我正在制作一个 iPhone 应用程序，它允许用户设计一个音频过滤器并在一些录制的声音上对其进行测试。我尝试执行以下操作:

1. 我创建了两个名为“recordeAudio.aiff”和“filteredAudio.aiff”的音频文件
2. 我用麦克风录制声音并将其保存在“recordedAudio.aiff”中
3. 我将音频数据从“recordedAudio.aiff”复制到缓冲区
4. 稍后，我将对此时缓冲区中的数据执行一些音频过滤，但出于测试目的，我只想将每个样本的值减半(这只会将音量减半)所以我确信我能够处理单个样本
5. 我将结果写入第二个缓冲区
6. 我将该缓冲区的数据写入第二个文件“filteredAudio.aiff”
7. 我播放第二个文件

问题如下:只要我将数据从一个缓冲区复制到另一个缓冲区，然后将其写入第二个音频文件，一切正常。但是一旦我尝试对样本执行任何类型的操作(比如将它们除以 2)，结果只是随机噪声。这让我怀疑我没有正确解释音频数据的值，但我已经尝试了五天，但我就是不明白。如果您知道如何访问和操作单个音频样本，请帮助我，我将不胜感激! 谢谢!

这是稍后将执行过滤的代码(现在它应该只是将所有音频样本除以 2)；

OSStatus status = noErr;
UInt32 propertySizeDataPacketCount;
UInt32 writabilityDataPacketCount;
UInt32 numberOfPackets;
UInt32 propertySizeMaxPacketSize;
UInt32 writabilityMaxPacketSize;
UInt32 maxPacketSize;
UInt32 numberOfBytesRead;
UInt32 numberOfBytesToWrite;
UInt32 propertySizeDataByteCount;
SInt64 currentPacket;
double x0;
double x1;


status = AudioFileOpenURL(audioFiles->recordedFile, 
                          kAudioFileReadPermission, 
                          kAudioFileAIFFType, 
                          &audioFiles->inputFile);
status = AudioFileOpenURL(audioFiles->filteredFile, 
                          kAudioFileReadWritePermission, 
                          kAudioFileAIFFType, 
                          &audioFiles->outputFile);

status = AudioFileGetPropertyInfo(audioFiles->inputFile, 
                                  kAudioFilePropertyAudioDataPacketCount, 
                                  &propertySizeDataPacketCount, 
                                  &writabilityDataPacketCount);

status = AudioFileGetProperty(audioFiles->inputFile, 
                              kAudioFilePropertyAudioDataPacketCount, 
                              &propertySizeDataPacketCount, 
                              &numberOfPackets);

status = AudioFileGetPropertyInfo (audioFiles->inputFile, 
                                   kAudioFilePropertyMaximumPacketSize, 
                                   &propertySizeMaxPacketSize, 
                                   &writabilityMaxPacketSize);

status = AudioFileGetProperty(audioFiles->inputFile, 
                              kAudioFilePropertyMaximumPacketSize, 
                              &propertySizeMaxPacketSize, 
                              &maxPacketSize);


SInt16 *inputBuffer = (SInt16 *)malloc(numberOfPackets * maxPacketSize);
SInt16 *outputBuffer = (SInt16 *)malloc(numberOfPackets * maxPacketSize);


currentPacket = 0;
status = AudioFileReadPackets(audioFiles->inputFile, 
                              false, &numberOfBytesRead, 
                              NULL, 
                              currentPacket, 
                              &numberOfPackets, 
                              inputBuffer);


for (int i = 0; i < numberOfPackets; i++) {

    x0 = (double)inputBuffer[i];
    x1 = 0.5 * x0; //This is supposed to reduce the value of the sample by half
    //x1 = x0;     //This just copies the value of the sample and works fine
    outputBuffer[i] = (SInt16)x1;
}



numberOfBytesToWrite = numberOfBytesRead;
currentPacket = 0;
status = AudioFileWritePackets(audioFiles->outputFile, 
                               false, 
                               numberOfBytesToWrite, 
                               NULL, 
                               currentPacket, 
                               &numberOfPackets, 
                               outputBuffer);

status = AudioFileClose(audioFiles->inputFile);
status = AudioFileClose(audioFiles->outputFile);

我使用以下代码创建音频文件:

 #import "AudioFiles.h"

 #define SAMPLE_RATE         44100

 #define FRAMES_PER_PACKET   1
 #define CHANNELS_PER_FRAME  1
 #define BYTES_PER_FRAME     2
 #define BYTES_PER_PACKET    2
 #define BITS_PER_CHANNEL    16

