海思3536輸出PCM多路混音
阿新 • • 發佈：2021-01-22
由於專案需求，需要將進行多路混音的功能。
對於海思音訊庫的處理程式
//初始化 int AudioInit(AIO_MODE_E enWorkmode) { HI_S32 s32Ret = HI_SUCCESS; ADEC_CHN AdChn = 0; AUDIO_DEV AoDev = 0; ADEC_CHN_ATTR_S stAdecAttr; ADEC_ATTR_AAC_S stAdecAac; stAdecAttr.enType = PT_AAC; stAdecAttr.u32BufSize = 20; stAdecAttr.enMode = ADEC_MODE_STREAM;/* propose use pack mode in your app */ stAdecAttr.pValue = &stAdecAac; stAdecAttr.enMode = ADEC_MODE_STREAM; /* aac should be stream mode */ stAdecAac.enTransType = AAC_TRANS_TYPE_ADTS; /* create adec chn*/ s32Ret = HI_MPI_ADEC_CreateChn(AdChn, &stAdecAttr); if (s32Ret) { printf("%s: HI_MPI_ADEC_CreateChn(%d) failed with %#x!\n", __FUNCTION__,AdChn,s32Ret); return s32Ret; } AIO_ATTR_S stAioAttr; stAioAttr.enSamplerate = 44100; // 輸出取樣率可以是32k, 44.1k或者48k stAioAttr.enBitwidth = AUDIO_BIT_WIDTH_16; stAioAttr.enWorkmode = enWorkmode; stAioAttr.enSoundmode = AUDIO_SOUND_MODE_STEREO; stAioAttr.u32EXFlag = 1; stAioAttr.u32FrmNum = MAX_AUDIO_FRAME_NUM; stAioAttr.u32PtNumPerFrm = 1024; stAioAttr.u32ChnCnt = 2; //立體聲模式需要設定為2 stAioAttr.u32ClkChnCnt = 1; stAioAttr.u32ClkSel = 0; if (stAioAttr.u32ClkChnCnt == 0) { stAioAttr.u32ClkChnCnt = stAioAttr.u32ChnCnt; } //A0屬性設定 s32Ret = HI_MPI_AO_SetPubAttr(AoDev, &stAioAttr); if (HI_SUCCESS != s32Ret) { printf("%sAoDev HI_MPI_AO_SetPubAttr(%d) failed with %#x!\n", __FUNCTION__, AoDev, s32Ret); return HI_FAILURE; } //使能AO s32Ret = HI_MPI_AO_Enable(AoDev); if (HI_SUCCESS != s32Ret) { printf("%s: HI_MPI_AO_Enable(%d) failed with %#x!\n", __FUNCTION__, AoDev, s32Ret); return HI_FAILURE; } //使能AO通道 s32Ret = HI_MPI_AO_EnableChn(AoDev, 0); if (HI_SUCCESS != s32Ret) { printf("%s: HI_MPI_AO_EnableChn(0) failed with %#x!\n", __FUNCTION__, s32Ret); return HI_FAILURE; } } //ADEC傳送幀函式 int AdecSendFrame(unsigned int dwAdeChn, unsigned long long u64FrameNo, unsigned char * pbyBuffer, unsigned int dwDataLen,HI_BOOL bBlock) { HI_S32 s32Ret = HI_SUCCESS; AUDIO_STREAM_S stAudioStream; stAudioStream.pStream = pbyBuffer; stAudioStream.u32Len = dwDataLen; stAudioStream.u64TimeStamp = u64FrameNo; s32Ret = HI_MPI_ADEC_SendStream(dwAdeChn, &stAudioStream, bBlock); if(HI_SUCCESS != s32Ret) { printf("%s: HI_MPI_ADEC_SendStream(%d) failed with %#x!\n",\ __FUNCTION__, dwAdeChn, s32Ret); } return s32Ret; } //ADEC獲取幀函式 int GetAdecEncodedFrame(unsigned int dwAencChn,AUDIO_FRAME_INFO_S *ptFrmInfo,HI_BOOL bBlock) { HI_S32 s32Ret = HI_SUCCESS; if(ptFrmInfo == NULL) { return HI_FAILURE; } s32Ret = HI_MPI_ADEC_GetFrame(dwAencChn, ptFrmInfo, bBlock); if (s32Ret != HI_SUCCESS && s32Ret != HI_ERR_AENC_BUF_EMPTY) { printf("Failed to get frame from ADEC, s32Ret: %#x\n", s32Ret); return s32Ret; } return HI_SUCCESS; } //釋放ADEC幀函式 void ReleaseADECFrame(unsigned int dwAencChn,AUDIO_FRAME_INFO_S *ptFrmInfo) { HI_MPI_ADEC_ReleaseFrame(dwAencChn,ptFrmInfo); } //A0傳送幀函式 int MPP_SendAudioFrameToAo(short *pLeftPcmBuffer, short *pRightPcmBuffer, int length,int sdwTimeoutMs) { AUDIO_DEV AoDev = 0; AUDIO_FRAME_S aoFrame; aoFrame.enBitwidth = AUDIO_BIT_WIDTH_16; aoFrame.enSoundmode = AUDIO_SOUND_MODE_STEREO; aoFrame.pVirAddr[0] = pLeftPcmBuffer; aoFrame.pVirAddr[1] = pRightPcmBuffer; aoFrame.u32Len = length; HI_S32 ret = HI_MPI_AO_SendFrame(AoDev, 0, &aoFrame, sdwTimeoutMs); if (ret != HI_SUCCESS) { printf("HI_MPI_AO_SendFrame %#x\n", ret); return ret; } return HI_SUCCESS; }
