Я записываю поток поплавков, идущих от микрофона (должно быть тишина)
Я установил аудио-очереди с размером буфера 256 поплавков
Типичный буфер выглядит так:
ПАКЕТ
0.004791,
0.012512,0.008423,0.000122, -0,000519, -0,002991, -0.000031,0.001801, -0,000641,
0.001190, -0,003143, -0.001587,0.001587, -0,015503, -0,019440, -0,015167, -0,017670,
-0,018158, -0,019928, -0,019409, -0,024017, -0,019684, -0,024719, -0,044128, -0,043579,
-0,043152, -0,046417, -0,045380, -0,050079, -0,050262, -0,049164, -0,040710, -0,036713,
-0,051056, -0,045868, -0,035034, -0,033722, -0,028534, -0,027161, -0,022186, -0,018036,
-0.012207,0.004303, -0,000824, -0.000610,0.014496,0.018005,0.019745,0.019226,
0.016144,0.013184,0.009003,0.014557,0.003357, -0,011353, -0,007751, -0,007660,
-0,006409, -0,003357, -0,003510, -0,001038, -0.000092,0.007690,0.002655,0.000000,
0.000000,0.000000,0.000000,0.000000,0.000000,0.000000,0.000000,0.000000,
0.000000,0.000000,0.000000,0.000000,0.000000,0.000000,0.000000,0.000000,
0.000000,0.000000,0.000000,0.000000,0.000000,0.000000,0.000000,0.000000,
0.000000,0.000000,0.000000,0.000000,0.000000,0.000000,0.000000,0.000000,
0.000000,0.000000,0.000000,0.000000,0.000000,0.000000,0.000000,0.000000,
0.000000,0.000000,0.000000,0.000000,0.000000,0.000000,0.000000,0.000000,
0.000000,0.000000,0.000000,0.000000,0.000000,0.000000,0.000000,0.000000,
0.000000,0.000000,0.000000,0.000000,0.000000,0.000000,0.000000,0.000000,
0.000000,0.000000,0.000000, -0,006897, -0.000549,0.003174,0.003540,0.003632,
0.004578,0.005280,0.001831,0.014771,0.014954,0.001801,0.009247,0.011139,
0.005249,0.008087,0.008636,0.007385,0.007263,0.016571,0.020264,0.010590,
0.014801,0.023132,0.027039,0.031128,0.031799,0.037109,0.038757,0.049438,
0.057098,0.042786,0.045593,0.052032,0.045380,0.045227,0.045837,0.043793,
0.041931,0.043976,0.046570,0.030182,0.024475,0.029877,0.026184,0.026001,
0.026611,0.031921,0.035736,0.040710,0.053070,0.042572,0.039917,0.051636,
0.053009,0.053528,0.053009,0.054962,0.055603,0.053833,0.060638,0.050171,
0.041779,0.049194,0.046356,0.000000,0.000000,0.000000,0.000000,0.000000,
0.000000,0.000000,0.000000,0.000000,0.000000,0.000000,0.000000,0.000000,
0.000000,0.000000,0.000000,0.000000,0.000000,0.000000,0.000000,0.000000,
0.000000,0.000000,0.000000,0.000000,0.000000,0.000000,0.000000,0.000000,
0.000000,0.000000,0.000000,0.000000,0.000000,0.000000,0.000000,0.041931,
0.038879,0.034515,0.031494,0.026337,0.034576,0.028992,0.014038,0.018127,
0.017822,0.015137,0.015778,0.013519,0.015564,0.014832,0.023285,0.022034,
0,006317,0.010254,0.010742,0,004303,0,003784, -0,000153, -0,002502, ~
Я не могу понять, почему у меня, кажется, есть случайные группы нулей во входном сигнале. Кажется, что-то прерывистое.
Сначала я подумал, что, возможно, у меня был левый и правый канал, а правый канал всегда записывал ноль. Но, просматривая мой код, я четко настроил его для одного канала.
Тогда я подумал, может быть, это просто места молчания в сигнале. Но это не имеет смысла. Если бы у меня только что была дюжина нулей, то, конечно, 0,000000, я бы ожидал, что последуют очень очень маленькие числа, например, .000007 .000014, но ненулевые числа, кажется, находятся около отметки 0,01.
Я только что попытался переключить свой аудиовход на внешний USB-микрофон, и это улучшает разрешение. Ненулевые числа теперь, кажется, находятся вокруг отметки .001. но все же есть заметный разрыв ...
Интересно, выполняется ли какой-либо расчет на чипе, который округляется до 0. Если это так, можно ли его откалибровать? Что здесь происходит?
Вторая действительно странная проблема, которую я замечаю, это мошеннические ценности.
