fix #250, support push MPEGTS over UDP to SRS. 2.0.111

@@ -486,7 +486,7 @@ Supported operating systems and hardware:
 ).
 1. Support HLS(h.264+mp3) streaming, read 
 [#301](https://github.com/winlinvip/simple-rtmp-server/issues/301).
-1. [dev] Support push MPEG-TS over UDP to SRS, read 
+1. Support push MPEG-TS over UDP to SRS, read 
 [#250](https://github.com/winlinvip/simple-rtmp-server/issues/250).
 1. [no-plan] Support <500ms latency, FRSC(Fast RTMP-compatible Stream Channel tech).
 1. [no-plan] Support RTMP 302 redirect [#92](https://github.com/winlinvip/simple-rtmp-server/issues/92).
@@ -525,6 +525,7 @@ Supported operating systems and hardware:
 
 ### SRS 2.0 history
 
+* v2.0, 2015-01-31, for [#250](https://github.com/winlinvip/simple-rtmp-server/issues/250), support push MPEGTS over UDP to SRS. 2.0.111
 * v2.0, 2015-01-29, build libfdk-aac in ffmpeg. 2.0.108
 * v2.0, 2015-01-25, for [#301](https://github.com/winlinvip/simple-rtmp-server/issues/301), hls support h.264+mp3, ok for vlc. 2.0.107
 * v2.0, 2015-01-25, for [#301](https://github.com/winlinvip/simple-rtmp-server/issues/301), http ts stream support h.264+mp3. 2.0.106

@@ -76,11 +76,21 @@ int SrsMpegtsQueue::push(SrsSharedPtrMessage* msg)
 {
     int ret = ERROR_SUCCESS;
 
-    if (msgs.find(msg->timestamp) != msgs.end()) {
+    // TODO: FIXME: use right way.
+    for (int i = 0; i < 10; i++) {
+        if (msgs.find(msg->timestamp) == msgs.end()) {
+            break;
+        }
+
+        // adjust the ts, add 1ms.
+        msg->timestamp += 1;
+
+        if (i >= 5) {
             srs_warn("mpegts: free the msg for dts exists, dts=%"PRId64, msg->timestamp);
             srs_freep(msg);
             return ret;
         }
+    }
 
     if (msg->is_audio()) {
         nb_audios++;
@@ -114,6 +124,8 @@ SrsSharedPtrMessage* SrsMpegtsQueue::dequeue()
         if (msg->is_video()) {
             nb_videos--;
         }
+
+        return msg;
     }
 
     return NULL;
@@ -131,6 +143,7 @@ SrsMpegtsOverUdp::SrsMpegtsOverUdp(SrsConfDirective* c)
     stfd = NULL;
     stream_id = 0;
     avc = new SrsRawH264Stream();
+    aac = new SrsRawAacStream();
     h264_sps_changed = false;
     h264_pps_changed = false;
     h264_sps_pps_sent = false;
@@ -145,6 +158,7 @@ SrsMpegtsOverUdp::~SrsMpegtsOverUdp()
     srs_freep(stream);
     srs_freep(context);
     srs_freep(avc);
+    srs_freep(aac);
     srs_freep(queue);
 }
 
@@ -309,6 +323,9 @@ int SrsMpegtsOverUdp::on_ts_message(SrsTsMessage* msg)
     if (msg->channel->stream == SrsTsStreamVideoH264) {
         return on_ts_video(msg, &avs);
     }
+    if (msg->channel->stream == SrsTsStreamAudioAAC) {
+        return on_ts_audio(msg, &avs);
+    }
 
     // TODO: FIXME: implements it.
     return ret;
@@ -327,6 +344,10 @@ int SrsMpegtsOverUdp::on_ts_video(SrsTsMessage* msg, SrsStream* avs)
     u_int32_t dts = msg->dts / 90; 
     u_int32_t pts = msg->dts / 90;
 
+    // the whole ts pes video packet must be a flv frame packet.
+    char* ibpframe = avs->data() + avs->pos();
+    int ibpframe_size = avs->size() - avs->pos();
+    
     // send each frame.
     while (!avs->empty()) {
         char* frame = NULL;
@@ -342,25 +363,6 @@ int SrsMpegtsOverUdp::on_ts_video(SrsTsMessage* msg, SrsStream* avs)
             continue;
         }
 
-        // it may be return error, but we must process all packets.
-        if ((ret = write_h264_raw_frame(frame, frame_size, dts, pts)) != ERROR_SUCCESS) {
-            if (ret == ERROR_H264_DROP_BEFORE_SPS_PPS) {
-                continue;
-            }
-            return ret;
-        }
-
-        // for video, drop others with same pts/dts.
-        break;
-    }
-
-    return ret;
-}
-
-int SrsMpegtsOverUdp::write_h264_raw_frame(char* frame, int frame_size, u_int32_t dts, u_int32_t pts)
-{
-    int ret = ERROR_SUCCESS;
-
         // for sps
         if (avc->is_sps(frame, frame_size)) {
             std::string sps;
@@ -369,12 +371,15 @@ int SrsMpegtsOverUdp::write_h264_raw_frame(char* frame, int frame_size, u_int32_
             }
         
             if (h264_sps == sps) {
-            return ret;
+                continue;
             }
             h264_sps_changed = true;
             h264_sps = sps;
         
