support build-in speex to aac,controled by the root config keyword: speex2aac on/off

+[submodule "trunk/3rdparty/speex"]
+	path = trunk/3rdparty/speex
+	url = git@123.56.226.173:hubin/speex.git
+[submodule "trunk/3rdparty/fdk-aac"]
+	path = trunk/3rdparty/fdk-aac
+	url = git@123.56.226.173:hubin/fdk-aac.git

+ Subproject commit db608feeed80cb1e7d6db6818293fa24900d967a

+ Subproject commit 6da384f4f06d8db9510937aa0206c340dcacc7a2

@@ -62,6 +62,9 @@ work_dir ./;
 # @reamrk do not support reload.
 # default: off
 asprocess off;
+# whether using build-in speex to aac transcoding
+# default: off
+speex2aac off;
 
 #############################################################################################
 # heartbeat/stats sections
@@ -928,6 +931,16 @@ vhost same.vhost.forward.srs.com {
     # active-active for cdn to build high available fault tolerance system.
     # format: {ip}:{port} {ip_N}:{port_N}
     forward         127.0.0.1:1936 127.0.0.1:1937;
+    # forward_in_turn all publish stream to the specified server in turn.
+    # this used to split/forward the current stream for transcode and make load balance  
+    # active-active for cdn to build high available fault tolerance system.
+    # format: {ip}:{port} {ip_N}:{port_N}
+    forward_in_turn         127.0.0.1:19351 127.0.0.1:19352;
+    #if the forward server is same as this server , a origin,try use forward_peer,
+    #the stream pushed from other forward peer will not forward any more in this server
+    forward_peer 127.0.0.1:1936;
+    #list out the servers which is not srs ,used in forward or forward_in_turn,and forward_peer for example,forward to some CDN
+    forward_server_other    127.0.0.1:19351;
 }
 
 # the main comments for transcode

+# main config for srs.
+# @see full.conf for detail config.
+
+listen              1935;
+max_connections     1000;
+srs_log_tank        file;
+srs_log_file        ./objs/srs.log;
+speex2aac           on;
+vhost __defaultVhost__ {
+}

