rtp send rtp g711 audio can config duration (#3325)

optimization for this
[issue](https://github.com/ZLMediaKit/ZLMediaKit/issues/3316)

conf/config.ini

@@ -329,6 +329,10 @@ opus_pt=100
 #如果不调用startSendRtp相关接口，可以置0节省内存
 gop_cache=1
 
+#国标发送g711 rtp 打包时，每个包的语音时长是多少，默认是100 ms，范围为20~180ms (gb28181-2016，c.2.4规定)，
+#最好为20 的倍数，程序自动向20的倍数取整
+rtp_g711_dur_ms = 100
+
 [rtc]
 #rtc播放推流、播放超时时间
 timeoutSec=15

ext-codec/G711Rtp.cpp

@@ -8,13 +8,27 @@ G711RtpEncoder::G711RtpEncoder(CodecId codec, uint32_t channels){
     _channels = channels;
 }
 
+void G711RtpEncoder::setOpt(int opt, const toolkit::Any &param) {
+    if (opt == RTP_ENCODER_PKT_DUR_MS) {
+        if (param.is<uint32_t>()) {
+            auto dur = param.get<uint32_t>();
+            if (dur < 20 || dur > 180) {
+                WarnL << "set g711 rtp encoder  duration ms failed for " << dur;
+                return;
+            }
+            // 向上 20ms 取整
+            _pkt_dur_ms = (dur + 19) / 20 * 20;
+        }
+    }
+}
+
 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
+        if ((next_pts + _pkt_dur_ms) < frame->pts()) { // 有丢包超过20ms
             _cache_frame->_pts = frame->pts() - dur;
         }
     }
@@ -24,9 +38,9 @@ bool G711RtpEncoder::inputFrame(const Frame::Ptr &frame) {
     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;
+    auto max_size = 160 * _channels * _pkt_dur_ms / 20; // 20 ms per 160 byte
+    uint32_t n = 0;
+    bool mark = true;
     while (remain_size >= max_size) {
         size_t rtp_size;
         if (remain_size >= max_size) {
@@ -35,13 +49,13 @@ bool G711RtpEncoder::inputFrame(const Frame::Ptr &frame) {
             break;
         }
         n++;
-        stamp += 20;
-        RtpCodec::inputRtp(getRtpInfo().makeRtp(TrackAudio, ptr, rtp_size, mark, stamp), false);
+        stamp += _pkt_dur_ms;
+        RtpCodec::inputRtp(getRtpInfo().makeRtp(TrackAudio, ptr, rtp_size, mark, stamp), true);
         ptr += rtp_size;
         remain_size -= rtp_size;
     }
     _cache_frame->_buffer.erase(0, n * max_size);
-    _cache_frame->_pts += 20 * n;
+    _cache_frame->_pts += (uint64_t)_pkt_dur_ms * n;
     return len > 0;
 }
 

ext-codec/G711Rtp.h

@@ -36,8 +36,11 @@ public:
      */
     bool inputFrame(const Frame::Ptr &frame) override;
 
+    void setOpt(int opt, const toolkit::Any &param) override;
+
 private:
     uint32_t _channels = 1;
+    uint32_t _pkt_dur_ms = 20;
     FrameImp::Ptr _cache_frame;
 };
 

src/Common/config.cpp

@@ -338,6 +338,7 @@ const string kH265PT = RTP_PROXY_FIELD "h265_pt";
 const string kPSPT = RTP_PROXY_FIELD "ps_pt";
 const string kOpusPT = RTP_PROXY_FIELD "opus_pt";
 const string kGopCache = RTP_PROXY_FIELD "gop_cache";
+const string kRtpG711DurMs = RTP_PROXY_FIELD "rtp_g711_dur_ms";
 
 static onceToken token([]() {
     mINI::Instance()[kDumpDir] = "";
@@ -348,6 +349,7 @@ static onceToken token([]() {
     mINI::Instance()[kPSPT] = 96;
     mINI::Instance()[kOpusPT] = 100;
     mINI::Instance()[kGopCache] = 1;
+    mINI::Instance()[kRtpG711DurMs] = 100;
 });
 } // namespace RtpProxy
 

src/Common/config.h

@@ -382,6 +382,9 @@ extern const std::string kPSPT;
 extern const std::string kOpusPT;
 // RtpSender相关功能是否提前开启gop缓存优化级联秒开体验，默认开启
 extern const std::string kGopCache;
+//国标发送g711 rtp 打包时，每个包的语音时长是多少，默认是100 ms，范围为20~180ms (gb28181-2016，c.2.4规定)，
+//最好为20 的倍数，程序自动向20的倍数取整
+extern const std::string kRtpG711DurMs;
 } // namespace RtpProxy
 
 /**

src/Rtp/RawEncoder.cpp

@@ -34,6 +34,12 @@ bool RawEncoderImp::addTrack(const Track::Ptr &track) {
         auto ring = std::make_shared<RtpRing::RingType>();
         ring->setDelegate(std::make_shared<RingDelegateHelper>([this](RtpPacket::Ptr rtp, bool is_key) { onRTP(std::move(rtp), true); }));
         _rtp_encoder->setRtpRing(std::move(ring));
+        if (track->getCodecId() == CodecG711A || track->getCodecId() == CodecG711U) {
+            GET_CONFIG(uint32_t, dur_ms, RtpProxy::kRtpG711DurMs);
+            Any param;
+            param.set<uint32_t>(dur_ms);
+            _rtp_encoder->setOpt(RtpCodec::RTP_ENCODER_PKT_DUR_MS, param);
+        }
         return true;
     }
 

src/Rtsp/RtpCodec.h

@@ -93,6 +93,17 @@ public:
 
     RtpInfo &getRtpInfo() { return *_rtp_info; }
 
+    enum {
+        RTP_ENCODER_PKT_DUR_MS = 1 // 主要应用于g711 rtp 打包器每个包的时间长度，option_value 为int*, option_len 为4
+    };
+    /**
+     * @brief 设置rtp打包器与解包器的相关参数，主要应用与g711 rtp 打包器，使用方法类似setsockopt
+     *
+     * @param opt 设置的选项
+     * @param param 设置的参数
+     */
+    virtual void setOpt(int opt, const toolkit::Any &param) {};
+
 private:
     std::unique_ptr<RtpInfo> _rtp_info;
 };