-        return write_h264_sps_pps(dts, pts);
+            if ((ret = write_h264_sps_pps(dts, pts)) != ERROR_SUCCESS) {
+                return ret;
+            }
+            continue;
         }
 
         // for pps
@@ -385,16 +390,23 @@ int SrsMpegtsOverUdp::write_h264_raw_frame(char* frame, int frame_size, u_int32_
             }
         
             if (h264_pps == pps) {
-            return ret;
+                continue;
             }
             h264_pps_changed = true;
             h264_pps = pps;
         
-        return write_h264_sps_pps(dts, pts);
+            if ((ret = write_h264_sps_pps(dts, pts)) != ERROR_SUCCESS) {
+                return ret;
+            }
+            continue;
+        }
+
+        break;
     }
 
     // ibp frame.
-    return write_h264_ipb_frame(frame, frame_size, dts, pts);
+    srs_info("mpegts: demux avc ibp frame size=%d, dts=%d", ibpframe_size, dts);
+    return write_h264_ipb_frame(ibpframe, ibpframe_size, dts, pts);
 }
 
 int SrsMpegtsOverUdp::write_h264_sps_pps(u_int32_t dts, u_int32_t pts)
@@ -421,14 +433,18 @@ int SrsMpegtsOverUdp::write_h264_sps_pps(u_int32_t dts, u_int32_t pts)
         return ret;
     }
     
+    // the timestamp in rtmp message header is dts.
+    u_int32_t timestamp = dts;
+    if ((ret = rtmp_write_packet(SrsCodecFlvTagVideo, timestamp, flv, nb_flv)) != ERROR_SUCCESS) {
+        return ret;
+    }
+    
     // reset sps and pps.
     h264_sps_changed = false;
     h264_pps_changed = false;
     h264_sps_pps_sent = true;
 
-    // the timestamp in rtmp message header is dts.
-    u_int32_t timestamp = dts;
-    return rtmp_write_packet(SrsCodecFlvTagVideo, timestamp, flv, nb_flv);
+    return ret;
 }
 
 int SrsMpegtsOverUdp::write_h264_ipb_frame(char* frame, int frame_size, u_int32_t dts, u_int32_t pts) 
@@ -459,6 +475,72 @@ int SrsMpegtsOverUdp::write_h264_ipb_frame(char* frame, int frame_size, u_int32_
     return rtmp_write_packet(SrsCodecFlvTagVideo, timestamp, flv, nb_flv);
 }
 
+int SrsMpegtsOverUdp::on_ts_audio(SrsTsMessage* msg, SrsStream* avs)
+{
+    int ret = ERROR_SUCCESS;
+
+    // ensure rtmp connected.
+    if ((ret = connect()) != ERROR_SUCCESS) {
+        return ret;
+    }
+
+    // ts tbn to flv tbn.
+    u_int32_t dts = msg->dts / 90;
+    
+    // send each frame.
+    while (!avs->empty()) {
+        char* frame = NULL;
+        int frame_size = 0;
+        SrsRawAacStreamCodec codec;
+        if ((ret = aac->adts_demux(avs, &frame, &frame_size, codec)) != ERROR_SUCCESS) {
+            return ret;
+        }
+    
+        // ignore invalid frame,
+        //  * atleast 1bytes for aac to decode the data.
+        if (frame_size <= 0) {
+            continue;
+        }
+        srs_info("mpegts: demux aac frame size=%d, dts=%d", frame_size, dts);
+
+        // generate sh.
+        if (aac_specific_config.empty()) {
+            std::string sh;
+            if ((ret = aac->mux_sequence_header(&codec, sh)) != ERROR_SUCCESS) {
+                return ret;
+            }
+            aac_specific_config = sh;
+
+            codec.aac_packet_type = 0;
+
+            if ((ret = write_audio_raw_frame((char*)sh.data(), (int)sh.length(), &codec, dts)) != ERROR_SUCCESS) {
+                return ret;
+            }
+        }
+
+        // audio raw data.
+        codec.aac_packet_type = 1;
+        if ((ret = write_audio_raw_frame(frame, frame_size, &codec, dts)) != ERROR_SUCCESS) {
+            return ret;
+        }
+    }
+
+    return ret;
+}
+
+int SrsMpegtsOverUdp::write_audio_raw_frame(char* frame, int frame_size, SrsRawAacStreamCodec* codec, u_int32_t dts)
+{
+    int ret = ERROR_SUCCESS;
+
+    char* data = NULL;
+    int size = 0;
+    if ((ret = aac->mux_aac2flv(frame, frame_size, codec, dts, &data, &size)) != ERROR_SUCCESS) {
+        return ret;
+    }
+    
+    return rtmp_write_packet(SrsCodecFlvTagAudio, dts, data, size);
+}
+
 int SrsMpegtsOverUdp::rtmp_write_packet(char type, u_int32_t timestamp, char* data, int size)
 {
     int ret = ERROR_SUCCESS;
@@ -483,6 +565,10 @@ int SrsMpegtsOverUdp::rtmp_write_packet(char type, u_int32_t timestamp, char* da
             break;
         }
 
