAudioUnitProperty_StreamFormat,
kAudioUnitScope_Input,
kOutputBus,
&audioFormat,
sizeof(audioFormat));
/**
We need to define a callback structure which holds
a pointer to the recordingCallback and a reference to
the audio processor object
*/
AURenderCallbackStruct callbackStruct;
// set recording callback
callbackStruct.inputProc = recordingCallback; // recordingCallback pointer
callbackStruct.inputProcRefCon = self;
// set input callback to recording callback on the input bus
status = AudioUnitSetProperty(audioUnit,
kAudioOutputUnitProperty_SetInputCallback,
kAudioUnitScope_Global,
kInputBus,
&callbackStruct,
sizeof(callbackStruct));
/*
We do the same on the output stream to hear what is coming
from the input stream
*/
callbackStruct.inputProc = playbackCallback;
callbackStruct.inputProcRefCon = self;
// set playbackCallback as callback on our renderer for the output bus
status = AudioUnitSetProperty(audioUnit,
kAudioUnitProperty_SetRenderCallback,
kAudioUnitScope_Global,
kOutputBus,
&callbackStruct,
sizeof(callbackStruct));
// reset flag to 0
flag = 0;
/*
we need to tell the audio unit to allocate the render buffer,
that we can directly write into it.
*/
status = AudioUnitSetProperty(audioUnit,
kAudioUnitProperty_ShouldAllocateBuffer,
kAudioUnitScope_Output,
kInputBus,
&flag,
sizeof(flag));
/*
we set the number of channels to mono and allocate our block size to
1024 bytes.
*/
inAudioBuffer.mNumberChannels = 1;
inAudioBuffer.mDataByteSize = 512 * 2;
inAudioBuffer.mData = malloc( 512 * 2 );
// Initialize the Audio Unit and cross fingers =)
status = AudioUnitInitialize(audioUnit);
NSLog(@"Started");
}
#pragma mark controll stream
-(void)start;
{
// start the audio unit. You should hear something, hopefully :)
OSStatus status = AudioOutputUnitStart(audioUnit);
}
-(void)stop;
{
// stop the audio unit
OSStatus status = AudioOutputUnitStop(audioUnit);
}
#pragma mark processing
-(void)processBuffer: (AudioBufferList*) audioBufferList
{
AudioBuffer sourceBuffer = audioBufferList->mBuffers[0];
// we check here if the input data byte size has changed
if (inAudioBuffer.mDataByteSize != sourceBuffer.mDataByteSize) {
// clear old buffer
free(inAudioBuffer.mData);
// assing new byte size and allocate them on mData
inAudioBuffer.mDataByteSize = sourceBuffer.mDataByteSize;
inAudioBuffer.mData = malloc(sourceBuffer.mDataByteSize);
}
int currentBuffer =0;
int maxBuf = 800;
NSMutableData *data=[[NSMutableData alloc] init];
// CMBlockBufferRef blockBuffer;
// CMSampleBufferGetAudioBufferListWithRetainedBlockBuffer(ref, NULL, &audioBufferList, sizeof(audioBufferList), NULL, NULL, 0, &blockBuffer);
// NSLog(@"%@",blockBuffer);
// audioBufferList->mBuffers[0].mData, audioBufferList->mBuffers[0].mDataByteSize
for( int y=0; y<audioBufferList->mNumberBuffers; y++ )
{
if (currentBuffer < maxBuf){
AudioBuffer audioBuff = audioBufferList->mBuffers[y];
Float32 *frame = (Float32*)audioBuff.mData;
[data appendBytes:frame length:inAudioBuffer.mDataByteSize];
currentBuffer += audioBuff.mDataByteSize;
}
else{
break;
}
}
[[PTTClient getDefaultInstance] setAudioBufferData: data];//This is call to send buffer data to the server
// copy incoming audio data to the audio buffer (no need since we are not using playback)
//memcpy(inAudioBuffer.mData, audioBufferList->mBuffers[0].mData, audioBufferList->mBuffers[0].mDataByteSize);
}
@end
最后这是向服务器发送音频数据的方法
-(void) setAudioBufferData: (NSData*) data{
[gcdSocket writeData:data withTimeout:timeout tag:tag];
}
所有这些工作都很好，我可以在我的服务器中聆听以 Java 运行的声音。现在我需要弄清楚如何调整这个音频单元来播放NSData我不断从服务器接收到的数据包（我看过一些播放远程文件的例子，这不是我需要的。我需要播放语音）。来源不是文件，而是有人在说话，所以我有点困惑。