需求
iOS中将压缩音频数据(如AAC)进行解码以得到原始音频数据类型:线性PCM.
本例最终实现的是通过Audio Queue采集到AAC压缩数据,将其解码为PCM数据,并将解码后的PCM数据以录制的形式保存在沙盒中.可调整解码后采样率,解码器类型等参数.
本例可拓展,不仅仅解码AAC音频数据流,还可以是音频文件,视频文件中的音频等等.
实现原理
利用Audio Toolbox Framework中的Audio Converter可以实现音频数据解码,即将AAC数据转为原始音频数据PCM.
阅读前提:
GitHub地址(附代码) : 音频解码
简书地址 : 音频解码
掘金地址 : 音频解码
博客地址 : 音频解码
1.初始化
1.1. 初始化解码器
初始化解码器实例, 通过指定原始数据格式,最终解码后的格式,采样率,以及使用硬编还是软编,以下是具体步骤.
- (instancetype)initWithSourceFormat:(AudioStreamBasicDescription)sourceFormat destFormatID:(AudioFormatID)destFormatID sampleRate:(float)sampleRate isUseHardwareDecode:(BOOL)isUseHardwareDecode {
if (self = [super init]) {
mSourceFormat = sourceFormat;
mAudioConverter = [self configureDecoderBySourceFormat:sourceFormat
destFormat:&mDestinationFormat
destFormatID:destFormatID
sampleRate:sampleRate
isUseHardwareDecode:isUseHardwareDecode];
}
return self;
}
复制代码1.2. 配置解码后ASBD音频流信息
AudioStreamBasicDescription destinationFormat = {0};
destinationFormat.mSampleRate = sampleRate;
if (destFormatID != kAudioFormatLinearPCM) {
NSLog(@"Not get compression format after decoding !");
return NULL;
} else {
destinationFormat.mFormatID = destFormatID;
destinationFormat.mChannelsPerFrame = sourceFormat.mChannelsPerFrame;
destinationFormat.mFormatID = kAudioFormatLinearPCM;
destinationFormat.mFormatFlags = (kAudioFormatFlagIsSignedInteger | kAudioFormatFlagIsPacked);
destinationFormat.mFramesPerPacket = kXDXAudioPCMFramesPerPacket;
destinationFormat.mBitsPerChannel = KXDXAudioBitsPerChannel;
destinationFormat.mBytesPerFrame = destinationFormat.mBitsPerChannel / 8 *destinationFormat.mChannelsPerFrame;
destinationFormat.mBytesPerPacket = destinationFormat.mBytesPerFrame * destinationFormat.mFramesPerPacket;
destinationFormat.mReserved = 0;
}
memcpy(destFormat, &destinationFormat, sizeof(AudioStreamBasicDescription));
复制代码对音频做解码操作,实际就是将压缩数据格式如AAC格式转为线性PCM原始音频数据,通过kAudioFormatProperty_FormatInfo属性可以自动获取指定音频格式的参数信息.
1.3. 选择解码器类型
AudioClassDescription结构体描述了系统使用音频解码器信息,其中最重要的就是使用硬编或软编。然后解码器的数量，即数组的个数，由当前的声道数决定。
//获取解码器的描述信息
AudioClassDescription *audioClassDesc = [self getAudioCalssDescriptionWithType:destFormatID fromManufacture:kAppleHardwareAudioCodecManufacturer];
...
- (AudioClassDescription *)getAudioCalssDescriptionWithType:(AudioFormatID)type fromManufacture:(uint32_t)manufacture {
static AudioClassDescription desc;
UInt32 decoderSpecific = type;
UInt32 size;
OSStatus status = AudioFormatGetPropertyInfo(kAudioFormatProperty_Decoders,
sizeof(decoderSpecific),
&decoderSpecific,
&size);
if (status != noErr) {
NSLog(@"Error！：硬解码AAC get info 失败, status= %d", (int)status);
return nil;
}
//计算aac解码器的个数
unsigned int count = size / sizeof(AudioClassDescription);
//创建一个包含count个解码器的数组
AudioClassDescription description[count];
//将满足aac解码的解码器的信息写入数组
status = AudioFormatGetProperty(kAudioFormatProperty_Encoders,
sizeof(decoderSpecific),
&decoderSpecific,
&size,
&description);
if (status != noErr) {
NSLog(@"Error！：硬解码AAC get propery 失败, status= %d", (int)status);
return nil;
}
for (unsigned int i = 0; i < count; i++) {
if (type == description[i].mSubType && manufacture == description[i].mManufacturer) {
desc = description[i];
return &desc;
}
}
return nil;
}
复制代码注意：硬解即利用设备GPU硬件完成高效解码，降低CPU消耗. 软解就是传统的通过CPU计算。
1.4. 创建解码器
AudioConverterNewSpecific: 通过指定解码器来创建audio converter实例对象.第3,4个 分别是解码器的数量与解码器描述,同上,与声道数保持一致.
