音頻輸入延遲

Question

我正在從iphone音頻輸入讀取緩衝區數據。 我有一個延遲，我不能走在50ms的變化的決議。

這裏是我的課，我稱之爲得到緩衝音頻數據 - 我需要知道如何設置延遲更低，是可以通過某種方式改變緩衝區大小，或我需要的remoteIO
到去做 。 非常感謝：

#import "SoundSensor.h" 

@interface SoundSensor (Private) 

- (void)updateLevels; 
- (void)setupQueue; 
- (void)setupFormat; 
- (void)setupBuffers; 
- (void)setupMetering; 

@end 


static SoundSensor *sharedListener = nil; 

static void listeningCallback(void *inUserData, AudioQueueRef inAQ, AudioQueueBufferRef inBuffer, const AudioTimeStamp *inStartTime, UInt32 inNumberPacketsDescriptions, const AudioStreamPacketDescription *inPacketDescs) 
{ 
    NSLog(@"listeningCallback "); 
    SoundSensor *listener = (SoundSensor *)inUserData; 
    if ([listener isListening]) 
     AudioQueueEnqueueBuffer(inAQ, inBuffer, 0, NULL); 

} 

@implementation SoundSensor 





+ (SoundSensor *)sharedListener { 
    @synchronized(self) { 
     if (sharedListener == nil) 
      [[self alloc] init]; 
    } 
    NSLog(@"sharedListener "); 
    return sharedListener; 
} 

- (void)dealloc { 
    [sharedListener stop]; 
    [super dealloc]; 
} 

#pragma mark - 
#pragma mark Listening 






- (void)listen { 
    NSLog(@"listen"); 
    if (queue == nil) 
     [self setupQueue]; 
    AudioQueueStart(queue, NULL); 
} 

- (void)pause { 
    NSLog(@"pause "); 
    if (![self isListening]) 

     return; 

    AudioQueueStop(queue, true); 
} 

- (void)stop { 
    NSLog(@"stop "); 
    if (queue == nil) 
     return; 

    AudioQueueDispose(queue, true); 
    queue = nil; 
} 

- (BOOL)isListening { 
    if (queue == nil) 
     return NO; 

    UInt32 isListening, ioDataSize = sizeof(UInt32); 
    OSStatus result = AudioQueueGetProperty(queue, kAudioQueueProperty_IsRunning, &isListening, &ioDataSize); 
    return (result != noErr) ? NO : isListening; 
} 

#pragma mark - 
#pragma mark Levels getters 

- (Float32)averagePower { 
    if (![self isListening]) 
     return 0.0; 
    float tmp = [self levels][0].mAveragePower; 
    tmp = tmp * 100;  
    return tmp; 
} 

- (Float32)peakPower { 
    if (![self isListening]) 
     return 0.0; 

    return [self levels][0].mPeakPower; 
} 

- (AudioQueueLevelMeterState *)levels { 
    if (![self isListening]) 
     return nil; 

    [self updateLevels]; 
    return levels; 
} 

- (void)updateLevels { 
    UInt32 ioDataSize = format.mChannelsPerFrame * sizeof(AudioQueueLevelMeterState); 
    AudioQueueGetProperty(queue, (AudioQueuePropertyID)kAudioQueueProperty_CurrentLevelMeter, levels, &ioDataSize); 
} 

#pragma mark - 
#pragma mark Setup 

- (void)setupQueue { 
    NSLog(@"setupQueue"); 
    if (queue) 
     return; 

    [self setupFormat]; 
    [self setupBuffers]; 
    AudioQueueNewInput(&format, listeningCallback, self, NULL, NULL, 0, &queue); 
    [self setupMetering]; 
} 

- (void)setupFormat { 
    NSLog(@"setupFormat "); 
#if TARGET_IPHONE_SIMULATOR 
    format.mSampleRate = 44100.0; 
#else 
    UInt32 ioDataSize = sizeof(sampleRate); 
    AudioSessionGetProperty(kAudioSessionProperty_CurrentHardwareSampleRate, &ioDataSize, &sampleRate); 
    format.mSampleRate = sampleRate; 
#endif 
    format.mFormatID = kAudioFormatLinearPCM; 
    format.mFormatFlags = kAudioFormatFlagIsSignedInteger | kAudioFormatFlagIsPacked; 
    format.mFramesPerPacket = format.mChannelsPerFrame = 1; 
    format.mBitsPerChannel = 16; 
    format.mBytesPerPacket = format.mBytesPerFrame = 2; 
} 

- (void)setupBuffers { 
    NSLog(@"setupBuffers "); 
    AudioQueueBufferRef buffers[3]; 
    for (NSInteger i = 0; i < 3; ++i) { 
     AudioQueueAllocateBuffer(queue, 735, &buffers[i]); 
     AudioQueueEnqueueBuffer(queue, buffers[i], 0, NULL); 
    } 
} 

- (void)setupMetering { 
    NSLog(@"setupMetering"); 
    levels = (AudioQueueLevelMeterState *)calloc(sizeof(AudioQueueLevelMeterState), format.mChannelsPerFrame); 
    UInt32 trueValue = true; 
    AudioQueueSetProperty(queue, kAudioQueueProperty_EnableLevelMetering, &trueValue, sizeof(UInt32)); 
} 

#pragma mark - 
#pragma mark Singleton Pattern 

+ (id)allocWithZone:(NSZone *)zone { 
    NSLog(@"allocWithZone"); 
    @synchronized(self) { 
     if (sharedListener == nil) { 
      sharedListener = [super allocWithZone:zone]; 
      return sharedListener; 
     } 
    } 

    return nil; 
} 

- (id)copyWithZone:(NSZone *)zone { 
    NSLog(@"copyWithZone"); 
    return self; 
} 

- (id)init { 
    if ([super init] == nil) 
     return nil; 

    return self; 
} 

- (id)retain { 
    return self; 
} 

- (unsigned)retainCount { 
    return UINT_MAX; 
} 

- (void)release { 
    // Do nothing. 
} 

- (id)autorelease { 
    return self; 
} 

@end 

Answer 1

使用RemoteIO音頻單元記錄，連同設置合適的較小值（試行5.8毫秒爲單位）kAudioSessionProperty_PreferredHardwareIOBufferDuration音頻會話屬性，似乎提供了更低的延遲比使用音頻用於記錄的隊列API。使用RemoteIO音頻單元進行輸入時，必須使用某種形式的DSP包絡跟蹤代碼（可能甚至是光譜加權），從原始PCM樣本計算自己的電平測量值。沒有內置API來直接爲RemoteIO提供這些API。

使用kAudioQueueProperty_CurrentLevelMeter似乎沒有爲低延遲設計或配置。