修复拼写错误

src/Extension/AAC.h

@@ -173,21 +173,21 @@ public:
      * 构造函数
      * @param aac_cfg aac两个字节的配置描述
      * @param sample_rate 音频采样率
-     * @param playload_type rtp playload type 默认98
+     * @param payload_type rtp payload type 默认98
      * @param bitrate 比特率
      */
     AACSdp(const string &aac_cfg,
            int sample_rate,
            int channels,
-           int playload_type = 98,
+           int payload_type = 98,
-           int bitrate = 128) : Sdp(sample_rate,playload_type){
+           int bitrate = 128) : Sdp(sample_rate,payload_type){
-        _printer << "m=audio 0 RTP/AVP " << playload_type << "\r\n";
+        _printer << "m=audio 0 RTP/AVP " << payload_type << "\r\n";
         _printer << "b=AS:" << bitrate << "\r\n";
-        _printer << "a=rtpmap:" << playload_type << " MPEG4-GENERIC/" << sample_rate << "/" << channels << "\r\n";
+        _printer << "a=rtpmap:" << payload_type << " MPEG4-GENERIC/" << sample_rate << "/" << channels << "\r\n";
 
         char configStr[32] = {0};
         snprintf(configStr, sizeof(configStr), "%02X%02X", (uint8_t)aac_cfg[0], (uint8_t)aac_cfg[1]);
-        _printer << "a=fmtp:" << playload_type << " streamtype=5;profile-level-id=1;mode=AAC-hbr;"
+        _printer << "a=fmtp:" << payload_type << " streamtype=5;profile-level-id=1;mode=AAC-hbr;"
                  << "sizelength=13;indexlength=3;indexdeltalength=3;config=" << configStr << "\r\n";
         _printer << "a=control:trackID=" << (int)TrackAudio << "\r\n";
     }

src/Extension/AACRtp.cpp

@@ -16,12 +16,12 @@ namespace mediakit{
 AACRtpEncoder::AACRtpEncoder(uint32_t ui32Ssrc,
                              uint32_t ui32MtuSize,
                              uint32_t ui32SampleRate,
-                             uint8_t ui8PlayloadType,
+                             uint8_t ui8PayloadType,
                              uint8_t ui8Interleaved) :
         RtpInfo(ui32Ssrc,
                 ui32MtuSize,
                 ui32SampleRate,
-                ui8PlayloadType,
+                ui8PayloadType,
                 ui8Interleaved){
 }
 

src/Extension/AACRtp.h

@@ -56,13 +56,13 @@ public:
      * @param ui32Ssrc ssrc
      * @param ui32MtuSize mtu 大小
      * @param ui32SampleRate 采样率
-     * @param ui8PlayloadType pt类型
+     * @param ui8PayloadType pt类型
      * @param ui8Interleaved rtsp interleaved 值
      */
     AACRtpEncoder(uint32_t ui32Ssrc,
                   uint32_t ui32MtuSize,
                   uint32_t ui32SampleRate,
-                  uint8_t ui8PlayloadType = 97,
+                  uint8_t ui8PayloadType = 97,
                   uint8_t ui8Interleaved = TrackAudio * 2);
     ~AACRtpEncoder() {}
 

src/Extension/Factory.cpp

@@ -115,7 +115,7 @@ RtpCodec::Ptr Factory::getRtpEncoderBySdp(const Sdp::Ptr &sdp) {
     }
     auto mtu = (sdp->getTrackType() == TrackVideo ? video_mtu : audio_mtu);
     auto sample_rate = sdp->getSampleRate();
-    auto pt = sdp->getPlayloadType();
+    auto pt = sdp->getPayloadType();
     auto interleaved = sdp->getTrackType() * 2;
     auto codec_id = sdp->getCodecId();
     switch (codec_id){