-#!/bin/bash
-
 #####################################################################################
 # the main output dir, all configure and make output are in this dir.
 #####################################################################################
@@ -125,6 +123,8 @@ if [ $SRS_SSL = YES ]; then if [ $SRS_USE_SYS_SSL = NO ]; then LibSSLRoot="${SRS
 # gperftools-2.1, for mem check and mem/cpu profile
 LibGperfRoot=""; LibGperfFile=""
 if [ $SRS_GPERF = YES ]; then LibGperfRoot="${SRS_OBJS_DIR}/gperf/include"; LibGperfFile="${SRS_OBJS_DIR}/gperf/lib/libtcmalloc_and_profiler.a"; fi
+# codec libs:speex fdk-aac for SPEEX2AAC
+LibSpeex2aacRoot="${SRS_OBJS_DIR}/libs/include"; LibSpeex2aacfile="${SRS_OBJS_DIR}/libs/lib/libspeex.a ${SRS_OBJS_DIR}/libs/lib/libfdk-aac.a"
 # the link options, always use static link
 SrsLinkOptions="-ldl"; 
 if [ $SRS_SSL = YES ]; then if [ $SRS_USE_SYS_SSL = YES ]; then SrsLinkOptions="${SrsLinkOptions} -lssl -lcrypto"; fi fi
@@ -166,10 +166,18 @@ MODULE_FILES=("srs_rtmp_amf0" "srs_rtmp_io" "srs_rtmp_stack"
 PROTOCOL_INCS="src/protocol"; MODULE_DIR=${PROTOCOL_INCS} . auto/modules.sh
 PROTOCOL_OBJS="${MODULE_OBJS[@]}"
 #
+#SPEEX2AAC speex to aac
+MODULE_ID="SPEEX2AAC"
+MODULE_DEPENDS=("CORE" "KERNEL") 
+ModuleLibIncs=(${SRS_OBJS_DIR} ${LibSpeex2aacRoot})
+MODULE_FILES=("srs_tc_adc_speex"  "srs_tc_aec_aac" "srs_tc_common")
+SPEEX2AAC_INCS="src/transcode"; MODULE_DIR=${SPEEX2AAC_INCS} . auto/modules.sh
+SPEEX2AAC_OBJS="${MODULE_OBJS[@]}"
+#
 #App Module
 if [ $SRS_EXPORT_LIBRTMP_PROJECT = NO ]; then
     MODULE_ID="APP" 
-    MODULE_DEPENDS=("CORE" "KERNEL" "PROTOCOL") 
+    MODULE_DEPENDS=("CORE" "KERNEL" "PROTOCOL" "SPEEX2AAC") 
     ModuleLibIncs=(${LibSTRoot} ${LibHttpParserRoot} ${LibSSLRoot} ${SRS_OBJS_DIR})
     MODULE_FILES=("srs_app_server" "srs_app_conn" "srs_app_rtmp_conn" "srs_app_source" 
             "srs_app_refer" "srs_app_hls" "srs_app_forward" "srs_app_encoder" "srs_app_http_stream"
@@ -202,8 +210,8 @@ LIBS_OBJS="${MODULE_OBJS[@]}"
 #Main Module
 if [ $SRS_EXPORT_LIBRTMP_PROJECT = NO ]; then
     MODULE_ID="MAIN" 
-    MODULE_DEPENDS=("CORE" "KERNEL" "PROTOCOL" "APP")
-    ModuleLibIncs=(${LibSTRoot} ${SRS_OBJS_DIR} ${LibGperfRoot} ${LibHttpParserRoot} ${LibSSLRoot})
+    MODULE_DEPENDS=("CORE" "KERNEL" "PROTOCOL" "APP" "SPEEX2AAC")
+    ModuleLibIncs=(${LibSTRoot} ${SRS_OBJS_DIR} ${LibGperfRoot} ${LibHttpParserRoot} ${LibSSLRoot} ${LibSpeex2aacRoot})
     MODULE_FILES=("srs_main_server" "srs_main_ingest_hls")
     # add each modules for main
     for SRS_MODULE in ${SRS_MODULES[*]}; do
@@ -229,9 +237,9 @@ if [ $SRS_EXPORT_LIBRTMP_PROJECT = NO ]; then
     done
     # 
     # all depends libraries
-    ModuleLibFiles=(${LibSTfile} ${LibHttpParserfile} ${LibSSLfile} ${LibGperfFile})
+    ModuleLibFiles=(${LibSTfile} ${LibHttpParserfile} ${LibSSLfile} ${LibGperfFile} ${LibSpeex2aacfile})
     # all depends objects
-    MODULE_OBJS="${CORE_OBJS[@]} ${KERNEL_OBJS[@]} ${PROTOCOL_OBJS[@]} ${APP_OBJS[@]} ${MAIN_OBJS[@]}"
+    MODULE_OBJS="${CORE_OBJS[@]} ${KERNEL_OBJS[@]} ${PROTOCOL_OBJS[@]} ${APP_OBJS[@]} ${MAIN_OBJS[@]} ${SPEEX2AAC_OBJS[@]}"
     LINK_OPTIONS="${SrsLinkOptions}${SrsGprofLink}${SrsGperfLink}"
     #
     # srs: srs(simple rtmp server) over st(state-threads)
@@ -258,10 +266,10 @@ if [ $SRS_UTEST = YES ]; then
     MODULE_FILES=("srs_utest" "srs_utest_amf0" "srs_utest_protocol" 
             "srs_utest_kernel" "srs_utest_core" "srs_utest_config" 
             "srs_utest_reload")
-    ModuleLibIncs=(${SRS_OBJS_DIR} ${LibSTRoot} ${LibSSLRoot})
-    ModuleLibFiles=(${LibSTfile} ${LibHttpParserfile} ${LibSSLfile})
+    ModuleLibIncs=(${SRS_OBJS_DIR} ${LibSTRoot} ${LibSSLRoot} ${LibSpeex2aacRoot})
+    ModuleLibFiles=(${LibSTfile} ${LibHttpParserfile} ${LibSSLfile} ${LibSpeex2aacfile})
     MODULE_DEPENDS=("CORE" "KERNEL" "PROTOCOL" "APP")
-    MODULE_OBJS="${CORE_OBJS[@]} ${KERNEL_OBJS[@]} ${PROTOCOL_OBJS[@]} ${APP_OBJS[@]}"
+    MODULE_OBJS="${CORE_OBJS[@]} ${KERNEL_OBJS[@]} ${PROTOCOL_OBJS[@]} ${APP_OBJS[@]} ${SPEEX2AAC_OBJS[@]}"
     LINK_OPTIONS="-lpthread ${SrsLinkOptions}" MODULE_DIR="src/utest" APP_NAME="srs_utest" . auto/utest.sh
 fi
 
