for #293, move the simple buffer to kernel.

@@ -366,7 +366,7 @@ ModuleLibIncs=(${SRS_OBJS_DIR})
 MODULE_FILES=("srs_kernel_error" "srs_kernel_log" "srs_kernel_stream"
 MODULE_FILES=("srs_kernel_error" "srs_kernel_log" "srs_kernel_stream"
         "srs_kernel_utility" "srs_kernel_flv" "srs_kernel_codec" "srs_kernel_file" 
         "srs_kernel_utility" "srs_kernel_flv" "srs_kernel_codec" "srs_kernel_file" 
         "srs_kernel_consts" "srs_kernel_aac" "srs_kernel_mp3" "srs_kernel_ts"
         "srs_kernel_consts" "srs_kernel_aac" "srs_kernel_mp3" "srs_kernel_ts"
-        "srs_kernel_avc")
+        "srs_kernel_avc" "srs_kernel_buffer")
 KERNEL_INCS="src/kernel"; MODULE_DIR=${KERNEL_INCS} . auto/modules.sh
 KERNEL_INCS="src/kernel"; MODULE_DIR=${KERNEL_INCS} . auto/modules.sh
 KERNEL_OBJS="${MODULE_OBJS[@]}"
 KERNEL_OBJS="${MODULE_OBJS[@]}"
 #
 #

@@ -22,6 +22,8 @@ file
 	../../src/kernel/srs_kernel_aac.cpp,
 	../../src/kernel/srs_kernel_aac.cpp,
 	../../src/kernel/srs_kernel_avc.hpp,
 	../../src/kernel/srs_kernel_avc.hpp,
 	../../src/kernel/srs_kernel_avc.cpp,
 	../../src/kernel/srs_kernel_avc.cpp,
+	../../src/kernel/srs_kernel_buffer.hpp,
+	../../src/kernel/srs_kernel_buffer.cpp,
 	../../src/kernel/srs_kernel_codec.hpp,
 	../../src/kernel/srs_kernel_codec.hpp,
 	../../src/kernel/srs_kernel_codec.cpp,
 	../../src/kernel/srs_kernel_codec.cpp,
 	../../src/kernel/srs_kernel_consts.hpp,
 	../../src/kernel/srs_kernel_consts.hpp,

@@ -24,25 +24,6 @@ CONNECTION WITH THE SOFTWARE OR THE USE OR OTHER DEALINGS IN THE SOFTWARE.
 #include <srs_app_hls.hpp>
 #include <srs_app_hls.hpp>
 
 
 /**
 /**
-* the public data, event HLS disable, others can use it.
-*/
-// 0 = 5.5 kHz = 5512 Hz
-// 1 = 11 kHz = 11025 Hz
-// 2 = 22 kHz = 22050 Hz
-// 3 = 44 kHz = 44100 Hz
-int flv_sample_rates[] = {5512, 11025, 22050, 44100};
-
-// the sample rates in the codec,
-// in the sequence header.
-int aac_sample_rates[] = 
-{
-    96000, 88200, 64000, 48000,
-    44100, 32000, 24000, 22050,
-    16000, 12000, 11025,  8000,
-    7350,     0,     0,    0
-};
-
-/**
 * the HLS section, only available when HLS enabled.
 * the HLS section, only available when HLS enabled.
 */
 */
 #ifdef SRS_AUTO_HLS
 #ifdef SRS_AUTO_HLS
@@ -80,25 +61,6 @@ using namespace std;
 // 63000: 700ms, ts_tbn=90000
 // 63000: 700ms, ts_tbn=90000
 #define SRS_AUTO_HLS_DELAY 63000
 #define SRS_AUTO_HLS_DELAY 63000
 
 
-// the mpegts header specifed the video/audio pid.
-#define TS_VIDEO_PID 256
-#define TS_AUDIO_PID 257
-
-// ts aac stream id.
-#define TS_AUDIO_AAC 0xc0
-// ts avc stream id.
-#define TS_VIDEO_AVC 0xe0
-
-// @see: ngx_rtmp_hls_audio
-/* We assume here AAC frame size is 1024
- * Need to handle AAC frames with frame size of 960 */
-#define _SRS_AAC_SAMPLE_SIZE 1024
-
-// in ms, for HLS aac sync time.
-#define SRS_CONF_DEFAULT_AAC_SYNC 100
-// in ms, for HLS aac flush the audio
-#define SRS_CONF_DEFAULT_AAC_DELAY 100
-
 // @see: ngx_rtmp_mpegts_header
 // @see: ngx_rtmp_mpegts_header
 u_int8_t mpegts_header[] = {
 u_int8_t mpegts_header[] = {
     /* TS */
     /* TS */
@@ -162,25 +124,6 @@ u_int8_t mpegts_header[] = {
     0xff, 0xff, 0xff, 0xff, 0xff, 0xff, 0xff
     0xff, 0xff, 0xff, 0xff, 0xff, 0xff, 0xff
 };
 };
 
 
-// @see: ngx_rtmp_SrsMpegtsFrame_t
-class SrsMpegtsFrame
-{
-public:
-    int64_t         pts;
-    int64_t         dts;
-    int             pid;
-    int             sid;
-    int             cc;
-    bool            key;
-    
-    SrsMpegtsFrame()
-    {
-        pts = dts = 0;
-        pid = sid = cc = 0;
-        key = false;
-    }
-};
-
 // @see: ngx_rtmp_mpegts.c
 // @see: ngx_rtmp_mpegts.c
 // TODO: support full mpegts feature in future.
 // TODO: support full mpegts feature in future.
 class SrsMpegtsWriter
 class SrsMpegtsWriter
@@ -401,69 +344,6 @@ private:
     }
     }
 };
 };
 