+        // TODO: FIXME: use pithy print.
+        srs_info("mpegts: send msg %s dts=%"PRId64", size=%d",
+            msg->is_audio()? "A":msg->is_video()? "V":"N", msg->timestamp, msg->size);
+    
         // send out encoded msg.
         if ((ret = client->send_and_free_message(msg, stream_id)) != ERROR_SUCCESS) {
             return ret;

@@ -45,6 +45,8 @@ class SrsStSocket;
 class SrsRequest;
 class SrsRawH264Stream;
 class SrsSharedPtrMessage;
+class SrsRawAacStream;
+class SrsRawAacStreamCodec;
 
 #include <srs_app_st.hpp>
 #include <srs_kernel_ts.hpp>
@@ -114,6 +116,10 @@ private:
     std::string h264_pps;
     bool h264_pps_changed;
     bool h264_sps_pps_sent;
+private:
+    SrsRawAacStream* aac;
+    std::string aac_specific_config;
+private:
     SrsMpegtsQueue* queue;
 public:
     SrsMpegtsOverUdp(SrsConfDirective* c);
@@ -126,9 +132,11 @@ public:
     virtual int on_ts_message(SrsTsMessage* msg);
 private:
     virtual int on_ts_video(SrsTsMessage* msg, SrsStream* avs);
-    virtual int write_h264_raw_frame(char* frame, int frame_size, u_int32_t dts, u_int32_t pts);
     virtual int write_h264_sps_pps(u_int32_t dts, u_int32_t pts);
     virtual int write_h264_ipb_frame(char* frame, int frame_size, u_int32_t dts, u_int32_t pts);
+    virtual int on_ts_audio(SrsTsMessage* msg, SrsStream* avs);
+    virtual int write_audio_raw_frame(char* frame, int frame_size, SrsRawAacStreamCodec* codec, u_int32_t dts);
+private:
     virtual int rtmp_write_packet(char type, u_int32_t timestamp, char* data, int size);
 private:
     // connect to rtmp output url. 

@@ -31,7 +31,7 @@ CONNECTION WITH THE SOFTWARE OR THE USE OR OTHER DEALINGS IN THE SOFTWARE.
 // current release version
 #define VERSION_MAJOR       2
 #define VERSION_MINOR       0
-#define VERSION_REVISION    110
+#define VERSION_REVISION    111
 
 // server info.
 #define RTMP_SIG_SRS_KEY "SRS"

@@ -81,9 +81,10 @@ struct Context
     SimpleSocketStream* skt;
     int stream_id;
     
-    // for h264 raw stream, 
-    // @see: https://github.com/winlinvip/simple-rtmp-server/issues/66#issuecomment-62240521
+    // the remux raw codec.
     SrsRawH264Stream avc_raw;
+    SrsRawAacStream aac_raw;
+
     // for h264 raw stream, 
     // @see: https://github.com/winlinvip/simple-rtmp-server/issues/66#issuecomment-62240521
     SrsStream h264_raw_stream;
@@ -1073,33 +1074,16 @@ srs_bool srs_rtmp_is_onMetaData(char type, char* data, int size)
 * directly write a audio frame.
 */
 int __srs_write_audio_raw_frame(Context* context,
-    char sound_format, char sound_rate, char sound_size, char sound_type,
-    char aac_packet_type, char* frame, int frame_size, u_int32_t timestamp
+    char* frame, int frame_size, SrsRawAacStreamCodec* codec, u_int32_t timestamp
 ) {
+    int ret = ERROR_SUCCESS;
 
-    // for audio frame, there is 1 or 2 bytes header:
-    //      1bytes, SoundFormat|SoundRate|SoundSize|SoundType
-    //      1bytes, AACPacketType for SoundFormat == 10, 0 is sequence header.
-    int size = frame_size + 1;
-    if (sound_format == SrsCodecAudioAAC) {
-        size += 1;
-    }
-    char* data = new char[size];
-    char* p = data;
-    
-    u_int8_t audio_header = sound_type & 0x01;
-    audio_header |= (sound_size << 1) & 0x02;
-    audio_header |= (sound_rate << 2) & 0x0c;
-    audio_header |= (sound_format << 4) & 0xf0;
-    
-    *p++ = audio_header;
-    
-    if (sound_format == SrsCodecAudioAAC) {
-        *p++ = aac_packet_type;
+    char* data = NULL;
+    int size = 0;
+    if ((ret = context->aac_raw.mux_aac2flv(frame, frame_size, codec, timestamp, &data, &size)) != ERROR_SUCCESS) {
+        return ret;
     }
     
-    memcpy(p, frame, frame_size);
-    
     return srs_rtmp_write_packet(context, SRS_RTMP_TYPE_AUDIO, timestamp, data, size);
 }
 
