diff --git a/src/Common/Device.cpp b/src/Common/Device.cpp
index 06de6d10..4ca91fac 100644
--- a/src/Common/Device.cpp
+++ b/src/Common/Device.cpp
@@ -196,7 +196,7 @@ void DevChannel::initAudio(const AudioInfo& info) {
         }
 
         case CodecG711A :
-        case CodecG711U : addTrack(std::make_shared<G711Track>(info.codecId)); break;
+        case CodecG711U : addTrack(std::make_shared<G711Track>(info.codecId, info.iSampleRate, info.iChannel, info.iSampleBit)); break;
         default: WarnL << "不支持该类型的音频编码类型:" << info.codecId; break;
     }
 }
diff --git a/src/Common/MediaSink.cpp b/src/Common/MediaSink.cpp
index aa5e023c..53697a56 100644
--- a/src/Common/MediaSink.cpp
+++ b/src/Common/MediaSink.cpp
@@ -42,8 +42,8 @@ void MediaSink::addTrack(const Track::Ptr &track_in) {
         if (_allTrackReady) {
             //运行至这里说明Track状态由未就绪切换为已就绪状态,那么这帧就不应该丢弃
             onTrackFrame(frame);
-        } else {
-            ErrorL << "some track is unready，drop frame of: " << frame->getCodecName();
+        } else if(frame->keyFrame()){
+            WarnL << "some track is unready，drop key frame of: " << frame->getCodecName();
         }
     }));
 }
diff --git a/src/Extension/AACRtmp.cpp b/src/Extension/AACRtmp.cpp
index ac66f306..6a22e45e 100644
--- a/src/Extension/AACRtmp.cpp
+++ b/src/Extension/AACRtmp.cpp
@@ -104,7 +104,7 @@ void AACRtmpEncoder::inputFrame(const Frame::Ptr &frame) {
 
         //header
         uint8_t is_config = false;
-        rtmpPkt->strBuf.push_back(_ui8AudioFlags);
+        rtmpPkt->strBuf.push_back(_audio_flv_flags);
         rtmpPkt->strBuf.push_back(!is_config);
 
         //aac data
@@ -117,44 +117,16 @@ void AACRtmpEncoder::inputFrame(const Frame::Ptr &frame) {
         rtmpPkt->typeId = MSG_AUDIO;
         RtmpCodec::inputRtmp(rtmpPkt, false);
     }
-
 }
 
 void AACRtmpEncoder::makeAudioConfigPkt() {
-    makeAdtsHeader(_aac_cfg,*_adts);
-    int iSampleRate , iChannel , iSampleBit = 16;
-    getAACInfo(*_adts,iSampleRate,iChannel);
-
-    uint8_t flvStereoOrMono = (iChannel > 1);
-    uint8_t flvSampleRate;
-    switch (iSampleRate) {
-        case 48000:
-        case 44100:
-            flvSampleRate = 3;
-            break;
-        case 24000:
-        case 22050:
-            flvSampleRate = 2;
-            break;
-        case 12000:
-        case 11025:
-            flvSampleRate = 1;
-            break;
-        default:
-            flvSampleRate = 0;
-            break;
-    }
-    uint8_t flvSampleBit = iSampleBit == 16;
-    uint8_t flvAudioType = FLV_CODEC_AAC;
-
-    _ui8AudioFlags = (flvAudioType << 4) | (flvSampleRate << 2) | (flvSampleBit << 1) | flvStereoOrMono;
-
+    _audio_flv_flags = getAudioRtmpFlags(std::make_shared<AACTrack>(_aac_cfg));
     RtmpPacket::Ptr rtmpPkt = ResourcePoolHelper<RtmpPacket>::obtainObj();
     rtmpPkt->strBuf.clear();
 
