调用了输入队列回调但没有数据
Input queue callback called but no data
为了开始学习 macOS 编程,我尝试制作一个简单的程序来记录来自输入设备(例如,我的 MacBook Pro 上的内置麦克风)的音频。我在 Xcode 中创建了一个 Objective-C Cocoa 项目,代码是 this tutorial from developer.apple.com.
的稍微改编的版本
这是我的代码:
// AppDelegate.m:
#include <AudioToolbox/AudioToolbox.h>
- (void)applicationDidFinishLaunching:(NSNotification *)aNotification {
struct AQRecorderState S;
#define PRINT_R do{\
printf("%d: r = %d\n",__LINE__, r);\
}while(0)
AudioStreamBasicDescription *fmt = &S.mDataFormat;
fmt->mFormatID = kAudioFormatLinearPCM;
fmt->mSampleRate = 44100.0;
fmt->mChannelsPerFrame = 1;
fmt->mBitsPerChannel = 32;
fmt->mBytesPerFrame = fmt->mChannelsPerFrame * sizeof (float);
fmt->mFramesPerPacket = 1;
fmt->mBytesPerPacket = fmt->mBytesPerFrame * fmt->mFramesPerPacket;
fmt->mFormatFlags = kAudioFormatFlagIsFloat | kAudioFormatFlagIsNonInterleaved;
OSStatus r = 0;
r = AudioQueueNewInput(&S.mDataFormat, HandleInputBuffer, &S, NULL, kCFRunLoopCommonModes, 0, &S.mQueue);
PRINT_R;
UInt32 dataFormatSize = sizeof (S.mDataFormat);
r = AudioQueueGetProperty (
S.mQueue,
kAudioConverterCurrentInputStreamDescription,
&S.mDataFormat,
&dataFormatSize
);
S.bufferByteSize = 22050;
for (int i = 0; i < NUM_BUFFERS; ++i) {
r = AudioQueueAllocateBuffer(S.mQueue, S.bufferByteSize, &S.mBuffers[i]);
PRINT_R;
r = AudioQueueEnqueueBuffer(S.mQueue, S.mBuffers[i], 0, NULL);
PRINT_R;
}
S.mCurrentPacket = 0;
S.mIsRunning = true;
r = AudioQueueStart(S.mQueue, NULL);
PRINT_R;
r = AudioQueueStop(S.mQueue, true);
S.mIsRunning = false;
PRINT_R;
r = AudioQueueDispose(S.mQueue, true);
}
这是我的输入回调函数(在单独的 C 文件中定义):
void HandleInputBuffer (
void *aqData,
AudioQueueRef inAQ,
AudioQueueBufferRef inBuffer,
const AudioTimeStamp *inStartTime,
UInt32 inNumPackets,
const AudioStreamPacketDescription *inPacketDesc
) {
struct AQRecorderState *pAqData = (struct AQRecorderState *) aqData;
if (inNumPackets == 0 && pAqData->mDataFormat.mBytesPerPacket != 0) {
inNumPackets =
inBuffer->mAudioDataByteSize / pAqData->mDataFormat.mBytesPerPacket;
}
printf("%f\n", *(float*)inBuffer->mAudioData);
if (pAqData->mIsRunning == 0)
return;
AudioQueueEnqueueBuffer(pAqData->mQueue, inBuffer, 0, NULL);
}
当程序是运行时,所有Core Audio函数调用return 0,(我相信)代表"no error",HandleInputBuffer被调用NUM_BUFFERS非常快速连续或几乎立即的时间(绝对不是每 0.5 秒一次,就像 22050 的缓冲区大小在这个采样率下所建议的那样),并且所有第一个样本都是 0.0.0.0。我在这里错过了什么?
S.bufferByteSize
以字节为单位,而不是帧,所以 22050 字节不是半秒,而是 22050/sizeof(float)
帧,所以大约八分之一秒。
如果你想要半秒,试试
S.bufferByteSize = fmt->mSampleRate * fmt->mBytesPerFrame / 2;
在您上面的代码中(以及在您链接的 git 回购中),您在 AudioQueueStart
之后立即 AudioQueueStop
和 AudioQueueDispose
音频队列。不要那样做。
为了开始学习 macOS 编程,我尝试制作一个简单的程序来记录来自输入设备(例如,我的 MacBook Pro 上的内置麦克风)的音频。我在 Xcode 中创建了一个 Objective-C Cocoa 项目,代码是 this tutorial from developer.apple.com.
