内容简介:iOS中将压缩音频数据(PCM)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.
iOS中将压缩音频数据(PCM)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:
- Core Audio基本原理:简书, 掘金 , 博客
- Audio Queue解析:掘金, 简书 , 博客
- Audio Queue实战:简书, 博客 , 掘金
- 音频文件录制:简书, 掘金 , 博客
- 音视频基础知识
- C,C++基本知识
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 ioOutputDataPackets outputPacketDescriptions
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
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