Вот пример пакета, который содержит некоторые из этих значений (на этот раз с помощью микрофона USB; вы можете увидеть, как улучшается разрешение):
~ ПАКЕТ -0.001343, -0.001190, -0.001526, -0.001373, -0.000946, -0.001526, -0.001221, -0.001190, -0.001221, -0.001251, -0.001373, -0.001190, -0.001312, -0.001312, -0-0.001282, -0.001312, -0.001099, -0.001007, -0.001221, -0.001160, -0.001312, -0.001343, -0.001221, -0.001007, -0.001099, -0.001404, -0.001068, -0.001038, -0.001404, -0.00104038, -0.00104038, -0.00104038, -0.00104038, -0.00104038, -0.001404, -0.001099, -0.001282, -0.001221, -0.001007, -0.001007, -0.001099, -0.001221, -0.001160, -0.001038, -0.001038, -0.001007, -0.000946, -0.001129, -0.000946, -0.000916, -0.000.0000.000946, -0.000946, -0.000824, -0.000824, -0.001007, -0.000763, -0.001038, -0.000854, -0.000977, -0.000916, -0.000641, -0.000977, -0.000916, -0.000946, -0.000732, -0.0008240000, 0,000000,0,000000,0,000000,0,000000,0,000000,0,000000,0,000000, 0,000000,0,000000,0,000000,0,000000,0,000000,0,000000,0,000000,0,000000, 0,000000,0,000000,0,000000,0,000000,0,000000,0,000000,0,000000,0,0000,00,00,0000,000,000000,0,000000,0,000000,0,000000,0,000000,0,000000 0,000000,0,000000,0,000000,0,000000, 0,000000,0,000000,0,000000,0,000000, 0,000000,0,000000,0,000000,0,000000,0,000000,0,000000,0,000000,0,000000, 0,000000,0,000000,0,000000,0,000000,0,000000,0,000000,0,000000,0,000000,00,00,00,00,00,00, 0,000000,0,000000,0,000000, 0,000000,0,000000,0,000000,0,000000,0,000000,0,000000,0,000000,0,000000, 0,000000,0,000000,0,000000,0,000000,0,000000,0,000000,0,000000,0,0000,00,00,00,00,00,0000,00,00,00,00, 0,000000,0,000000, 0,000000,0,000000,0,000000,0,000000,0,000000,0,000000,0,000000,0,000000, 0,000000,0,000000,0,000000,0,000000,0,000000,0,000000,0,000000,0,000000,0,0000,00,00,00,00,00,000000,00,00,00,00000000,0000,00000000,00 чтобы зачитать, 0.000000, -0.000000,2.000000, -2.000000,0.000000, -0.000000,36893488147419103232.000000, -36893488147419103232.000000,0.000000, -0.000000,8589934592.000000, -8589934592.000000,0.000000, -0.000000,158456325028528675187087900672.000000, -158456325028528675187087900672.000000,0.000000, -0.000000,131072.000000, -131072,000000,0.000000, -0.000000,2417851639229258349412352.000000, -2417851639229258349412352.000000,0.000000, -0.000000,562949953421312.000000, -562949953421312.000000,0.000031, -0.000031,10384593717069655257060992658440192.000000, -10384593717069655257060992658440192.000000,0.000000, 0.000000,0.000000,0.000000,0.000000,0.000000,0.000000,0.000000,0.000000,0,000000,0,000000,0,000000,0,000000,0,000000,0,000000,0,000000,0,000000, 0,000000,0,000000,0,000000,0,000000,0,000000,0,000000,0,000000,0,000000, 0,000000,0,000000,0,000000,0,000000,0,000000,0,000000,00,0000,00,00,0000,00,00,000,000000,0,000000,0,000000,0,000000,0,000000,0,000000,0,000000, 0,000000,0,000000,0,000000,0,000000,0,000000,0,000000,0,000000,
Меня это смущает.встречаемость этих глюков низкая;меньше, чем один из 10 кадров.
Означает ли это, что я должен предварительно обработать свой аудиопоток?
В прошлый раз, когда я использовал аудиоустройства, я никогда этого не делал.Я просто передал звук прямо в процедуру определения высоты тона.И с этим проблем не было.Вот мне и интересно, получал ли я там глюки ..
Я получаю глюки как со встроенным микрофоном MacBook, так и с внешним USB-микрофоном
Вот мой код:
//
// MicRecord.m
// PitchDetect
//
// Created by Pi on 05/01/2011.