     //header
     uint8_t is_config = true;
-    rtmpPkt->strBuf.push_back(_ui8AudioFlags);
+    rtmpPkt->strBuf.push_back(_audio_flv_flags);
     rtmpPkt->strBuf.push_back(!is_config);
     //aac config
     rtmpPkt->strBuf.append(_aac_cfg);
diff --git a/src/Extension/AACRtmp.h b/src/Extension/AACRtmp.h
index 4d245dd7..20f424d5 100644
--- a/src/Extension/AACRtmp.h
+++ b/src/Extension/AACRtmp.h
@@ -79,7 +79,7 @@ public:
 private:
     void makeAudioConfigPkt();
 private:
-    uint8_t _ui8AudioFlags;
+    uint8_t _audio_flv_flags;
     AACTrack::Ptr _track;
 };
 
diff --git a/src/Extension/Factory.cpp b/src/Extension/Factory.cpp
index 5f5e3491..4bdb965c 100644
--- a/src/Extension/Factory.cpp
+++ b/src/Extension/Factory.cpp
@@ -48,11 +48,11 @@ Track::Ptr Factory::getTrackBySdp(const SdpTrack::Ptr &track) {
     }
 
     if (strcasecmp(track->_codec.data(), "PCMA") == 0) {
-        return std::make_shared<G711Track>(CodecG711A);
+        return std::make_shared<G711Track>(CodecG711A, track->_samplerate, track->_channel, 16);
     }
 
     if (strcasecmp(track->_codec.data(), "PCMU") == 0) {
-        return std::make_shared<G711Track>(CodecG711U);
+        return std::make_shared<G711Track>(CodecG711U,  track->_samplerate, track->_channel, 16);
     }
 
     if (strcasecmp(track->_codec.data(), "h264") == 0) {
@@ -88,29 +88,6 @@ Track::Ptr Factory::getTrackBySdp(const SdpTrack::Ptr &track) {
     return nullptr;
 }
 
-Track::Ptr Factory::getTrackByCodecId(CodecId codecId) {
-    switch (codecId){
-        case CodecH264:{
-            return std::make_shared<H264Track>();
-        }
-        case CodecH265:{
-            return std::make_shared<H265Track>();
-        }
-        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;
-    }
-}
-
 RtpCodec::Ptr Factory::getRtpEncoderBySdp(const Sdp::Ptr &sdp) {
     GET_CONFIG(uint32_t,audio_mtu,Rtp::kAudioMtuSize);
     GET_CONFIG(uint32_t,video_mtu,Rtp::kVideoMtuSize);
@@ -134,35 +111,23 @@ RtpCodec::Ptr Factory::getRtpEncoderBySdp(const Sdp::Ptr &sdp) {
     auto interleaved = sdp->getTrackType() * 2;
     auto codec_id = sdp->getCodecId();
     switch (codec_id){
-        case CodecH264:
-            return std::make_shared<H264RtpEncoder>(ssrc,mtu,sample_rate,pt,interleaved);
-        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 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;
+        case CodecH264 : return std::make_shared<H264RtpEncoder>(ssrc,mtu,sample_rate,pt,interleaved);
+        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 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;
     }
 }
 
 RtpCodec::Ptr Factory::getRtpDecoderByTrack(const Track::Ptr &track) {
     switch (track->getCodecId()){
-        case CodecH264:
-            return std::make_shared<H264RtpDecoder>();
-        case CodecH265:
-            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;
+        case CodecH264 : return std::make_shared<H264RtpDecoder>();
+        case CodecH265 : 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;
     }
 }
 
@@ -190,15 +155,25 @@ static CodecId getVideoCodecIdByAmf(const AMFValue &val){
             case FLV_CODEC_H264: return CodecH264;
             case FLV_CODEC_AAC: return CodecAAC;
             case FLV_CODEC_H265: return CodecH265;
-            default:
-                WarnL << "暂不支持该Amf:" << type_id;
-                return CodecInvalid;
+            default : WarnL << "暂不支持该Amf:" << type_id; return CodecInvalid;
         }
     }
 
     return CodecInvalid;
 }
 
+
+Track::Ptr getTrackByCodecId(CodecId codecId, int sample_rate = 0, int channels = 0, int sample_bit = 0) {
+    switch (codecId){
+        case CodecH264 : return std::make_shared<H264Track>();
+        case CodecH265 : return std::make_shared<H265Track>();
+        case CodecAAC : return std::make_shared<AACTrack>();
+        case CodecG711A :
+        case CodecG711U : return (sample_rate && channels && sample_bit) ? std::make_shared<G711Track>(codecId, sample_rate, channels, sample_bit) : nullptr;
+        default : WarnL << "暂不支持该CodecId:" << codecId; return nullptr;
+    }
+}
+
 Track::Ptr Factory::getVideoTrackByAmf(const AMFValue &amf) {
     CodecId codecId = getVideoCodecIdByAmf(amf);
     if(codecId == CodecInvalid){
@@ -230,28 +205,22 @@ static CodecId getAudioCodecIdByAmf(const AMFValue &val) {
     return CodecInvalid;
 }
 
-Track::Ptr Factory::getAudioTrackByAmf(const AMFValue& amf){
+Track::Ptr Factory::getAudioTrackByAmf(const AMFValue& amf, int sample_rate, int channels, int sample_bit){
     CodecId codecId = getAudioCodecIdByAmf(amf);
     if (codecId == CodecInvalid) {
         return nullptr;
     }
-    return getTrackByCodecId(codecId);
+    return getTrackByCodecId(codecId, sample_rate, channels, sample_bit);
 }
 
 RtmpCodec::Ptr Factory::getRtmpCodecByTrack(const Track::Ptr &track) {
     switch (track->getCodecId()){
-        case CodecH264:
-            return std::make_shared<H264RtmpEncoder>(track);
-        case CodecAAC:
-            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;
+        case CodecH264 : return std::make_shared<H264RtmpEncoder>(track);
+        case CodecAAC : 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;
     }
 }
 
diff --git a/src/Extension/Factory.h b/src/Extension/Factory.h
index 07398637..6b678d34 100644
--- a/src/Extension/Factory.h
+++ b/src/Extension/Factory.h
@@ -24,13 +24,6 @@ namespace mediakit{
 
 class Factory {
 public:
-
-    /**
-     * 根据CodecId获取Track，该Track的ready()状态一般都为false
-     * @param codecId 编解码器id
-     */
-    static Track::Ptr getTrackByCodecId(CodecId codecId);
-
     ////////////////////////////////rtsp相关//////////////////////////////////
     /**
      * 根据sdp生成Track对象
@@ -61,7 +54,7 @@ public:
      * 根据amf对象获取音频相应的Track
      * @param amf rtmp metadata中的audiocodecid的值
      */
-    static Track::Ptr getAudioTrackByAmf(const AMFValue& amf);
+    static Track::Ptr getAudioTrackByAmf(const AMFValue& amf, int sample_rate, int channels, int sample_bit);
 
     /**
      * 根据Track获取Rtmp的编解码器
diff --git a/src/Extension/G711.cpp b/src/Extension/G711.cpp
index 23194e2b..15c3d74e 100644
--- a/src/Extension/G711.cpp
+++ b/src/Extension/G711.cpp
@@ -12,13 +12,12 @@
 
 namespace mediakit{
 
-
 Sdp::Ptr G711Track::getSdp() {
     if(!ready()){
         WarnL << getCodecName() << " Track未准备好";
         return nullptr;
     }
-    return std::make_shared<G711Sdp>(getCodecId());
+    return std::make_shared<G711Sdp>(getCodecId(), getAudioSampleRate(), getAudioChannel());
 }
 
 }//namespace mediakit
diff --git a/src/Extension/G711.h b/src/Extension/G711.h
index 056ac1dd..d1644201 100644
--- a/src/Extension/G711.h
+++ b/src/Extension/G711.h
@@ -102,8 +102,11 @@ public:
     /**
      * G711A G711U
      */
-    G711Track(CodecId codecId){
+    G711Track(CodecId codecId,int sample_rate, int channels, int sample_bit){
         _codecid = codecId;
+        _sample_rate = sample_rate;
+        _channels = channels;
+        _sample_bit = sample_bit;
     }
 
     /**
@@ -114,7 +117,7 @@ public:
     }
 
     /**
-     * G711的Track不需要初始化
+     * 是否已经初始化
      */
     bool ready() override {
         return true;
@@ -124,21 +127,21 @@ public:
      * 返回音频采样率
      */
     int getAudioSampleRate() const override{
-        return 8000;
+        return _sample_rate;
     }
 
     /**
      * 返回音频采样位数，一般为16或8
      */
     int getAudioSampleBit() const override{
-        return 16;
+        return _sample_bit;
     }
 
     /**
      * 返回音频通道数
      */
     int getAudioChannel() const override{
-        return 1;
+        return _channels;
     }
 
 private:
@@ -149,7 +152,10 @@ private:
     //生成sdp
     Sdp::Ptr getSdp() override ;
 private:
-    CodecId _codecid = CodecG711A;
+    CodecId _codecid;
+    int _sample_rate;
+    int _channels;
+    int _sample_bit;
 };
 
 /**
@@ -160,11 +166,17 @@ public:
     /**
      * G711采样率固定为8000
      * @param codecId G711A G711U
+     * @param sample_rate 音频采样率
+     * @param playload_type rtp playload
+     * @param bitrate 比特率
      */
-    G711Sdp(CodecId codecId) : Sdp(8000,payloadType(codecId)), _codecId(codecId){
-        int pt = payloadType(codecId);
-        _printer << "m=audio 0 RTP/AVP " << pt << "\r\n";
-        _printer << "a=rtpmap:" << pt << (codecId == CodecG711A ? " PCMA/" : " PCMU/") << 8000 << "\r\n";
+    G711Sdp(CodecId codecId,
+            int sample_rate,
+            int channels,
+            int playload_type = 98,
+            int bitrate = 128) : Sdp(sample_rate,playload_type), _codecId(codecId){
+        _printer << "m=audio 0 RTP/AVP " << playload_type << "\r\n";
+        _printer << "a=rtpmap:" << playload_type << (codecId == CodecG711A ? " PCMA/" : " PCMU/") << sample_rate  << "/" << channels << "\r\n";
         _printer << "a=control:trackID=" << getTrackType() << "\r\n";
     }
 
@@ -180,14 +192,6 @@ public:
         return _codecId;
     }
 
-    int payloadType(CodecId codecId){
-        switch (codecId){
-            case CodecG711A : return 8;
-            case CodecG711U : return 0;
-            default : return  -1;
-        }
-    }
-
 private:
     _StrPrinter _printer;
     CodecId _codecId;
diff --git a/src/Extension/G711Rtmp.cpp b/src/Extension/G711Rtmp.cpp
index 854772fc..11e46d61 100644
--- a/src/Extension/G711Rtmp.cpp
+++ b/src/Extension/G711Rtmp.cpp
@@ -27,7 +27,7 @@ G711Frame::Ptr G711RtmpDecoder::obtainFrame() {
 
 bool G711RtmpDecoder::inputRtmp(const RtmpPacket::Ptr &pkt, bool) {
     //拷贝G711负载
-    _frame->buffer.assign(pkt->strBuf.data() + 2, pkt->strBuf.size() - 2);
+    _frame->buffer.assign(pkt->strBuf.data() + 1, pkt->strBuf.size() - 1);
     _frame->timeStamp = pkt->timeStamp;
     //写入环形缓存
     RtmpCodec::inputFrame(_frame);
@@ -38,55 +38,17 @@ bool G711RtmpDecoder::inputRtmp(const RtmpPacket::Ptr &pkt, bool) {
 /////////////////////////////////////////////////////////////////////////////////////
 
