release 1.0r2

 #Simple-RTMP-Server
 
+SRS/1.0，开发代号：[HuKaiqun](https://github.com/winlinvip/simple-rtmp-server/wiki/v1_CN_Product#release10)
+
 SRS定位是运营级的互联网直播服务器集群，追求更好的概念完整性和最简单实现的代码。
 
 下载发布版(github): 
-[Centos6-x86_64](http://winlinvip.github.io/srs.release/releases/files/SRS-CentOS6-x86_64-1.0.21.zip) 
+[Centos6-x86_64](http://winlinvip.github.io/srs.release/releases/files/SRS-CentOS6-x86_64-1.0.27.zip) 
 其他[more...](http://winlinvip.github.io/srs.release/releases/) <br/>
 下载发布版(国内阿里云镜像): 
-[Centos6-x86_64](http://www.ossrs.net/srs.release/releases/files/SRS-CentOS6-x86_64-1.0.21.zip) 
+[Centos6-x86_64](http://www.ossrs.net/srs.release/releases/files/SRS-CentOS6-x86_64-1.0.27.zip) 
 其他[more...](http://www.ossrs.net/srs.release/releases/)<br/>
 QQ群: 365936885, by wenjie <br/>
 同类产品：[BLS](https://github.com/wenjiegit/Bull-Live-Server)/[BLE](https://github.com/wenjiegit/Bull-Live-Encoder), [NGINX-RTMP](https://github.com/arut/nginx-rtmp-module), [CRTMPD](http://www.rtmpd.com/), [RED5](http://www.red5.org/), [WOWZA](http://www.wowza.com/), [FMS/AMS](http://www.adobe.com/products/adobe-media-server-standard.html)
@@ -362,6 +364,7 @@ Supported operating systems and hardware:
 [2.0dev branch](https://github.com/winlinvip/simple-rtmp-server/tree/master).
 
 ## Releases
+* 2015-02-12, [Release v1.0r2](https://github.com/winlinvip/simple-rtmp-server/releases/tag/1.0r2), bug fixed, 1.0.27, 59507 lines.<br/>
 * 2015-01-15, [Release v1.0r1](https://github.com/winlinvip/simple-rtmp-server/releases/tag/1.0r1), bug fixed, 1.0.21, 59472 lines.<br/>
 * 2014-12-05, [Release v1.0](https://github.com/winlinvip/simple-rtmp-server/releases/tag/1.0), all bug fixed, 1.0.10, 59391 lines.<br/>
 * 2014-10-09, [Release v1.0-beta](https://github.com/winlinvip/simple-rtmp-server/releases/tag/1.0.beta), all bug fixed, 1.0.0, 59316 lines.<br/>
@@ -384,6 +387,14 @@ Supported operating systems and hardware:
 * 2013-10-17, Created.<br/>
 
 ## History
+
+* <strong>v1.0, 2015-02-12, [1.0r2 release(1.0.27)](https://github.com/winlinvip/simple-rtmp-server/releases/tag/1.0r2) released. 59507 lines.</strong>
+* v1.0, 2015-02-11, dev code HuKaiqun for 1.0.27.
+* v1.0, 2015-02-10, for [#310](https://github.com/winlinvip/simple-rtmp-server/issues/310), the aac profile must be object plus one. 1.0.26
+* v1.0, 2015-01-25, hotfix [#268](https://github.com/winlinvip/simple-rtmp-server/issues/268), refine the pcr start at 0, dts/pts plus delay. 1.0.25
+* v1.0, 2015-01-25, hotfix [#151](https://github.com/winlinvip/simple-rtmp-server/issues/151), refine pcr=dts-800ms and use dts/pts directly. 1.0.24
+* v1.0, 2015-01-23, hotfix [#151](https://github.com/winlinvip/simple-rtmp-server/issues/151), use absolutely overflow to make jwplayer happy. 1.0.23
+* v1.0, 2015-01-17, hotfix [#290](https://github.com/winlinvip/simple-rtmp-server/issues/290), use iformat only for rtmp input. 1.0.22
 * <strong>v1.0, 2015-01-15, [1.0r1 release(1.0.21)](https://github.com/winlinvip/simple-rtmp-server/releases/tag/1.0r1) released. 59472 lines.</strong>
 * v1.0, 2015-01-08, hotfix [#281](https://github.com/winlinvip/simple-rtmp-server/issues/281), fix hls bug ignore type-9 send aud. 1.0.20
 * v1.0, 2015-01-03, hotfix to remove the pageUrl for http callback. 1.0.19