 
-SrsHlsAacJitter::SrsHlsAacJitter()
-{
-    base_pts = 0;
-    nb_samples = 0;
-
-    // TODO: config it, 0 means no adjust
-    sync_ms = SRS_CONF_DEFAULT_AAC_SYNC;
-}
-
-SrsHlsAacJitter::~SrsHlsAacJitter()
-{
-}
-
-int64_t SrsHlsAacJitter::on_buffer_start(int64_t flv_pts, int sample_rate, int aac_sample_rate)
-{
-    // use sample rate in flv/RTMP.
-    int flv_sample_rate = flv_sample_rates[sample_rate & 0x03];
-
-    // override the sample rate by sequence header
-    if (aac_sample_rate != __SRS_AAC_SAMPLE_RATE_UNSET) {
-        flv_sample_rate = aac_sample_rates[aac_sample_rate];
-    }
-
-    // sync time set to 0, donot adjust the aac timestamp.
-    if (!sync_ms) {
-        return flv_pts;
-    }
-    
-    // @see: ngx_rtmp_hls_audio
-    // drop the rtmp audio packet timestamp, re-calc it by sample rate.
-    // 
-    // resample for the tbn of ts is 90000, flv is 1000,
-    // we will lost timestamp if use audio packet timestamp,
-    // so we must resample. or audio will corupt in IOS.
-    int64_t est_pts = base_pts + nb_samples * 90000LL * _SRS_AAC_SAMPLE_SIZE / flv_sample_rate;
-    int64_t dpts = (int64_t) (est_pts - flv_pts);
-
-    if (dpts <= (int64_t) sync_ms * 90 && dpts >= (int64_t) sync_ms * -90) {
-        srs_info("HLS correct aac pts "
-            "from %"PRId64" to %"PRId64", base=%"PRId64", nb_samples=%d, sample_rate=%d",
-            flv_pts, est_pts, nb_samples, flv_sample_rate, base_pts);
-
-        nb_samples++;
-        
-        return est_pts;
-    }
-    
-    // resync
-    srs_trace("HLS aac resync, dpts=%"PRId64", pts=%"PRId64
-        ", base=%"PRId64", nb_samples=%"PRId64", sample_rate=%d",
-        dpts, flv_pts, base_pts, nb_samples, flv_sample_rate);
-    
-    base_pts = flv_pts;
-    nb_samples = 1;
-    
-    return flv_pts;
-}
-
-void SrsHlsAacJitter::on_buffer_continue()
-{
-    nb_samples++;
-}
-
 SrsTSMuxer::SrsTSMuxer()
 SrsTSMuxer::SrsTSMuxer()
 {
 {
     writer = new SrsFileWriter();
     writer = new SrsFileWriter();
@@ -963,27 +843,12 @@ int SrsHlsMuxer::create_dir()
 
 
 SrsHlsCache::SrsHlsCache()
 SrsHlsCache::SrsHlsCache()
 {
 {
-    aac_jitter = new SrsHlsAacJitter();
-    
-    ab = new SrsSimpleBuffer();
-    vb = new SrsSimpleBuffer();
-    
-    af = new SrsMpegtsFrame();
-    vf = new SrsMpegtsFrame();
+    cache = new SrsTsCache();
 }
 }
 
 
 SrsHlsCache::~SrsHlsCache()
 SrsHlsCache::~SrsHlsCache()
 {
 {
-    srs_freep(aac_jitter);
-    
-    ab->erase(ab->length());
-    vb->erase(vb->length());
-    
-    srs_freep(ab);
-    srs_freep(vb);
-    
-    srs_freep(af);
-    srs_freep(vf);
+    srs_freep(cache);
 }
 }
 
 
 int SrsHlsCache::on_publish(SrsHlsMuxer* muxer, SrsRequest* req, int64_t segment_start_dts)
 int SrsHlsCache::on_publish(SrsHlsMuxer* muxer, SrsRequest* req, int64_t segment_start_dts)
@@ -1021,7 +886,7 @@ int SrsHlsCache::on_unpublish(SrsHlsMuxer* muxer)
 {
 {
     int ret = ERROR_SUCCESS;
     int ret = ERROR_SUCCESS;
     
     
-    if ((ret = muxer->flush_audio(af, ab)) != ERROR_SUCCESS) {
+    if ((ret = muxer->flush_audio(cache->af, cache->ab)) != ERROR_SUCCESS) {
         srs_error("m3u8 muxer flush audio failed. ret=%d", ret);
         srs_error("m3u8 muxer flush audio failed. ret=%d", ret);
         return ret;
         return ret;
     }
     }
@@ -1047,26 +912,17 @@ int SrsHlsCache::on_sequence_header(SrsHlsMuxer* muxer)
 int SrsHlsCache::write_audio(SrsAvcAacCodec* codec, SrsHlsMuxer* muxer, int64_t pts, SrsCodecSample* sample)
 int SrsHlsCache::write_audio(SrsAvcAacCodec* codec, SrsHlsMuxer* muxer, int64_t pts, SrsCodecSample* sample)
 {
 {
     int ret = ERROR_SUCCESS;
     int ret = ERROR_SUCCESS;
-    
-    // start buffer, set the af
-    if (ab->length() == 0) {
-        pts = aac_jitter->on_buffer_start(pts, sample->sound_rate, codec->aac_sample_rate);
-        
-        af->dts = af->pts = audio_buffer_start_pts = pts;
-        af->pid = TS_AUDIO_PID;
-        af->sid = TS_AUDIO_AAC;
-    } else {
-        aac_jitter->on_buffer_continue();
-    }
+
+    audio_buffer_start_pts = pts;
     
     
     // write audio to cache.
     // write audio to cache.
-    if ((ret = cache_audio(codec, sample)) != ERROR_SUCCESS) {
+    if ((ret = cache->cache_audio(codec, pts, sample)) != ERROR_SUCCESS) {
         return ret;
         return ret;
     }
     }
     
     
     // flush if buffer exceed max size.
     // flush if buffer exceed max size.
-    if (ab->length() > SRS_AUTO_HLS_AUDIO_CACHE_SIZE) {
-        if ((ret = muxer->flush_audio(af, ab)) != ERROR_SUCCESS) {
+    if (cache->ab->length() > SRS_AUTO_HLS_AUDIO_CACHE_SIZE) {
+        if ((ret = muxer->flush_audio(cache->af, cache->ab)) != ERROR_SUCCESS) {
             return ret;
             return ret;
         }
         }
     }
     }
@@ -1075,7 +931,7 @@ int SrsHlsCache::write_audio(SrsAvcAacCodec* codec, SrsHlsMuxer* muxer, int64_t
     int64_t audio_delay = SRS_CONF_DEFAULT_AAC_DELAY;
     int64_t audio_delay = SRS_CONF_DEFAULT_AAC_DELAY;
     // flush if audio delay exceed
     // flush if audio delay exceed
     if (pts - audio_buffer_start_pts > audio_delay * 90) {
     if (pts - audio_buffer_start_pts > audio_delay * 90) {
-        if ((ret = muxer->flush_audio(af, ab)) != ERROR_SUCCESS) {
+        if ((ret = muxer->flush_audio(cache->af, cache->ab)) != ERROR_SUCCESS) {
             return ret;
             return ret;
         }
         }
     }
     }
@@ -1087,7 +943,7 @@ int SrsHlsCache::write_audio(SrsAvcAacCodec* codec, SrsHlsMuxer* muxer, int64_t
     // so we reap event when the audio incoming when segment overflow.
     // so we reap event when the audio incoming when segment overflow.
     // @see https://github.com/winlinvip/simple-rtmp-server/issues/151
     // @see https://github.com/winlinvip/simple-rtmp-server/issues/151
     if (muxer->is_segment_overflow()) {
     if (muxer->is_segment_overflow()) {
-        if ((ret = reap_segment("audio", muxer, af->pts)) != ERROR_SUCCESS) {
+        if ((ret = reap_segment("audio", muxer, cache->af->pts)) != ERROR_SUCCESS) {
             return ret;
             return ret;
         }
         }
     }
     }
@@ -1095,33 +951,26 @@ int SrsHlsCache::write_audio(SrsAvcAacCodec* codec, SrsHlsMuxer* muxer, int64_t
     return ret;
     return ret;
 }
 }
     
