我似乎无法理解为什么这种方法不起作用，我试图为给定功能获取声音样本。
我的代码基于使用了Objective-c的函数版本。
但是下面用C ++编写的代码并不意味着通过将浮点缓冲区传递给OXY_DecodeAudioBuffer函数来工作，该函数随后会尝试并在缓冲区中查找数据。
问题：我是否将正确的缓冲区大小和输出从缓冲区传递给函数？我总是在缓冲区中找不到任何数据。谁能看到这个问题？
我正在使用的硬件是带有USB麦克风的Raspberry Pi 2。
我还在说明中包括了该功能：
//OXY_DecodeAudioBuffer function, receives an audiobuffer of specified size and outputs if encoded data is found
//* Parameters:
// audioBuffer: float array of bufferSize size with audio data to be decoded
// size: size of audioBuffer
// oxyingObject: OXY object instance, created in OXY_Create()
//* Returns: -1 if no decoded data is found, -2 if start token is found, -3 if complete word has been decoded, positive number if character is decoded (number is the token idx)
OXY_DLLEXPORT int32_t OXY_DecodeAudioBuffer(float *audioBuffer, int size, void *oxyingObject);
以下代码中的float_buffer输出：
1. -0.00354004 -0.00369263 -0.00338745 -0.00354004 -0.00341797 -0.00402832
程序代码：
#include <stdio.h>
#include <stdlib.h>
#include <alsa/asoundlib.h>
#include <unistd.h>
#include <math.h>
#include <time.h>
#include <stdio.h>
#include <stdlib.h>
#include <iostream>
#include "Globals.h"
#include "OxyCoreLib_api.h"
void* mCore;
using namespace std;
void GetDecodedMode(){
std::cerr << "DECODE_MODE ---> " << OXY_GetDecodedMode(mCore) << std::endl << std::endl;
}
int main(void)
{
int i,j;
int err;
int mode = 3;
int16_t *buffer;
float* float_buffer;
// Allocate our own buffers (1 channel, 16 bits per sample, thus 16 bits per frame, thus 2 bytes per frame).
// Practice learns the buffers used contain 512 frames, if this changes it will be fixed in processAudio.
int buffer_frames = 512; //Not sure this correct but reason above
unsigned int rate = 44100;
float sampleRate = 44100.f; //to configure
snd_pcm_t *capture_handle;
snd_pcm_hw_params_t *hw_params;
snd_pcm_format_t format = SND_PCM_FORMAT_S16_LE;
if ((err = snd_pcm_open(&capture_handle, "hw:1,0", SND_PCM_STREAM_CAPTURE, 0)) < 0) {
fprintf(stderr, "cannot open audio device %s (%s)\n","device",snd_strerror(err));
exit(1);
} else {fprintf(stdout, "audio interface opened\n");}
if ((err = snd_pcm_hw_params_malloc(&hw_params)) < 0) {
fprintf(stderr, "cannot allocate hardware parameter structure (%s)\n",
snd_strerror(err));
exit(1);
} else { fprintf(stdout, "hw_params allocated\n"); }
if ((err = snd_pcm_hw_params_any(capture_handle, hw_params)) < 0) {
fprintf(stderr, "cannot initialize hardware parameter structure (%s)\n",
snd_strerror(err));
exit(1);
} else { fprintf(stdout, "hw_params initialized\n"); }
if ((err = snd_pcm_hw_params_set_access(capture_handle, hw_params, SND_PCM_ACCESS_RW_INTERLEAVED)) < 0) {
fprintf(stderr, "cannot set access type (%s)\n",
snd_strerror(err));
exit(1);
} else { fprintf(stdout, "hw_params access set\n"); }
if ((err = snd_pcm_hw_params_set_format(capture_handle, hw_params, format)) < 0) {
fprintf(stderr, "cannot set sample format (%s)\n",
snd_strerror(err));
exit(1);
} else { fprintf(stdout, "hw_params format set\n"); }
if ((err = snd_pcm_hw_params_set_rate_near(capture_handle, hw_params, &rate, 0)) < 0) {
fprintf(stderr, "cannot set sample rate (%s)\n",
snd_strerror(err));
exit(1);
} else { fprintf(stdout, "hw_params rate set\n"); }
if ((err = snd_pcm_hw_params_set_channels(capture_handle, hw_params, 1)) < 0) {