@@ -1107,72 +1091,27 @@ int __srs_write_audio_raw_frame(Context* context,
 * write aac frame in adts.
 */
 int __srs_write_aac_adts_frame(Context* context, 
-    char sound_format, char sound_rate, char sound_size, char sound_type,
-    char aac_profile, char aac_samplerate, char aac_channel,
-    char* frame, int frame_size, u_int32_t timestamp
+    SrsRawAacStreamCodec* codec, char* frame, int frame_size, u_int32_t timestamp
 ) {
     int ret = ERROR_SUCCESS;
     
-    // override the aac samplerate by user specified.
-    // @see https://github.com/winlinvip/simple-rtmp-server/issues/212#issuecomment-64146899
-    switch (sound_rate) {
-        case SrsCodecAudioSampleRate11025: 
-            aac_samplerate = 0x0a; break;
-        case SrsCodecAudioSampleRate22050: 
-            aac_samplerate = 0x07; break;
-        case SrsCodecAudioSampleRate44100: 
-            aac_samplerate = 0x04; break;
-        default:
-            break;
-    }
-    
     // send out aac sequence header if not sent.
     if (context->aac_specific_config.empty()) {
-        char ch = 0;
-        // @see aac-mp4a-format-ISO_IEC_14496-3+2001.pdf
-        // AudioSpecificConfig (), page 33
-        // 1.6.2.1 AudioSpecificConfig
-        // audioObjectType; 5 bslbf
-        ch = (aac_profile << 3) & 0xf8;
-        // 3bits left.
-        
-        // samplingFrequencyIndex; 4 bslbf
-        ch |= (aac_samplerate >> 1) & 0x07;
-        context->aac_specific_config += ch;
-        ch = (aac_samplerate << 7) & 0x80;
-        if (aac_samplerate == 0x0f) {
-            return ERROR_AAC_DATA_INVALID;
-        }
-        // 7bits left.
-        
-        // channelConfiguration; 4 bslbf
-        ch |= (aac_channel << 3) & 0x78;
-        // 3bits left.
-        
-        // only support aac profile 1-4.
-        if (aac_profile < 1 || aac_profile > 4) {
-            return ERROR_AAC_DATA_INVALID;
-        }
-        // GASpecificConfig(), page 451
-        // 4.4.1 Decoder configuration (GASpecificConfig)
-        // frameLengthFlag; 1 bslbf
-        // dependsOnCoreCoder; 1 bslbf
-        // extensionFlag; 1 bslbf
-        context->aac_specific_config += ch;
-        
-        char* sh = (char*)context->aac_specific_config.data();
-        int nb_sh = (int)context->aac_specific_config.length();
-        if ((ret = __srs_write_audio_raw_frame(context, 
-            sound_format, sound_rate, sound_size, sound_type, 
-            0, sh, nb_sh, timestamp)) != ERROR_SUCCESS
-        ) {
+        std::string sh;
+        if ((ret = context->aac_raw.mux_sequence_header(codec, sh)) != ERROR_SUCCESS) {
+            return ret;
+        }
+        context->aac_specific_config = sh;
+
+        codec->aac_packet_type = 0;
+
+        if ((ret = __srs_write_audio_raw_frame(context, (char*)sh.data(), (int)sh.length(), codec, timestamp)) != ERROR_SUCCESS) {
             return ret;
         }
     }
     
-    return __srs_write_audio_raw_frame(context, 
-        sound_format, sound_rate, sound_size, sound_type, 
-        1, frame, frame_size, timestamp);
+    codec->aac_packet_type = 1;
+    return __srs_write_audio_raw_frame(context, frame, frame_size, codec, timestamp);
 }
 
 /**
@@ -1180,126 +1119,32 @@ int __srs_write_aac_adts_frame(Context* context,
 */
 int __srs_write_aac_adts_frames(Context* context,
     char sound_format, char sound_rate, char sound_size, char sound_type,
-    char* frame, int frame_size, u_int32_t timestamp
+    char* frames, int frames_size, u_int32_t timestamp
 ) {
     int ret = ERROR_SUCCESS;
     
     SrsStream* stream = &context->aac_raw_stream;
-    if ((ret = stream->initialize(frame, frame_size)) != ERROR_SUCCESS) {
+    if ((ret = stream->initialize(frames, frames_size)) != ERROR_SUCCESS) {
         return ret;
     }
     
     while (!stream->empty()) {
-        int adts_header_start = stream->pos();
-        
-        // decode the ADTS.
-        // @see aac-mp4a-format-ISO_IEC_14496-3+2001.pdf, page 75,
-        //      1.A.2.2 Audio_Data_Transport_Stream frame, ADTS
-        // @see https://github.com/winlinvip/simple-rtmp-server/issues/212#issuecomment-64145885
-        // byte_alignment()
-        
-        // adts_fixed_header:
-        //      12bits syncword,
-        //      16bits left.
-        // adts_variable_header:
-        //      28bits
-        //      12+16+28=56bits
-        // adts_error_check:
-        //      16bits if protection_absent
-        //      56+16=72bits
-        // if protection_absent:
-        //      require(7bytes)=56bits
-        // else
-        //      require(9bytes)=72bits
-        if (!stream->require(7)) {
-            return ERROR_AAC_ADTS_HEADER;
+        char* frame = NULL;
+        int frame_size = 0;
+        SrsRawAacStreamCodec codec;
+        if ((ret = context->aac_raw.adts_demux(stream, &frame, &frame_size, codec)) != ERROR_SUCCESS) {
+            return ret;
         }
 