// Create the AudioConverterRef.
AudioConverterRef converter = NULL;
if (![self checkError:AudioConverterNewSpecific(&sourceFormat, &destinationFormat, destinationFormat.mChannelsPerFrame, audioClassDesc, &converter) withErrorString:@"Audio Converter New failed"]) {
return NULL;
}else {
printf("Audio converter create successful \n");
}
复制代码2.解码
2.1. 计算解码数据大小
注意,当使用Audio Convert无论做编解码,每次都需要1024个采样点才能完成一次转换,此值是固定的.
根据解码器的采样点,计算解码出音频数据的大小.因为线性PCM的数据可以通过公式算出,即数据包数量*声道数*每个数据包中字节数.
// Note: audio convert must set 1024.
UInt32 ioOutputDataPackets = kIOOutputDataPackets;
UInt32 outputBufferSize = (UInt32)(ioOutputDataPackets * destFormat.mChannelsPerFrame * destFormat.mBytesPerFrame);
复制代码2.2. 为解码后音频数据预分配内存
我们可以将2.1中算出的size为这个Buffer list分配内存.
// Set up output buffer list.
// Set up output buffer list.
AudioBufferList fillBufferList = {0};
fillBufferList.mNumberBuffers = 1;
fillBufferList.mBuffers[0].mNumberChannels = destFormat.mChannelsPerFrame;
fillBufferList.mBuffers[0].mDataByteSize = outputBufferSize;
fillBufferList.mBuffers[0].mData = malloc(outputBufferSize * sizeof(char));
复制代码2.3. 解码音频数据
解析AudioConverterFillComplexBuffer:用来解码音频数据.同时需要指定回调函数(C语言函数),
第二个参数即指定回调函数,此回调函数中主要做的是为即将解码的数据进行赋值,即我们要把原始音频数据赋值给回调函数中的ioData参数,这是我们在解码前最后一次控制原始音频数据,此回调函数执行后即完成了解码的过程,新的数据会填充到第五个参数中,也就是我们上面预定义的fillBufferList.
userInfo: 自定义一个结构体,用来与解码回调函数间交互以传递数据.在这里是将原始音频数据信息传给解码回调函数中.ioOutputDataPackets: 填入函数中时表示原始音频数据包的数量,而函数调用完成时表示转换后输出的音频数据包总数,注意,当我们做解码时,输出肯定为PCM类型数据,所以需要提供1024个AAC采样点.而做编码时会将PCM数据压缩成很多音频数据包,仅仅需要1个完整的PCM数据包即可.outputPacketDescriptions: 转换完成后,如果此参数非空,表示转换器输出使用的音频数据包描述,它必须提前分配好内存,以让转换器赋值到其中.
最终,我们将转换后得到的AAC数据以回调函数的形式传给调用者.
OSStatus DecodeConverterComplexInputDataProc(AudioConverterRef inAudioConverter,
UInt32 *ioNumberDataPackets,
AudioBufferList *ioData,
AudioStreamPacketDescription **outDataPacketDescription,
void *inUserData) {
XDXConverterInfoType *info = (XDXConverterInfoType *)inUserData;
if (info->sourceDataSize <= 0) {
ioNumberDataPackets = 0;
return -1;
}
*outDataPacketDescription = &info->packetDesc;
(*outDataPacketDescription)[0].mStartOffset = 0;
(*outDataPacketDescription)[0].mDataByteSize = info->sourceDataSize;
(*outDataPacketDescription)[0].mVariableFramesInPacket = 0;
ioData->mNumberBuffers = 1;
ioData->mBuffers[0].mData = info->sourceBuffer;
ioData->mBuffers[0].mNumberChannels = info->sourceChannelsPerFrame;
ioData->mBuffers[0].mDataByteSize = info->sourceDataSize;
return noErr;
}
- (void)decodeFormatByConverter:(AudioConverterRef)audioConverter sourceBuffer:(void *)sourceBuffer sourceBufferSize:(UInt32)sourceBufferSize sourceFormat:(AudioStreamBasicDescription)sourceFormat dest:(AudioStreamBasicDescription)destFormat completeHandler:(void(^)(AudioBufferList *destBufferList, UInt32 outputPackets, AudioStreamPacketDescription *outputPacketDescriptions))completeHandler {
...