fprintf(stderr, "cannot set channel count (%s)\n",
snd_strerror(err));
exit(1);
} else { fprintf(stdout, "hw_params channels set\n"); }
if ((err = snd_pcm_hw_params(capture_handle, hw_params)) < 0) {
fprintf(stderr, "cannot set parameters (%s)\n",
snd_strerror(err));
exit(1);
} else { fprintf(stdout, "hw_params set\n"); }
snd_pcm_hw_params_free(hw_params);
fprintf(stdout, "hw_params freed\n");
if ((err = snd_pcm_prepare(capture_handle)) < 0) {
fprintf(stderr, "cannot prepare audio interface for use (%s)\n",
snd_strerror(err));
exit(1);
} else { fprintf(stdout, "audio interface prepared\n"); }
//allocate buffer of 16bit ints, as specified in PCM_FORMAT
//initialise
mCore = OXY_Create();
//Configure - Mode 3 inaudible, 44100, bufferSize
OXY_Configure(mode, sampleRate, buffer_frames, mCore);
//Debug to make sure
GetDecodedMode();
buffer = static_cast<int16_t*>(malloc(buffer_frames * snd_pcm_format_width(format) / 8 * 2));
//buffer = malloc(buffer_frames * snd_pcm_format_width(format) / 8 * 2);
float_buffer = static_cast<float*>(malloc(buffer_frames*sizeof(float)));
//float_buffer = malloc(buffer_frames*sizeof(float));
fprintf(stdout, "buffer allocated\n");
//where did 10000 come from doubt its correct
for (i = 0; i < 10000; ++i) {
//read from audio device into buffer
if ((err = snd_pcm_readi(capture_handle, buffer, buffer_frames)) != buffer_frames) {
fprintf(stderr, "read from audio interface failed (%s)\n",
err, snd_strerror(err));
exit(1);
}
//try to change buffer from short ints to floats for transformation
for (i = 0; i < buffer_frames; i++){
//norm
float_buffer[i] = (float)buffer[i]/32768.0;
//Example output of float_buffer
/*
-0.00354004
-0.00369263
-0.00338745
-0.00354004
-0.00341797
-0.00402832
-0.00341797
-0.00427246
-0.00375366
-0.00378418
-0.00408936
-0.00332642
-0.00369263
-0.00350952
-0.00369263
-0.00369263
-0.00344849
-0.00354004
*/
}
//send to float_to be tested
int ret = OXY_DecodeAudioBuffer(float_buffer, buffer_frames, mCore);
if (ret == -2)
{
std::cerr << "FOUND_TOKEN ---> -2 " << std::endl << std::endl;
}
else if(ret>=0)
{
std::cerr << "Decode started ---> -2 " << ret << std::endl << std::endl;
}
else if (ret == -3)
{
//int sizeStringDecoded = OXY_GetDecodedData(mStringDecoded, mCore);
std::cerr << "STRING DECODED ---> -2 " << std::endl << std::endl;
// ...
}
else
{
std::cerr << "No data found in this buffer" << std::endl << std::endl;
//no data found in this buffer
}
}
free(buffer);
snd_pcm_close(capture_handle);
std::cerr << "memory freed\n" << std::endl << std::endl;
//snd_pcm_close(capture_handle);
return(0);
//exit(0);
}
使用相同API的工作Objective-C版本：
//
// IosAudioController.m
//
#import "IosAudioController.h"
#import <AudioToolbox/AudioToolbox.h>
#import "OxyCoreLib_api.h"
#define kOutputBus 0
#define kInputBus 1
IosAudioController* iosAudio;
void checkStatus(int status){
if (status) {
printf("Status not 0! %d\n", status);
exit(1);
}
}
static OSStatus recordingCallback(void *inRefCon,
AudioUnitRenderActionFlags *ioActionFlags,
const AudioTimeStamp *inTimeStamp,
UInt32 inBusNumber,
UInt32 inNumberFrames,
AudioBufferList *ioData) {
if (iosAudio->mOxyObject->mDecoding == 0)
return noErr;
// Because of the way our audio format (setup below) is chosen:
// we only need 1 buffer, since it is mono
// Samples are 16 bits = 2 bytes.