-        // for aac, the frame must be ADTS format.
-        if (!srs_aac_startswith_adts(stream)) {
-            return ERROR_AAC_REQUIRED_ADTS;
-        }
+        // override by user specified.
+        codec.sound_format = sound_format;
+        codec.sound_rate = sound_rate;
+        codec.sound_size = sound_size;
+        codec.sound_type = sound_type;
 
-        // Syncword 12 bslbf
-        stream->read_1bytes();
-        // 4bits left.
-        // adts_fixed_header(), 1.A.2.2.1 Fixed Header of ADTS
-        // ID 1 bslbf
-        // Layer 2 uimsbf
-        // protection_absent 1 bslbf
-        int8_t fh0 = (stream->read_1bytes() & 0x0f);
-        /*int8_t fh_id = (fh0 >> 3) & 0x01;*/
-        /*int8_t fh_layer = (fh0 >> 1) & 0x03;*/
-        int8_t fh_protection_absent = fh0 & 0x01;
-        
-        int16_t fh1 = stream->read_2bytes();
-        // Profile_ObjectType 2 uimsbf
-        // sampling_frequency_index 4 uimsbf
-        // private_bit 1 bslbf
-        // channel_configuration 3 uimsbf
-        // original/copy 1 bslbf
-        // home 1 bslbf
-        int8_t fh_Profile_ObjectType = (fh1 >> 14) & 0x03;
-        int8_t fh_sampling_frequency_index = (fh1 >> 10) & 0x0f;
-        /*int8_t fh_private_bit = (fh1 >> 9) & 0x01;*/
-        int8_t fh_channel_configuration = (fh1 >> 6) & 0x07;
-        /*int8_t fh_original = (fh1 >> 5) & 0x01;*/
-        /*int8_t fh_home = (fh1 >> 4) & 0x01;*/
-        // @remark, Emphasis is removed, 
-        //      @see https://github.com/winlinvip/simple-rtmp-server/issues/212#issuecomment-64154736
-        //int8_t fh_Emphasis = (fh1 >> 2) & 0x03;
-        // 4bits left.
-        // adts_variable_header(), 1.A.2.2.2 Variable Header of ADTS
-        // copyright_identification_bit 1 bslbf
-        // copyright_identification_start 1 bslbf
-        /*int8_t fh_copyright_identification_bit = (fh1 >> 3) & 0x01;*/
-        /*int8_t fh_copyright_identification_start = (fh1 >> 2) & 0x01;*/
-        // aac_frame_length 13 bslbf: Length of the frame including headers and error_check in bytes.
-        // use the left 2bits as the 13 and 12 bit,
-        // the aac_frame_length is 13bits, so we move 13-2=11.
-        int16_t fh_aac_frame_length = (fh1 << 11) & 0x0800;
-        
-        int32_t fh2 = stream->read_3bytes();
-        // aac_frame_length 13 bslbf: consume the first 13-2=11bits
-        // the fh2 is 24bits, so we move right 24-11=13.
-        fh_aac_frame_length |= (fh2 >> 13) & 0x07ff;
-        // adts_buffer_fullness 11 bslbf
-        /*int16_t fh_adts_buffer_fullness = (fh2 >> 2) & 0x7ff;*/
-        // no_raw_data_blocks_in_frame 2 uimsbf
-        /*int16_t fh_no_raw_data_blocks_in_frame = fh2 & 0x03;*/
-        // adts_error_check(), 1.A.2.2.3 Error detection
-        if (!fh_protection_absent) {
-            if (!stream->require(2)) {
-                return ERROR_AAC_ADTS_HEADER;
-            }
-            // crc_check 16 Rpchof
-            /*int16_t crc_check = */stream->read_2bytes();
-        }
-        
-        // TODO: check the fh_sampling_frequency_index
-        // TODO: check the fh_channel_configuration
-        
-        // raw_data_blocks
-        int adts_header_size = stream->pos() - adts_header_start;
-        int raw_data_size = fh_aac_frame_length - adts_header_size;
-        if (!stream->require(raw_data_size)) {
-            return ERROR_AAC_ADTS_HEADER;
-        }
-        
-        // the profile = object_id + 1
-        // @see aac-mp4a-format-ISO_IEC_14496-3+2001.pdf, page 78,
-        //      Table 1. A.9 – MPEG-2 Audio profiles and MPEG-4 Audio object types
-        char aac_profile = fh_Profile_ObjectType + 1;
-        
-        char* raw_data = stream->data() + stream->pos();
-        if ((ret = __srs_write_aac_adts_frame(context,
-            sound_format, sound_rate, sound_size, sound_type,
-            aac_profile, fh_sampling_frequency_index, fh_channel_configuration,
-            raw_data, raw_data_size, timestamp)) != ERROR_SUCCESS
-        ) {
+        if ((ret = __srs_write_aac_adts_frame(context, &codec, frame, frame_size, timestamp)) != ERROR_SUCCESS) {
             return ret;
         }
-        stream->skip(raw_data_size);
     }
     
