Merge branch 'dev' of https://gitee.com/xiongguangjie/ZLMediaKit into dev

src/Extension/H264Rtp.cpp

@@ -189,8 +189,52 @@ void H264RtpEncoder::inputFrame(const Frame::Ptr &frame) {
     auto len = frame->size() - frame->prefixSize();
     auto pts = frame->pts();
     auto nal_type = H264_TYPE(ptr[0]);
-    auto packet_size = getMaxSize() - 2;
+    if(nal_type == H264Frame::NAL_SEI || nal_type == H264Frame::NAL_AUD){
+        return;
+    }
 
+    if(nal_type == H264Frame::NAL_SPS){
+        _sps = std::string(ptr,len);
+        return;
+    }
+
+    if(nal_type == H264Frame::NAL_PPS){
+        _pps = std::string(ptr,len);
+        return;
+    }
+
+    if(!_last_frame){
+        _last_frame = frame;
+        return;
+    }
+    // 上一帧打包，保证rtp 的mark是正确的
+    bool isMark = _last_frame->pts() != frame->pts();
+    ptr = _last_frame->data() + _last_frame->prefixSize();
+    len = _last_frame->size() - _last_frame->prefixSize();
+    pts = _last_frame->pts();
+    nal_type = H264_TYPE(ptr[0]);
+    if(nal_type == H264Frame::NAL_IDR && (ptr[1]&0x80))
+    {// 保证每一个I帧前都有SPS与PPS ,为了兼容webrtc 需要在一个rtp包中，并且只能是 STAP-A 
+     // https://blog.csdn.net/momo0853/article/details/88872873
+     // 多slice 一帧的情况下检查 first_mb_in_slice 是否为0 表示其为一帧的开始，SPS PPS 只有在帧开始时，才插入
+        auto rtp  = makeRtp(getTrackType(), nullptr,_sps.size()+_pps.size()+2*2+1,false,pts);
+        uint8_t *payload = rtp->getPayload();
+        payload[0] = 24;
+        payload[1] = _sps.size() >> 8;
+        payload[2] = _sps.size() & 0xff;
+        memcpy(payload+3,(uint8_t *) _sps.data(),_sps.size());
+
+        payload[_sps.size()+3] = _pps.size() >> 8;
+        payload[_sps.size()+4] = _pps.size() & 0xff;
+
+        memcpy(payload+3+_sps.size()+2,(uint8_t *) _pps.data(),_pps.size());
+        RtpCodec::inputRtp(rtp,true);
+    }
+
+
+
+    auto packet_size = getMaxSize() - 2;
+    //InfoL<<"nal type = "<<nal_type<<" pts="<<pts<<" len="<<len;
     //末尾5bit为nalu type，固定为28(FU-A)
     auto fu_char_0 = (ptr[0] & (~0x1F)) | 28;
     auto fu_char_1 = nal_type;
@@ -208,7 +252,7 @@ void H264RtpEncoder::inputFrame(const Frame::Ptr &frame) {
             }
 
             //传入nullptr先不做payload的内存拷贝
-            auto rtp = makeRtp(getTrackType(), nullptr, packet_size + 2, fu_flags->end_bit, pts);
+            auto rtp = makeRtp(getTrackType(), nullptr, packet_size + 2, fu_flags->end_bit && isMark, pts);
             //rtp payload 负载部分
             uint8_t *payload = rtp->getPayload();
             //FU-A 第1个字节
@@ -218,15 +262,25 @@ void H264RtpEncoder::inputFrame(const Frame::Ptr &frame) {
             //H264 数据
             memcpy(payload + 2, (uint8_t *) ptr + offset, packet_size);
             //输入到rtp环形缓存
-            RtpCodec::inputRtp(rtp, fu_flags->start_bit && nal_type == H264Frame::NAL_IDR);
+            RtpCodec::inputRtp(rtp, false);
 
             offset += packet_size;
             fu_flags->start_bit = 0;
         }
     } else {
         //如果帧长度不超过mtu, 则按照Single NAL unit packet per H.264 方式打包
-        makeH264Rtp(ptr, len, false, false, pts);
+        //为了兼容性 webrtc使用 STAP-A 打包
+        auto rtp  = makeRtp(getTrackType(), nullptr,len+3,isMark,pts);
+        uint8_t *payload = rtp->getPayload();
+        payload[0] = (ptr[0] & (~0x1F)) | 24;
+        payload[1] = len >> 8;
+        payload[2] = len & 0xff;
+        memcpy(payload+3,(uint8_t *) ptr,len);
+        RtpCodec::inputRtp(rtp,false);
+        //makeH264Rtp(ptr, len, false, false, pts);
     }
+
+    _last_frame = frame;
 }
 
 void H264RtpEncoder::makeH264Rtp(const void* data, size_t len, bool mark, bool gop_pos, uint32_t uiStamp) {

src/Extension/H264Rtp.h

@@ -83,6 +83,9 @@ public:
 
 private:
     void makeH264Rtp(const void *pData, size_t uiLen, bool bMark,  bool gop_pos, uint32_t uiStamp);
+    string _sps;
+    string _pps;
+    Frame::Ptr _last_frame;
 };
 
 }//namespace mediakit{