// 1 frame includes only 1 sample
AudioBuffer buffer;
buffer.mNumberChannels = 1;
buffer.mDataByteSize = inNumberFrames * 2;
buffer.mData = malloc( inNumberFrames * 2 );
// Put buffer in a AudioBufferList
AudioBufferList bufferList;
bufferList.mNumberBuffers = 1;
bufferList.mBuffers[0] = buffer;
// Then:
// Obtain recorded samples
OSStatus status;
status = AudioUnitRender([iosAudio audioUnit],
ioActionFlags,
inTimeStamp,
inBusNumber,
inNumberFrames,
&bufferList);
checkStatus(status);
// Now, we have the samples we just read sitting in buffers in bufferList
// Process the new data
[iosAudio processAudio:&bufferList];
//Now Decode Audio *******************
//convert from AudioBuffer format to *float buffer
iosAudio->floatBuffer = (float *)malloc(inNumberFrames * sizeof(float));
//UInt16 *frameBuffer = bufferList.mBuffers[0].mData;
SInt16 *frameBuffer = bufferList.mBuffers[0].mData;
for(int j=0;j<inNumberFrames;j++)
{
iosAudio->floatBuffer[j] = frameBuffer[j]/32768.0;
}
int ret = OXY_DecodeAudioBuffer(iosAudio->floatBuffer, inNumberFrames, (void*)iosAudio->mOxyObject->mOxyCore);
if (ret == -2)
{
// NSLog(@"BEGIN TOKEN FOUND!");
[iosAudio->mObject performSelector:iosAudio->mSelector withObject:[NSNumber numberWithInt:0]];
}
else if (ret >= 0)
{
NSLog(@"Decode started %@",@(ret).stringValue);
}
else if (ret == -3)
{
int sizeStringDecoded = OXY_GetDecodedData(iosAudio->mStringDecoded, (void*)iosAudio->mOxyObject->mOxyCore);
NSString *tmpString = [NSString stringWithUTF8String:iosAudio->mStringDecoded];
iosAudio->mOxyObject->mDecodedString = [NSString stringWithUTF8String:iosAudio->mStringDecoded];
if (sizeStringDecoded > 0)
{
iosAudio->mOxyObject->mDecodedOK = 1;
NSLog(@"Decoded OK! %@ ", tmpString);
[iosAudio->mObject performSelector:iosAudio->mSelector withObject:[NSNumber numberWithInt:1]];
}
else
{
iosAudio->mOxyObject->mDecodedOK = -1;
NSLog(@"END DECODING BAD! %@ ", tmpString);
[iosAudio->mObject performSelector:iosAudio->mSelector withObject:[NSNumber numberWithInt:2]];
}
}
else
{
//no data found in this buffer
}
// release the malloc'ed data in the buffer we created earlier
free(bufferList.mBuffers[0].mData);
free(iosAudio->floatBuffer);
return noErr;
}
static OSStatus playbackCallback(void *inRefCon,
AudioUnitRenderActionFlags *ioActionFlags,
const AudioTimeStamp *inTimeStamp,
UInt32 inBusNumber,
UInt32 inNumberFrames,
AudioBufferList *ioData) {
// Notes: ioData contains buffers (may be more than one!)