src/Extension/G711.h

@@ -83,16 +83,16 @@ public:
      * G711采样率固定为8000
      * @param codecId G711A G711U
      * @param sample_rate 音频采样率
-     * @param playload_type rtp playload
+     * @param payload_type rtp payload
      * @param bitrate 比特率
      */
     G711Sdp(CodecId codecId,
             int sample_rate,
             int channels,
-            int playload_type = 98,
+            int payload_type = 98,
-            int bitrate = 128) : Sdp(sample_rate,playload_type), _codecId(codecId){
+            int bitrate = 128) : Sdp(sample_rate,payload_type), _codecId(codecId){
-        _printer << "m=audio 0 RTP/AVP " << playload_type << "\r\n";
+        _printer << "m=audio 0 RTP/AVP " << payload_type << "\r\n";
-        _printer << "a=rtpmap:" << playload_type << (codecId == CodecG711A ? " PCMA/" : " PCMU/") << sample_rate  << "/" << channels << "\r\n";
+        _printer << "a=rtpmap:" << payload_type << (codecId == CodecG711A ? " PCMA/" : " PCMU/") << sample_rate  << "/" << channels << "\r\n";
         _printer << "a=control:trackID=" << (int)TrackAudio << "\r\n";
     }
 

src/Extension/G711Rtp.cpp

@@ -62,12 +62,12 @@ void G711RtpDecoder::onGetG711(const G711Frame::Ptr &frame) {
 G711RtpEncoder::G711RtpEncoder(uint32_t ui32Ssrc,
                                uint32_t ui32MtuSize,
                                uint32_t ui32SampleRate,
-                               uint8_t ui8PlayloadType,
+                               uint8_t ui8PayloadType,
                                uint8_t ui8Interleaved) :
         RtpInfo(ui32Ssrc,
                 ui32MtuSize,
                 ui32SampleRate,
-                ui8PlayloadType,
+                ui8PayloadType,
                 ui8Interleaved) {
 }
 

src/Extension/G711Rtp.h

@@ -58,13 +58,13 @@ public:
      * @param ui32Ssrc ssrc
      * @param ui32MtuSize mtu 大小
      * @param ui32SampleRate 采样率
-     * @param ui8PlayloadType pt类型
+     * @param ui8PayloadType pt类型
      * @param ui8Interleaved rtsp interleaved 值
      */
     G711RtpEncoder(uint32_t ui32Ssrc,
                    uint32_t ui32MtuSize,
                    uint32_t ui32SampleRate,
-                   uint8_t ui8PlayloadType = 0,
+                   uint8_t ui8PayloadType = 0,
                    uint8_t ui8Interleaved = TrackAudio * 2);
     ~G711RtpEncoder() {}
 

src/Extension/H264.h

@@ -289,18 +289,18 @@ public:
      *
      * @param sps 264 sps,不带0x00000001头
      * @param pps 264 pps,不带0x00000001头
-     * @param playload_type  rtp playload type 默认96
+     * @param payload_type  rtp payload type 默认96
      * @param bitrate 比特率
      */
     H264Sdp(const string &strSPS,
             const string &strPPS,
-            int playload_type = 96,
+            int payload_type = 96,
-            int bitrate = 4000) : Sdp(90000,playload_type) {
+            int bitrate = 4000) : Sdp(90000,payload_type) {
         //视频通道
-        _printer << "m=video 0 RTP/AVP " << playload_type << "\r\n";
+        _printer << "m=video 0 RTP/AVP " << payload_type << "\r\n";
         _printer << "b=AS:" << bitrate << "\r\n";
-        _printer << "a=rtpmap:" << playload_type << " H264/" << 90000 << "\r\n";
+        _printer << "a=rtpmap:" << payload_type << " H264/" << 90000 << "\r\n";
-        _printer << "a=fmtp:" << playload_type << " packetization-mode=1; profile-level-id=";
+        _printer << "a=fmtp:" << payload_type << " packetization-mode=1; profile-level-id=";
 
         char strTemp[100];
         uint32_t profile_level_id = 0;

