Question

我想将PCM（CMSampleBufferRef（s）从AVCaptureAudioDataOutputSampleBufferDelegate生成）转换为AAC。

当第一个CMSampleBufferRef到达时，我根据文档设置了（进/出）AudioStreamBasicDescription（s），“out”

AudioStreamBasicDescription inAudioStreamBasicDescription = *CMAudioFormatDescriptionGetStreamBasicDescription((CMAudioFormatDescriptionRef)CMSampleBufferGetFormatDescription(sampleBuffer));

AudioStreamBasicDescription outAudioStreamBasicDescription = {0}; // Always initialize the fields of a new audio stream basic description structure to zero, as shown here: ...
outAudioStreamBasicDescription.mSampleRate = 44100; // The number of frames per second of the data in the stream, when the stream is played at normal speed. For compressed formats, this field indicates the number of frames per second of equivalent decompressed data. The mSampleRate field must be nonzero, except when this structure is used in a listing of supported formats (see “kAudioStreamAnyRate”).
outAudioStreamBasicDescription.mFormatID = kAudioFormatMPEG4AAC; // kAudioFormatMPEG4AAC_HE does not work. Can't find `AudioClassDescription`. `mFormatFlags` is set to 0.
outAudioStreamBasicDescription.mFormatFlags = kMPEG4Object_AAC_SSR; // Format-specific flags to specify details of the format. Set to 0 to indicate no format flags. See “Audio Data Format Identifiers” for the flags that apply to each format.
outAudioStreamBasicDescription.mBytesPerPacket = 0; // The number of bytes in a packet of audio data. To indicate variable packet size, set this field to 0. For a format that uses variable packet size, specify the size of each packet using an AudioStreamPacketDescription structure.
outAudioStreamBasicDescription.mFramesPerPacket = 1024; // The number of frames in a packet of audio data. For uncompressed audio, the value is 1. For variable bit-rate formats, the value is a larger fixed number, such as 1024 for AAC. For formats with a variable number of frames per packet, such as Ogg Vorbis, set this field to 0.
outAudioStreamBasicDescription.mBytesPerFrame = 0; // The number of bytes from the start of one frame to the start of the next frame in an audio buffer. Set this field to 0 for compressed formats. ...
outAudioStreamBasicDescription.mChannelsPerFrame = 1; // The number of channels in each frame of audio data. This value must be nonzero.
outAudioStreamBasicDescription.mBitsPerChannel = 0; // ... Set this field to 0 for compressed formats.
outAudioStreamBasicDescription.mReserved = 0; // Pads the structure out to force an even 8-byte alignment. Must be set to 0.

和AudioConverterRef。

AudioClassDescription audioClassDescription;
memset(&audioClassDescription, 0, sizeof(audioClassDescription));
UInt32 size;
NSAssert(AudioFormatGetPropertyInfo(kAudioFormatProperty_Encoders, sizeof(outAudioStreamBasicDescription.mFormatID), &outAudioStreamBasicDescription.mFormatID, &size) == noErr, nil);
uint32_t count = size / sizeof(AudioClassDescription);
AudioClassDescription descriptions[count];
NSAssert(AudioFormatGetProperty(kAudioFormatProperty_Encoders, sizeof(outAudioStreamBasicDescription.mFormatID), &outAudioStreamBasicDescription.mFormatID, &size, descriptions) == noErr, nil);
for (uint32_t i = 0; i < count; i++) {

    if ((outAudioStreamBasicDescription.mFormatID == descriptions[i].mSubType) && (kAppleSoftwareAudioCodecManufacturer == descriptions[i].mManufacturer)) {

        memcpy(&audioClassDescription, &descriptions[i], sizeof(audioClassDescription));

    }
}
NSAssert(audioClassDescription.mSubType == outAudioStreamBasicDescription.mFormatID && audioClassDescription.mManufacturer == kAppleSoftwareAudioCodecManufacturer, nil);
AudioConverterRef audioConverter;
memset(&audioConverter, 0, sizeof(audioConverter));
NSAssert(AudioConverterNewSpecific(&inAudioStreamBasicDescription, &outAudioStreamBasicDescription, 1, &audioClassDescription, &audioConverter) == 0, nil);