 G711RtmpEncoder::G711RtmpEncoder(const Track::Ptr &track) : G711RtmpDecoder(track->getCodecId()) {
-    auto g711_track = dynamic_pointer_cast<AudioTrack>(track);
-    if(!g711_track){
-        WarnL << "无效的G711 track, 将忽略打包为RTMP";
-        return;
-    }
-
-    auto iSampleRate = g711_track->getAudioSampleRate() ;
-    auto iChannel =  g711_track->getAudioChannel();
-    auto iSampleBit = g711_track->getAudioSampleBit();
-    uint8_t flvStereoOrMono = (iChannel > 1);
-    uint8_t flvSampleRate;
-    switch (iSampleRate) {
-        case 48000:
-        case 44100:
-            flvSampleRate = 3;
-            break;
-        case 24000:
-        case 22050:
-            flvSampleRate = 2;
-            break;
-        case 12000:
-        case 11025:
-            flvSampleRate = 1;
-            break;
-        default:
-            flvSampleRate = 0;
-            break;
-    }
-    uint8_t flvSampleBit = iSampleBit == 16;
-    uint8_t flvAudioType ;
-    switch (g711_track->getCodecId()){
-        case CodecG711A : flvAudioType = FLV_CODEC_G711A; break;
-        case CodecG711U : flvAudioType = FLV_CODEC_G711U; break;
-        default: WarnL << "无效的G711 track, 将忽略打包为RTMP"; return ;
-    }
-
-    _g711_flags = (flvAudioType << 4) | (flvSampleRate << 2) | (flvSampleBit << 1) | flvStereoOrMono;
+    _audio_flv_flags = getAudioRtmpFlags(track);
 }
 
 void G711RtmpEncoder::inputFrame(const Frame::Ptr &frame) {
-    if(!_g711_flags){
+    if(!_audio_flv_flags){
         return;
     }
     RtmpPacket::Ptr rtmpPkt = ResourcePoolHelper<RtmpPacket>::obtainObj();
     rtmpPkt->strBuf.clear();
     //header
-    uint8_t is_config = false;
-    rtmpPkt->strBuf.push_back(_g711_flags);
-    rtmpPkt->strBuf.push_back(!is_config);
+    rtmpPkt->strBuf.push_back(_audio_flv_flags);
 
     //g711 data
     rtmpPkt->strBuf.append(frame->data() + frame->prefixSize(), frame->size() - frame->prefixSize());
diff --git a/src/Extension/G711Rtmp.h b/src/Extension/G711Rtmp.h
index 74b697e4..71f3e069 100644
--- a/src/Extension/G711Rtmp.h
+++ b/src/Extension/G711Rtmp.h
@@ -62,7 +62,7 @@ public:
      */
     void inputFrame(const Frame::Ptr &frame) override;
 private:
-    uint8_t _g711_flags = 0;
+    uint8_t _audio_flv_flags = 0;
 };
 
 }//namespace mediakit
diff --git a/src/Rtmp/Rtmp.cpp b/src/Rtmp/Rtmp.cpp
index 52a9111b..4749f3ea 100644
--- a/src/Rtmp/Rtmp.cpp
+++ b/src/Rtmp/Rtmp.cpp
@@ -36,11 +36,68 @@ AudioMeta::AudioMeta(const AudioTrack::Ptr &audio,int datarate){
         _metadata.set("audiosamplesize", audio->getAudioSampleBit());
     }
     if(audio->getAudioChannel() > 0){
-        _metadata.set("audiochannels", audio->getAudioChannel());
         _metadata.set("stereo", audio->getAudioChannel() > 1);
     }
     _codecId = audio->getCodecId();
     _metadata.set("audiocodecid", Factory::getAmfByCodecId(_codecId));
 }
 