@@ -349,6 +357,7 @@ else
 server: _prepare_dir
 	@echo "build the srs(simple rtmp server) over st(state-threads)"
 	./scripts/version.sh
+	./scripts/build_libs.sh
 	\$(MAKE) -f ${SRS_OBJS_DIR}/${SRS_MAKEFILE} srs
 srs_ingest_hls: _prepare_dir
 	@echo "build the srs_ingest_hls for srs"

+#!/bin/bash
+mkdir -p objs
+root=$PWD
+if [ ! -d "$root/3rdparty/speex/.git" ]; then
+  cd ..
+  git submodule update --init
+  cd trunk
+fi
+if [ "$1" == "-f" ]; then
+  rm -rf objs/libs
+fi 
+if [ ! -d "$root/objs/libs/include/speex" ]; then
+  cd 3rdparty/speex && ./autogen.sh && ./configure --prefix=$root/objs/libs --enable-shared=no && make && make install
+  cd $root
+fi
+if [ ! -d "$root/objs/libs/include/fdk-aac" ]; then
+  cd 3rdparty/fdk-aac && ./autogen.sh && ./configure --prefix=$root/objs/libs --enable-shared=no && make && make install
+fi
+

@@ -463,6 +463,8 @@ SrsConfig::SrsConfig()
     root = new SrsConfDirective();
     root->conf_line = 0;
     root->name = "root";
+
+    _speex2aac = false;
 }
 
 SrsConfig::~SrsConfig()
@@ -1005,6 +1007,8 @@ int SrsConfig::reload_conf(SrsConfig* conf)
         return ret;
     }
     
+    set_config_static();
+    
     return ret;
 }
 
@@ -1579,7 +1583,7 @@ int SrsConfig::check_config()
             && n != "max_connections" && n != "daemon" && n != "heartbeat"
             && n != "http_api" && n != "stats" && n != "vhost" && n != "pithy_print_ms"
             && n != "http_stream" && n != "http_server" && n != "stream_caster"
-            && n != "utc_time" && n != "work_dir" && n != "asprocess"
+            && n != "utc_time" && n != "work_dir" && n != "asprocess" && n != "speex2aac"
         ) {
             ret = ERROR_SYSTEM_CONFIG_INVALID;
             srs_error("unsupported directive %s, ret=%d", n.c_str(), ret);
@@ -2080,9 +2084,27 @@ int SrsConfig::check_config()
         return ret;
     }
     
+    
+    set_config_static();
+    
     return ret;
 }
 
+void SrsConfig::set_config_static()
+{
+	 SrsConfDirective* conf = root->get("speex2aac");
+	 if(conf){
+	 	if(conf->arg0() == "on"){
+	 		_speex2aac = true;
+	 	}
+	 	else
+	 	    _speex2aac = false;
+	 }
+	 else
+	     _speex2aac = false;
+
+}
+
 int SrsConfig::parse_buffer(SrsConfigBuffer* buffer)
 {
     int ret = ERROR_SUCCESS;