 @implementation AudioFiles

 -(void)setupAudioFormat:(AudioStreamBasicDescription *)format {
format->mSampleRate = SAMPLE_RATE;
format->mFormatID = kAudioFormatLinearPCM;
format->mFramesPerPacket = FRAMES_PER_PACKET;
format->mChannelsPerFrame = CHANNELS_PER_FRAME;
format->mBytesPerFrame = BYTES_PER_FRAME;
format->mBytesPerPacket = BYTES_PER_PACKET;
format->mBitsPerChannel = BITS_PER_CHANNEL;
format->mReserved = 0;
format->mFormatFlags = kLinearPCMFormatFlagIsBigEndian |
    kLinearPCMFormatFlagIsSignedInteger | kLinearPCMFormatFlagIsPacked;
 }


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

    char path[256];
    NSArray *dirPaths;
    NSString *docsDir;

    dirPaths = NSSearchPathForDirectoriesInDomains(NSDocumentDirectory, NSUserDomainMask, YES);
    docsDir = [dirPaths objectAtIndex:0];

    NSString *recordedFilePath = [docsDir    stringByAppendingPathComponent:@"/recordedAudio.aiff"];
    [recordedFilePath getCString:path maxLength:sizeof(path) encoding:NSUTF8StringEncoding];
    recordedFile = CFURLCreateFromFileSystemRepresentation(NULL, (UInt8 *)path, strlen(path), false);
    recordedFileURL = [NSURL fileURLWithPath:recordedFilePath];

    dirPaths = NSSearchPathForDirectoriesInDomains(NSDocumentDirectory, NSUserDomainMask, YES);
    docsDir = [dirPaths objectAtIndex:0];

    NSString *filteredFilePath = [docsDir stringByAppendingPathComponent:@"/filteredAudio.aiff"];
    [filteredFilePath getCString:path maxLength:sizeof(path) encoding:NSUTF8StringEncoding];
    filteredFile = CFURLCreateFromFileSystemRepresentation(NULL, (UInt8 *)path, strlen(path), false);
    filteredFileURL = [NSURL fileURLWithPath:filteredFilePath];

    AudioStreamBasicDescription audioFileFormat;
    [self setupAudioFormat:&audioFileFormat];

    OSStatus status = noErr;
    status = AudioFileCreateWithURL(recordedFile, 
                                    kAudioFileAIFFType, 
                                    &audioFileFormat, 
                                    kAudioFileFlags_EraseFile, 
                                    &inputFile);
    status = AudioFileCreateWithURL(filteredFile, 
                                    kAudioFileAIFFType, 
                                    &audioFileFormat, 
                                    kAudioFileFlags_EraseFile, 
                                    &outputFile);

}

return self;
}
@end

对于录音，我使用具有以下设置的 AVAudioRecorder:

 NSDictionary *recordSettings =
 [[NSDictionary alloc] initWithObjectsAndKeys:
 [NSNumber numberWithFloat: 8000.0], AVSampleRateKey,
 [NSNumber numberWithInt: kAudioFormatLinearPCM], AVFormatIDKey,
 [NSNumber numberWithInt: 1], AVNumberOfChannelsKey,
 [NSNumber numberWithInt: AVAudioQualityMax], AVEncoderAudioQualityKey,
 [NSNumber numberWithInt:16], AVEncoderBitRateKey,
 [NSNumber numberWithBool:YES],AVLinearPCMIsBigEndianKey,
 [NSNumber numberWithBool:NO],AVLinearPCMIsFloatKey,
 [NSNumber numberWithInt:16],AVLinearPCMBitDepthKey,
 [NSNumber numberWithBool:YES], AVLinearPCMIsNonInterleaved,
 nil];

NSError *error = nil;

audioRecorder = [[AVAudioRecorder alloc] initWithURL:audioFiles->recordedFileURL settings:recordSettings error:&error];

if (error)
{
    NSLog(@"error: %@", [error localizedDescription]);

} else {
    [audioRecorder prepareToRecord];
}

最佳答案

您的输入数据是 BigEndian，但您假设它是 LittleEndian。

处理此问题的一种方法是:

SInt16 inVal = OSSwapBigToHostInt16(inputBuffer[i]);
SInt16 outVal = inVal / 2;
outputBuffer[i] = OSSwapHostToBigInt16(outVal);

关于ios - 如何访问音频文件中的样本，我们在Stack Overflow上找到一个类似的问题： https://stackoverflow.com/questions/8400716/