然后，我将每个CMSampleBufferRef转换为原始AAC数据。

AudioBufferList inAaudioBufferList;
CMBlockBufferRef blockBuffer;
CMSampleBufferGetAudioBufferListWithRetainedBlockBuffer(sampleBuffer, NULL, &inAaudioBufferList, sizeof(inAaudioBufferList), NULL, NULL, 0, &blockBuffer);
NSAssert(inAaudioBufferList.mNumberBuffers == 1, nil);

uint32_t bufferSize = inAaudioBufferList.mBuffers[0].mDataByteSize;
uint8_t *buffer = (uint8_t *)malloc(bufferSize);
memset(buffer, 0, bufferSize);
AudioBufferList outAudioBufferList;
outAudioBufferList.mNumberBuffers = 1;
outAudioBufferList.mBuffers[0].mNumberChannels = inAaudioBufferList.mBuffers[0].mNumberChannels;
outAudioBufferList.mBuffers[0].mDataByteSize = bufferSize;
outAudioBufferList.mBuffers[0].mData = buffer;

UInt32 ioOutputDataPacketSize = 1;

NSAssert(AudioConverterFillComplexBuffer(audioConverter, inInputDataProc, &inAaudioBufferList, &ioOutputDataPacketSize, &outAudioBufferList, NULL) == 0, nil);

NSData *data = [NSData dataWithBytes:outAudioBufferList.mBuffers[0].mData length:outAudioBufferList.mBuffers[0].mDataByteSize];

free(buffer);
CFRelease(blockBuffer);

inInputDataProc()实施：

OSStatus inInputDataProc(AudioConverterRef inAudioConverter, UInt32 *ioNumberDataPackets, AudioBufferList *ioData, AudioStreamPacketDescription **outDataPacketDescription, void *inUserData)
{
    AudioBufferList audioBufferList = *(AudioBufferList *)inUserData;

    ioData->mBuffers[0].mData = audioBufferList.mBuffers[0].mData;
    ioData->mBuffers[0].mDataByteSize = audioBufferList.mBuffers[0].mDataByteSize;

    return  noErr;
}

现在，data保存了我的原始AAC，我用适当的ADTS头封装到ADTS帧中，这些ADTS帧的序列是可播放的AAC文档。

但是我并不像我想的那样理解这段代码。一般来说，我不懂音频...我刚刚在博客，论坛和文档之后以某种方式写了它，现在它已经有效但我不知道为什么以及如何更改一些参数。所以这是我的问题：

我需要在HW编码器占用期间使用此转换器（AVAssetWriter）。这就是我通过AudioConverterNewSpecific()而不是AudioConverterNew()制作SW转换器的原因。但现在设置outAudioStreamBasicDescription.mFormatID = kAudioFormatMPEG4AAC_HE;不起作用。找不到AudioClassDescription。即使mFormatFlags设置为0.使用kAudioFormatMPEG4AAC（kMPEG4Object_AAC_SSR）而不是kAudioFormatMPEG4AAC_HE，我又失去了什么？我应该将什么用于直播？ kMPEG4Object_AAC_SSR或kMPEG4Object_AAC_Main？
如何正确更改采样率？例如，如果我将outAudioStreamBasicDescription.mSampleRate设置为22050或8000，则音频播放就会变慢。我在ADTS标头中设置了与outAudioStreamBasicDescription.mSampleRate相同的频率的采样频率索引。
如何更改比特率？ ffmpeg -i显示了生成的aac的这个信息： Stream #0:0: Audio: aac, 44100 Hz, mono, fltp, 64 kb/s。例如，如何将其更改为16 kbps？随着我降低频率，比特率正在下降，但我相信这不是唯一的方法吗？正如我在2中提到的那样，通过降低频率来损坏播放。
如何计算buffer的大小？现在我把它设置为uint32_t bufferSize = inAaudioBufferList.mBuffers[0].mDataByteSize;因为我认为压缩格式不会比未压缩格式大......但不是不必要地太多了吗？
如何正确设置ioOutputDataPacketSize？如果我正确获得文档，我应将其设置为UInt32 ioOutputDataPacketSize = bufferSize / outAudioStreamBasicDescription.mBytesPerPacket;但mBytesPerPacket为0.如果我将其设置为0，AudioConverterFillComplexBuffer()将返回错误。如果我将它设置为1，它可以工作，但我不知道为什么......
在inInputDataProc()中有3个“out”参数。我只设置ioData。我还应该设置ioNumberDataPackets和outDataPacketDescription吗？为什么以及如何？

Answer 1

在将音频馈送到AAC转换器之前，您可能需要使用重采样音频设备更改原始音频数据的采样率。否则，AAC标头和音频数据之间将不匹配。

将PCM（CMSampleBufferRef）编码到iOS上的AAC - 如何设置频率和比特率？

1 个答案: