添加对g711A/U rtsp支持相关文件，api添加windows静态库编译选项，g711 RTMP支持尚未修改

api/CMakeLists.txt

 include_directories(include source)
 
 file(GLOB api_src_list include/*.h source/*.cpp source/*.h source/*.c)
+
+set(ENABLE_API_STATIC false)
+
 if (IOS)
     add_library(mk_api STATIC ${api_src_list})
     target_link_libraries(mk_api ${LINK_LIB_LIST})
 else ()
-    add_library(mk_api SHARED ${api_src_list})
-    if (WIN32)
-        add_definitions(-DMediaKitApi_EXPORTS)
-    endif ()
-
+    if(ENABLE_API_STATIC)
+        add_library(mk_api STATIC ${api_src_list})
+        if (WIN32)
+            add_definitions(-DMediaKitApi_STATIC)
+        endif ()
+    else ()
+        add_library(mk_api SHARED ${api_src_list})
+        if (WIN32)
+            add_definitions(-DMediaKitApi_EXPORTS)
+        endif ()
+    endif()
     target_link_libraries(mk_api ${LINK_LIB_LIST})
     add_subdirectory(tests)
 

api/include/mk_common.h

@@ -14,13 +14,20 @@
 #include <stdint.h>
 
 #if defined(_WIN32)
+
+#ifndef MediaKitApi_STATIC
 #if defined(MediaKitApi_EXPORTS)
-        #define API_EXPORT __declspec(dllexport)
-    #else
-        #define API_EXPORT __declspec(dllimport)
-    #endif
+		#define API_EXPORT __declspec(dllexport)
+	#else
+		#define API_EXPORT __declspec(dllimport)
+	#endif
+
+	#define API_CALL __cdecl
+#else
+#define API_EXPORT
+#define API_CALL
+#endif
 
-    #define API_CALL __cdecl
 #else
 #define API_EXPORT
 #define API_CALL

api/include/mk_media.h

@@ -66,6 +66,16 @@ API_EXPORT void API_CALL mk_media_init_h265(mk_media ctx, int width, int height,
 API_EXPORT void API_CALL mk_media_init_aac(mk_media ctx, int channel, int sample_bit, int sample_rate, int profile);
 
 /**
+ * 添加g711音频轨道
+ * @param ctx 对象指针
+ * @param au 1.G711A 2.G711U
+ * @param channel 通道数
+ * @param sample_bit 采样位数，只支持16
+ * @param sample_rate 采样率
+ */
+API_EXPORT void API_CALL mk_media_init_g711(mk_media ctx, int au, int sample_bit, int sample_rate);
+
+/**
  * 初始化h264/h265/aac完毕后调用此函数，
  * 在单track(只有音频或视频)时，因为ZLMediaKit不知道后续是否还要添加track，所以会多等待3秒钟
  * 如果产生的流是单Track类型，请调用此函数以便加快流生成速度，当然不调用该函数，影响也不大(会多等待3秒)
@@ -114,6 +124,15 @@ API_EXPORT void API_CALL mk_media_input_aac(mk_media ctx, void *data, int len, u
 API_EXPORT void API_CALL mk_media_input_aac1(mk_media ctx, void *data, int len, uint32_t dts, void *adts);
 
 /**
+ * 输入单帧G711音频
+ * @param ctx 对象指针
+ * @param data 单帧G711数据
+ * @param len 单帧G711数据字节数
+ * @param dts 时间戳，毫秒
+ */
+API_EXPORT void API_CALL mk_media_input_g711(mk_media ctx, void* data, int len, uint32_t dts);
+
+/**
  * MediaSource.close()回调事件
  * 在选择关闭一个关联的MediaSource时，将会最终触发到该回调
  * 你应该通过该事件调用mk_media_release函数并且释放其他资源

api/source/mk_media.cpp

@@ -140,6 +140,20 @@ API_EXPORT void API_CALL mk_media_init_aac(mk_media ctx, int channel, int sample
     (*obj)->getChannel()->initAudio(info);
 }
 