src/Extension/H264Rtp.cpp

@@ -204,12 +204,12 @@ void H264RtpDecoder::onGetH264(const H264Frame::Ptr &frame) {
 H264RtpEncoder::H264RtpEncoder(uint32_t ui32Ssrc,
                                uint32_t ui32MtuSize,
                                uint32_t ui32SampleRate,
-                               uint8_t ui8PlayloadType,
+                               uint8_t ui8PayloadType,
                                uint8_t ui8Interleaved) :
         RtpInfo(ui32Ssrc,
                 ui32MtuSize,
                 ui32SampleRate,
-                ui8PlayloadType,
+                ui8PayloadType,
                 ui8Interleaved) {
 }
 

src/Extension/H264Rtp.h

@@ -62,13 +62,13 @@ public:
      * @param ui32Ssrc ssrc
      * @param ui32MtuSize mtu大小
      * @param ui32SampleRate 采样率，强制为90000
-     * @param ui8PlayloadType pt类型
+     * @param ui8PayloadType pt类型
      * @param ui8Interleaved rtsp interleaved
      */
     H264RtpEncoder(uint32_t ui32Ssrc,
                    uint32_t ui32MtuSize = 1400,
                    uint32_t ui32SampleRate = 90000,
-                   uint8_t ui8PlayloadType = 96,
+                   uint8_t ui8PayloadType = 96,
                    uint8_t ui8Interleaved = TrackVideo * 2);
     ~H264RtpEncoder() {}
 

src/Extension/H265.h

@@ -315,19 +315,19 @@ public:
      * 构造函数
      * @param sps 265 sps,不带0x00000001头
      * @param pps 265 pps,不带0x00000001头
-     * @param playload_type  rtp playload type 默认96
+     * @param payload_type  rtp payload type 默认96
      * @param bitrate 比特率
      */
     H265Sdp(const string &strVPS,
             const string &strSPS,
             const string &strPPS,
-            int playload_type = 96,
+            int payload_type = 96,
-            int bitrate = 4000) : Sdp(90000,playload_type) {
+            int bitrate = 4000) : Sdp(90000,payload_type) {
         //视频通道
-        _printer << "m=video 0 RTP/AVP " << playload_type << "\r\n";
+        _printer << "m=video 0 RTP/AVP " << payload_type << "\r\n";
         _printer << "b=AS:" << bitrate << "\r\n";
-        _printer << "a=rtpmap:" << playload_type << " H265/" << 90000 << "\r\n";
+        _printer << "a=rtpmap:" << payload_type << " H265/" << 90000 << "\r\n";
-        _printer << "a=fmtp:" << playload_type << " ";
+        _printer << "a=fmtp:" << payload_type << " ";
         _printer << "sprop-vps=";
         _printer << encodeBase64(strVPS) << "; ";
         _printer << "sprop-sps=";

src/Extension/H265Rtp.cpp

@@ -140,12 +140,12 @@ void H265RtpDecoder::onGetH265(const H265Frame::Ptr &frame) {
 H265RtpEncoder::H265RtpEncoder(uint32_t ui32Ssrc,
                                uint32_t ui32MtuSize,
                                uint32_t ui32SampleRate,
-                               uint8_t ui8PlayloadType,
+                               uint8_t ui8PayloadType,
                                uint8_t ui8Interleaved) :
         RtpInfo(ui32Ssrc,
                 ui32MtuSize,
                 ui32SampleRate,
-                ui8PlayloadType,
+                ui8PayloadType,
                 ui8Interleaved) {
 }
 

src/Extension/H265Rtp.h

@@ -63,13 +63,13 @@ public:
      * @param ui32Ssrc ssrc
      * @param ui32MtuSize mtu大小
      * @param ui32SampleRate 采样率，强制为90000
-     * @param ui8PlayloadType pt类型
+     * @param ui8PayloadType pt类型
      * @param ui8Interleaved rtsp interleaved
      */
     H265RtpEncoder(uint32_t ui32Ssrc,
                    uint32_t ui32MtuSize = 1400,
                    uint32_t ui32SampleRate = 90000,
-                   uint8_t ui8PlayloadType = 96,
+                   uint8_t ui8PayloadType = 96,
                    uint8_t ui8Interleaved = TrackVideo * 2);
     ~H265RtpEncoder() {}
 