XDXConverterInfoType userInfo = {0};
userInfo.sourceBuffer = sourceBuffer;
userInfo.sourceDataSize = sourceBufferSize;
userInfo.sourceChannelsPerFrame = sourceFormat.mChannelsPerFrame;
userInfo.packetDesc.mDataByteSize = (UInt32)sourceBufferSize;
userInfo.packetDesc.mStartOffset = 0;
userInfo.packetDesc.mVariableFramesInPacket = 0;
AudioStreamPacketDescription outputPacketDesc;
OSStatus status = AudioConverterFillComplexBuffer(audioConverter,
DecodeConverterComplexInputDataProc,
&userInfo,
&ioOutputDataPackets,
&fillBufferList,
&outputPacketDesc);
// if interrupted in the process of the conversion call, we must handle the error appropriately
if (status != noErr) {
if (status == kAudioConverterErr_HardwareInUse) {
printf("Audio Converter returned kAudioConverterErr_HardwareInUse!\n");
} else {
if (![self checkError:status withErrorString:@"AudioConverterFillComplexBuffer error!"]) {
return;
}
}
} else {
if (ioOutputDataPackets == 0) {
// This is the EOF condition.
status = noErr;
}
if (completeHandler) {
completeHandler(&fillBufferList, ioOutputDataPackets, &outputPacketDesc);
}
}
}
复制代码3. 模块对接
因为音频解码要依赖音频采集,所以我们这里以audio unit采集为例作示范,即使用audio unit采集pcm数据然后使用此模块解码得到aac数据.如需了解请参考如下链接
- GitHub地址(附代码) : Audio Unit Capture
- 简书地址 : Audio Unit Capture
- 掘金地址 : Audio Unit Capture
- 博客地址 : Audio Unit Capture
3.1. 初始化解码器
如下,在音频采集的类中声明一个解码器实例变量,然后初始化它. 仅仅需要设置原始数据格式,解码后的格式,采样率,使用硬编,软编即可.
@property (nonatomic, strong) XDXAduioDecoder *audioDecoder;
...
// audio decode: aac->pcm
self.audioDecoder = [[XDXAduioDecoder alloc] initWithSourceFormat:m_audioInfo->mDataFormat
destFormatID:kAudioFormatLinearPCM
sampleRate:48000
isUseHardwareDecode:YES];
复制代码3.2. 解码音频数据
在Audio Queue采集AAC音频数据的回调中将AAC数据送入解码器,然后在回调函数中将得到的PCM数据其写入文件.
注意: 直接用Audio Queue采集AAC类型音频数据,实际系统在其内部做了一次转换,即直接采集其实只能采原始PCM数据,直接用Audio Queue设置采集AAC相当于系统在内部为我们做了一次转换.
static void CaptureAudioDataCallback(void * inUserData,
AudioQueueRef inAQ,
AudioQueueBufferRef inBuffer,
const AudioTimeStamp * inStartTime,
UInt32 inNumPackets,
const AudioStreamPacketDescription* inPacketDesc) {
XDXAudioQueueCaptureManager *instance = (__bridge XDXAudioQueueCaptureManager *)inUserData;
[instance.audioDecoder decodeAudioWithSourceBuffer:inBuffer->mAudioData
sourceBufferSize:inBuffer->mAudioDataByteSize
completeHandler:^(AudioBufferList * _Nonnull destBufferList, UInt32 outputPackets, AudioStreamPacketDescription * _Nonnull outputPacketDescriptions) {
if (instance.isRecordVoice) {
[[XDXAudioFileHandler getInstance] writeFileWithInNumBytes:destBufferList->mBuffers->mDataByteSize
ioNumPackets:outputPackets
inBuffer:destBufferList->mBuffers->mData
inPacketDesc:outputPacketDescriptions];
}
free(destBufferList->mBuffers->mData);
}];
if (instance.isRunning) {
AudioQueueEnqueueBuffer(inAQ, inBuffer, 0, NULL);
}
}
复制代码4. 文件录制
此部分可参考另一篇文章: 音频文件录制
- 简书地址 : Audio File Record
- 掘金地址 : Audio File Record
- 博客地址 : Audio File Record
5. 释放解码器资源
如需释放内存,请保证解码器工作彻底结束后再释放内存.
- (void)freeEncoder {
if (mAudioConverter) {
AudioConverterDispose(mAudioConverter);
mAudioConverter = NULL;
}
}
复制代码