+
+API_EXPORT void API_CALL mk_media_init_g711(mk_media ctx, int au, int sample_bit, int sample_rate)
+{
+    assert(ctx);
+    MediaHelper::Ptr* obj = (MediaHelper::Ptr*) ctx;
+    AudioInfo info;
+    info.iSampleRate = sample_rate;
+    info.iChannel = 1;
+    info.iSampleBit = sample_bit;
+    info.iProfile = 0;
+    info.codecId = (CodecId)au;
+    (*obj)->getChannel()->initAudio(info);
+}
+
 API_EXPORT void API_CALL mk_media_init_complete(mk_media ctx){
     assert(ctx);
     MediaHelper::Ptr *obj = (MediaHelper::Ptr *) ctx;
@@ -170,6 +184,12 @@ API_EXPORT void API_CALL mk_media_input_aac1(mk_media ctx, void *data, int len,
     (*obj)->getChannel()->inputAAC((char *) data, len, dts, (char *) adts);
 }
 
+API_EXPORT void API_CALL mk_media_input_g711(mk_media ctx, void* data, int len, uint32_t dts)
+{
+    assert(ctx && data && len > 0);
+    MediaHelper::Ptr* obj = (MediaHelper::Ptr*) ctx;
+    (*obj)->getChannel()->inputG711((char*)data, len, dts);
+}
 
 
 

src/Common/Device.cpp

@@ -16,6 +16,7 @@
 #include "Util/base64.h"
 #include "Util/TimeTicker.h"
 #include "Extension/AAC.h"
+#include "Extension/G711.h"
 #include "Extension/H264.h"
 #include "Extension/H265.h"
 
@@ -148,6 +149,14 @@ void DevChannel::inputAAC(const char *pcDataWithoutAdts,int iDataLen, uint32_t u
 }
 
 
+void DevChannel::inputG711(const char* pcData, int iDataLen, uint32_t uiStamp)
+{
+    if (uiStamp == 0) {
+        uiStamp = (uint32_t)_aTicker[1].elapsedTime();
+    }
+    inputFrame(std::make_shared<G711FrameNoCacheAble>(_audio->codecId, (char*)pcData, iDataLen, uiStamp, 0));
+}
+
 void DevChannel::initVideo(const VideoInfo& info) {
     _video = std::make_shared<VideoInfo>(info);
     addTrack(std::make_shared<H264Track>());
@@ -159,32 +168,39 @@ void DevChannel::initH265Video(const VideoInfo &info){
 }
 
 void DevChannel::initAudio(const AudioInfo& info) {
-    _audio = std::make_shared<AudioInfo>(info);
-    addTrack(std::make_shared<AACTrack>());
-
-    AACFrame adtsHeader;
-    adtsHeader.syncword = 0x0FFF;
-    adtsHeader.id = 0;
-    adtsHeader.layer = 0;
-    adtsHeader.protection_absent = 1;
-    adtsHeader.profile =  info.iProfile;//audioObjectType - 1;
-    int i = 0;
-    for(auto rate : samplingFrequencyTable){
-        if(rate == info.iSampleRate){
-            adtsHeader.sf_index = i;
-        };
-        ++i;
-    }
-    adtsHeader.private_bit = 0;
-    adtsHeader.channel_configuration = info.iChannel;
-    adtsHeader.original = 0;
-    adtsHeader.home = 0;
-    adtsHeader.copyright_identification_bit = 0;
-    adtsHeader.copyright_identification_start = 0;
-    adtsHeader.aac_frame_length = 7;
-    adtsHeader.adts_buffer_fullness = 2047;
-    adtsHeader.no_raw_data_blocks_in_frame = 0;
-    writeAdtsHeader(adtsHeader,_adtsHeader);
+	_audio = std::make_shared<AudioInfo>(info);
+    if (info.codecId == CodecAAC)
+    {
+        addTrack(std::make_shared<AACTrack>());
+
+        AACFrame adtsHeader;
+        adtsHeader.syncword = 0x0FFF;
+        adtsHeader.id = 0;
+        adtsHeader.layer = 0;
+        adtsHeader.protection_absent = 1;
+        adtsHeader.profile = info.iProfile;//audioObjectType - 1;
+        int i = 0;
+        for (auto rate : samplingFrequencyTable) {
+            if (rate == info.iSampleRate) {
+                adtsHeader.sf_index = i;
+            };
+            ++i;
+        }
+        adtsHeader.private_bit = 0;
+        adtsHeader.channel_configuration = info.iChannel;
+        adtsHeader.original = 0;
+        adtsHeader.home = 0;
+        adtsHeader.copyright_identification_bit = 0;
+        adtsHeader.copyright_identification_start = 0;
+        adtsHeader.aac_frame_length = 7;
+        adtsHeader.adts_buffer_fullness = 2047;
+        adtsHeader.no_raw_data_blocks_in_frame = 0;
+        writeAdtsHeader(adtsHeader, _adtsHeader);
+    }
+    else if (info.codecId == CodecG711A || info.codecId == CodecG711U)
+    {
+        addTrack(std::make_shared<G711Track>(info.codecId, info.iSampleBit, info.iSampleRate));
+    }
 }
 
 } /* namespace mediakit */

src/Common/Device.h

@@ -41,10 +41,11 @@ public:
 };
 class AudioInfo {
 public:
-    int iChannel;
-    int iSampleBit;
-    int iSampleRate;
-    int iProfile;
+    CodecId codecId;
+	int iChannel;
+	int iSampleBit;
+	int iSampleRate;
+	int iProfile;
 };
 
 /**
@@ -121,6 +122,13 @@ public:
      */
     void inputAAC(const char *pcDataWithoutAdts,int iDataLen, uint32_t uiStamp,const char *pcAdtsHeader);
 
+    /**
+     * G711音频帧
+     * @param pcData 音频帧
+     * @param iDataLen 帧数据长度
+     * @param uiStamp 时间戳，单位毫秒
+     */
+    void inputG711(const char* pcData, int iDataLen, uint32_t uiStamp);
 #ifdef ENABLE_X264
     /**
      * 输入yuv420p视频帧，内部会完成编码并调用inputH264方法

src/Extension/Factory.cpp

@@ -13,8 +13,10 @@
 #include "H264Rtmp.h"
 #include "H265Rtmp.h"
 #include "AACRtmp.h"
+#include "G711Rtmp.h"
 #include "H264Rtp.h"
 #include "AACRtp.h"
+#include "G711Rtp.h"
 #include "H265Rtp.h"
 #include "Common/Parser.h"
 
@@ -45,6 +47,14 @@ Track::Ptr Factory::getTrackBySdp(const SdpTrack::Ptr &track) {
         return std::make_shared<AACTrack>(aac_cfg);
     }
 
+    if (strcasecmp(track->_codec.data(), "PCMA") == 0) {
+        return std::make_shared<G711Track>(CodecG711A);
+    }
+
+    if (strcasecmp(track->_codec.data(), "PCMU") == 0) {
+        return std::make_shared<G711Track>(CodecG711U);
+    }
+
     if (strcasecmp(track->_codec.data(), "h264") == 0) {
         //a=fmtp:96 packetization-mode=1;profile-level-id=42C01F;sprop-parameter-sets=Z0LAH9oBQBboQAAAAwBAAAAPI8YMqA==,aM48gA==
         auto map = Parser::parseArgs(FindField(track->_fmtp.data()," ", nullptr),";","=");
@@ -90,6 +100,12 @@ Track::Ptr Factory::getTrackByCodecId(CodecId codecId) {
         case CodecAAC:{
             return std::make_shared<AACTrack>();
         }
+        case CodecG711A: {
+            return std::make_shared<G711Track>(CodecG711A);
+        }
+        case CodecG711U: {
+            return std::make_shared<G711Track>(CodecG711U);
+        }
         default:
             WarnL << "暂不支持该CodecId:" << codecId;
             return nullptr;
@@ -125,6 +141,9 @@ RtpCodec::Ptr Factory::getRtpEncoderBySdp(const Sdp::Ptr &sdp) {
             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 CodecG711A:
+        case CodecG711U:
+            return std::make_shared<G711RtpEncoder>(ssrc, mtu, sample_rate, pt, interleaved);
         default:
             WarnL << "暂不支持该CodecId:" << codec_id;
             return nullptr;
@@ -139,6 +158,9 @@ RtpCodec::Ptr Factory::getRtpDecoderByTrack(const Track::Ptr &track) {
             return std::make_shared<H265RtpDecoder>();
         case CodecAAC:
             return std::make_shared<AACRtpDecoder>(track->clone());
+        case CodecG711A:
+        case CodecG711U:
+            return std::make_shared<G711RtpDecoder>(track->clone());
         default:
             WarnL << "暂不支持该CodecId:" << track->getCodecName();
             return nullptr;
@@ -198,6 +220,9 @@ RtmpCodec::Ptr Factory::getRtmpCodecByTrack(const Track::Ptr &track) {
             return std::make_shared<AACRtmpEncoder>(track);
         case CodecH265:
             return std::make_shared<H265RtmpEncoder>(track);
+	    case CodecG711A:
+        case CodecG711U:
+            return std::make_shared<G711RtmpEncoder>(track);
         default:
             WarnL << "暂不支持该CodecId:" << track->getCodecName();
             return nullptr;
@@ -209,6 +234,8 @@ AMFValue Factory::getAmfByCodecId(CodecId codecId) {
         case CodecAAC: return AMFValue("mp4a");
         case CodecH264: return AMFValue("avc1");
         case CodecH265: return AMFValue(FLV_CODEC_H265);
+        case CodecG711A: return AMFValue(7);
+        case CodecG711U: return AMFValue(8);
         default: return AMFValue(AMF_NULL);
     }
 }

src/Extension/Frame.h

@@ -26,6 +26,8 @@ typedef enum {
     CodecH264 = 0,
     CodecH265,
     CodecAAC,
+    CodecG711A,
+    CodecG711U,
     CodecMax = 0x7FFF
 } CodecId;
 

src/Extension/G711.cpp

+/*
+ * Copyright (c) 2016 The ZLMediaKit project authors. All Rights Reserved.
+ *
+ * This file is part of ZLMediaKit(https://github.com/xiongziliang/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.
+ */
+
+#include "G711.h"
+
+namespace mediakit{
+
+
+Sdp::Ptr G711Track::getSdp() {
+    if(!ready()){
+        WarnL << "AAC Track未准备好";
+        return nullptr;
+    }
+    return std::make_shared<G711Sdp>(getCodecId(), getAudioSampleRate(), getCodecId() == CodecG711A ? 8 : 0, getAudioSampleBit());
+}
+
+}//namespace mediakit
+
+

src/Extension/G711.h

+/*
+ * Copyright (c) 2016 The ZLMediaKit project authors. All Rights Reserved.
+ *
+ * This file is part of ZLMediaKit(https://github.com/xiongziliang/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_G711_H
+#define ZLMEDIAKIT_G711_H
+
+#include "Frame.h"
+#include "Track.h"
+
+namespace mediakit{
+
+class G711Frame;
+
+unsigned const samplingFrequencyTableG711[16] = { 96000, 88200,
+                                              64000, 48000,
+                                              44100, 32000,
+                                              24000, 22050,
+                                              16000, 12000,
+                                              11025, 8000,
+                                              7350, 0, 0, 0 };
+
+void	makeAdtsHeader(const string &strAudioCfg,G711Frame &adts);
+void 	writeAdtsHeader(const G711Frame &adts, uint8_t *pcAdts) ;
+string 	makeG711AdtsConfig(const uint8_t *pcAdts);
+void 	getAACInfo(const G711Frame &adts,int &iSampleRate,int &iChannel);
+
+
+/**
+ * aac帧，包含adts头
+ */
+class G711Frame : public Frame {
+public:
+    typedef std::shared_ptr<G711Frame> Ptr;
+
+    char *data() const override{
+        return (char *)buffer;
+    }
+    uint32_t size() const override {
+        return frameLength;
+    }
+    uint32_t dts() const override {
+        return timeStamp;
+    }
+    uint32_t prefixSize() const override{
+        return iPrefixSize;
+    }
+
+    TrackType getTrackType() const override{
+        return TrackAudio;
+    }
+
+    CodecId getCodecId() const override{
+        return CodecAAC;
+    }
+
+    bool keyFrame() const override {
+        return false;
+    }
+
+    bool configFrame() const override{
+        return false;
+    }
+public:
+    unsigned int frameLength; // 一个帧的长度包括 raw data block
+    unsigned char buffer[2 * 1024 + 7];
+    uint32_t timeStamp;
+    uint32_t iPrefixSize = 0;
+} ;
+
+class G711FrameNoCacheAble : public FrameNoCacheAble {
+public:
+    typedef std::shared_ptr<G711FrameNoCacheAble> Ptr;
+
+    G711FrameNoCacheAble(CodecId codecId, char *ptr,uint32_t size,uint32_t dts,int prefixeSize = 7){
+        _codecId = codecId;
+        _ptr = ptr;
+        _size = size;
+        _dts = dts;
+        _prefixSize = prefixeSize;
+    }
+
+    TrackType getTrackType() const override{
+        return TrackAudio;
+    }
+
+    CodecId getCodecId() const override{
+        return _codecId;
+    }
+
+    bool keyFrame() const override {
+        return false;
+    }
+
+    bool configFrame() const override{
+        return false;
+    }
+
+private:
+    CodecId _codecId;
+} ;
+
+
+/**
+ * g711音频通道
+ */
+class G711Track : public AudioTrack{
+public:
+    typedef std::shared_ptr<G711Track> Ptr;
+
+    /**
+     * 延后获取adts头信息
+     * 在随后的inputFrame中获取adts头信息
+     */
+    G711Track(){}
+
+    /**
+     * G711A G711U
+     */
+    G711Track(CodecId codecId, int sampleBit = 16, int sampleRate = 8000){
+        _codecid = codecId;
+        _sampleBit = sampleBit;
+        _sampleRate = sampleRate;
+        onReady();
+    }
+
+    /**
+     * 返回编码类型
+     * @return
+     */
+    CodecId getCodecId() const override{
+        return _codecid;
+    }
+
+    /**
+     * 在获取aac_cfg前是无效的Track
+     * @return
+     */
+    bool ready() override {
+        return true;
+    }
+
+
+    /**
+    * 返回音频采样率
+    * @return
+    */
+    int getAudioSampleRate() const override{
+        return _sampleRate;
+    }
+    /**
+     * 返回音频采样位数，一般为16或8
+     * @return
+     */
+    int getAudioSampleBit() const override{
+        return _sampleBit;
+    }
+    /**
+     * 返回音频通道数
+     * @return
+     */
+    int getAudioChannel() const override{
+        return _channel;
+    }
+
+    /**
+    * 输入数据帧,并获取aac_cfg
+    * @param frame 数据帧
+    */
+    void inputFrame(const Frame::Ptr &frame) override{
+        AudioTrack::inputFrame(frame);
+    }
+private:
+    /**
+     * 
+     */
+    void onReady(){
+/*
+        if(_cfg.size() < 2){
+            return;
+        }
+        G711Frame aacFrame;
+        makeAdtsHeader(_cfg,aacFrame);
+        getAACInfo(aacFrame,_sampleRate,_channel);*/
+    }
+    Track::Ptr clone() override {
+        return std::make_shared<std::remove_reference<decltype(*this)>::type >(*this);
+    }
+
+    //生成sdp
+    Sdp::Ptr getSdp() override ;
+private:
+    string _cfg;
+    CodecId _codecid = CodecG711A;
+    int _sampleRate = 8000;
+    int _sampleBit = 16;
+    int _channel = 1;
+};
+
+
+/**
+* aac类型SDP
+*/
+class G711Sdp : public Sdp {
+public:
+
+    /**
+     *
+     * @param aac_codecId G711A G711U
+     * @param sample_rate 音频采样率
+     * @param playload_type rtp playload type 默认0为G711U， 8为G711A
+     * @param bitrate 比特率
+     */
+    G711Sdp(CodecId codecId,
+           int sample_rate,
+           int playload_type = 0,
+           int bitrate = 128) : Sdp(sample_rate,playload_type), _codecId(codecId){
+        _printer << "m=audio 0 RTP/AVP " << playload_type << "\r\n";
+        //_printer << "b=AS:" << bitrate << "\r\n";
+        _printer << "a=rtpmap:" << playload_type << (codecId == CodecG711A ? " PCMA/" : " PCMU/") << sample_rate << "\r\n";
+        _printer << "a=control:trackID=" << getTrackType() << "\r\n";
+    }
+
+    string getSdp() const override {
+        return _printer;
+    }
+
+    TrackType getTrackType() const override {
+        return TrackAudio;
+    }
+    CodecId getCodecId() const override {
+        return _codecId;
+    }
+private:
+    _StrPrinter _printer;
+    CodecId _codecId = CodecG711A;
+};
+
+}//namespace mediakit
+
+
+#endif //ZLMEDIAKIT_AAC_H