@@ -1141,6 +1141,18 @@ public:
     * @return the disk device name to stat. NULL if not configed.
     */
     virtual SrsConfDirective*   get_stats_disk_device();
+
+public:
+
+    /**
+     * the flag to indicate auto transcode with build-in modules
+     */
+    bool                 _speex2aac;
+private:
+    /**
+     * set the global static variables from config file
+     */
+    void 						set_config_static();
 };
 
 namespace _srs_internal

@@ -46,6 +46,7 @@ using namespace std;
 #include <srs_app_statistic.hpp>
 #include <srs_core_autofree.hpp>
 #include <srs_rtmp_utility.hpp>
+#include <srs_tc_av_codec.hpp>
 
 #define CONST_MAX_JITTER_MS         250
 #define CONST_MAX_JITTER_MS_NEG         -250
@@ -970,6 +971,8 @@ SrsSource::SrsSource()
     
     _srs_config->subscribe(this);
     atc = false;
+	_speex_opt = NULL;
+	_aac_opt = NULL;
 }
 
 SrsSource::~SrsSource()
@@ -1013,6 +1016,15 @@ SrsSource::~SrsSource()
 #endif
 
     srs_freep(_req);
+
+	if (_speex_opt){
+		adc_spx.close_codec((tc_audio_opt *)_speex_opt);
+		_speex_opt = NULL;
+	}
+	if (_aac_opt) {
+		aec_aac.close_codec((tc_audio_opt *)_aac_opt);
+		_aac_opt = NULL;
+	}
 }
 
 void SrsSource::dispose()
@@ -1662,6 +1674,101 @@ int SrsSource::on_data(SrsCommonMessage* shared_data)
 
 int SrsSource::on_audio(SrsCommonMessage* shared_audio)
 {
+	 int ret = ERROR_SUCCESS;
+
+	 if(_srs_config->_speex2aac && SrsFlvCodec::audio_is_speex(shared_audio->payload, shared_audio->size)){
+
+			if (NULL == _speex_opt){
+				tc_audio_opt * speex_opt = new tc_audio_opt();
+				_speex_opt = speex_opt;
+				speex_opt->sample_rate = 16000;
+				speex_opt->channels = 1;
+				speex_opt->frame_size = 320;
+				speex_opt->buffer_size = 640;
+				adc_spx.open_codec(speex_opt);
+				if (NULL == _aac_opt){
+					tc_audio_opt * aac_opt = new tc_audio_opt();
+					_aac_opt = aac_opt;
+					aac_opt->sample_rate = 16000;
+					aac_opt->channels = 1;
+					aac_opt->frame_size = 1024;
+					aac_opt->buffer_size = 2048;
+					aac_opt->audio_bitrate = 64000;
+
+					aec_aac.open_codec(aac_opt);
+					char * buf = new char[aac_opt->extradata_size + 2];
+					buf[0] = 0xAF;
+					buf[1] = 0x00;
+					memcpy(buf + 2, aac_opt->extradata, aac_opt->extradata_size);
+					SrsCommonMessage msg;
+					msg.header = shared_audio->header;
+					msg.payload = buf;
+					msg.size = aac_opt->extradata_size + 2;
+					ret = _on_audio(&msg);
+					freep(msg.payload);
+					if(ret != ERROR_SUCCESS){
+						freep(shared_audio->payload);
+						return ret;
+					}
+				}
+			}
+			uint8_t* packet_data = (uint8_t *)shared_audio->payload + 1;
+			int      packet_size = shared_audio->size - 1;
+
+			char * payload = shared_audio->payload;
+
+			do {
+				uint8_t  out_data[1024];
+				uint32_t  out_size;
+
+				int ret_spx = adc_spx.decode_frame((tc_audio_opt*)_speex_opt, packet_data, packet_size, out_data);
+				if (ret_spx >= 0) {
+#if 0
+					static FILE * fp = NULL;
+					if (!fp){
+						fp = fopen("testout.pcm", "wb");
+					}
+
+					if (fp){
+						fwrite(out_data, 1, 640, fp);
+					}
+#endif
+					((tc_audio_opt *)_aac_opt)->buffer_size = ((tc_audio_opt*)_speex_opt)->buffer_size;
+
+					shared_audio->payload = new char[1024];
+					out_size = 1022;
+					int ret_aac = aec_aac.encode_frame((tc_audio_opt *)_aac_opt, out_data, (uint8_t *)shared_audio->payload + 2 , &out_size);
+					if (ret_aac >= 0){
+						if (out_size){
+							shared_audio->payload[0] = 0xAF;
+							shared_audio->payload[1] = 0x01;
+							shared_audio->size = out_size +2;
+							ret = _on_audio(shared_audio);
+#if 0
+							static int max_out_size = 0;
+							if(out_size > max_out_size){
+								max_out_size = out_size;
+								srs_trace("max_speex_aac_outsize:%d",max_out_size);
+							}
+#endif
+						}
+					}
+				}
+				packet_data += ret_spx;
+				packet_size -= ret_spx;
+
+		} while (packet_size > 0);
+
+		freep(payload);
+	 }
+	 else {
+		 return _on_audio(shared_audio);
+	 }
+	 return ret;
+}
+
+int SrsSource::_on_audio(SrsCommonMessage* shared_audio)
+{
     int ret = ERROR_SUCCESS;
     
