fix non rtc push rtc play g711 audio loss

src/Extension/G711Rtp.cpp

@@ -0,0 +1,46 @@
+#include "G711Rtp.h"
+
+G711RtpEncoder::G711RtpEncoder(CodecId codec, uint32_t ssrc, uint32_t mtu_size,
+                                   uint32_t sample_rate,  uint8_t payload_type, uint8_t interleaved,uint32_t channels)
+        : CommonRtpDecoder(codec), RtpInfo(ssrc, mtu_size, sample_rate, payload_type, interleaved) {
+        _cache_frame = FrameImp::create();
+        _cache_frame->_codec_id = codec;
+        _channels = channels;
+}
+
+bool G711RtpEncoder::inputFrame(const Frame::Ptr &frame){
+    auto dur = (_cache_frame->size()-_cache_frame->prefixSize())/(8*_channels);
+    auto next_pts = _cache_frame->pts()+dur;
+    if(next_pts == 0){
+        _cache_frame->_pts = frame->pts();
+    }else{
+        if((next_pts+20) < frame->pts()){// 有丢包超过20ms
+            _cache_frame->_pts = frame->pts() - dur;
+        }
+    }
+    _cache_frame->_buffer.append(frame->data() + frame->prefixSize(),frame->size() - frame->prefixSize());
+
+    auto stamp = _cache_frame->pts();
+    auto ptr = _cache_frame->data() + _cache_frame->prefixSize();
+    auto len = _cache_frame->size() - _cache_frame->prefixSize();
+    auto remain_size = len;
+    auto max_size = 160*_channels; //20 ms per rtp 
+    int n = 0;
+    bool mark = false;
+    while (remain_size >= max_size) {
+        size_t rtp_size;
+        if (remain_size >= max_size) {
+            rtp_size = max_size;
+        } else {
+            break;
+        }
+        n++;
+        stamp += 20;
+        RtpCodec::inputRtp(makeRtp(getTrackType(), ptr, rtp_size, mark, stamp), false);
+        ptr += rtp_size;
+        remain_size -= rtp_size;
+    }
+    _cache_frame->_buffer.erase(0,n*max_size);
+    _cache_frame->_pts += 20*n;
+    return len > 0;
+}