     return ret;
@@ -1328,10 +1173,16 @@ int srs_audio_write_raw_frame(srs_rtmp_t rtmp,
             sound_format, sound_rate, sound_size, sound_type, 
             frame, frame_size, timestamp);
     } else {
+        // use codec info for aac.
+        SrsRawAacStreamCodec codec;
+        codec.sound_format = sound_format;
+        codec.sound_rate = sound_rate;
+        codec.sound_size = sound_size;
+        codec.sound_type = sound_type;
+        codec.aac_packet_type = 0;
+
         // for other data, directly write frame.
-        return __srs_write_audio_raw_frame(context, 
-            sound_format, sound_rate, sound_size, sound_type, 
-            0, frame, frame_size, timestamp);
+        return __srs_write_audio_raw_frame(context, frame, frame_size, &codec, timestamp);
     }
     
     

@@ -27,6 +27,7 @@ CONNECTION WITH THE SOFTWARE OR THE USE OR OTHER DEALINGS IN THE SOFTWARE.
 using namespace std;
 
 #include <srs_kernel_error.hpp>
+#include <srs_kernel_log.hpp>
 #include <srs_kernel_stream.hpp>
 #include <srs_kernel_utility.hpp>
 #include <srs_core_autofree.hpp>
@@ -312,3 +313,254 @@ int SrsRawH264Stream::mux_avc2flv(string video, int8_t frame_type, int8_t avc_pa
     return ret;
 }
 