     // monotically increase detect.

@@ -511,6 +511,8 @@ private:
     SrsSharedPtrMessage* cache_sh_video;
     // the cached audio sequence header.
     SrsSharedPtrMessage* cache_sh_audio;
+	void *  _speex_opt;
+	void *  _aac_opt;
 public:
     SrsSource();
     virtual ~SrsSource();
@@ -559,6 +561,7 @@ public:
 public:
     virtual int on_audio(SrsCommonMessage* audio);
 private:
+    virtual int _on_audio(SrsCommonMessage* audio);
     virtual int on_audio_imp(SrsSharedPtrMessage* audio);
 public:
     virtual int on_video(SrsCommonMessage* video);

@@ -266,6 +266,18 @@ bool SrsFlvCodec::audio_is_aac(char* data, int size)
     return sound_format == SrsCodecAudioAAC;
 }
 
+bool SrsFlvCodec::audio_is_speex(char* data, int size)
+{
+    // 1bytes required.
+    if (size < 1) {
+        return false;
+    }
+
+    char sound_format = (*data >> 4) & 0x0F;
+
+    return sound_format == SrsCodecAudioSpeex;
+}
+
 bool SrsFlvCodec::video_is_acceptable(char* data, int size)
 {
     // 1bytes required.

@@ -223,6 +223,10 @@ public:
     */
     static bool audio_is_aac(char* data, int size);
     /**
+    * check codec speex.
+    */
+    static bool audio_is_speex(char* data, int size);
+    /**
      * check the video RTMP/flv header info,
      * @return true if video RTMP/flv header is ok.
      * @remark all type of audio is possible, no need to check audio.

+#include "speex/speex.h"
+#include "srs_tc_av_codec.hpp"
+#include "srs_tc_common.hpp"
+
+
+typedef struct {
+    SpeexBits bits;
+    void *dec_state;
+    int dec_frame_size;
+    int dec_buffer_size;
+    spx_int16_t* output_buffer;
+} handle_adc_spx_t;
+
+
+static int open_codec_spx(tc_audio_opt* opt) {
+    handle_adc_spx_t* s = (handle_adc_spx_t*)calloc(1, sizeof(handle_adc_spx_t));
+    opt->handle = s;
+    speex_bits_init(&s->bits);
+    s->dec_state = speex_decoder_init(&speex_wb_mode);
+    speex_decoder_ctl(s->dec_state, SPEEX_GET_FRAME_SIZE, &s->dec_frame_size);
+    s->dec_buffer_size = s->dec_frame_size * sizeof(spx_int16_t);
+    opt->sample_rate = 16000;
+    opt->channels = 1;
+    opt->buffer_size = s->dec_buffer_size ;
+    opt->frame_size = s->dec_frame_size;
+    return 0;
+}
+
+
+static int decode_frame_spx(tc_audio_opt* opt, uint8_t* inData, uint32_t inDataSize, uint8_t* outData) {
+    handle_adc_spx_t* s = (handle_adc_spx_t*)opt->handle;
+    int consumed = 0;
+    if (speex_bits_remaining(&s->bits) < 5 ||
+        speex_bits_peek_unsigned(&s->bits, 5) == 0xF) {
+        speex_bits_read_from(&s->bits, (char*)inData, inDataSize);
+        consumed = inDataSize;
+    }
+    
+	int err = 0;
+	if ((err = speex_decode_int(s->dec_state, &s->bits, (spx_int16_t *)outData)) == 0) {
+        return consumed;
+    } else {
+		tc_log(LOG_LEVEL_ERROR, "speex_decode_int error: %d", err);
+        return -1;
+    }
+}
+
+static int close_codec_spx(tc_audio_opt* opt) {
+    handle_adc_spx_t* s = (handle_adc_spx_t*)opt->handle;
+    speex_decoder_destroy(s->dec_state);
+    speex_bits_destroy(&s->bits);
+    return 0;
+}
+
+
+const tc_av_codec_t adc_spx = {
+    11,
+	"adc_spx",
+    open_codec_spx,
+    decode_frame_spx,
+    NULL,
+    close_codec_spx
+};
\ No newline at end of file