// Fill them up as much as you can. Remember to set the size value in each buffer to match how
// much data is in the buffer.
for (int i=0; i < ioData->mNumberBuffers; i++)
{ // in practice we will only ever have 1 buffer, since audio format is mono
AudioBuffer buffer = ioData->mBuffers[i];
// NSLog(@" Buffer %d has %d channels and wants %d bytes of data.", i, buffer.mNumberChannels, buffer.mDataByteSize);
// copy temporary buffer data to output buffer
UInt32 size = min(buffer.mDataByteSize, [iosAudio tempBuffer].mDataByteSize); // dont copy more data than we have, or than fits
memcpy(buffer.mData, [iosAudio tempBuffer].mData, size);
buffer.mDataByteSize = size; // indicate how much data we wrote in the buffer
// uncomment to hear random noise
/*UInt16 *frameBuffer = buffer.mData;
for (int j = 0; j < inNumberFrames; j++)
frameBuffer[j] = rand();*/
// Play encoded buffer
if (iosAudio->mOxyObject->mEncoding > 0)
{
int sizeSamplesRead;
float audioBuffer[2048];
sizeSamplesRead = OXY_GetEncodedAudioBuffer(audioBuffer, (void*)iosAudio->mOxyObject->mOxyCore);
if (sizeSamplesRead == 0)
iosAudio->mOxyObject->mEncoding = 0;
SInt16 *frameBuffer = buffer.mData;
for(int j=0;j<sizeSamplesRead;j++)
{
frameBuffer[j] = audioBuffer[j]*32768.0;
}
}
else
{
SInt16 *frameBuffer = buffer.mData;
for (int j = 0; j < inNumberFrames; j++)
frameBuffer[j] = 0;
}
}
return noErr;
}
@implementation IosAudioController
@synthesize audioUnit, tempBuffer;
- (id) init {
self = [super init];
OSStatus status;
// Describe audio component
AudioComponentDescription desc;
desc.componentType = kAudioUnitType_Output;
desc.componentSubType = kAudioUnitSubType_RemoteIO;
desc.componentFlags = 0;
desc.componentFlagsMask = 0;
desc.componentManufacturer = kAudioUnitManufacturer_Apple;
// Get component
AudioComponent inputComponent = AudioComponentFindNext(NULL, &desc);
// Get audio units
status = AudioComponentInstanceNew(inputComponent, &audioUnit);
checkStatus(status);
// Enable IO for recording
UInt32 flag = 1;
status = AudioUnitSetProperty(audioUnit,
kAudioOutputUnitProperty_EnableIO,
kAudioUnitScope_Input,
kInputBus,
&flag,
sizeof(flag));
checkStatus(status);
// Enable IO for playback
status = AudioUnitSetProperty(audioUnit,
kAudioOutputUnitProperty_EnableIO,
kAudioUnitScope_Output,
kOutputBus,
&flag,
sizeof(flag));
checkStatus(status);
// Describe format
AudioStreamBasicDescription audioFormat;
audioFormat.mSampleRate = 44100.0;
audioFormat.mFormatID = kAudioFormatLinearPCM;
audioFormat.mFormatFlags = kAudioFormatFlagIsSignedInteger | kAudioFormatFlagIsPacked;
audioFormat.mFramesPerPacket = 1;
audioFormat.mChannelsPerFrame = 1;
audioFormat.mBitsPerChannel = 16;
audioFormat.mBytesPerPacket = 2;
audioFormat.mBytesPerFrame = 2;
// Apply format
status = AudioUnitSetProperty(audioUnit,
kAudioUnitProperty_StreamFormat,
kAudioUnitScope_Output,
kInputBus,
&audioFormat,
sizeof(audioFormat));
checkStatus(status);
status = AudioUnitSetProperty(audioUnit,
kAudioUnitProperty_StreamFormat,
kAudioUnitScope_Input,
kOutputBus,
&audioFormat,
sizeof(audioFormat));
checkStatus(status);
// Set input callback
AURenderCallbackStruct callbackStruct;
callbackStruct.inputProc = recordingCallback;
callbackStruct.inputProcRefCon = (__bridge void *)self;
status = AudioUnitSetProperty(audioUnit,
kAudioOutputUnitProperty_SetInputCallback,
kAudioUnitScope_Global,
kInputBus,
&callbackStruct,
sizeof(callbackStruct));
checkStatus(status);
// Set output callback
callbackStruct.inputProc = playbackCallback;
callbackStruct.inputProcRefCon = (__bridge void *)self;
status = AudioUnitSetProperty(audioUnit,
kAudioUnitProperty_SetRenderCallback,
kAudioUnitScope_Global,
kOutputBus,
&callbackStruct,
sizeof(callbackStruct));
checkStatus(status);
// Disable buffer allocation for the recorder (optional - do this if we want to pass in our own)
flag = 0;
status = AudioUnitSetProperty(audioUnit,
kAudioUnitProperty_ShouldAllocateBuffer,
kAudioUnitScope_Output,
kInputBus,
&flag,
sizeof(flag));
// Allocate our own buffers (1 channel, 16 bits per sample, thus 16 bits per frame, thus 2 bytes per frame).