src/Extension/Opus.h

@@ -80,15 +80,15 @@ public:
     /**
      * 构造opus sdp
      * @param sample_rate 音频采样率
-     * @param playload_type rtp playload
+     * @param payload_type rtp payload
      * @param bitrate 比特率
      */
     OpusSdp(int sample_rate,
             int channels,
-            int playload_type = 98,
+            int payload_type = 98,
-            int bitrate = 128) : Sdp(sample_rate,playload_type){
+            int bitrate = 128) : Sdp(sample_rate,payload_type){
-        _printer << "m=audio 0 RTP/AVP " << playload_type << "\r\n";
+        _printer << "m=audio 0 RTP/AVP " << payload_type << "\r\n";
-        _printer << "a=rtpmap:" << playload_type << " opus/" << sample_rate  << "/" << channels << "\r\n";
+        _printer << "a=rtpmap:" << payload_type << " opus/" << sample_rate  << "/" << channels << "\r\n";
         _printer << "a=control:trackID=" << (int)TrackAudio << "\r\n";
     }
 

src/Http/WebSocketClient.h

@@ -193,7 +193,7 @@ protected:
     //WebSocketSplitter override
 
     /**
-     * 收到一个webSocket数据包包头，后续将继续触发onWebSocketDecodePlayload回调
+     * 收到一个webSocket数据包包头，后续将继续触发onWebSocketDecodePayload回调
      * @param header 数据包头
      */
     void onWebSocketDecodeHeader(const WebSocketHeader &header) override{
@@ -205,9 +205,9 @@ protected:
      * @param header 数据包包头
      * @param ptr 负载数据指针
      * @param len 负载数据长度
-     * @param recved 已接收数据长度(包含本次数据长度)，等于header._playload_len时则接受完毕
+     * @param recved 已接收数据长度(包含本次数据长度)，等于header._payload_len时则接受完毕
      */
-    void onWebSocketDecodePlayload(const WebSocketHeader &header, const uint8_t *ptr, uint64_t len, uint64_t recved) override{
+    void onWebSocketDecodePayload(const WebSocketHeader &header, const uint8_t *ptr, uint64_t len, uint64_t recved) override{
         _payload.append((char *)ptr,len);
     }
 

src/Http/WebSocketSession.h

@@ -161,7 +161,7 @@ protected:
      * @param len
      * @param recved
      */
-    void onWebSocketDecodePlayload(const WebSocketHeader &packet,const uint8_t *ptr,uint64_t len,uint64_t recved) override {
+    void onWebSocketDecodePayload(const WebSocketHeader &packet,const uint8_t *ptr,uint64_t len,uint64_t recved) override {
         _remian_data.append((char *)ptr,len);
     }
 

src/Http/WebSocketSplitter.cpp

@@ -72,16 +72,16 @@ begin_decode:
 
         CHECK_LEN(1);
         _mask_flag = (*ptr & 0x80) >> 7;
-        _playload_len = (*ptr & 0x7F);
+        _payload_len = (*ptr & 0x7F);
         ptr += 1;
 
-        if (_playload_len == 126) {
+        if (_payload_len == 126) {
             CHECK_LEN(2);
-            _playload_len = (*ptr << 8) | *(ptr + 1);
+            _payload_len = (*ptr << 8) | *(ptr + 1);
             ptr += 2;
-        } else if (_playload_len == 127) {
+        } else if (_payload_len == 127) {
             CHECK_LEN(8);
-            _playload_len = ((uint64_t) ptr[0] << (8 * 7)) |
+            _payload_len = ((uint64_t) ptr[0] << (8 * 7)) |
                             ((uint64_t) ptr[1] << (8 * 6)) |
                             ((uint64_t) ptr[2] << (8 * 5)) |
                             ((uint64_t) ptr[3] << (8 * 4)) |
@@ -98,9 +98,9 @@ begin_decode:
         }
         _got_header = true;
         _mask_offset = 0;
-        _playload_offset = 0;
+        _payload_offset = 0;
         onWebSocketDecodeHeader(*this);
-        if(_playload_len == 0){
+        if(_payload_len == 0){
             onWebSocketDecodeComplete(*this);
         }
     }
@@ -109,19 +109,19 @@ begin_decode:
 