+#include "fdk-aac/aacenc_lib.h"
+#include "srs_tc_av_codec.hpp"
+#include "srs_tc_common.hpp"
+
+static const char *aac_get_error(AACENC_ERROR err)
+{
+    switch (err) {
+        case AACENC_OK:
+            return "No error";
+        case AACENC_INVALID_HANDLE:
+            return "Invalid handle";
+        case AACENC_MEMORY_ERROR:
+            return "Memory allocation error";
+        case AACENC_UNSUPPORTED_PARAMETER:
+            return "Unsupported parameter";
+        case AACENC_INVALID_CONFIG:
+            return "Invalid config";
+        case AACENC_INIT_ERROR:
+            return "Initialization error";
+        case AACENC_INIT_AAC_ERROR:
+            return "AAC library initialization error";
+        case AACENC_INIT_SBR_ERROR:
+            return "SBR library initialization error";
+        case AACENC_INIT_TP_ERROR:
+            return "Transport library initialization error";
+        case AACENC_INIT_META_ERROR:
+            return "Metadata library initialization error";
+        case AACENC_ENCODE_ERROR:
+            return "Encoding error";
+        case AACENC_ENCODE_EOF:
+            return "End of file";
+        default:
+            return "Unknown error";
+    }
+}
+
+typedef struct {
+    HANDLE_AACENCODER handle;
+    int afterburner;
+    int eld_sbr;
+    int signaling;
+    int latm;
+    int header_period;
+    int vbr;
+	uint8_t buf[2048];
+	int data_len;
+} handle_aec_aac_t;
+
+
+static int open_codec_aac(tc_audio_opt* opt) {
+    handle_aec_aac_t* s = (handle_aec_aac_t*)calloc(1, sizeof(handle_aec_aac_t));
+    opt->handle = s;
+    AACENC_ERROR err = AACENC_OK;
+    AACENC_InfoStruct info = { 0 };
+    CHANNEL_MODE mode;
+    int sce = 0, cpe = 0;
+    do{
+        if ((err = aacEncOpen(&s->handle, 0, opt->channels)) != AACENC_OK) {
+            tc_log(LOG_LEVEL_ERROR, "[aac] Unable to open the encoder.\n");
+            break;
+        }
+        
+        int aot;
+        if(opt->audio_profile == 0) {
+            aot = 2; //LC-AAC
+        }else {
+            aot = 5; //HE-AAC
+        }
+        
+        //set audio object type
+        if ((err = aacEncoder_SetParam(s->handle, AACENC_AOT, aot)) != AACENC_OK) {
+            tc_log(LOG_LEVEL_ERROR, "[aac] Unable to set the AOT %d: %s\n",aot, aac_get_error(err));
+            break;
+        }
+        
+        //set sample rate
+        if ((err = aacEncoder_SetParam(s->handle, AACENC_SAMPLERATE, opt->sample_rate)) != AACENC_OK) {
+            tc_log(LOG_LEVEL_ERROR, "[aac] Unable to set the sample rate %d: %s\n",opt->sample_rate, aac_get_error(err));
+            break;
+        }
+        
+        switch (opt->channels) {
+            case 1: mode = MODE_1;       sce = 1; cpe = 0; break;
+            case 2: mode = MODE_2;       sce = 0; cpe = 1; break;
+            default:mode = MODE_1;       sce = 1; cpe = 0; break;
+        }
+        
+        if ((err = aacEncoder_SetParam(s->handle, AACENC_CHANNELMODE, mode)) != AACENC_OK) {
+            tc_log(LOG_LEVEL_ERROR, "[aac] Unable to set channel mode %d: %s\n", mode, aac_get_error(err));
+            break;
+        }
+        
+        if ((err = aacEncoder_SetParam(s->handle, AACENC_CHANNELORDER, 1)) != AACENC_OK) {
+            tc_log(LOG_LEVEL_ERROR, "[aac] Unable to set channel order %d: %s\n", mode, aac_get_error(err));
+            break;
+        }
+        