+uint8_t getAudioRtmpFlags(const Track::Ptr &track){
+    switch (track->getTrackType()){
+        case TrackAudio : {
+            auto audioTrack = dynamic_pointer_cast<AudioTrack>(track);
+            if (!audioTrack) {
+                WarnL << "获取AudioTrack失败";
+                return 0;
+            }
+            auto iSampleRate = audioTrack->getAudioSampleRate();
+            auto iChannel = audioTrack->getAudioChannel();
+            auto iSampleBit = audioTrack->getAudioSampleBit();
+
+            uint8_t flvAudioType ;
+            switch (track->getCodecId()){
+                case CodecG711A : flvAudioType = FLV_CODEC_G711A; break;
+                case CodecG711U : flvAudioType = FLV_CODEC_G711U; break;
+                case CodecAAC : {
+                    flvAudioType = FLV_CODEC_AAC;
+                    //aac不通过flags获取音频相关信息
+                    iSampleRate = 44100;
+                    iSampleBit = 16;
+                    iChannel = 2;
+                    break;
+                }
+                default: WarnL << "该编码格式不支持转换为RTMP: " << track->getCodecName(); return 0;
+            }
+
+            uint8_t flvSampleRate;
+            switch (iSampleRate) {
+                case 44100:
+                    flvSampleRate = 3;
+                    break;
+                case 22050:
+                    flvSampleRate = 2;
+                    break;
+                case 11025:
+                    flvSampleRate = 1;
+                    break;
+                case 16000: // nellymoser only
+                case 8000: // nellymoser only
+                case 5512: // not MP3
+                    flvSampleRate = 0;
+                    break;
+                default:
+                    WarnL << "FLV does not support sample rate " << iSampleRate << " ,choose from (44100, 22050, 11025)";
+                    return 0;
+            }
+
+            uint8_t flvStereoOrMono = (iChannel > 1);
+            uint8_t flvSampleBit = iSampleBit == 16;
+            return (flvAudioType << 4) | (flvSampleRate << 2) | (flvSampleBit << 1) | flvStereoOrMono;
+        }
+
+        default : return 0;
+    }
+}
+
+
 }//namespace mediakit
\ No newline at end of file
diff --git a/src/Rtmp/Rtmp.h b/src/Rtmp/Rtmp.h
index 073f8b04..ec4c4e72 100644
--- a/src/Rtmp/Rtmp.h
+++ b/src/Rtmp/Rtmp.h
@@ -161,27 +161,26 @@ public:
         strBuf = std::move(that.strBuf);
     }
     bool isVideoKeyFrame() const {
-        return typeId == MSG_VIDEO && (uint8_t) strBuf[0] >> 4 == FLV_KEY_FRAME
-        && (uint8_t) strBuf[1] == 1;
+        return typeId == MSG_VIDEO && (uint8_t) strBuf[0] >> 4 == FLV_KEY_FRAME && (uint8_t) strBuf[1] == 1;
     }
     bool isCfgFrame() const {
-        return (typeId == MSG_VIDEO || typeId == MSG_AUDIO)
-        && (uint8_t) strBuf[1] == 0;
+        switch (typeId){
+            case MSG_VIDEO : return strBuf[1] == 0;
+            case MSG_AUDIO : {
+                switch (getMediaType()){
+                    case FLV_CODEC_AAC : return strBuf[1] == 0;
+                    default : return false;
+                }
+            }
+            default : return false;
+        }
     }
     int getMediaType() const {
         switch (typeId) {
-            case MSG_VIDEO: {
-                return (uint8_t) strBuf[0] & 0x0F;
-            }
-                break;
-            case MSG_AUDIO: {
-                return (uint8_t) strBuf[0] >> 4;
-            }
-                break;
-            default:
-                break;
+            case MSG_VIDEO : return (uint8_t) strBuf[0] & 0x0F;
+            case MSG_AUDIO : return (uint8_t) strBuf[0] >> 4;
+            default : return 0;
         }
-        return 0;
     }
     int getAudioSampleRate() const {
         if (typeId != MSG_AUDIO) {
@@ -314,6 +313,8 @@ private:
     CodecId _codecId;
 };
 