src/Extension/G711Rtmp.cpp

+/*
+ * Copyright (c) 2016 The ZLMediaKit project authors. All Rights Reserved.
+ *
+ * This file is part of ZLMediaKit(https://github.com/xiongziliang/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.
+ */
+
+#include "G711Rtmp.h"
+
+namespace mediakit{
+
+G711RtmpDecoder::G711RtmpDecoder() {
+    _adts = obtainFrame();
+}
+
+G711Frame::Ptr G711RtmpDecoder::obtainFrame() {
+    //从缓存池重新申请对象，防止覆盖已经写入环形缓存的对象
+    auto frame = ResourcePoolHelper<G711Frame>::obtainObj();
+    frame->frameLength = 0;
+    frame->iPrefixSize = 0;
+    return frame;
+}
+
+bool G711RtmpDecoder::inputRtmp(const RtmpPacket::Ptr &pkt, bool key_pos) {
+    onGetG711(pkt->strBuf.data() + 2, pkt->strBuf.size() - 2, pkt->timeStamp);
+    return false;
+}
+
+void G711RtmpDecoder::onGetG711(const char* pcData, int iLen, uint32_t ui32TimeStamp) {
+    if(iLen + 7 > sizeof(_adts->buffer)){
+        WarnL << "Illegal adts data, exceeding the length limit.";
+        return;
+    }
+
+    //拷贝aac负载
+    memcpy(_adts->buffer, pcData, iLen);
+    _adts->frameLength = iLen;
+    _adts->timeStamp = ui32TimeStamp;
+
+    //写入环形缓存
+    RtmpCodec::inputFrame(_adts);
+    _adts = obtainFrame();
+}
+/////////////////////////////////////////////////////////////////////////////////////
+
+G711RtmpEncoder::G711RtmpEncoder(const Track::Ptr &track) {
+    _track = dynamic_pointer_cast<G711Track>(track);
+}
+
+void G711RtmpEncoder::inputFrame(const Frame::Ptr& frame) {
+
+    RtmpPacket::Ptr rtmpPkt = ResourcePoolHelper<RtmpPacket>::obtainObj();
+    rtmpPkt->strBuf.clear();
+    rtmpPkt->strBuf.append(frame->data() + frame->prefixSize(), frame->size() - frame->prefixSize());
+
+    rtmpPkt->bodySize = rtmpPkt->strBuf.size();
+    rtmpPkt->chunkId = CHUNK_AUDIO;
+    rtmpPkt->streamId = STREAM_MEDIA;
+    rtmpPkt->timeStamp = frame->dts();
+    rtmpPkt->typeId = MSG_AUDIO;
+    RtmpCodec::inputRtmp(rtmpPkt, false);
+
+}
+
+}//namespace mediakit
\ No newline at end of file