混音處理程式碼：
PCM多路混音演算法參考：https://blog.csdn.net/daska110/article/details/80322696
#define AUDIO_DATA_TYPE short #define AUDIO_DATA_TYPE_MAX (32767) // 2^15（short） #define AUDIO_DATA_TYPE_MIN (-32768) #define WIDEN_TEMP_TYPE int // 4位元組有符號的中間變數，用於混音時防止溢位 #define AUDIO_MAX_CHN (32) typedef int SDWORD #include <math.h> #include <vector> #include <mutex> using namespace std; class AudioPcmMix { public: ~AudioPcmMix(); static AudioPcmMix* GetInstance(void); // 疊加，然後歸一化 void AddAndNormalization(vector< vector<AUDIO_DATA_TYPE> >& allMixingSounds, DWORD dwRawDataCnt, vector<AUDIO_DATA_TYPE>& rawDataBuffer); SDWORD PcmMixHandle(); void DelAudioChnID(SDWORD sdwAudioChnID); void AddAudioChnID(SDWORD sdwAudioChnID); private: AudioPcmMix(); public: vector<AUDIO_DATA_TYPE> m_leftRawDataBuffer; vector<AUDIO_DATA_TYPE> m_rightRawDataBuffer; SDWORD m_sdwFrameLen; private: vector<SDWORD> m_audioChnIDList; std::mutex m_audioMutex; SDWORD m_getFrameRet[AUDIO_MAX_CHN]; };
AudioPcmMix::AudioPcmMix() :m_sdwFrameLen(0),m_audioChnIDList(AUDIO_MAX_CHN,-1) { } AudioPcmMix::~AudioPcmMix() { } AudioPcmMix* AudioPcmMix::GetInstance(void) { static AudioPcmMix mixStance; return &mixStance; } void AudioPcmMix::AddAndNormalization(vector< vector<AUDIO_DATA_TYPE> >& allMixingSounds, DWORD dwRawDataCnt, vector<AUDIO_DATA_TYPE>& rawDataBuffer) { WIDEN_TEMP_TYPE Sum = 0; // 用更大的範圍來表示（用有符號的int，而不要用無符號的DWORD） double decayFactor = 1; // 衰減因子（防止溢位） for (int i = 0; i < dwRawDataCnt; ++i) { Sum = 0; // 復位疊加的值 for (int wavNum = 0; wavNum < allMixingSounds.size(); ++wavNum) { Sum += allMixingSounds[wavNum][i]; } Sum *= decayFactor; // 將衰減因子作用在疊加的音訊上 // 計算衰減因子 // 1. 疊加之後，會溢位，計算溢位的倍數（即衰減因子） if (Sum > AUDIO_DATA_TYPE_MAX) { decayFactor = static_cast<double>(AUDIO_DATA_TYPE_MAX) / static_cast<double>(Sum); // 算大了，就用小數0.8衰減 Sum = AUDIO_DATA_TYPE_MAX; } else if (Sum < AUDIO_DATA_TYPE_MIN) { decayFactor = static_cast<double>(AUDIO_DATA_TYPE_MIN) / static_cast<double>(Sum); // 算小了，就用大數1.2增加 Sum = AUDIO_DATA_TYPE_MIN; } // 2. 衰減因子的平滑（為了防止個別點偶然的溢位） if (decayFactor < 1) { decayFactor += static_cast<double>(1 - decayFactor) / static_cast<double>(32); } rawDataBuffer.push_back(AUDIO_DATA_TYPE(Sum)); // 把int再強制轉換回為short } } void AudioPcmMix::DelAudioChnID(SDWORD sdwAudioChnID) { m_audioMutex.lock(); for(auto it = m_audioChnIDList.begin();it != m_audioChnIDList.end();it++) { if(*it == sdwAudioChnID) { m_getFrameRet[sdwAudioChnID] = -1; //丟棄這一幀的資料 m_sdwFrameLen = 0; m_audioChnIDList.erase(it); break; } } m_audioMutex.unlock(); } void AudioPcmMix::AddAudioChnID(SDWORD sdwAudioChnID) { m_audioMutex.lock(); for(auto it = m_audioChnIDList.begin();it != m_audioChnIDList.end();it++) { if(*it == sdwAudioChnID) { m_audioMutex.unlock(); return; } } m_audioChnIDList.push_back(sdwAudioChnID); m_audioMutex.unlock(); } SDWORD