// Practice learns the buffers used contain 512 frames, if this changes it will be fixed in processAudio.
tempBuffer.mNumberChannels = 1;
int size = 512;
#if (TARGET_OS_SIMULATOR)
size = 256; //TODO check this value!! depends on play/record callback buffer size
#else
size = 512; //TODO check this value!! depends on play/record callback buffer size
#endif
tempBuffer.mDataByteSize = size * 2;
tempBuffer.mData = malloc( size * 2);
// Initialise
status = AudioUnitInitialize(audioUnit);
checkStatus(status);
return self;
}
- (void) start {
OSStatus status = AudioOutputUnitStart(audioUnit);
checkStatus(status);
}
- (void) stop {
OSStatus status = AudioOutputUnitStop(audioUnit);
checkStatus(status);
}
- (void) processAudio: (AudioBufferList*) bufferList{
AudioBuffer sourceBuffer = bufferList->mBuffers[0];
// fix tempBuffer size if it's the wrong size
if (tempBuffer.mDataByteSize != sourceBuffer.mDataByteSize) {
free(tempBuffer.mData);
tempBuffer.mDataByteSize = sourceBuffer.mDataByteSize;
tempBuffer.mData = malloc(sourceBuffer.mDataByteSize);
}
// copy incoming audio data to temporary buffer
memcpy(tempBuffer.mData, bufferList->mBuffers[0].mData, bufferList->mBuffers[0].mDataByteSize);
}
- (void) dealloc {
AudioUnitUninitialize(audioUnit);
free(tempBuffer.mData);
}
- (void) setOxyObject: (OxyCore*) oxyObject
{
mOxyObject = oxyObject;
}
- (void) setListenCallback:(id)object withSelector:(SEL)selector
{
mObject = object;
mSelector = selector;
}
@end
答案 0 :(得分：0)
我可以看到的一个问题是，您正在使用2个嵌套循环和相同的变量进行迭代。第一个循环for (i = 0; i < 10000; ++i)和第二个循环for (i = 0; i < buffer_frames; i++)，如果buffer_frames >= 10000 - 1，则第一个循环将执行一次并退出，否则将进入无限循环。
关于以下行，我还有两句话：
buffer = static_cast<int16_t*>(malloc(buffer_frames * snd_pcm_format_width(format) / 8 * 2));
根据API reference snd_pcm_format_width（format）返回每个样本的位数。由于每个样本有16位，并且每个帧仅包含一个样本，因此您应该分配buffer_frames * snd_pcm_format_width(format) / 8字节的内存（乘法运算中的2代表通道数，在您的情况下为1）。另外，我建议将缓冲区类型更改为char*，因为它是唯一不违反strict aliasing rule的类型。因此，该行变为：
static_cast<char*>(malloc(buffer_frames * (snd_pcm_format_width(format) / 8)));
当您完成从短整数转换为浮点数的技巧时，第二个for循环变为：
int16_t* sint_buffer = buffer;
for (j = 0; j < buffer_frames; ++j){
float_buffer[j] = (float)sint_buffer[j]/32768.0;
// everything else goes here
}