src/Extension/G711Rtmp.h

+/*
+ * Copyright (c) 2016 The ZLMediaKit project authors. All Rights Reserved.
+ *
+ * This file is part of ZLMediaKit(https://github.com/xiongziliang/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_G711RTMPCODEC_H
+#define ZLMEDIAKIT_G711RTMPCODEC_H
+
+#include "Rtmp/RtmpCodec.h"
+#include "Extension/Track.h"
+#include "Extension/G711.h"
+
+namespace mediakit{
+/**
+ * G711 Rtmp转adts类
+ */
+class G711RtmpDecoder : public RtmpCodec , public ResourcePoolHelper<G711Frame> {
+public:
+    typedef std::shared_ptr<G711RtmpDecoder> Ptr;
+
+    G711RtmpDecoder();
+    ~G711RtmpDecoder() {}
+
+    /**
+     * 输入Rtmp并解码
+     * @param Rtmp Rtmp数据包
+     * @param key_pos 此参数内部强制转换为false,请忽略之
+     */
+    bool inputRtmp(const RtmpPacket::Ptr &Rtmp, bool key_pos = false) override;
+
+    TrackType getTrackType() const override{
+        return TrackAudio;
+    }
+
+    void setCodecId(CodecId codecId)
+    {
+        _codecid = codecId;
+    }
+
+    CodecId getCodecId() const override{
+        return _codecid;
+    }
+
+protected:
+    void onGetG711(const char* pcData, int iLen, uint32_t ui32TimeStamp);
+    G711Frame::Ptr obtainFrame();
+protected:
+    G711Frame::Ptr _adts;
+    CodecId _codecid = CodecG711A;
+
+};
+
+
+/**
+ * aac adts转Rtmp类
+ */
+class G711RtmpEncoder : public G711RtmpDecoder ,  public ResourcePoolHelper<RtmpPacket> {
+public:
+    typedef std::shared_ptr<G711RtmpEncoder> Ptr;
+
+    /**
+     * 构造函数，track可以为空，此时则在inputFrame时输入adts头
+     * 如果track不为空且包含adts头相关信息，
+     * 那么inputFrame时可以不输入adts头
+     * @param track
+     */
+    G711RtmpEncoder(const Track::Ptr &track);
+    ~G711RtmpEncoder() {}
+
+    /**
+     * 输入aac 数据，可以不带adts头
+     * @param frame aac数据
+     */
+    void inputFrame(const Frame::Ptr &frame) override;
+private:
+    G711Track::Ptr _track;
+};
+
+}//namespace mediakit
+
+#endif //ZLMEDIAKIT_G711RTMPCODEC_H

src/Extension/G711Rtp.cpp

+/*
+ * Copyright (c) 2016 The ZLMediaKit project authors. All Rights Reserved.
+ *
+ * This file is part of ZLMediaKit(https://github.com/xiongziliang/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.
+ */
+
+#include "G711Rtp.h"
+
+namespace mediakit{
+
+G711RtpEncoder::G711RtpEncoder(uint32_t ui32Ssrc,
+                             uint32_t ui32MtuSize,
+                             uint32_t ui32SampleRate,
+                             uint8_t ui8PlayloadType,
+                             uint8_t ui8Interleaved) :
+        RtpInfo(ui32Ssrc,
+                ui32MtuSize,
+                ui32SampleRate,
+                ui8PlayloadType,
+                ui8Interleaved){
+}
+
+void G711RtpEncoder::inputFrame(const Frame::Ptr &frame) {
+    GET_CONFIG(uint32_t, cycleMS, Rtp::kCycleMS);
+    auto uiStamp = frame->dts();
+    auto pcData = frame->data() + frame->prefixSize();
+    auto iLen = frame->size() - frame->prefixSize();
+
+    uiStamp %= cycleMS;
+    char *ptr = (char *) pcData;
+    int iSize = iLen;
+    while (iSize > 0) {
+        if (iSize <= _ui32MtuSize - 20) {
+            makeG711Rtp(ptr, iSize, true, uiStamp);
+            break;
+        }
+        makeG711Rtp(ptr, _ui32MtuSize - 20, false, uiStamp);
+        ptr += (_ui32MtuSize - 20);
+        iSize -= (_ui32MtuSize - 20);
+    }
+}
+
+void G711RtpEncoder::makeG711Rtp(const void *data, unsigned int len, bool mark, uint32_t uiStamp) {
+    RtpCodec::inputRtp(makeRtp(getTrackType(),data,len,mark,uiStamp), false);
+}
+
+/////////////////////////////////////////////////////////////////////////////////////
+
+G711RtpDecoder::G711RtpDecoder(const Track::Ptr &track){
+    auto aacTrack = dynamic_pointer_cast<G711Track>(track);
+    _codecid = aacTrack->getCodecId();
+    if(!aacTrack || !aacTrack->ready()){
+        WarnL << "该g711 track无效!";
+    }else{
+        //_aac_cfg = aacTrack->getAacCfg();
+    }
+    _adts = obtainFrame();
+}
+G711RtpDecoder::G711RtpDecoder() {
+    _adts = obtainFrame();
+}
+
+G711Frame::Ptr G711RtpDecoder::obtainFrame() {
+    //从缓存池重新申请对象，防止覆盖已经写入环形缓存的对象
+    auto frame = ResourcePoolHelper<G711Frame>::obtainObj();
+    frame->frameLength = 0;
+    frame->iPrefixSize = 0;
+    return frame;
+}
+
+bool G711RtpDecoder::inputRtp(const RtpPacket::Ptr &rtppack, bool key_pos) {
+	// 获取rtp数据长度
+    int length = rtppack->size() - rtppack->offset;
+
+	// 获取rtp数据
+	const uint8_t *rtp_packet_buf = (uint8_t *)rtppack->data() + rtppack->offset;
+
+    _adts->frameLength = length;
+    memcpy(_adts->buffer, rtp_packet_buf, length);
+    if (rtppack->mark == true) {
+        _adts->timeStamp = rtppack->timeStamp;
+        onGetG711(_adts);
+    }
+    return false;
+}
+
+void G711RtpDecoder::onGetG711(const G711Frame::Ptr &frame) {
+    //写入环形缓存
+    RtpCodec::inputFrame(frame);
+    _adts = obtainFrame();
+}
+
+
+}//namespace mediakit
+
+
+