+        if ((err = aacEncoder_SetParam(s->handle, AACENC_BITRATE, opt->audio_bitrate)) != AACENC_OK) {
+            tc_log(LOG_LEVEL_ERROR, "[aac] Unable to set the bitrate %d: %s\n", opt->audio_bitrate, aac_get_error(err));
+            break;
+        }
+        if (aacEncoder_SetParam(s->handle, AACENC_TRANSMUX, 0) != AACENC_OK) {
+            tc_log(LOG_LEVEL_ERROR,  "Unable to set the ADTS transmux\n");
+            break;
+        }
+        if ((err = aacEncoder_SetParam(s->handle, AACENC_SIGNALING_MODE, 2)) != AACENC_OK) {
+            tc_log(LOG_LEVEL_ERROR,  "Unable to set signaling mode %d: %s\n",2, aac_get_error(err));
+            break;
+        }
+        
+        if (aacEncEncode(s->handle, NULL, NULL, NULL, NULL) != AACENC_OK) {
+            tc_log(LOG_LEVEL_ERROR,  "Unable to initialize the encoder\n");
+            return 1;
+        }
+        if (aacEncInfo(s->handle, &info) != AACENC_OK) {
+            tc_log(LOG_LEVEL_ERROR,  "Unable to get the encoder info\n");
+            return 1;
+        }
+        opt->extradata = (uint8_t *)malloc(info.confSize);
+        opt->extradata_size = info.confSize;
+        memcpy(opt->extradata, info.confBuf, info.confSize);
+    
+    }while (0);
+    
+    if( err != AACENC_OK) {
+        aacEncClose(&s->handle);
+    }
+    
+    return 0;
+}
+
+
+static int encode_frame_aac(tc_audio_opt* opt, uint8_t* inData, uint8_t* inOutData, uint32_t* inOutDataSize) {
+    handle_aec_aac_t* s = (handle_aec_aac_t*)opt->handle;
+    
+    AACENC_BufDesc in_buf = { 0 }, out_buf = { 0 };
+    AACENC_InArgs in_args = { 0 };
+    AACENC_OutArgs out_args = { 0 };
+    int in_identifier = IN_AUDIO_DATA;
+    int in_size, in_elem_size;
+    int out_identifier = OUT_BITSTREAM_DATA;
+    int out_size, out_elem_size;
+    void *in_ptr, *out_ptr;
+    AACENC_ERROR err;
+
+	if (s->data_len + opt->buffer_size < 2048){
+		memcpy(s->buf + s->data_len, inData, opt->buffer_size);
+		s->data_len += opt->buffer_size;
+		*inOutDataSize = 0;
+		return 0;
+	}
+	else{
+		memcpy(s->buf + s->data_len, inData, 2048 - s->data_len);
+
+		in_ptr = s->buf;
+		in_size = 2048;
+		in_elem_size = 2;
+
+		in_args.numInSamples = opt->frame_size;
+		in_buf.numBufs = 1;
+		in_buf.bufs = &in_ptr;
+		in_buf.bufferIdentifiers = &in_identifier;
+		in_buf.bufSizes = &in_size;
+		in_buf.bufElSizes = &in_elem_size;
+
+		out_ptr = inOutData;
+		out_size = *inOutDataSize;
+		out_elem_size = 1;
+		out_buf.numBufs = 1;
+		out_buf.bufs = &out_ptr;
+		out_buf.bufferIdentifiers = &out_identifier;
+		out_buf.bufSizes = &out_size;
+		out_buf.bufElSizes = &out_elem_size;
+
+		if ((err = aacEncEncode(s->handle, &in_buf, &out_buf, &in_args, &out_args)) != AACENC_OK) {
+			tc_log(LOG_LEVEL_ERROR, "[aac] Unable to encode frame: %s\n", aac_get_error(err));
+			return -1;
+		}
+
+		*inOutDataSize = out_args.numOutBytes;
+		memcpy(s->buf, inData + 2048 - s->data_len, opt->buffer_size -( 2048 - s->data_len));
+		s->data_len = opt->buffer_size - (2048 - s->data_len);
+	}
+
+    return 0;
+}
+
+static int close_codec_aac(tc_audio_opt* opt) {
+    handle_aec_aac_t* s = (handle_aec_aac_t*)opt->handle;
+    if (s->handle) {
+        aacEncClose(&s->handle);
+    }
+    freep(opt->extradata);
+    freep(opt->handle);
+    return 0;
+}
+
+
+const tc_av_codec_t aec_aac = {
+    10,
+	"aec_aac",
+    open_codec_aac,
+    NULL,
+    encode_frame_aac,
+    close_codec_aac
+};

