大家好，我用rtmp推送H.264+AAC数据到服务器，然后登录客服端看视频的时候发现没声音，然后视频播放延时了十几秒吧，播放速度加快了，就好像快进一样。不知道什么问题。下面是程序 #define AAC_ADTS_HEADER_SIZE 7 static uint32_t find_start_code(uint8_t *buf, uint32_t zeros_in_startcode)    {      uint32_t info;      uint32_t i;          info = 1;      if ((info = (buf[zeros_in_startcode] != 1)? 0: 1) == 0)          return 0;             for (i = 0; i < zeros_in_startcode; i++)        if (buf != 0)        {         info = 0;         break;     };            return info;    }    uint8_t * get_nal(uint32_t *len, uint8_t **offset, uint8_t *start, uint32_t total) {     uint32_t info;     uint8_t *q ;     uint8_t *p  =  *offset;     *len = 0;     while(1) {         info =  find_start_code(p, 3);         if (info == 1)             break;         p++;         if ((p - start) >= total)             return NULL;     }     q = p + 4;     p = q;     while(1) {         info =  find_start_code(p, 3);         if (info == 1)             break;         p++;         if ((p - start) >= total)             return NULL;     }         *len = (p - q);     *offset = p;     return q; } uint8_t *get_adts(uint32_t *len, uint8_t **offset, uint8_t *start, uint32_t total) {     uint8_t *p  =  *offset;     uint32_t frame_len_1;     uint32_t frame_len_2;     uint32_t frame_len_3;     uint32_t frame_length;         if (total < AAC_ADTS_HEADER_SIZE) {         return NULL;     }     if ((p - start) >= total) {         return NULL;     }         if (p[0] != 0xff) {         return NULL;     }     if ((p[1] & 0xf0) != 0xf0) {         return NULL;     }     frame_len_1 = p[3] & 0x03;     frame_len_2 = p[4];     frame_len_3 = (p[5] & 0xe0) >> 5;     frame_length = (frame_len_1 << 11) | (frame_len_2 << 3) | frame_len_3;     *offset = p + frame_length;     *len = frame_length;     return p; } typedef unsigned long ULONG; typedef unsigned int UINT; typedef unsigned char BYTE; int main(int argc, char *argv[]) {         char serverStrBuf[100];         sprintf(serverStrBuf, "rtmp://%s:%s/%s/livestream_chn_%s",argv[1]/*server*/,argv[2]/*port*/,argv[3],/*myapp*/argv[4]/*channel*/);         printf("serverStrBuf is %s\n",serverStrBuf);         char vpipe[15];         char apipe[15];         sprintf(vpipe,"/tmp/vfifoChn%s",argv[4]);                sprintf(apipe,"/tmp/afifoChn%s",argv[4]);                 printf("vpipe is %s\n",vpipe);                printf("apipe is %s\n",apipe);     void*prtmp = rtmp_sender_alloc(serverStrBuf/*"rtmp://192.168.2.102:1936/myapp/livestream_chn_1"*/); //return handle     rtmp_sender_start_publish(prtmp, 0, 0);     uint8_t *audio_buf_offset ;     uint32_t audio_len;     uint8_t *p_audio;     int ret = 0;     int vpipeFD = open(vpipe/*"/tmp/vfifoChn1"*/, O_RDONLY|O_NONBLOCK);         if (vpipeFD < 0)         {                 printf("open FOR READ failed\n");         }     int apipeFD = open(apipe/*"/tmp/afifoChn1"*/, O_RDONLY|O_NONBLOCK);         if (apipeFD < 0)         {                 printf("open FOR aREAD failed\n");         }     ULONG nSampleRate = 8000;  // 采样率     UINT nChannels = 1;         // 声道数     UINT nBit = 16;             // 单样本位数     ULONG nInputSamples = 0;        //输入样本数     ULONG nMaxOutputBytes = 0;        //输出所需最大空间         ULONG nMaxInputBytes=0;     //输入最大字节     faacEncHandle hEncoder;                //aac句柄     faacEncConfigurationPtr pConfiguration;//aac设置指针     char* abuffer;         char* vbuffer;     BYTE* pbAACBuffer;   // (1) Open FAAC engine     hEncoder = faacEncOpen(nSampleRate, nChannels, &nInputSamples, &nMaxOutputBytes);//初始化aac句柄，同时获取最大输入样本，及编码所需最小字节         nMaxInputBytes=nInputSamples*nBit/8;//计算最大输入字节,跟据最大输入样本数         printf("nInputSamples:%d nMaxInputBytes:%d nMaxOutputBytes:%d\n", nInputSamples, nMaxInputBytes,nMaxOutputBytes);     if(hEncoder == NULL)     {         printf("[ERROR] Failed to call faacEncOpen()\n");         return -1;     }     pbAACBuffer = (char*)malloc(nMaxOutputBytes);     // (2.1) Get current encoding configuration     pConfiguration = faacEncGetCurrentConfiguration(hEncoder);//获取配置结构指针     pConfiguration->inputFormat = FAAC_INPUT_16BIT;         pConfiguration->outputFormat=1;         pConfiguration->useTns=true;         pConfiguration->useLfe=false;         pConfiguration->aacObjectType=LOW;         pConfiguration->mpegVersion = MPEG4;         pConfiguration->shortctl=SHORTCTL_NORMAL;         pConfiguration->quantqual=60;         pConfiguration->bandWidth=80000;         pConfiguration->bitRate=0;         printf("!!!!!!!!!!!!!pConfiguration->outputFormat is %d\n",pConfiguration->outputFormat);     // (2.2) Set encoding configuration     ret = faacEncSetConfiguration(hEncoder, pConfiguration);//设置配置，根据不同设置，耗时不一样         abuffer = (char*)malloc(nMaxInputBytes);         if(abuffer == NULL)                 exit(0);         vbuffer = (char*)malloc(25*1024);         if(vbuffer == NULL)                 exit(0);                char* output = (char*)malloc(10*1024);           static unsigned int audioCount = 0;         uint32_t start_time = RTMP_GetTime();         uint32_t timeCount = 0;         timeCount = 0;//RTMP_GetTime();     while (1) {         //        timeCount += 0.01;         //        start_time += timeCount;                 ret = read(vpipeFD, vbuffer, 25 * 1024);                 if(ret > 0){                              rtmp_sender_write_video_frame(prtmp, vbuffer, ret, timeCount, 0,start_time);                         timeCount += 2;                 }                 ret = read(apipeFD, abuffer, nMaxInputBytes - audioCount);                 if(ret >0){                         if(audioCount < 1*2048)        {                                 memcpy(output + audioCount,abuffer,ret);                                 audioCount += ret;                                 ret = 0;                                 if(audioCount >= 1*2048){                                         ret = audioCount;                                         audioCount = 0;                                 }                         }else{                                 ret = audioCount;                                 audioCount = 0;                         }                 }                 if(ret > 0){                         nInputSamples = ret/ (nBit / 8);                         // (3) Encode                         ret = faacEncEncode(hEncoder, (int*) output, nInputSamples, pbAACBuffer, nMaxOutputBytes);                         audio_buf_offset = pbAACBuffer;                         if (ret > 0){                                 p_audio = get_adts(&audio_len, &audio_buf_offset, pbAACBuffer, ret);                                 if (p_audio == NULL){                                         printf("p_audio is null,this should not happen!!!!\n");                                         continue;                                 }                                 rtmp_sender_write_audio_frame(prtmp, p_audio, audio_len, timeCount,start_time);                         }                                        }                 usleep(1);     } }