+//根据音频track获取flags
+uint8_t getAudioRtmpFlags(const Track::Ptr &track);
 
 }//namespace mediakit
 
diff --git a/src/Rtmp/RtmpDemuxer.cpp b/src/Rtmp/RtmpDemuxer.cpp
index 26a3106f..4848a960 100644
--- a/src/Rtmp/RtmpDemuxer.cpp
+++ b/src/Rtmp/RtmpDemuxer.cpp
@@ -15,14 +15,55 @@ namespace mediakit {
 
 void RtmpDemuxer::loadMetaData(const AMFValue &val){
     try {
-        makeVideoTrack(val["videocodecid"]);
-        makeAudioTrack(val["audiocodecid"]);
+        int audiosamplerate = 0;
+        int audiochannels = 0;
+        int audiosamplesize = 0;
+        const AMFValue *audiocodecid = nullptr;
+        const AMFValue *videocodecid = nullptr;
+
         val.object_for_each([&](const string &key, const AMFValue &val) {
             if (key == "duration") {
                 _fDuration = val.as_number();
                 return;
             }
+
+            if(key == "audiosamplerate"){
+                audiosamplerate = val.as_integer();
+                return;
+            }
+
+            if(key == "audiosamplesize"){
+                audiosamplesize = val.as_integer();
+                return;
+            }
+
+            if(key == "stereo"){
+                audiochannels = val.as_boolean() ? 2 : 1;
+                return;
+            }
+
+            if(key == "videocodecid"){
+                //找到视频
+                videocodecid = &val;
+                return;
+            }
+
+            if(key == "audiocodecid"){
+                //找到音频
+                audiocodecid = &val;
+                return;
+            }
         });
+
+        if(videocodecid){
+            //有视频
+            makeVideoTrack(*videocodecid);
+        }
+
+        if(audiocodecid){
+            //有音频
+            makeAudioTrack(*audiocodecid, audiosamplerate, audiochannels, audiosamplesize);
+        }
     }catch (std::exception &ex){
         WarnL << ex.what();
     }
@@ -46,7 +87,7 @@ bool RtmpDemuxer::inputRtmp(const RtmpPacket::Ptr &pkt) {
             if(!_tryedGetAudioTrack) {
                 _tryedGetAudioTrack = true;
                 auto codec = AMFValue(pkt->getMediaType());
-                makeAudioTrack(codec);
+                makeAudioTrack(codec, pkt->getAudioSampleRate(), pkt->getAudioChannel(), pkt->getAudioSampleBit());
             }
             if(_audioRtmpDecoder){
                 _audioRtmpDecoder->inputRtmp(pkt, false);
@@ -69,6 +110,7 @@ void RtmpDemuxer::makeVideoTrack(const AMFValue &videoCodec) {
             //设置rtmp解码器代理，生成的frame写入该Track
             _videoRtmpDecoder->addDelegate(_videoTrack);
             onAddTrack(_videoTrack);
+            _tryedGetVideoTrack = true;
         } else {
             //找不到相应的rtmp解码器，该track无效
             _videoTrack.reset();
@@ -76,9 +118,9 @@ void RtmpDemuxer::makeVideoTrack(const AMFValue &videoCodec) {
     }
 }
 