的稍微改编的版本这是我的代码:
// AppDelegate.m:
#include <AudioToolbox/AudioToolbox.h>
- (void)applicationDidFinishLaunching:(NSNotification *)aNotification {
struct AQRecorderState S;
#define PRINT_R do{\
printf("%d: r = %d\n",__LINE__, r);\
}while(0)
AudioStreamBasicDescription *fmt = &S.mDataFormat;
fmt->mFormatID = kAudioFormatLinearPCM;
fmt->mSampleRate = 44100.0;
fmt->mChannelsPerFrame = 1;
fmt->mBitsPerChannel = 32;
fmt->mBytesPerFrame = fmt->mChannelsPerFrame * sizeof (float);
fmt->mFramesPerPacket = 1;
fmt->mBytesPerPacket = fmt->mBytesPerFrame * fmt->mFramesPerPacket;
fmt->mFormatFlags = kAudioFormatFlagIsFloat | kAudioFormatFlagIsNonInterleaved;
OSStatus r = 0;
r = AudioQueueNewInput(&S.mDataFormat, HandleInputBuffer, &S, NULL, kCFRunLoopCommonModes, 0, &S.mQueue);
PRINT_R;
UInt32 dataFormatSize = sizeof (S.mDataFormat);
r = AudioQueueGetProperty (
S.mQueue,
kAudioConverterCurrentInputStreamDescription,
&S.mDataFormat,
&dataFormatSize
);
S.bufferByteSize = 22050;
for (int i = 0; i < NUM_BUFFERS; ++i) {
r = AudioQueueAllocateBuffer(S.mQueue, S.bufferByteSize, &S.mBuffers[i]);
PRINT_R;
r = AudioQueueEnqueueBuffer(S.mQueue, S.mBuffers[i], 0, NULL);
PRINT_R;
}
S.mCurrentPacket = 0;
S.mIsRunning = true;
r = AudioQueueStart(S.mQueue, NULL);
PRINT_R;
r = AudioQueueStop(S.mQueue, true);
S.mIsRunning = false;
PRINT_R;
r = AudioQueueDispose(S.mQueue, true);
}
这是我的输入回调函数(在单独的 C 文件中定义):
void HandleInputBuffer (
void *aqData,
AudioQueueRef inAQ,
AudioQueueBufferRef inBuffer,
const AudioTimeStamp *inStartTime,
UInt32 inNumPackets,
const AudioStreamPacketDescription *inPacketDesc
) {
struct AQRecorderState *pAqData = (struct AQRecorderState *) aqData;
if (inNumPackets == 0 && pAqData->mDataFormat.mBytesPerPacket != 0) {
inNumPackets =
inBuffer->mAudioDataByteSize / pAqData->mDataFormat.mBytesPerPacket;
}
printf("%f\n", *(float*)inBuffer->mAudioData);
if (pAqData->mIsRunning == 0)
return;
AudioQueueEnqueueBuffer(pAqData->mQueue, inBuffer, 0, NULL);
}
当程序是运行时,所有Core Audio函数调用return 0,(我相信)代表"no error",HandleInputBuffer被调用NUM_BUFFERS非常快速连续或几乎立即的时间(绝对不是每 0.5 秒一次,就像 22050 的缓冲区大小在这个采样率下所建议的那样),并且所有第一个样本都是 0.0.0.0。我在这里错过了什么?
S.bufferByteSize
以字节为单位,而不是帧,所以 22050 字节不是半秒,而是 22050/sizeof(float)
帧,所以大约八分之一秒。
如果你想要半秒,试试
S.bufferByteSize = fmt->mSampleRate * fmt->mBytesPerFrame / 2;
在您上面的代码中(以及在您链接的 git 回购中),您在 AudioQueueStart
之后立即 AudioQueueStop
和 AudioQueueDispose
音频队列。不要那样做。