src/Extension/G711Rtp.h

+/*
+ * Copyright (c) 2016 The ZLMediaKit project authors. All Rights Reserved.
+ *
+ * This file is part of ZLMediaKit(https://github.com/xiongziliang/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_G711RTPCODEC_H
+#define ZLMEDIAKIT_G711RTPCODEC_H
+
+#include "Rtsp/RtpCodec.h"
+#include "Extension/G711.h"
+namespace mediakit{
+/**
+ * G711 rtp转adts类
+ */
+class G711RtpDecoder : public RtpCodec , public ResourcePoolHelper<G711Frame> {
+public:
+    typedef std::shared_ptr<G711RtpDecoder> Ptr;
+
+    G711RtpDecoder(const Track::Ptr &track);
+    ~G711RtpDecoder() {}
+
+    /**
+     * 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
+     */
+    bool inputRtp(const RtpPacket::Ptr &rtp, bool key_pos = false) override;
+
+    TrackType getTrackType() const override{
+        return TrackAudio;
+    }
+    CodecId getCodecId() const override{
+        return _codecid;
+    }
+protected:
+    G711RtpDecoder();
+private:
+    void onGetG711(const G711Frame::Ptr &frame);
+    G711Frame::Ptr obtainFrame();
+private:
+    G711Frame::Ptr _adts;
+    CodecId _codecid = CodecG711A;
+};
+
+
+/**
+ * g711 转rtp类
+ */
+class G711RtpEncoder : public G711RtpDecoder , public RtpInfo {
+public:
+    typedef std::shared_ptr<G711RtpEncoder> Ptr;
+
+    /**
+     * @param ui32Ssrc ssrc
+     * @param ui32MtuSize mtu 大小
+     * @param ui32SampleRate 采样率
+     * @param ui8PlayloadType pt类型
+     * @param ui8Interleaved rtsp interleaved 值
+     */
+    G711RtpEncoder(uint32_t ui32Ssrc,
+                  uint32_t ui32MtuSize,
+                  uint32_t ui32SampleRate,
+                  uint8_t ui8PlayloadType = 0,
+                  uint8_t ui8Interleaved = TrackAudio * 2);
+    ~G711RtpEncoder() {}
+
+    /**
+     * @param frame g711数据
+     */
+    void inputFrame(const Frame::Ptr &frame) override;
+private:
+    void makeG711Rtp(const void *pData, unsigned int uiLen, bool bMark, uint32_t uiStamp);
+private:
+    unsigned char _aucSectionBuf[1600];
+};
+
+}//namespace mediakit
+
+#endif //ZLMEDIAKIT_G711RTPCODEC_H

src/Rtsp/RtspSession.cpp

@@ -779,7 +779,7 @@ void RtspSession::handleReq_Play(const Parser &parser) {
         rtp_info.pop_back();
 
 		sendRtspResponse("200 OK",
-						 {"Range", StrPrinter << "npt=" << setiosflags(ios::fixed) << setprecision(2) <<  /*pMediaSrc->getTimeStamp(TrackInvalid) / 1000.0*/iStartTime/1000,
+						 {"Range", StrPrinter << "npt=" << setiosflags(ios::fixed) << setprecision(2) << (useBuf? iStartTime / 1000:pMediaSrc->getTimeStamp(TrackInvalid) / 1000.0),
 						  "RTP-Info",rtp_info
 						 });