@@ -247,6 +247,13 @@ int SrsAvcAacCodec::audio_aac_demux(char* data, int size, SrsCodecSample* sample
             return ret;
         }
         
+        // 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
+        // so the aac_profile should plus 1, not minus 1, and nginx-rtmp used it to 
+        // downcast aac SSR to LC.
+        // @see https://github.com/winlinvip/simple-rtmp-server/issues/310
+        // TODO: FIXME: fix the following in future version.
         // aac_profile = audioObjectType - 1
         aac_profile--;
         

@@ -222,7 +222,7 @@ int SrsEdgeIngester::connect_app(string ep_server, string ep_port)
     // @see https://github.com/winlinvip/simple-rtmp-server/issues/147
     SrsAmf0Object* data = req->args;
     data->set("srs_sig", SrsAmf0Any::str(RTMP_SIG_SRS_KEY));
-    data->set("srs_server", SrsAmf0Any::str(RTMP_SIG_SRS_KEY" "RTMP_SIG_SRS_VERSION" ("RTMP_SIG_SRS_URL_SHORT")"));
+    data->set("srs_server", SrsAmf0Any::str(RTMP_SIG_SRS_SERVER));
     data->set("srs_license", SrsAmf0Any::str(RTMP_SIG_SRS_LICENSE));
     data->set("srs_role", SrsAmf0Any::str(RTMP_SIG_SRS_ROLE));
     data->set("srs_url", SrsAmf0Any::str(RTMP_SIG_SRS_URL));
@@ -654,7 +654,7 @@ int SrsEdgeForwarder::connect_app(string ep_server, string ep_port)
     // @see https://github.com/winlinvip/simple-rtmp-server/issues/147
     SrsAmf0Object* data = req->args;
     data->set("srs_sig", SrsAmf0Any::str(RTMP_SIG_SRS_KEY));
-    data->set("srs_server", SrsAmf0Any::str(RTMP_SIG_SRS_KEY" "RTMP_SIG_SRS_VERSION" ("RTMP_SIG_SRS_URL_SHORT")"));
+    data->set("srs_server", SrsAmf0Any::str(RTMP_SIG_SRS_SERVER));
     data->set("srs_license", SrsAmf0Any::str(RTMP_SIG_SRS_LICENSE));
     data->set("srs_role", SrsAmf0Any::str(RTMP_SIG_SRS_ROLE));
     data->set("srs_url", SrsAmf0Any::str(RTMP_SIG_SRS_URL));

@@ -209,6 +209,13 @@ int SrsFFMPEG::initialize_transcode(SrsConfDirective* engine)
         return ret;
     }
     
+    // for not rtmp input, donot append the iformat,
+    // for example, "-f flv" before "-i udp://192.168.1.252:2222"
+    // @see https://github.com/winlinvip/simple-rtmp-server/issues/290
+    if (input.find("rtmp://") != 0) {
+        iformat = "";
+    }
+    
     return ret;
 }
 