AudioPcmMix::PcmMixHandle() { m_sdwFrameLen = 0; //資料處理時加鎖 m_audioMutex.lock() if(m_audioChnIDList.size() == 0) { m_audioMutex.unlock(); return -1; } memset(m_getFrameRet,-1,AUDIO_MAX_CHN); AUDIO_FRAME_INFO_S stFrmInfo[AUDIO_MAX_CHN]; vector< vector<AUDIO_DATA_TYPE> > allMixingSounds; m_leftRawDataBuffer.clear(); m_rightRawDataBuffer.clear(); for (auto &it : m_audioChnIDList) { SDWORD sdwChnID = it; if( (sdwChnID < 0) || (sdwChnID >= AUDIO_MAX_CHN)) { continue; } printf("sdwChnID = %d size() %d\n",sdwChnID,m_audioChnIDList.size()); //需要進行阻塞處理，否則混的聲音將不對 m_getFrameRet[sdwChnID] = GetAdecEncodedFrame(sdwChnID, stFrmInfo + sdwChnID, Block); printf("=========== channel[%d] get frame finish: %#x================", sdwChnID, m_getFrameRet[sdwChnID]); if (m_getFrameRet[sdwChnID] == 0) { printf("channel[%d] frame len: %#x\n", sdwChnID, stFrmInfo[sdwChnID].pstFrame->u32Len); if ((m_sdwFrameLen == 0) || (m_sdwFrameLen > stFrmInfo[sdwChnID].pstFrame->u32Len)) { m_sdwFrameLen = stFrmInfo[sdwChnID].pstFrame->u32Len; } } } for (auto &it : m_audioChnIDList) { SDWORD sdwChnID = it; if( (sdwChnID < 0) || (sdwChnID >= AUDIO_MAX_CHN) ) { continue; } if ((m_getFrameRet[sdwChnID] == 0)) { AUDIO_DATA_TYPE *tmp = (AUDIO_DATA_TYPE *)stFrmInfo[sdwChnID].pstFrame->pVirAddr[0]; if(tmp != nullptr) { allMixingSounds.push_back(vector<AUDIO_DATA_TYPE>(tmp,tmp + m_sdwFrameLen/sizeof(AUDIO_DATA_TYPE))); } else { printf("Get Left Frame Data Error\n"); } } } //左聲道進行混音 AddAndNormalization(allMixingSounds,m_sdwFrameLen/2,m_leftRawDataBuffer); allMixingSounds.clear(); for (auto &it : m_audioChnIDList) { SDWORD sdwChnID = it; if( (sdwChnID < 0) || (sdwChnID >= AUDIO_MAX_CHN) ) { continue; } if ((m_getFrameRet[sdwChnID] == 0)) { AUDIO_DATA_TYPE *tmp = (AUDIO_DATA_TYPE *)stFrmInfo[sdwChnID].pstFrame->pVirAddr[1]; if(tmp != nullptr) { allMixingSounds.push_back(vector<AUDIO_DATA_TYPE>(tmp,tmp + m_sdwFrameLen/sizeof(AUDIO_DATA_TYPE))); } else { printf("Get Right Frame Data Error\n"); } } } //右聲道進行混音 AddAndNormalization(allMixingSounds,m_sdwFrameLen/2,m_rightRawDataBuffer); for (auto &it : m_audioChnIDList) { SDWORD sdwChnID = it; if( (sdwChnID < 0) || (sdwChnID >= AUDIO_MAX_CHN) ) { continue; } if ((m_getFrameRet[sdwChnID] != 0)) { continue; } //釋放獲取的幀 ReleaseADECFrame(sdwChnID, stFrmInfo + sdwChnID); } m_audioMutex.unlock(); return 0; }
混音處理執行緒：
void AudioPcmMixThread(void)
{
prctl(PR_SET_NAME,"AudioPcmMixThread");
SDWORD sdwTimeoutMs = 0;
SDWORD sdwRet = -1;
while(1)
{
//獲取PCM混音資料
if( AudioPcmMix::GetInstance()->PcmMixHandle() == 0)
{
//將PCM資料傳送到AO
if(AudioPcmMix::GetInstance()->m_sdwFrameLen > 0)
{
sdwRet = SendAudioFrameToAo(AudioPcmMix::GetInstance()->m_leftRawDataBuffer.data(),AudioPcmMix::GetInstance()->m_rightRawDataBuffer.data(),AudioPcmMix::GetInstance()->m_sdwFrameLen,sdwTimeoutMs);
}
usleep(1*1000);
}
else
{
sleep(1);
}
}
}
由於我們專案使用的是AAC音訊進行資料傳輸，因此需要使用ADEC庫將AAC解碼為PCM，然後在將PCM混音，最後傳送到A0中去，實現多路混音輸出的目的。
在程式當中，對於每個AAC的音訊連線都建立一個執行緒進行處理，將收到的AAC資料，通過AdecSendFrame進行傳送解碼，然後在混音處理執行緒當中統一通過GetAdecEncodedFrame取出解碼後的PCM資料，再進行多路混音處理。