//
#import "MicRecord.h"
void AudioInputCallback(
void * inUserData,
AudioQueueRef inAQ,
AudioQueueBufferRef inBuffer,
const AudioTimeStamp * inStartTime,
UInt32 inNumberPacketDescriptions,
const AudioStreamPacketDescription * inPacketDescs) ;
@implementation MicRecord
@synthesize fftFrame;
/*
- (id) init
{
if (self = [super init])
{
[self setupWithSampleRate: 44100
buffers: 12
bufLen: 512 ];
}
return self;
}
*/
// - - - - - - - -
- (void) setupWithSampleRate: (int) in_sampRate
buffers: (int) in_nBuffers
step: (int) in_step
frameSize: (int) in_frameSize
target: (id) in_target
sel: (SEL) in_sel
{
sampRate = in_sampRate;
nBuffers = in_nBuffers;
bufLen = in_step;
frameSize = in_frameSize;
targ = in_target;
sel = in_sel;
audioBuffer = calloc(nBuffers, sizeof(AudioQueueBufferRef *) );
[self setupAudioFormat];
[self setupAudioQueue];
fftFrame = calloc(frameSize, sizeof(float) );
}
// - - - - - - - - - - - - - - - - - - - - - - - - - - - - - -
- (void) setupAudioFormat
{
// Set the format to 32 bit, single channel, floating point, linear PCM
const int four_bytes_per_float = 4;
const int eight_bits_per_byte = 8;
memset(& dataFormat,
(int) 0x00,
sizeof(dataFormat) );
dataFormat.mSampleRate = sampRate;
dataFormat.mFormatID = kAudioFormatLinearPCM;
dataFormat.mFormatFlags = kAudioFormatFlagsNativeFloatPacked | kAudioFormatFlagIsNonInterleaved;
dataFormat.mBytesPerPacket = four_bytes_per_float;
dataFormat.mFramesPerPacket = 1;
dataFormat.mBytesPerFrame = four_bytes_per_float;
dataFormat.mChannelsPerFrame = 1;
dataFormat.mBitsPerChannel = four_bytes_per_float * eight_bits_per_byte;
}
// - - - - - - - - - - - - - - - - -
- (void) setupAudioQueue
{
currentPacket = 0;
OSStatus status;
status = AudioQueueNewInput(& dataFormat,
AudioInputCallback,
self,
CFRunLoopGetCurrent(),
kCFRunLoopCommonModes,
0,
& queue);
for(int i = 0; i < nBuffers; i++)
{
status = AudioQueueAllocateBuffer(queue,
bufLen,
& audioBuffer[i]);
status = AudioQueueEnqueueBuffer(queue,
audioBuffer[i], 0, NULL);
}
status = AudioQueueFlush (queue);
printf("Status: %d", (int) status);
}
// - - - - - - - - - - - - - - - - -
- (void) start
{
OSStatus status = AudioQueueStart(queue, NULL);
printf("Status: %d", (int) status);
}
// - - - - - - - - - - - - - - -
- (void) stop
{
AudioQueueStop(queue, true);
for(int i = 0; i < nBuffers; i++)
AudioQueueFreeBuffer(queue, audioBuffer[i]);
AudioQueueDispose(queue, true);
}
// - - - - - - - - - -
- (void) dealloc
{
[self stop];
free (audioBuffer);
[super dealloc];
}
@end
// = = = = = = = = = = = = = = = = = = = = = = = = = = = = = = = =
void AudioInputCallback(
void * inUserData,
AudioQueueRef inAQ,
AudioQueueBufferRef inBuffer,
const AudioTimeStamp * inStartTime,
UInt32 inNumberPacketDescriptions,
const AudioStreamPacketDescription * inPacketDescs
)
{
MicRecord * x = (MicRecord *) inUserData;
//if(inNumberPacketDescriptions == 0 && recordState->dataFormat.mBytesPerPacket != 0)
//{
// inNumberPacketDescriptions = inBuffer->mAudioDataByteSize / recordState->dataFormat.mBytesPerPacket;
//}
if (0)
printf("Handling buffer %d\n", (int) x->currentPacket);
int step = x->bufLen;
if (inBuffer->mAudioDataBytesCapacity != step)
{
printf("---");
}
static int k = -1;
k++;
static float lastVal = 0;
static int count = 0;
if (k < 32) {
if (k == 0)
printf("\nfloat buf[32*%d=%d] = {", step, 32*step);
float * in_buf = (float *) inBuffer->mAudioData;
printf("\n ~\nPACKET\n");
for (int i = 0; i < step; i++)
{
/*
if (fabs(in_buf[i]) < .0001 && fabs(lastVal) > .0001)
{
printf("%d Nonzeros\n",count);
count = 0;
}
if (fabs(in_buf[i]) > .0001 && fabs(lastVal) < .0001)
{
printf("%d Zeros\n",count);
count = 0;
}
count++;
lastVal = in_buf[i];*/
printf("%f,", in_buf[i] );
if (i % 8 == 0)
printf("\n");
//if (count % (8 * 64) == 0)
// printf("\n");
count++;
}
if (k == 31)
printf("}\n");
}
// shifty frame data down by 'step' elements
// to make room for new data
// imagine cutting out elts [0] thru [step-1] (ie 'step' of them)
// first new elt at pos [0] will be [step]
memmove(& x->fftFrame[0], // dest first
& x->fftFrame[step], // src
x->frameSize - step
);
memcpy(& x->fftFrame[x->frameSize - step],
inBuffer->mAudioData,
step * sizeof(float)
);
x->currentPacket += inNumberPacketDescriptions;
// }
AudioQueueEnqueueBuffer(x->queue, inBuffer, 0, NULL);
[x->targ performSelector: x->sel];
}