From 341459fbf5e65a2350470a28b12d22fcca9b0967 Mon Sep 17 00:00:00 2001
From: xgj <xiong_panda@163.com>
Date: Mon, 24 Jan 2022 11:57:24 +0800
Subject: [PATCH] fix non rtc push  rtc play g711 audio loss

---
 src/Extension/Factory.cpp |  9 +++++--
 src/Extension/G711Rtp.cpp | 46 +++++++++++++++++++++++++++++++++++
 src/Extension/G711Rtp.h   | 51 +++++++++++++++++++++++++++++++++++++++
 webrtc/WebRtcPlayer.cpp   |  2 +-
 4 files changed, 105 insertions(+), 3 deletions(-)
 create mode 100644 src/Extension/G711Rtp.cpp
 create mode 100644 src/Extension/G711Rtp.h

diff --git a/src/Extension/Factory.cpp b/src/Extension/Factory.cpp
index 6d047940..b1614bf9 100644
--- a/src/Extension/Factory.cpp
+++ b/src/Extension/Factory.cpp
@@ -18,6 +18,7 @@
 #include "AACRtp.h"
 #include "H265Rtp.h"
 #include "CommonRtp.h"
+#include "G711Rtp.h"
 #include "Opus.h"
 #include "G711.h"
 #include "L16.h"
@@ -119,9 +120,13 @@ RtpCodec::Ptr Factory::getRtpEncoderBySdp(const Sdp::Ptr &sdp) {
         case CodecH265 : return std::make_shared<H265RtpEncoder>(ssrc, mtu, sample_rate, pt, interleaved);
         case CodecAAC : return std::make_shared<AACRtpEncoder>(ssrc, mtu, sample_rate, pt, interleaved);
         case CodecL16 :
-        case CodecOpus :
+        case CodecOpus : return std::make_shared<CommonRtpEncoder>(codec_id, ssrc, mtu, sample_rate, pt, interleaved);
         case CodecG711A :
-        case CodecG711U : return std::make_shared<CommonRtpEncoder>(codec_id, ssrc, mtu, sample_rate, pt, interleaved);
+        case CodecG711U : 
+            if(pt == Rtsp::PT_PCMA || pt == Rtsp::PT_PCMU){
+                return std::make_shared<G711RtpEncoder>(codec_id, ssrc, mtu, sample_rate, pt, interleaved,1);
+            }
+            return std::make_shared<CommonRtpEncoder>(codec_id, ssrc, mtu, sample_rate, pt, interleaved);
         default : WarnL << "暂不支持该CodecId:" << codec_id; return nullptr;
     }
 }
diff --git a/src/Extension/G711Rtp.cpp b/src/Extension/G711Rtp.cpp
new file mode 100644
index 00000000..14b256b6
--- /dev/null
+++ b/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;
+}
\ No newline at end of file
diff --git a/src/Extension/G711Rtp.h b/src/Extension/G711Rtp.h
new file mode 100644
index 00000000..3b66bdd8
--- /dev/null
+++ b/src/Extension/G711Rtp.h
@@ -0,0 +1,51 @@
+/*
+ * Copyright (c) 2016 The ZLMediaKit project authors. All Rights Reserved.
+ *
+ * This file is part of ZLMediaKit(https://github.com/xia-chu/ZLMediaKit).
+ *
+ * Use of this source code is governed by MIT license that can be found in the
+ * LICENSE file in the root of the source tree. All contributing project authors
+ * may be found in the AUTHORS file in the root of the source tree.
+ */
+
+#ifndef ZLMEDIAKIT_G711RTP_H
+#define ZLMEDIAKIT_G711RTP_H
+
+#include "Frame.h"
+#include "CommonRtp.h"
+#include "Rtsp/RtpCodec.h"
+
+namespace mediakit{
+
+
+/**
+ * G711 rtp编码类
+ */
+class G711RtpEncoder : public CommonRtpDecoder, public RtpInfo {
+public:
+    typedef std::shared_ptr <G711RtpEncoder> Ptr;
+
+    ~G711RtpEncoder() override {}
+
+    /**
+     * 构造函数
+     * @param codec 编码类型
+     * @param ssrc ssrc
+     * @param mtu_size mtu 大小
+     * @param sample_rate 采样率
+     * @param payload_type pt类型
+     * @param interleaved rtsp interleaved 值
+     */
+    G711RtpEncoder(CodecId codec, uint32_t ssrc, uint32_t mtu_size, uint32_t sample_rate, uint8_t payload_type, uint8_t interleaved,uint32_t channels);
+
+    /**
+     * 输入帧数据并编码成rtp
+     */
+    bool inputFrame(const Frame::Ptr &frame) override;
+private:
+    FrameImp::Ptr _cache_frame;
+    uint32_t _channels = 1;
+};
+
+}//namespace mediakit
+#endif //ZLMEDIAKIT_G711RTP_H
diff --git a/webrtc/WebRtcPlayer.cpp b/webrtc/WebRtcPlayer.cpp
index 1bd8b513..4447ab8e 100644
--- a/webrtc/WebRtcPlayer.cpp
+++ b/webrtc/WebRtcPlayer.cpp
@@ -43,6 +43,7 @@ void WebRtcPlayer::onStartWebRTC() {
             }
             size_t i = 0;
             pkt->for_each([&](const RtpPacket::Ptr &rtp) {
+                //TraceL<<"send track type:"<<rtp->type<<" ts:"<<rtp->getStamp()<<" ntp:"<<rtp->ntp_stamp<<" size:"<<rtp->getPayloadSize()<<" i:"<<i;
                 strongSelf->onSendRtp(rtp, ++i == pkt->size());
             });
         });
@@ -57,7 +58,6 @@ void WebRtcPlayer::onStartWebRTC() {
     //使用完毕后，释放强引用，这样确保推流器断开后能及时注销媒体
     _play_src = nullptr;
 }
-
 void WebRtcPlayer::onDestory() {
     WebRtcTransportImp::onDestory();