     
-int SrsHlsCache::write_video(
-    SrsAvcAacCodec* codec, SrsHlsMuxer* muxer, int64_t dts, SrsCodecSample* sample)
+int SrsHlsCache::write_video(SrsAvcAacCodec* codec, SrsHlsMuxer* muxer, int64_t dts, SrsCodecSample* sample)
 {
 {
     int ret = ERROR_SUCCESS;
     int ret = ERROR_SUCCESS;
     
     
     // write video to cache.
     // write video to cache.
-    if ((ret = cache_video(codec, sample)) != ERROR_SUCCESS) {
+    if ((ret = cache->cache_video(codec, dts, sample)) != ERROR_SUCCESS) {
         return ret;
         return ret;
     }
     }
     
     
-    vf->dts = dts;
-    vf->pts = vf->dts + sample->cts * 90;
-    vf->pid = TS_VIDEO_PID;
-    vf->sid = TS_VIDEO_AVC;
-    vf->key = sample->frame_type == SrsCodecVideoAVCFrameKeyFrame;
-    
     // new segment when:
     // new segment when:
     // 1. base on gop.
     // 1. base on gop.
     // 2. some gops duration overflow.
     // 2. some gops duration overflow.
-    if (vf->key && muxer->is_segment_overflow()) {
-        if ((ret = reap_segment("video", muxer, vf->dts)) != ERROR_SUCCESS) {
+    if (cache->vf->key && muxer->is_segment_overflow()) {
+        if ((ret = reap_segment("video", muxer, cache->vf->dts)) != ERROR_SUCCESS) {
             return ret;
             return ret;
         }
         }
     }
     }
     
     
     // flush video when got one
     // flush video when got one
-    if ((ret = muxer->flush_video(af, ab, vf, vb)) != ERROR_SUCCESS) {
+    if ((ret = muxer->flush_video(cache->af, cache->ab, cache->vf, cache->vb)) != ERROR_SUCCESS) {
         srs_error("m3u8 muxer flush video failed. ret=%d", ret);
         srs_error("m3u8 muxer flush video failed. ret=%d", ret);
         return ret;
         return ret;
     }
     }
@@ -1147,7 +996,7 @@ int SrsHlsCache::reap_segment(string log_desc, SrsHlsMuxer* muxer, int64_t segme
     // segment open, flush the audio.
     // segment open, flush the audio.
     // @see: ngx_rtmp_hls_open_fragment
     // @see: ngx_rtmp_hls_open_fragment
     /* start fragment with audio to make iPhone happy */
     /* start fragment with audio to make iPhone happy */
-    if ((ret = muxer->flush_audio(af, ab)) != ERROR_SUCCESS) {
+    if ((ret = muxer->flush_audio(cache->af, cache->ab)) != ERROR_SUCCESS) {
         srs_error("m3u8 muxer flush audio failed. ret=%d", ret);
         srs_error("m3u8 muxer flush audio failed. ret=%d", ret);
         return ret;
         return ret;
     }
     }
@@ -1155,185 +1004,6 @@ int SrsHlsCache::reap_segment(string log_desc, SrsHlsMuxer* muxer, int64_t segme
     return ret;
     return ret;
 }
 }
 
 
-int SrsHlsCache::cache_audio(SrsAvcAacCodec* codec, SrsCodecSample* sample)
-{
-    int ret = ERROR_SUCCESS;
-    
-    for (int i = 0; i < sample->nb_sample_units; i++) {
-        SrsCodecSampleUnit* sample_unit = &sample->sample_units[i];
-        int32_t size = sample_unit->size;
-        
-        if (!sample_unit->bytes || size <= 0 || size > 0x1fff) {
-            ret = ERROR_HLS_AAC_FRAME_LENGTH;
-            srs_error("invalid aac frame length=%d, ret=%d", size, ret);
-            return ret;
-        }
-        
-        // the frame length is the AAC raw data plus the adts header size.
-        int32_t frame_length = size + 7;
-        
-        // AAC-ADTS
-        // 6.2 Audio Data Transport Stream, ADTS
-        // in aac-iso-13818-7.pdf, page 26.
-        // fixed 7bytes header
-        static u_int8_t adts_header[7] = {0xff, 0xf1, 0x00, 0x00, 0x00, 0x0f, 0xfc};
-        /*
-        // adts_fixed_header
-        // 2B, 16bits
-        int16_t syncword; //12bits, '1111 1111 1111'
-        int8_t ID; //1bit, '0'
-        int8_t layer; //2bits, '00'
-        int8_t protection_absent; //1bit, can be '1'
-        // 12bits
-        int8_t profile; //2bit, 7.1 Profiles, page 40
-        TSAacSampleFrequency sampling_frequency_index; //4bits, Table 35, page 46
-        int8_t private_bit; //1bit, can be '0'
-        int8_t channel_configuration; //3bits, Table 8
-        int8_t original_or_copy; //1bit, can be '0'
-        int8_t home; //1bit, can be '0'
-        
-        // adts_variable_header
-        // 28bits
-        int8_t copyright_identification_bit; //1bit, can be '0'
-        int8_t copyright_identification_start; //1bit, can be '0'
-        int16_t frame_length; //13bits
-        int16_t adts_buffer_fullness; //11bits, 7FF signals that the bitstream is a variable rate bitstream.
-        int8_t number_of_raw_data_blocks_in_frame; //2bits, 0 indicating 1 raw_data_block()
-        */
-        // profile, 2bits
-        adts_header[2] = (codec->aac_profile << 6) & 0xc0;
-        // sampling_frequency_index 4bits
-        adts_header[2] |= (codec->aac_sample_rate << 2) & 0x3c;
-        // channel_configuration 3bits
-        adts_header[2] |= (codec->aac_channels >> 2) & 0x01;
-        adts_header[3] = (codec->aac_channels << 6) & 0xc0;
-        // frame_length 13bits
-        adts_header[3] |= (frame_length >> 11) & 0x03;
-        adts_header[4] = (frame_length >> 3) & 0xff;
-        adts_header[5] = ((frame_length << 5) & 0xe0);
-        // adts_buffer_fullness; //11bits
-        adts_header[5] |= 0x1f;
-
-        // copy to audio buffer
-        ab->append((const char*)adts_header, sizeof(adts_header));
-        ab->append(sample_unit->bytes, sample_unit->size);
-    }
-    
-    return ret;
-}
-
-int SrsHlsCache::cache_video(SrsAvcAacCodec* codec, SrsCodecSample* sample)
-{
-    int ret = ERROR_SUCCESS;
-    
-    // for type1/5/6, insert aud packet.
-    static u_int8_t aud_nal[] = { 0x00, 0x00, 0x00, 0x01, 0x09, 0xf0 };
-    
-    bool sps_pps_sent = false;
-    bool aud_sent = false;
-    /**
-    * a ts sample is format as:
-    * 00 00 00 01 // header
-    *       xxxxxxx // data bytes
-    * 00 00 01 // continue header
-    *       xxxxxxx // data bytes.
-    * so, for each sample, we append header in aud_nal, then appends the bytes in sample.
-    */
-    for (int i = 0; i < sample->nb_sample_units; i++) {
-        SrsCodecSampleUnit* sample_unit = &sample->sample_units[i];
-        int32_t size = sample_unit->size;
-        
-        if (!sample_unit->bytes || size <= 0) {
-            ret = ERROR_HLS_AVC_SAMPLE_SIZE;
-            srs_error("invalid avc sample length=%d, ret=%d", size, ret);
-            return ret;
-        }
-        
-        /**
-        * step 1:
-        * first, before each "real" sample, 
-        * we add some packets according to the nal_unit_type,
-        * for example, when got nal_unit_type=5, insert SPS/PPS before sample.
-        */
-        
-        // 5bits, 7.3.1 NAL unit syntax, 
-        // H.264-AVC-ISO_IEC_14496-10.pdf, page 44.
-        u_int8_t nal_unit_type;
-        nal_unit_type = *sample_unit->bytes;
-        nal_unit_type &= 0x1f;
-        
-        // @see: ngx_rtmp_hls_video
-        // Table 7-1 – NAL unit type codes, page 61
-        // 1: Coded slice
-        if (nal_unit_type == 1) {
-            sps_pps_sent = false;
-        }
-        
-        // 6: Supplemental enhancement information (SEI) sei_rbsp( ), page 61
-        // @see: ngx_rtmp_hls_append_aud
-        if (!aud_sent) {
-            // @remark, when got type 9, we donot send aud_nal, but it will make 
-            //      ios unhappy, so we remove it.
-            // @see https://github.com/winlinvip/simple-rtmp-server/issues/281
-            /*if (nal_unit_type == 9) {
-                aud_sent = true;
-            }*/
-            
-            if (nal_unit_type == 1 || nal_unit_type == 5 || nal_unit_type == 6) {
-                // for type 6, append a aud with type 9.
-                vb->append((const char*)aud_nal, sizeof(aud_nal));
-                aud_sent = true;
-            }
-        }
-        
-        // 5: Coded slice of an IDR picture.
-        // insert sps/pps before IDR or key frame is ok.
-        if (nal_unit_type == 5 && !sps_pps_sent) {
-            sps_pps_sent = true;
-            
-            // @see: ngx_rtmp_hls_append_sps_pps
-            if (codec->sequenceParameterSetLength > 0) {
-                // AnnexB prefix, for sps always 4 bytes header
-                vb->append((const char*)aud_nal, 4);
-                // sps
-                vb->append(codec->sequenceParameterSetNALUnit, codec->sequenceParameterSetLength);
-            }
-            if (codec->pictureParameterSetLength > 0) {
-                // AnnexB prefix, for pps always 4 bytes header
-                vb->append((const char*)aud_nal, 4);
-                // pps
-                vb->append(codec->pictureParameterSetNALUnit, codec->pictureParameterSetLength);
-            }
-        }
-        
-        // 7-9, ignore, @see: ngx_rtmp_hls_video
-        if (nal_unit_type >= 7 && nal_unit_type <= 9) {
-            continue;
-        }
-        
-        /**
-        * step 2:
-        * output the "real" sample, in buf.
-        * when we output some special assist packets according to nal_unit_type
-        */
-        
-        // sample start prefix, '00 00 00 01' or '00 00 01'
-        u_int8_t* p = aud_nal + 1;
-        u_int8_t* end = p + 3;
-        
-        // first AnnexB prefix is long (4 bytes)
-        if (vb->length() == 0) {
-            p = aud_nal;
-        }
-        vb->append((const char*)p, end - p);
-        
-        // sample data
-        vb->append(sample_unit->bytes, sample_unit->size);
-    }
-    
-    return ret;
-}
-
 SrsHls::SrsHls(SrsSource* _source)
 SrsHls::SrsHls(SrsSource* _source)
 {
 {
     hls_enabled = false;
     hls_enabled = false;

@@ -30,19 +30,6 @@ CONNECTION WITH THE SOFTWARE OR THE USE OR OTHER DEALINGS IN THE SOFTWARE.
 #include <srs_core.hpp>
 #include <srs_core.hpp>
 
 
 /**
 /**
-* the public data, event HLS disable, others can use it.
-*/
-/**
-* the flv sample rate map
-*/
-extern int flv_sample_rates[];
-
-/**
-* the aac sample rate map
-*/
-extern int aac_sample_rates[];
-
-/**
 * the HLS section, only available when HLS enabled.
 * the HLS section, only available when HLS enabled.
 */
 */
 #ifdef SRS_AUTO_HLS
 #ifdef SRS_AUTO_HLS
@@ -62,38 +49,8 @@ class SrsPithyPrint;
 class SrsSource;
 class SrsSource;
 class SrsFileWriter;
 class SrsFileWriter;
 class SrsSimpleBuffer;
 class SrsSimpleBuffer;
-
-/**
-* jitter correct for audio,
-* the sample rate 44100/32000 will lost precise,
-* when mp4/ts(tbn=90000) covert to flv/rtmp(1000),
-* so the Hls on ipad or iphone will corrupt,
-* @see nginx-rtmp: est_pts
-*/
-class SrsHlsAacJitter
-{
-private:
-    int64_t base_pts;
-    int64_t nb_samples;
-    int sync_ms;
-public:
-    SrsHlsAacJitter();
-    virtual ~SrsHlsAacJitter();
-    /**
-    * when buffer start, calc the "correct" pts for ts,
-    * @param flv_pts, the flv pts calc from flv header timestamp,
-    * @param sample_rate, the sample rate in format(flv/RTMP packet header).
-    * @param aac_sample_rate, the sample rate in codec(sequence header).
-    * @return the calc correct pts.
-    */
-    virtual int64_t on_buffer_start(int64_t flv_pts, int sample_rate, int aac_sample_rate);
-    /**
-    * when buffer continue, muxer donot write to file,
-    * the audio buffer continue grow and donot need a pts,
-    * for the ts audio PES packet only has one pts at the first time.
-    */
-    virtual void on_buffer_continue();
-};
+class SrsTsAacJitter;
+class SrsTsCache;
 
 
 /**
 /**
 * write data from frame(header info) and buffer(data) to ts file.
 * write data from frame(header info) and buffer(data) to ts file.
@@ -223,22 +180,15 @@ private:
 * about the flv tbn problem:
 * about the flv tbn problem:
 *   flv tbn is 1/1000, ts tbn is 1/90000,
 *   flv tbn is 1/1000, ts tbn is 1/90000,
 *   when timestamp convert to flv tbn, it will loose precise,
 *   when timestamp convert to flv tbn, it will loose precise,
-*   so we must gather audio frame together, and recalc the timestamp @see SrsHlsAacJitter,
+*   so we must gather audio frame together, and recalc the timestamp @see SrsTsAacJitter,
 *   we use a aac jitter to correct the audio pts.
 *   we use a aac jitter to correct the audio pts.
 */
 */
 class SrsHlsCache
 class SrsHlsCache
 {
 {
 private:
 private:
-    // current frame and buffer
-    SrsMpegtsFrame* af;
-    SrsSimpleBuffer* ab;
-    SrsMpegtsFrame* vf;
-    SrsSimpleBuffer* vb;
-private:
     // the audio cache buffer start pts, to flush audio if full.
     // the audio cache buffer start pts, to flush audio if full.
     int64_t audio_buffer_start_pts;
     int64_t audio_buffer_start_pts;
-    // time jitter for aac
-    SrsHlsAacJitter* aac_jitter;
+    SrsTsCache* cache;
 public:
 public:
     SrsHlsCache();
     SrsHlsCache();
     virtual ~SrsHlsCache();
     virtual ~SrsHlsCache();
@@ -271,8 +221,6 @@ private:
     * so, user must reap_segment then flush_video to hls muxer.
     * so, user must reap_segment then flush_video to hls muxer.
     */
     */
     virtual int reap_segment(std::string log_desc, SrsHlsMuxer* muxer, int64_t segment_start_dts);
     virtual int reap_segment(std::string log_desc, SrsHlsMuxer* muxer, int64_t segment_start_dts);
-    virtual int cache_audio(SrsAvcAacCodec* codec, SrsCodecSample* sample);
-    virtual int cache_video(SrsAvcAacCodec* codec, SrsCodecSample* sample);
 };
 };
 