     uint64_t remain = len - (ptr - data);
     if(remain > 0){
-        uint64_t playload_slice_len = remain;
+        uint64_t payload_slice_len = remain;
-        if(playload_slice_len + _playload_offset > _playload_len){
+        if(payload_slice_len + _payload_offset > _payload_len){
-            playload_slice_len = _playload_len - _playload_offset;
+            payload_slice_len = _payload_len - _payload_offset;
         }
-        _playload_offset += playload_slice_len;
+        _payload_offset += payload_slice_len;
-        onPlayloadData(ptr,playload_slice_len);
+        onPayloadData(ptr, payload_slice_len);
 
-        if(_playload_offset == _playload_len){
+        if(_payload_offset == _payload_len){
             onWebSocketDecodeComplete(*this);
 
             //这是下一个包
-            remain -= playload_slice_len;
+            remain -= payload_slice_len;
-            ptr += playload_slice_len;
+            ptr += payload_slice_len;
             _got_header = false;
 
             if(remain > 0){
@@ -138,14 +138,14 @@ begin_decode:
     _remain_data.clear();
 }
 
-void WebSocketSplitter::onPlayloadData(uint8_t *ptr, uint64_t len) {
+void WebSocketSplitter::onPayloadData(uint8_t *data, uint64_t len) {
     if(_mask_flag){
-        for(int i = 0; i < len ; ++i,++ptr){
+        for(int i = 0; i < len ; ++i,++data){
-            *(ptr) ^= _mask[(i + _mask_offset) % 4];
+            *(data) ^= _mask[(i + _mask_offset) % 4];
         }
         _mask_offset = (_mask_offset + len) % 4;
     }
-    onWebSocketDecodePlayload(*this, _mask_flag ? ptr - len : ptr, len, _playload_offset);
+    onWebSocketDecodePayload(*this, _mask_flag ? data - len : data, len, _payload_offset);
 }
 
 void WebSocketSplitter::encode(const WebSocketHeader &header,const Buffer::Ptr &buffer) {

src/Http/WebSocketSplitter.h

@@ -51,7 +51,7 @@ public:
     uint8_t _reserved;
     Type _opcode;
     bool _mask_flag;
-    uint64_t _playload_len;
+    uint64_t _payload_len;
     vector<uint8_t > _mask;
 };
 
@@ -62,7 +62,7 @@ public:
 
     /**
      * 输入数据以便解包webSocket数据以及处理粘包问题
-     * 可能触发onWebSocketDecodeHeader和onWebSocketDecodePlayload回调
+     * 可能触发onWebSocketDecodeHeader和onWebSocketDecodePayload回调
      * @param data 需要解包的数据，可能是不完整的包或多个包
      * @param len 数据长度
      */
@@ -77,7 +77,7 @@ public:
     void encode(const WebSocketHeader &header,const Buffer::Ptr &buffer);
 protected:
     /**
-     * 收到一个webSocket数据包包头，后续将继续触发onWebSocketDecodePlayload回调
+     * 收到一个webSocket数据包包头，后续将继续触发onWebSocketDecodePayload回调
      * @param header 数据包头
      */
     virtual void onWebSocketDecodeHeader(const WebSocketHeader &header) {};
@@ -87,9 +87,9 @@ protected:
      * @param header 数据包包头
      * @param ptr 负载数据指针
      * @param len 负载数据长度
-     * @param recved 已接收数据长度(包含本次数据长度)，等于header._playload_len时则接受完毕
+     * @param recved 已接收数据长度(包含本次数据长度)，等于header._payload_len时则接受完毕
      */
-    virtual void onWebSocketDecodePlayload(const WebSocketHeader &header, const uint8_t *ptr, uint64_t len, uint64_t recved) {};
+    virtual void onWebSocketDecodePayload(const WebSocketHeader &header, const uint8_t *ptr, uint64_t len, uint64_t recved) {};
 