+#ifndef TC_AV_CODEC_H
+#define TC_AV_CODEC_H
+
+#include <stdint.h>
+#include <stdio.h>
+#include <stdlib.h>
+#include <string.h>
+
+#define MAX_EXTRA 3
+
+typedef void* handle_t;
+
+typedef struct {
+    int sample_rate;
+    int channels;
+    int frame_size;
+    int buffer_size;
+    int audio_profile;
+    int audio_bitrate;
+
+    handle_t handle;
+    
+    uint32_t extradata_size;
+    uint8_t* extradata;
+    
+}tc_audio_opt;
+
+typedef struct {
+    int codec_id;
+    
+    const char* codec_name;
+    
+    int (*open_codec)(tc_audio_opt* opt);
+    
+    int (*decode_frame)(tc_audio_opt* opt, uint8_t* inData, uint32_t inDataSize, uint8_t* outData);
+    
+    int (*encode_frame)(tc_audio_opt* opt, uint8_t* inData, uint8_t* inoutData, uint32_t* inoutDataSize);
+    
+    int (*close_codec)(tc_audio_opt* opt);
+    
+}tc_av_codec_t;
+
+
+extern const tc_av_codec_t adc_aac;
+extern const tc_av_codec_t aec_aac;
+
+extern const tc_av_codec_t adc_spx;
+
+#define freep(x) if(x){ free(x); x = NULL;}
+#endif /* tc_av_codec_h */

+#include "srs_tc_common.hpp"
+#include <stdarg.h>
+#include <srs_kernel_log.hpp>
+
+void tc_log(int log_level, const char *format, ...)
+{
+	va_list args;
+	va_start(args, format);
+	
+	if (log_level == LOG_LEVEL_INFO){
+		srs_info(format, args);
+	}
+	else if (log_level == LOG_LEVEL_WARN){
+		srs_warn(format, args);
+	}
+	else if (log_level == LOG_LEVEL_ERROR){
+		srs_error(format, args);
+	}
+
+	va_end(args);
+}

+#ifndef TC_COMMON_H_
+#define TC_COMMON_H_
+
+enum LOG_LEVEL
+{
+	LOG_LEVEL_INFO,
+	LOG_LEVEL_WARN,
+	LOG_LEVEL_ERROR,
+};
+void tc_log(int log_level, const char *format, ...);
+
+#endif
\ No newline at end of file