整理代码 添加注释

src/RTP/AACRtpCodec.h

@@ -5,44 +5,62 @@
 #ifndef ZLMEDIAKIT_AACRTPCODEC_H
 #define ZLMEDIAKIT_AACRTPCODEC_H
 
-
 #include "RtpCodec.h"
 
+/**
+ * aac rtp转adts类
+ */
 class AACRtpDecoder : public RtpCodec {
 public:
-    AACRtpDecoder() {
-        m_framePool.setSize(32);
-        m_adts = m_framePool.obtain();
-    }
-
+    /**
+     * @param ui32SampleRate 采样率，用于时间戳转换用
+     */
+    AACRtpDecoder(uint32_t ui32SampleRate);
     ~AACRtpDecoder() {}
 
-    void inputRtp(const RtpPacket::Ptr &rtp, bool key_pos) override;
-
+    /**
+     * 输入rtp并解码
+     * @param rtp rtp数据包
+     * @param key_pos 此参数内部强制转换为false,请忽略之
+     */
+    void inputRtp(const RtpPacket::Ptr &rtp, bool key_pos = false) override;
 private:
     void onGetAdts(const AdtsFrame::Ptr &frame);
-
+    AdtsFrame::Ptr obtainFrame();
 private:
     AdtsFrame::Ptr m_adts;
     ResourcePool<AdtsFrame> m_framePool;
+    uint32_t m_sampleRate;
 };
 
 
+/**
+ * aac adts转rtp类
+ */
 class AACRtpEncoder : public RtpInfo, public RtpCodec {
 public:
+    /**
+     * @param ui32Ssrc ssrc
+     * @param ui32MtuSize mtu 大小
+     * @param ui32SampleRate 采样率
+     * @param ui8PlayloadType pt类型
+     * @param ui8Interleaved rtsp interleaved 值
+     */
     AACRtpEncoder(uint32_t ui32Ssrc,
                   uint32_t ui32MtuSize,
                   uint32_t ui32SampleRate,
                   uint8_t ui8PlayloadType = 97,
                   uint8_t ui8Interleaved = TrackAudio * 2);
-
     ~AACRtpEncoder() {}
 
-    void inputFame(const Frame::Ptr &frame, bool key_pos) override;
-
+    /**
+     * 输入aac 数据，必须带dats头
+     * @param frame 带dats头的aac数据
+     * @param key_pos 此参数内部强制转换为false,请忽略之
+     */
+    void inputFame(const Frame::Ptr &frame, bool key_pos = false) override;
 private:
     void makeAACRtp(const void *pData, unsigned int uiLen, bool bMark, uint32_t uiStamp);
-
 private:
     unsigned char m_aucSectionBuf[1600];
 };