 
     /**
@@ -105,12 +105,12 @@ protected:
      */
     virtual void onWebSocketEncodeData(const Buffer::Ptr &buffer){};
 private:
-    void onPlayloadData(uint8_t *data,uint64_t len);
+    void onPayloadData(uint8_t *data, uint64_t len);
 private:
     string _remain_data;
     int _mask_offset = 0;
     bool _got_header = false;
-    uint64_t _playload_offset = 0;
+    uint64_t _payload_offset = 0;
 };
 
 } /* namespace mediakit */

src/Rtsp/RtpCodec.cpp

@@ -28,18 +28,18 @@ RtpPacket::Ptr RtpInfo::makeRtp(TrackType type, const void* data, unsigned int l
     pucRtp[2] = ui16RtpLen >> 8;
     pucRtp[3] = ui16RtpLen & 0x00FF;
     pucRtp[4] = 0x80;
-    pucRtp[5] = (mark << 7) | _ui8PlayloadType;
+    pucRtp[5] = (mark << 7) | _ui8PayloadType;
     memcpy(&pucRtp[6], &sq, 2);
     memcpy(&pucRtp[8], &ts, 4);
     //ssrc
     memcpy(&pucRtp[12], &sc, 4);
 
     if(data){
-        //playload
+        //payload
         memcpy(&pucRtp[16], data, len);
     }
 
-    rtppkt->PT = _ui8PlayloadType;
+    rtppkt->PT = _ui8PayloadType;
     rtppkt->interleaved = _ui8Interleaved;
     rtppkt->mark = mark;
     rtppkt->sequence = _ui16Sequence;

src/Rtsp/RtpCodec.h

@@ -66,7 +66,7 @@ public:
     RtpInfo(uint32_t ui32Ssrc,
             uint32_t ui32MtuSize,
             uint32_t ui32SampleRate,
-            uint8_t ui8PlayloadType,
+            uint8_t ui8PayloadType,
             uint8_t ui8Interleaved) {
         if(ui32Ssrc == 0){
             ui32Ssrc = ((uint64_t)this) & 0xFFFFFFFF;
@@ -74,7 +74,7 @@ public:
         _ui32Ssrc = ui32Ssrc;
         _ui32SampleRate = ui32SampleRate;
         _ui32MtuSize = ui32MtuSize;
-        _ui8PlayloadType = ui8PlayloadType;
+        _ui8PayloadType = ui8PayloadType;
         _ui8Interleaved = ui8Interleaved;
     }
 
@@ -84,8 +84,8 @@ public:
         return _ui8Interleaved;
     }
 
-    int getPlayloadType() const {
+    int getPayloadType() const {
-        return _ui8PlayloadType;
+        return _ui8PayloadType;
     }
 
     int getSampleRate() const {
@@ -110,7 +110,7 @@ protected:
     uint32_t _ui32Ssrc;
     uint32_t _ui32SampleRate;
     uint32_t _ui32MtuSize;
-    uint8_t _ui8PlayloadType;
+    uint8_t _ui8PayloadType;
     uint8_t _ui8Interleaved;
     uint16_t _ui16Sequence = 0;
     uint32_t _ui32TimeStamp = 0;

src/Rtsp/Rtsp.h

@@ -188,11 +188,11 @@ public:
     /**
      * 构造sdp
      * @param sample_rate 采样率
-     * @param playload_type pt类型
+     * @param payload_type pt类型
      */
-    Sdp(uint32_t sample_rate, uint8_t playload_type){
+    Sdp(uint32_t sample_rate, uint8_t payload_type){
         _sample_rate = sample_rate;
-        _playload_type = playload_type;
+        _payload_type = payload_type;
     }
 
     virtual ~Sdp(){}
@@ -207,8 +207,8 @@ public:
      * 获取pt
      * @return
      */
-    uint8_t getPlayloadType() const{
+    uint8_t getPayloadType() const{
-        return _playload_type;
+        return _payload_type;
     }
 
     /**
@@ -219,7 +219,7 @@ public:
         return _sample_rate;
     }
 private:
-    uint8_t _playload_type;
+    uint8_t _payload_type;
     uint32_t _sample_rate;
 };