+SrsRawAacStream::SrsRawAacStream()
+{
+}
+
+SrsRawAacStream::~SrsRawAacStream()
+{
+}
+
+int SrsRawAacStream::adts_demux(SrsStream* stream, char** pframe, int* pnb_frame, SrsRawAacStreamCodec& codec)
+{
+    int ret = ERROR_SUCCESS;
+    
+    while (!stream->empty()) {
+        int adts_header_start = stream->pos();
+        
+        // decode the ADTS.
+        // @see aac-mp4a-format-ISO_IEC_14496-3+2001.pdf, page 75,
+        //      1.A.2.2 Audio_Data_Transport_Stream frame, ADTS
+        // @see https://github.com/winlinvip/simple-rtmp-server/issues/212#issuecomment-64145885
+        // byte_alignment()
+        
+        // adts_fixed_header:
+        //      12bits syncword,
+        //      16bits left.
+        // adts_variable_header:
+        //      28bits
+        //      12+16+28=56bits
+        // adts_error_check:
+        //      16bits if protection_absent
+        //      56+16=72bits
+        // if protection_absent:
+        //      require(7bytes)=56bits
+        // else
+        //      require(9bytes)=72bits
+        if (!stream->require(7)) {
+            return ERROR_AAC_ADTS_HEADER;
+        }
+        
+        // for aac, the frame must be ADTS format.
+        if (!srs_aac_startswith_adts(stream)) {
+            return ERROR_AAC_REQUIRED_ADTS;
+        }
+        
+        // Syncword 12 bslbf
+        stream->read_1bytes();
+        // 4bits left.
+        // adts_fixed_header(), 1.A.2.2.1 Fixed Header of ADTS
+        // ID 1 bslbf
+        // Layer 2 uimsbf
+        // protection_absent 1 bslbf
+        int8_t fh0 = (stream->read_1bytes() & 0x0f);
+        /*int8_t fh_id = (fh0 >> 3) & 0x01;*/
+        /*int8_t fh_layer = (fh0 >> 1) & 0x03;*/
+        int8_t fh_protection_absent = fh0 & 0x01;
+        
+        int16_t fh1 = stream->read_2bytes();
+        // Profile_ObjectType 2 uimsbf
+        // sampling_frequency_index 4 uimsbf
+        // private_bit 1 bslbf
+        // channel_configuration 3 uimsbf
+        // original/copy 1 bslbf
+        // home 1 bslbf
+        int8_t fh_Profile_ObjectType = (fh1 >> 14) & 0x03;
+        int8_t fh_sampling_frequency_index = (fh1 >> 10) & 0x0f;
+        /*int8_t fh_private_bit = (fh1 >> 9) & 0x01;*/
+        int8_t fh_channel_configuration = (fh1 >> 6) & 0x07;
+        /*int8_t fh_original = (fh1 >> 5) & 0x01;*/
+        /*int8_t fh_home = (fh1 >> 4) & 0x01;*/
+        // @remark, Emphasis is removed, 
+        //      @see https://github.com/winlinvip/simple-rtmp-server/issues/212#issuecomment-64154736
+        //int8_t fh_Emphasis = (fh1 >> 2) & 0x03;
+        // 4bits left.
+        // adts_variable_header(), 1.A.2.2.2 Variable Header of ADTS
+        // copyright_identification_bit 1 bslbf
+        // copyright_identification_start 1 bslbf
+        /*int8_t fh_copyright_identification_bit = (fh1 >> 3) & 0x01;*/
+        /*int8_t fh_copyright_identification_start = (fh1 >> 2) & 0x01;*/
+        // aac_frame_length 13 bslbf: Length of the frame including headers and error_check in bytes.
+        // use the left 2bits as the 13 and 12 bit,
+        // the aac_frame_length is 13bits, so we move 13-2=11.
+        int16_t fh_aac_frame_length = (fh1 << 11) & 0x0800;
+        
+        int32_t fh2 = stream->read_3bytes();
+        // aac_frame_length 13 bslbf: consume the first 13-2=11bits
+        // the fh2 is 24bits, so we move right 24-11=13.
+        fh_aac_frame_length |= (fh2 >> 13) & 0x07ff;
+        // adts_buffer_fullness 11 bslbf
+        /*int16_t fh_adts_buffer_fullness = (fh2 >> 2) & 0x7ff;*/
+        // no_raw_data_blocks_in_frame 2 uimsbf
+        /*int16_t fh_no_raw_data_blocks_in_frame = fh2 & 0x03;*/
+        // adts_error_check(), 1.A.2.2.3 Error detection
+        if (!fh_protection_absent) {
+            if (!stream->require(2)) {
+                return ERROR_AAC_ADTS_HEADER;
+            }
+            // crc_check 16 Rpchof
+            /*int16_t crc_check = */stream->read_2bytes();
+        }
+        
+        // TODO: check the fh_sampling_frequency_index
+        // TODO: check the fh_channel_configuration
+        
+        // raw_data_blocks
+        int adts_header_size = stream->pos() - adts_header_start;
+        int raw_data_size = fh_aac_frame_length - adts_header_size;
+        if (!stream->require(raw_data_size)) {
+            return ERROR_AAC_ADTS_HEADER;
+        }
+        
+        // the profile = object_id + 1
+        // @see aac-mp4a-format-ISO_IEC_14496-3+2001.pdf, page 78,
+        //      Table 1. A.9 ¨C MPEG-2 Audio profiles and MPEG-4 Audio object types
+        char aac_profile = fh_Profile_ObjectType + 1;
+        
+        // the codec info.
+        codec.protection_absent = fh_protection_absent;
+        codec.Profile_ObjectType = fh_Profile_ObjectType;
+        codec.sampling_frequency_index = fh_sampling_frequency_index;
+        codec.channel_configuration = fh_channel_configuration;
+        codec.aac_frame_length = fh_aac_frame_length;
+
+        codec.aac_profile = aac_profile;
+        codec.aac_samplerate = fh_sampling_frequency_index;
+        codec.aac_channel = fh_channel_configuration;
+        
+        // @see srs_audio_write_raw_frame().
+        codec.sound_format = 10; // AAC
+        if (fh_sampling_frequency_index <= 0x0c && fh_sampling_frequency_index > 0x0a) {
+            codec.sound_rate = SrsCodecAudioSampleRate5512;
+        } else if (fh_sampling_frequency_index <= 0x0a && fh_sampling_frequency_index > 0x07) {
+            codec.sound_rate = SrsCodecAudioSampleRate11025;
+        } else if (fh_sampling_frequency_index <= 0x07 && fh_sampling_frequency_index > 0x04) {
+            codec.sound_rate = SrsCodecAudioSampleRate22050;
+        } else if (fh_sampling_frequency_index <= 0x04) {
+            codec.sound_rate = SrsCodecAudioSampleRate44100;
+        } else {
+            codec.sound_rate = SrsCodecAudioSampleRate44100;
+            srs_warn("adts invalid sample rate for flv, rate=%#x", fh_sampling_frequency_index);
+        }
+        codec.sound_size = srs_max(0, srs_min(1, fh_channel_configuration - 1));
+        // TODO: FIXME: finger it out the sound size by adts.
+        codec.sound_size = 1; // 0(8bits) or 1(16bits).
+
+        // frame data.
+        *pframe = stream->data() + stream->pos();
+        *pnb_frame = raw_data_size;
+        stream->skip(raw_data_size);
+
+        break;
+    }
+
+    return ret;
+}
+
+int SrsRawAacStream::mux_sequence_header(SrsRawAacStreamCodec* codec, string& sh)
+{
+    int ret = ERROR_SUCCESS;
+    
+    char aac_channel = codec->aac_channel;
+    char aac_profile = codec->aac_profile;
+    char aac_samplerate = codec->aac_samplerate;
+
+    // override the aac samplerate by user specified.
+    // @see https://github.com/winlinvip/simple-rtmp-server/issues/212#issuecomment-64146899
+    switch (codec->sound_rate) {
+        case SrsCodecAudioSampleRate11025: 
+            aac_samplerate = 0x0a; break;
+        case SrsCodecAudioSampleRate22050: 
+            aac_samplerate = 0x07; break;
+        case SrsCodecAudioSampleRate44100: 
+            aac_samplerate = 0x04; break;
+        default:
+            break;
+    }
+
+    sh = "";
+
+    char ch = 0;
+    // @see aac-mp4a-format-ISO_IEC_14496-3+2001.pdf
+    // AudioSpecificConfig (), page 33
+    // 1.6.2.1 AudioSpecificConfig
+    // audioObjectType; 5 bslbf
+    ch = (aac_profile << 3) & 0xf8;
+    // 3bits left.
+        
+    // samplingFrequencyIndex; 4 bslbf
+    ch |= (aac_samplerate >> 1) & 0x07;
+    sh += ch;
+    ch = (aac_samplerate << 7) & 0x80;
+    if (aac_samplerate == 0x0f) {
+        return ERROR_AAC_DATA_INVALID;
+    }
+    // 7bits left.
+        
+    // channelConfiguration; 4 bslbf
+    ch |= (aac_channel << 3) & 0x78;
+    // 3bits left.
+        
+    // only support aac profile 1-4.
+    if (aac_profile < 1 || aac_profile > 4) {
+        return ERROR_AAC_DATA_INVALID;
+    }
+    // GASpecificConfig(), page 451
+    // 4.4.1 Decoder configuration (GASpecificConfig)
+    // frameLengthFlag; 1 bslbf
+    // dependsOnCoreCoder; 1 bslbf
+    // extensionFlag; 1 bslbf
+    sh += ch;
+
+    return ret;
+}
+
+int SrsRawAacStream::mux_aac2flv(char* frame, int nb_frame, SrsRawAacStreamCodec* codec, u_int32_t dts, char** flv, int* nb_flv)
+{
+    int ret = ERROR_SUCCESS;
+
+    char sound_format = codec->sound_format;
+    char sound_type = codec->sound_type;
+    char sound_size = codec->sound_size;
+    char sound_rate = codec->sound_rate;
+    char aac_packet_type = codec->aac_packet_type;
+
+    // for audio frame, there is 1 or 2 bytes header:
+    //      1bytes, SoundFormat|SoundRate|SoundSize|SoundType
+    //      1bytes, AACPacketType for SoundFormat == 10, 0 is sequence header.
+    int size = nb_frame + 1;
+    if (sound_format == SrsCodecAudioAAC) {
+        size += 1;
+    }
+    char* data = new char[size];
+    char* p = data;
+    
+    u_int8_t audio_header = sound_type & 0x01;
+    audio_header |= (sound_size << 1) & 0x02;
+    audio_header |= (sound_rate << 2) & 0x0c;
+    audio_header |= (sound_format << 4) & 0xf0;
+    
+    *p++ = audio_header;
+    
+    if (sound_format == SrsCodecAudioAAC) {
+        *p++ = aac_packet_type;
+    }
+    
+    memcpy(p, frame, nb_frame);
+
+    *flv = data;
+    *nb_flv = size;
+
+    return ret;
+}
+