-void RtmpDemuxer::makeAudioTrack(const AMFValue &audioCodec) {
+void RtmpDemuxer::makeAudioTrack(const AMFValue &audioCodec,int sample_rate, int channels, int sample_bit) {
     //生成Track对象
-    _audioTrack = dynamic_pointer_cast<AudioTrack>(Factory::getAudioTrackByAmf(audioCodec));
+    _audioTrack = dynamic_pointer_cast<AudioTrack>(Factory::getAudioTrackByAmf(audioCodec, sample_rate, channels, sample_bit));
     if (_audioTrack) {
         //生成rtmpCodec对象以便解码rtmp
         _audioRtmpDecoder = Factory::getRtmpCodecByTrack(_audioTrack);
@@ -86,6 +128,7 @@ void RtmpDemuxer::makeAudioTrack(const AMFValue &audioCodec) {
             //设置rtmp解码器代理，生成的frame写入该Track
             _audioRtmpDecoder->addDelegate(_audioTrack);
             onAddTrack(_audioTrack);
+            _tryedGetAudioTrack = true;
         } else {
             //找不到相应的rtmp解码器，该track无效
             _audioTrack.reset();
diff --git a/src/Rtmp/RtmpDemuxer.h b/src/Rtmp/RtmpDemuxer.h
index 6d1c264d..fbff5d2e 100644
--- a/src/Rtmp/RtmpDemuxer.h
+++ b/src/Rtmp/RtmpDemuxer.h
@@ -40,7 +40,7 @@ public:
     bool inputRtmp(const RtmpPacket::Ptr &pkt);
 private:
     void makeVideoTrack(const AMFValue &val);
-    void makeAudioTrack(const AMFValue &val);
+    void makeAudioTrack(const AMFValue &val, int sample_rate, int channels, int sample_bit);
 private:
     bool _tryedGetVideoTrack = false;
     bool _tryedGetAudioTrack = false;
diff --git a/src/Rtmp/RtmpSession.cpp b/src/Rtmp/RtmpSession.cpp
index 8d0d479e..8c9028fc 100644
--- a/src/Rtmp/RtmpSession.cpp
+++ b/src/Rtmp/RtmpSession.cpp
@@ -193,7 +193,6 @@ void RtmpSession::onCmd_deleteStream(AMFDecoder &dec) {
     throw std::runtime_error(StrPrinter << "Stop publishing" << endl);
 }
 
-
 void RtmpSession::sendPlayResponse(const string &err,const RtmpMediaSource::Ptr &src){
     bool authSuccess = err.empty();
     bool ok = (src.operator bool() && authSuccess);
diff --git a/src/Rtsp/Rtsp.cpp b/src/Rtsp/Rtsp.cpp
index 8ae86f84..7828b5dc 100644
--- a/src/Rtsp/Rtsp.cpp
+++ b/src/Rtsp/Rtsp.cpp
@@ -172,6 +172,7 @@ void SdpParser::load(const string &sdp) {
                     if (4 == sscanf(opt_val.data(), " %15[^ ] %d %15[^ ] %d", type, &port, rtp, &pt)) {
                         track->_pt = pt;
                         track->_samplerate = RtpPayload::getClockRate(pt) ;
+                        track->_channel = RtpPayload::getAudioChannel(pt);
                         track->_type = toTrackType(type);
                         track->_m = opt_val;
                         track->_port = port;
@@ -214,9 +215,14 @@ void SdpParser::load(const string &sdp) {
         it = track._attr.find("rtpmap");
         if(it != track._attr.end()){
             auto rtpmap = it->second;
-            int pt, samplerate;
+            int pt, samplerate, channel;
             char codec[16] = {0};
-            if (3 == sscanf(rtpmap.data(), "%d %15[^/]/%d", &pt, codec, &samplerate)) {
+            if (4 == sscanf(rtpmap.data(), "%d %15[^/]/%d/%d", &pt, codec, &samplerate, &channel)) {
+                track._pt = pt;
+                track._codec = codec;
+                track._samplerate = samplerate;
+                track._channel = channel;
+            }else if (3 == sscanf(rtpmap.data(), "%d %15[^/]/%d", &pt, codec, &samplerate)) {
                 track._pt = pt;
                 track._codec = codec;
                 track._samplerate = samplerate;
diff --git a/src/Rtsp/Rtsp.h b/src/Rtsp/Rtsp.h
index a1debdf4..bf8040a3 100644
--- a/src/Rtsp/Rtsp.h
+++ b/src/Rtsp/Rtsp.h
@@ -128,6 +128,7 @@ public:
     int _pt;
     string _codec;
     int _samplerate;
+    int _channel;
     string _fmtp;
     string _control;
     string _control_surffix;