 
 /**
 /**

@@ -27,6 +27,358 @@ CONNECTION WITH THE SOFTWARE OR THE USE OR OTHER DEALINGS IN THE SOFTWARE.
 #include <srs_kernel_log.hpp>
 #include <srs_kernel_log.hpp>
 #include <srs_kernel_stream.hpp>
 #include <srs_kernel_stream.hpp>
 #include <srs_kernel_utility.hpp>
 #include <srs_kernel_utility.hpp>
+#include <srs_kernel_buffer.hpp>
+
+// in ms, for HLS aac sync time.
+#define SRS_CONF_DEFAULT_AAC_SYNC 100
+
+// @see: ngx_rtmp_hls_audio
+/* We assume here AAC frame size is 1024
+ * Need to handle AAC frames with frame size of 960 */
+#define _SRS_AAC_SAMPLE_SIZE 1024
+
+// the mpegts header specifed the video/audio pid.
+#define TS_VIDEO_PID 256
+#define TS_AUDIO_PID 257
+
+// ts aac stream id.
+#define TS_AUDIO_AAC 0xc0
+// ts avc stream id.
+#define TS_VIDEO_AVC 0xe0
+
+/**
+* the public data, event HLS disable, others can use it.
+*/
+// 0 = 5.5 kHz = 5512 Hz
+// 1 = 11 kHz = 11025 Hz
+// 2 = 22 kHz = 22050 Hz
+// 3 = 44 kHz = 44100 Hz
+int flv_sample_rates[] = {5512, 11025, 22050, 44100};
+
+// the sample rates in the codec,
+// in the sequence header.
+int aac_sample_rates[] = 
+{
+    96000, 88200, 64000, 48000,
+    44100, 32000, 24000, 22050,
+    16000, 12000, 11025,  8000,
+    7350,     0,     0,    0
+};
+
+SrsMpegtsFrame::SrsMpegtsFrame()
+{
+    pts = dts = 0;
+    pid = sid = cc = 0;
+    key = false;
+}
+
+SrsTsAacJitter::SrsTsAacJitter()
+{
+    base_pts = 0;
+    nb_samples = 0;
+
+    // TODO: config it, 0 means no adjust
+    sync_ms = SRS_CONF_DEFAULT_AAC_SYNC;
+}
+
+SrsTsAacJitter::~SrsTsAacJitter()
+{
+}
+
+int64_t SrsTsAacJitter::on_buffer_start(int64_t flv_pts, int sample_rate, int aac_sample_rate)
+{
+    // use sample rate in flv/RTMP.
+    int flv_sample_rate = flv_sample_rates[sample_rate & 0x03];
+
+    // override the sample rate by sequence header
+    if (aac_sample_rate != __SRS_AAC_SAMPLE_RATE_UNSET) {
+        flv_sample_rate = aac_sample_rates[aac_sample_rate];
+    }
+
+    // sync time set to 0, donot adjust the aac timestamp.
+    if (!sync_ms) {
+        return flv_pts;
+    }
+    
+    // @see: ngx_rtmp_hls_audio
+    // drop the rtmp audio packet timestamp, re-calc it by sample rate.
+    // 
+    // resample for the tbn of ts is 90000, flv is 1000,
+    // we will lost timestamp if use audio packet timestamp,
+    // so we must resample. or audio will corupt in IOS.
+    int64_t est_pts = base_pts + nb_samples * 90000LL * _SRS_AAC_SAMPLE_SIZE / flv_sample_rate;
+    int64_t dpts = (int64_t) (est_pts - flv_pts);
+
+    if (dpts <= (int64_t) sync_ms * 90 && dpts >= (int64_t) sync_ms * -90) {
+        srs_info("HLS correct aac pts "
+            "from %"PRId64" to %"PRId64", base=%"PRId64", nb_samples=%d, sample_rate=%d",
+            flv_pts, est_pts, nb_samples, flv_sample_rate, base_pts);
+
+        nb_samples++;
+        
+        return est_pts;
+    }
+    
+    // resync
+    srs_trace("HLS aac resync, dpts=%"PRId64", pts=%"PRId64
+        ", base=%"PRId64", nb_samples=%"PRId64", sample_rate=%d",
+        dpts, flv_pts, base_pts, nb_samples, flv_sample_rate);
+    
+    base_pts = flv_pts;
+    nb_samples = 1;
+    
+    return flv_pts;
+}
+
+void SrsTsAacJitter::on_buffer_continue()
+{
+    nb_samples++;
+}
+
+SrsTsCache::SrsTsCache()
+{
+    aac_jitter = new SrsTsAacJitter();
+    
+    ab = new SrsSimpleBuffer();
+    vb = new SrsSimpleBuffer();
+    
+    af = new SrsMpegtsFrame();
+    vf = new SrsMpegtsFrame();
+}
+
+SrsTsCache::~SrsTsCache()
+{
+    srs_freep(aac_jitter);
+    
+    ab->erase(ab->length());
+    vb->erase(vb->length());
+    
+    srs_freep(ab);
+    srs_freep(vb);
+    
+    srs_freep(af);
+    srs_freep(vf);
+}
+    
+int SrsTsCache::cache_audio(SrsAvcAacCodec* codec, int64_t pts, SrsCodecSample* sample)
+{
+    int ret = ERROR_SUCCESS;
+    
+    // start buffer, set the af
+    if (ab->length() == 0) {
+        pts = aac_jitter->on_buffer_start(pts, sample->sound_rate, codec->aac_sample_rate);
+        
+        af->dts = af->pts = pts;
+        af->pid = TS_AUDIO_PID;
+        af->sid = TS_AUDIO_AAC;
+    } else {
+        aac_jitter->on_buffer_continue();
+    }
+    
+    // write audio to cache.
+    if ((ret = do_cache_audio(codec, sample)) != ERROR_SUCCESS) {
+        return ret;
+    }
+    
+    return ret;
+}
+    
+int SrsTsCache::cache_video(SrsAvcAacCodec* codec, int64_t dts, SrsCodecSample* sample)
+{
+    int ret = ERROR_SUCCESS;
+    
+    // write video to cache.
+    if ((ret = do_cache_video(codec, sample)) != ERROR_SUCCESS) {
+        return ret;
+    }
+    
+    vf->dts = dts;
+    vf->pts = vf->dts + sample->cts * 90;
+    vf->pid = TS_VIDEO_PID;
+    vf->sid = TS_VIDEO_AVC;
+    vf->key = sample->frame_type == SrsCodecVideoAVCFrameKeyFrame;
+    
+    return ret;
+}
+
+int SrsTsCache::do_cache_audio(SrsAvcAacCodec* codec, SrsCodecSample* sample)
+{
+    int ret = ERROR_SUCCESS;
+    
+    for (int i = 0; i < sample->nb_sample_units; i++) {
+        SrsCodecSampleUnit* sample_unit = &sample->sample_units[i];
+        int32_t size = sample_unit->size;
+        
+        if (!sample_unit->bytes || size <= 0 || size > 0x1fff) {
+            ret = ERROR_HLS_AAC_FRAME_LENGTH;
+            srs_error("invalid aac frame length=%d, ret=%d", size, ret);
+            return ret;
+        }
+        
+        // the frame length is the AAC raw data plus the adts header size.
+        int32_t frame_length = size + 7;
+        
+        // AAC-ADTS
+        // 6.2 Audio Data Transport Stream, ADTS
+        // in aac-iso-13818-7.pdf, page 26.
+        // fixed 7bytes header
+        static u_int8_t adts_header[7] = {0xff, 0xf1, 0x00, 0x00, 0x00, 0x0f, 0xfc};
+        /*
+        // adts_fixed_header
+        // 2B, 16bits
+        int16_t syncword; //12bits, '1111 1111 1111'
+        int8_t ID; //1bit, '0'
+        int8_t layer; //2bits, '00'
+        int8_t protection_absent; //1bit, can be '1'
+        // 12bits
+        int8_t profile; //2bit, 7.1 Profiles, page 40
+        TSAacSampleFrequency sampling_frequency_index; //4bits, Table 35, page 46
+        int8_t private_bit; //1bit, can be '0'
+        int8_t channel_configuration; //3bits, Table 8
+        int8_t original_or_copy; //1bit, can be '0'
+        int8_t home; //1bit, can be '0'
+        
+        // adts_variable_header
+        // 28bits
+        int8_t copyright_identification_bit; //1bit, can be '0'
+        int8_t copyright_identification_start; //1bit, can be '0'
+        int16_t frame_length; //13bits
+        int16_t adts_buffer_fullness; //11bits, 7FF signals that the bitstream is a variable rate bitstream.
+        int8_t number_of_raw_data_blocks_in_frame; //2bits, 0 indicating 1 raw_data_block()
+        */
+        // profile, 2bits
+        adts_header[2] = (codec->aac_profile << 6) & 0xc0;
+        // sampling_frequency_index 4bits
+        adts_header[2] |= (codec->aac_sample_rate << 2) & 0x3c;
+        // channel_configuration 3bits
+        adts_header[2] |= (codec->aac_channels >> 2) & 0x01;
+        adts_header[3] = (codec->aac_channels << 6) & 0xc0;
+        // frame_length 13bits
+        adts_header[3] |= (frame_length >> 11) & 0x03;
+        adts_header[4] = (frame_length >> 3) & 0xff;
+        adts_header[5] = ((frame_length << 5) & 0xe0);
+        // adts_buffer_fullness; //11bits
+        adts_header[5] |= 0x1f;
+
+        // copy to audio buffer
+        ab->append((const char*)adts_header, sizeof(adts_header));
+        ab->append(sample_unit->bytes, sample_unit->size);
+    }
+    
+    return ret;
+}
+
+int SrsTsCache::do_cache_video(SrsAvcAacCodec* codec, SrsCodecSample* sample)
+{
+    int ret = ERROR_SUCCESS;
+    
+    // for type1/5/6, insert aud packet.
+    static u_int8_t aud_nal[] = { 0x00, 0x00, 0x00, 0x01, 0x09, 0xf0 };
+    
+    bool sps_pps_sent = false;
+    bool aud_sent = false;
+    /**
+    * a ts sample is format as:
+    * 00 00 00 01 // header
+    *       xxxxxxx // data bytes
+    * 00 00 01 // continue header
+    *       xxxxxxx // data bytes.
+    * so, for each sample, we append header in aud_nal, then appends the bytes in sample.
+    */
+    for (int i = 0; i < sample->nb_sample_units; i++) {
+        SrsCodecSampleUnit* sample_unit = &sample->sample_units[i];
+        int32_t size = sample_unit->size;
+        
+        if (!sample_unit->bytes || size <= 0) {
+            ret = ERROR_HLS_AVC_SAMPLE_SIZE;
+            srs_error("invalid avc sample length=%d, ret=%d", size, ret);
+            return ret;
+        }
+        
+        /**
+        * step 1:
+        * first, before each "real" sample, 
+        * we add some packets according to the nal_unit_type,
+        * for example, when got nal_unit_type=5, insert SPS/PPS before sample.
+        */
+        
+        // 5bits, 7.3.1 NAL unit syntax, 
+        // H.264-AVC-ISO_IEC_14496-10.pdf, page 44.
+        u_int8_t nal_unit_type;
+        nal_unit_type = *sample_unit->bytes;
+        nal_unit_type &= 0x1f;
+        
+        // @see: ngx_rtmp_hls_video
+        // Table 7-1 ¨C NAL unit type codes, page 61
+        // 1: Coded slice
+        if (nal_unit_type == 1) {
+            sps_pps_sent = false;
+        }
+        
+        // 6: Supplemental enhancement information (SEI) sei_rbsp( ), page 61
+        // @see: ngx_rtmp_hls_append_aud
+        if (!aud_sent) {
+            // @remark, when got type 9, we donot send aud_nal, but it will make 
+            //      ios unhappy, so we remove it.
+            // @see https://github.com/winlinvip/simple-rtmp-server/issues/281
+            /*if (nal_unit_type == 9) {
+                aud_sent = true;
+            }*/
+            
+            if (nal_unit_type == 1 || nal_unit_type == 5 || nal_unit_type == 6) {
+                // for type 6, append a aud with type 9.
+                vb->append((const char*)aud_nal, sizeof(aud_nal));
+                aud_sent = true;
+            }
+        }
+        
+        // 5: Coded slice of an IDR picture.
+        // insert sps/pps before IDR or key frame is ok.
+        if (nal_unit_type == 5 && !sps_pps_sent) {
+            sps_pps_sent = true;
+            
+            // @see: ngx_rtmp_hls_append_sps_pps
+            if (codec->sequenceParameterSetLength > 0) {
+                // AnnexB prefix, for sps always 4 bytes header
+                vb->append((const char*)aud_nal, 4);
+                // sps
+                vb->append(codec->sequenceParameterSetNALUnit, codec->sequenceParameterSetLength);
+            }
+            if (codec->pictureParameterSetLength > 0) {
+                // AnnexB prefix, for pps always 4 bytes header
+                vb->append((const char*)aud_nal, 4);
+                // pps
+                vb->append(codec->pictureParameterSetNALUnit, codec->pictureParameterSetLength);
+            }
+        }
+        
+        // 7-9, ignore, @see: ngx_rtmp_hls_video
+        if (nal_unit_type >= 7 && nal_unit_type <= 9) {
+            continue;
+        }
+        
+        /**
+        * step 2:
+        * output the "real" sample, in buf.
+        * when we output some special assist packets according to nal_unit_type
+        */
+        
+        // sample start prefix, '00 00 00 01' or '00 00 01'
+        u_int8_t* p = aud_nal + 1;
+        u_int8_t* end = p + 3;
+        
+        // first AnnexB prefix is long (4 bytes)
+        if (vb->length() == 0) {
+            p = aud_nal;
+        }
+        vb->append((const char*)p, end - p);
+        
+        // sample data
+        vb->append(sample_unit->bytes, sample_unit->size);
+    }
+    
+    return ret;
+}
 
 
 SrsCodecSampleUnit::SrsCodecSampleUnit()
 SrsCodecSampleUnit::SrsCodecSampleUnit()
 {
 {

@@ -33,11 +33,30 @@ CONNECTION WITH THE SOFTWARE OR THE USE OR OTHER DEALINGS IN THE SOFTWARE.
 #include <srs_kernel_codec.hpp>
 #include <srs_kernel_codec.hpp>
 
 
 class SrsStream;
 class SrsStream;
-class SrsAmf0Object;
+class SrsMpegtsFrame;
+class SrsSimpleBuffer;
+class SrsAvcAacCodec;
+class SrsCodecSample;
+
+/**
+* the public data, event HLS disable, others can use it.
+*/
+/**
+* the flv sample rate map
+*/
+extern int flv_sample_rates[];
+
+/**
+* the aac sample rate map
+*/
+extern int aac_sample_rates[];
 
 
 #define __SRS_SRS_MAX_CODEC_SAMPLE 128
 #define __SRS_SRS_MAX_CODEC_SAMPLE 128
 #define __SRS_AAC_SAMPLE_RATE_UNSET 15
 #define __SRS_AAC_SAMPLE_RATE_UNSET 15
 
 
+// in ms, for HLS aac flush the audio
+#define SRS_CONF_DEFAULT_AAC_DELAY 100
+
 /**
 /**
 * the FLV/RTMP supported audio sample size.
 * the FLV/RTMP supported audio sample size.
 * Size of each audio sample. This parameter only pertains to
 * Size of each audio sample. This parameter only pertains to
@@ -70,6 +89,90 @@ enum SrsCodecAudioSoundType
     SrsCodecAudioSoundTypeStereo                    = 1,
     SrsCodecAudioSoundTypeStereo                    = 1,
 };
 };
 
 
+// @see: ngx_rtmp_SrsMpegtsFrame_t
+class SrsMpegtsFrame
+{
+public:
+    int64_t         pts;
+    int64_t         dts;
+    int             pid;
+    int             sid;
+    int             cc;
+    bool            key;
+    
+    SrsMpegtsFrame();
+};
+
+/**
+* jitter correct for audio,
+* the sample rate 44100/32000 will lost precise,
+* when mp4/ts(tbn=90000) covert to flv/rtmp(1000),
+* so the Hls on ipad or iphone will corrupt,
+* @see nginx-rtmp: est_pts
+*/
+class SrsTsAacJitter
+{
+private:
+    int64_t base_pts;
+    int64_t nb_samples;
+    int sync_ms;
+public:
+    SrsTsAacJitter();
+    virtual ~SrsTsAacJitter();
+    /**
+    * when buffer start, calc the "correct" pts for ts,
+    * @param flv_pts, the flv pts calc from flv header timestamp,
+    * @param sample_rate, the sample rate in format(flv/RTMP packet header).
+    * @param aac_sample_rate, the sample rate in codec(sequence header).
+    * @return the calc correct pts.
+    */
+    virtual int64_t on_buffer_start(int64_t flv_pts, int sample_rate, int aac_sample_rate);
+    /**
+    * when buffer continue, muxer donot write to file,
+    * the audio buffer continue grow and donot need a pts,
+    * for the ts audio PES packet only has one pts at the first time.
+    */
+    virtual void on_buffer_continue();
+};
+
+/**
+* ts stream cache, 
+* use to cache ts stream.
+* 
+* about the flv tbn problem:
+*   flv tbn is 1/1000, ts tbn is 1/90000,
+*   when timestamp convert to flv tbn, it will loose precise,
+*   so we must gather audio frame together, and recalc the timestamp @see SrsTsAacJitter,
+*   we use a aac jitter to correct the audio pts.
+*/
+class SrsTsCache
+{
+public:
+    // current frame and buffer
+    SrsMpegtsFrame* af;
+    SrsSimpleBuffer* ab;
+    SrsMpegtsFrame* vf;
+    SrsSimpleBuffer* vb;
+protected:
+    // time jitter for aac
+    SrsTsAacJitter* aac_jitter;
+public:
+    SrsTsCache();
+    virtual ~SrsTsCache();
+public:
+    /**
+    * write audio to cache
+    */
+    virtual int cache_audio(SrsAvcAacCodec* codec, int64_t pts, SrsCodecSample* sample);
+    /**
+    * write video to muxer.
+    */
+    virtual int cache_video(SrsAvcAacCodec* codec, int64_t dts, SrsCodecSample* sample);
+private:
+    virtual int do_cache_audio(SrsAvcAacCodec* codec, SrsCodecSample* sample);
+    virtual int do_cache_video(SrsAvcAacCodec* codec, SrsCodecSample* sample);
+};
+
 /**
 /**
 * the codec sample unit.
 * the codec sample unit.
 * for h.264 video packet, a NALU is a sample unit.
 * for h.264 video packet, a NALU is a sample unit.

+/*
+The MIT License (MIT)
+
+Copyright (c) 2013-2015 winlin
+
+Permission is hereby granted, free of charge, to any person obtaining a copy of
+this software and associated documentation files (the "Software"), to deal in
+the Software without restriction, including without limitation the rights to
+use, copy, modify, merge, publish, distribute, sublicense, and/or sell copies of
+the Software, and to permit persons to whom the Software is furnished to do so,
+subject to the following conditions:
+
+The above copyright notice and this permission notice shall be included in all
+copies or substantial portions of the Software.
+
+THE SOFTWARE IS PROVIDED "AS IS", WITHOUT WARRANTY OF ANY KIND, EXPRESS OR
+IMPLIED, INCLUDING BUT NOT LIMITED TO THE WARRANTIES OF MERCHANTABILITY, FITNESS
+FOR A PARTICULAR PURPOSE AND NONINFRINGEMENT. IN NO EVENT SHALL THE AUTHORS OR
+COPYRIGHT HOLDERS BE LIABLE FOR ANY CLAIM, DAMAGES OR OTHER LIABILITY, WHETHER
+IN AN ACTION OF CONTRACT, TORT OR OTHERWISE, ARISING FROM, OUT OF OR IN
+CONNECTION WITH THE SOFTWARE OR THE USE OR OTHER DEALINGS IN THE SOFTWARE.
+*/
+
+#include <srs_kernel_buffer.hpp>
+
+#include <srs_kernel_error.hpp>
+#include <srs_kernel_log.hpp>
+#include <srs_kernel_utility.hpp>
+#include <srs_core_performance.hpp>
+
+SrsSimpleBuffer::SrsSimpleBuffer()
+{
+}
+
+SrsSimpleBuffer::~SrsSimpleBuffer()
+{
+}
+
+int SrsSimpleBuffer::length()
+{
+    int len = (int)data.size();
+    srs_assert(len >= 0);
+    return len;
+}
+
+char* SrsSimpleBuffer::bytes()
+{
+    return (length() == 0)? NULL : &data.at(0);
+}
+
+void SrsSimpleBuffer::erase(int size)
+{
+    if (size <= 0) {
+        return;
+    }
+    
+    if (size >= length()) {
+        data.clear();
+        return;
+    }
+    
+    data.erase(data.begin(), data.begin() + size);
+}
+
+void SrsSimpleBuffer::append(const char* bytes, int size)
+{
+    srs_assert(size > 0);
+
+    data.insert(data.end(), bytes, bytes + size);
+}

+/*
+The MIT License (MIT)
+
+Copyright (c) 2013-2015 winlin
+
+Permission is hereby granted, free of charge, to any person obtaining a copy of
+this software and associated documentation files (the "Software"), to deal in
+the Software without restriction, including without limitation the rights to
+use, copy, modify, merge, publish, distribute, sublicense, and/or sell copies of
+the Software, and to permit persons to whom the Software is furnished to do so,
+subject to the following conditions:
+
+The above copyright notice and this permission notice shall be included in all
+copies or substantial portions of the Software.
+
+THE SOFTWARE IS PROVIDED "AS IS", WITHOUT WARRANTY OF ANY KIND, EXPRESS OR
+IMPLIED, INCLUDING BUT NOT LIMITED TO THE WARRANTIES OF MERCHANTABILITY, FITNESS
+FOR A PARTICULAR PURPOSE AND NONINFRINGEMENT. IN NO EVENT SHALL THE AUTHORS OR
+COPYRIGHT HOLDERS BE LIABLE FOR ANY CLAIM, DAMAGES OR OTHER LIABILITY, WHETHER
+IN AN ACTION OF CONTRACT, TORT OR OTHERWISE, ARISING FROM, OUT OF OR IN
+CONNECTION WITH THE SOFTWARE OR THE USE OR OTHER DEALINGS IN THE SOFTWARE.
+*/
+
+#ifndef SRS_KERNEL_BUFFER_HPP
+#define SRS_KERNEL_BUFFER_HPP
+
+/*
+#include <srs_kernel_buffer.hpp>
+*/
+
+#include <srs_core.hpp>
+
+#include <vector>
+
+/**
+* the simple buffer use vector to append bytes,
+* it's for hls and http, and need to be refined in future.
+*/
+class SrsSimpleBuffer
+{
+private:
+    std::vector<char> data;
+public:
+    SrsSimpleBuffer();
+    virtual ~SrsSimpleBuffer();
+public:
+    /**
+    * get the length of buffer. empty if zero.
+    * @remark assert length() is not negative.
+    */
+    virtual int length();
+    /**
+    * get the buffer bytes.
+    * @return the bytes, NULL if empty.
+    */
+    virtual char* bytes();
+    /**
+    * erase size of bytes from begin.
+    * @param size to erase size of bytes. 
+    *       clear if size greater than or equals to length()
+    * @remark ignore size is not positive.
+    */
+    virtual void erase(int size);
+    /**
+    * append specified bytes to buffer.
+    * @param size the size of bytes
+    * @remark assert size is positive.
+    */
+    virtual void append(const char* bytes, int size);
+};
+
+#endif

@@ -40,47 +40,6 @@ CONNECTION WITH THE SOFTWARE OR THE USE OR OTHER DEALINGS IN THE SOFTWARE.
 // @see SrsProtocol::read_message_header().
 // @see SrsProtocol::read_message_header().
 #define SRS_RTMP_MAX_MESSAGE_HEADER 11
 #define SRS_RTMP_MAX_MESSAGE_HEADER 11
 
 
-SrsSimpleBuffer::SrsSimpleBuffer()
-{
-}
-
-SrsSimpleBuffer::~SrsSimpleBuffer()
-{
-}
-
-int SrsSimpleBuffer::length()
-{
-    int len = (int)data.size();
-    srs_assert(len >= 0);
-    return len;
-}
-
-char* SrsSimpleBuffer::bytes()
-{
-    return (length() == 0)? NULL : &data.at(0);
-}
-
-void SrsSimpleBuffer::erase(int size)
-{
-    if (size <= 0) {
-        return;
-    }
-    
-    if (size >= length()) {
-        data.clear();
-        return;
-    }
-    
-    data.erase(data.begin(), data.begin() + size);
-}
-
-void SrsSimpleBuffer::append(const char* bytes, int size)
-{
-    srs_assert(size > 0);
-
-    data.insert(data.end(), bytes, bytes + size);
-}
-
 #ifdef SRS_PERF_MERGED_READ
 #ifdef SRS_PERF_MERGED_READ
 IMergeReadHandler::IMergeReadHandler()
 IMergeReadHandler::IMergeReadHandler()
 {
 {

@@ -30,47 +30,9 @@ CONNECTION WITH THE SOFTWARE OR THE USE OR OTHER DEALINGS IN THE SOFTWARE.
 
 
 #include <srs_core.hpp>
 #include <srs_core.hpp>
 
 
-#include <vector>
-
 #include <srs_protocol_io.hpp>
 #include <srs_protocol_io.hpp>
 #include <srs_core_performance.hpp>
 #include <srs_core_performance.hpp>
-
-/**
-* the simple buffer use vector to append bytes,
-* it's for hls and http, and need to be refined in future.
-*/
-class SrsSimpleBuffer
-{
-private:
-    std::vector<char> data;
-public:
-    SrsSimpleBuffer();
-    virtual ~SrsSimpleBuffer();
-public:
-    /**
-    * get the length of buffer. empty if zero.
-    * @remark assert length() is not negative.
-    */
-    virtual int length();
-    /**
-    * get the buffer bytes.
-    * @return the bytes, NULL if empty.
-    */
-    virtual char* bytes();
-    /**
-    * erase size of bytes from begin.
-    * @param size to erase size of bytes. 
-    *       clear if size greater than or equals to length()
-    * @remark ignore size is not positive.
-    */
-    virtual void erase(int size);
-    /**
-    * append specified bytes to buffer.
-    * @param size the size of bytes
-    * @remark assert size is positive.
-    */
-    virtual void append(const char* bytes, int size);
-};
+#include <srs_kernel_buffer.hpp>
 
 
 #ifdef SRS_PERF_MERGED_READ
 #ifdef SRS_PERF_MERGED_READ
 /**
 /**