@@ -86,4 +86,62 @@ public:
     virtual int mux_avc2flv(std::string video, int8_t frame_type, int8_t avc_packet_type, u_int32_t dts, u_int32_t pts, char** flv, int* nb_flv);
 };
 
+/**
+* the header of adts sample.
+*/
+struct SrsRawAacStreamCodec
+{
+    int8_t protection_absent;
+    int8_t Profile_ObjectType;
+    int8_t sampling_frequency_index;
+    int8_t channel_configuration;
+    int16_t aac_frame_length;
+
+    // calc by Profile_ObjectType+1
+    char aac_profile;
+    char aac_samplerate;
+    char aac_channel;
+
+    char sound_format;
+    char sound_rate;
+    char sound_size;
+    char sound_type;
+    // 0 for sh; 1 for raw data.
+    int8_t aac_packet_type;
+};
+
+/**
+* the raw aac stream, in adts.
+*/
+class SrsRawAacStream
+{
+public:
+    SrsRawAacStream();
+    virtual ~SrsRawAacStream();
+public:
+    /**
+    * demux the stream in adts format.
+    * @param stream the input stream bytes.
+    * @param pframe the output aac frame in stream. user should never free it.
+    * @param pnb_frame the output aac frame size.
+    * @param codec the output codec info.
+    */
+    virtual int adts_demux(SrsStream* stream, char** pframe, int* pnb_frame, SrsRawAacStreamCodec& codec);
+    /**
+    * aac raw data to aac packet, without flv payload header.
+    * mux the aac specific config to flv sequence header packet.
+    * @param sh output the sequence header.
+    */
+    virtual int mux_sequence_header(SrsRawAacStreamCodec* codec, std::string& sh);
+    /**
+    * mux the aac audio packet to flv audio packet.
+    * @param frame the aac raw data.
+    * @param nb_frame the count of aac frame.
+    * @param codec the codec info of aac.
+    * @param flv output the muxed flv packet.
+    * @param nb_flv output the muxed flv size.
+    */
+    virtual int mux_aac2flv(char* frame, int nb_frame, SrsRawAacStreamCodec* codec, u_int32_t dts, char** flv, int* nb_flv);
+};
+
 #endif