大家好，要想用hi3520生成aac音频文件，大概要怎么做呢，mdk有提供api吗

大家好，我想问下，hi3520能产生aac格式的音频吗

由于开发需要，我想添加自己的函数进去，但是一大堆Makefile好晕啊，不知道怎么改，有什么好的方法吗

大家好，运行mdk给的例程有错误， # ./sample_audio 23 /************************************/ press sample command as follows! 1:  start AI to AO loop 2:  send audio frame to AENC channel from AI, save them 3:  read audio stream from file,decode and send AO 4:  start AI to AO(Hdmi) loop q:  quit whole audio sample sample command:2 hello world ************* 1 AEC_Init error framelen(320) [Func]:MPI_AI_AecInit [Line]:369 [Info]:aec open fail. hello world ************* 2 hello world ************* 1 AEC_Init error framelen(320) [Func]:MPI_AI_AecInit [Line]:369 [Info]:aec open fail. hello world ************* 2 Ai(0,0) bind to AencChn:0 ok! open stream file:"audio_chn0.pcm" for aenc ok bind adec:0 to ao(0,0) ok please press twice ENTER to exit this sample 就这样卡住了

大家好，最近在搞rtmp推流，实现直播。想问下大家，vi -> vpss -> venc -> rtmp这个过程是怎样的呢。rtmp怎样读取编码后的数据才能实现直播呢，是不是编码后的数据放到一个文件里面，然后再用rtmp读取这个文件上传。麻烦大家帮帮忙，小弟不懂

各路大神好，小弟最近有个项目，实现1080P的网络摄像头，hi35系列的芯片哪一款可以呢