我想我越来越近了……还是很迷茫。我最终使用了 The Amazing Audio Engine (TAAE)。我现在正在查看 AEAudioReceiver,我的回调代码如下所示。我认为逻辑上是正确的,但我认为它实施正确。
手头的任务:以 AAC 格式记录约 5 秒的片段。
尝试:使用 AEAudioReciever 回调并将 AudioBufferList 存储在循环缓冲区中。跟踪在录音机类中接收到的音频秒数;一旦超过 5 秒标记(它可能会超过一点,但不会超过 6 秒)。调用 Obj-c 方法使用 AEAudioFileWriter 写入文件
结果:没用,录音听起来很慢,而且经常有很多噪音;我可以听到一些录音;所以我知道那里有一些数据,但就像我丢失了很多数据一样。我什至不确定如何调试它(我会继续尝试,但现在很迷茫)。
另一个项目是转换为 AAC,我是先以 PCM 格式写入文件而不是转换为 AAC,还是可以仅将音频段转换为 AAC?
提前感谢您的帮助!
----- 循环缓冲区初始化 -----
//trying to get 5 seconds audio, how do I know what the length is if I don't know the frame size yet? and is that even the right question to ask?
TPCircularBufferInit(&_buffer, 1024 * 256);
----- AEAudioReceiver 回调 ------
static void receiverCallback(__unsafe_unretained MyAudioRecorder *THIS,
__unsafe_unretained AEAudioController *audioController,
void *source,
const AudioTimeStamp *time,
UInt32 frames,
AudioBufferList *audio) {
//store the audio into the buffer
TPCircularBufferCopyAudioBufferList(&THIS->_buffer, audio, time, kTPCircularBufferCopyAll, NULL);
//increase the time interval to track by THIS
THIS.numberOfSecondInCurrentRecording += AEConvertFramesToSeconds(THIS.audioController, frames);
//if number of seconds passed an interval of 5 seconds, than write the last 5 seconds of the buffer to a file
if (THIS.numberOfSecondInCurrentRecording > 5 * THIS->_currentSegment + 1) {
NSLog(@"Segment %d is full, writing file", THIS->_currentSegment);
[THIS writeBufferToFile];
//segment tracking variables
THIS->_numberOfReceiverLoop = 0;
THIS.lastTimeStamp = nil;
THIS->_currentSegment += 1;
} else {
THIS->_numberOfReceiverLoop += 1;
}
// Do something with 'audio'
if (!THIS.lastTimeStamp) {
THIS.lastTimeStamp = (AudioTimeStamp *)time;
}
}
---- 写入文件(MyAudioRecorderClass 中的方法)----
- (void)writeBufferToFileHandler {
NSString *documentsFolder = [NSSearchPathForDirectoriesInDomains(NSDocumentDirectory, NSUserDomainMask, YES)
objectAtIndex:0];
NSString *filePath = [documentsFolder stringByAppendingPathComponent:[NSString stringWithFormat:@"Segment_%d.aiff", _currentSegment]];
NSError *error = nil;
//setup audio writer, should the buffer be converted to aac first or save the file than convert; and how the heck do you do that?
AEAudioFileWriter *writeFile = [[AEAudioFileWriter alloc] initWithAudioDescription:_audioController.inputAudioDescription];
[writeFile beginWritingToFileAtPath:filePath fileType:kAudioFileAIFFType error:&error];
if (error) {
NSLog(@"Error in init. the file: %@", error);
return;
}
int i = 1;
//loop to write all the AudioBufferLists that is in the Circular Buffer; retrieve the ones based off of the _lastTimeStamp; but I had it in NULL too and worked the same way.
while (1) {
//NSLog(@"Processing buffer file list for segment [%d] and buffer index [%d]", _currentSegment, i);
i += 1;
// Discard any buffers with an incompatible format, in the event of a format change
AudioBufferList *nextBuffer = TPCircularBufferNextBufferList(&_buffer, _lastTimeStamp);
Float32 *frame = (Float32*) &nextBuffer->mBuffers[0].mData;
//if buffer runs out, than we are done writing it and exit loop to close the file
if ( !nextBuffer ) {
NSLog(@"Ran out of frames, there were [%d] AudioBufferList", i - 1);
break;
}
//Adding audio using AudioFileWriter, is the length correct?
OSStatus status = AEAudioFileWriterAddAudio(writeFile, nextBuffer, sizeof(nextBuffer->mBuffers[0].mDataByteSize));
if (status) {
NSLog(@"Writing Error? %d", status);
}
//consume/clear the buffer
TPCircularBufferConsumeNextBufferList(&_buffer);
}
//close the file and hope it worked
[writeFile finishWriting];
}
----- 音频控制器 AudioStreamBasicDescription ------
//interleaved16BitStereoAudioDescription
AudioStreamBasicDescription audioDescription;
memset(&audioDescription, 0, sizeof(audioDescription));
audioDescription.mFormatID = kAudioFormatLinearPCM;
audioDescription.mFormatFlags = kAudioFormatFlagIsSignedInteger | kAudioFormatFlagIsPacked | kAudioFormatFlagsNativeEndian;
audioDescription.mChannelsPerFrame = 2;
audioDescription.mBytesPerPacket = sizeof(SInt16)*audioDescription.mChannelsPerFrame;
audioDescription.mFramesPerPacket = 1;
audioDescription.mBytesPerFrame = sizeof(SInt16)*audioDescription.mChannelsPerFrame;
audioDescription.mBitsPerChannel = 8 * sizeof(SInt16);
audioDescription.mSampleRate = 44100.0;