@@ -336,7 +336,7 @@ int SrsForwarder::connect_app(string ep_server, string ep_port)
     // @see https://github.com/winlinvip/simple-rtmp-server/issues/147
     SrsAmf0Object* data = req->args;
     data->set("srs_sig", SrsAmf0Any::str(RTMP_SIG_SRS_KEY));
-    data->set("srs_server", SrsAmf0Any::str(RTMP_SIG_SRS_KEY" "RTMP_SIG_SRS_VERSION" ("RTMP_SIG_SRS_URL_SHORT")"));
+    data->set("srs_server", SrsAmf0Any::str(RTMP_SIG_SRS_SERVER));
     data->set("srs_license", SrsAmf0Any::str(RTMP_SIG_SRS_LICENSE));
     data->set("srs_role", SrsAmf0Any::str(RTMP_SIG_SRS_ROLE));
     data->set("srs_url", SrsAmf0Any::str(RTMP_SIG_SRS_URL));

@@ -78,7 +78,9 @@ using namespace std;
 
 // @see: NGX_RTMP_HLS_DELAY, 
 // 63000: 700ms, ts_tbn=90000
-#define SRS_AUTO_HLS_DELAY 63000
+// 72000: 800ms, ts_tbn=90000
+// @see https://github.com/winlinvip/simple-rtmp-server/issues/151#issuecomment-71352511
+#define SRS_AUTO_HLS_DELAY 72000
 
 // the mpegts header specifed the video/audio pid.
 #define TS_VIDEO_PID 256
@@ -237,7 +239,8 @@ public:
                     p[-1] |= 0x20; // Both Adaption and Payload
                     *p++ = 7;    // size
                     *p++ = 0x50; // random access + PCR
-                    p = write_pcr(p, frame->dts - SRS_AUTO_HLS_DELAY);
+                    // about the pcr, read https://github.com/winlinvip/simple-rtmp-server/issues/151#issuecomment-71352511
+                    p = write_pcr(p, frame->dts);
                 }
                 
                 // PES header
@@ -368,10 +371,12 @@ private:
     }
     static char* write_pcr(char* p, int64_t pcr)
     {
-        // the pcr=dts-delay
-        // and the pcr maybe negative
+        // the pcr=dts-delay, where dts = frame->dts + delay
+        // and the pcr should never be negative
         // @see https://github.com/winlinvip/simple-rtmp-server/issues/268
-        int64_t v = srs_max(0, pcr);
+        srs_assert(pcr >= 0);
+        
+        int64_t v = pcr;
         
         *p++ = (char) (v >> 25);
         *p++ = (char) (v >> 17);
@@ -655,10 +660,15 @@ int SrsHlsMuxer::on_sequence_header()
 bool SrsHlsMuxer::is_segment_overflow()
 {
     srs_assert(current);
-    
     return current->duration >= hls_fragment;
 }
 
+bool SrsHlsMuxer::is_segment_absolutely_overflow()
+{
+    srs_assert(current);
+    return current->duration >= 2 * hls_fragment;
+}
+
 int SrsHlsMuxer::flush_audio(SrsMpegtsFrame* af, SrsBuffer* ab)
 {
     int ret = ERROR_SUCCESS;
@@ -1090,7 +1100,9 @@ int SrsHlsCache::write_audio(SrsAvcAacCodec* codec, SrsHlsMuxer* muxer, int64_t
     // pure audio again for audio disabled.
     // so we reap event when the audio incoming when segment overflow.
     // @see https://github.com/winlinvip/simple-rtmp-server/issues/151
-    if (muxer->is_segment_overflow()) {
+    // we use absolutely overflow of segment to make jwplayer/ffplay happy
+    // @see https://github.com/winlinvip/simple-rtmp-server/issues/151#issuecomment-71155184
+    if (muxer->is_segment_absolutely_overflow()) {
         if ((ret = reap_segment("audio", muxer, af->pts)) != ERROR_SUCCESS) {
             return ret;
         }

@@ -192,10 +192,16 @@ public:
     virtual int segment_open(int64_t segment_start_dts);
     virtual int on_sequence_header();
     /**
-    * whether video overflow,
-    * that is whether the current segment duration >= the segment in config
+    * whether segment overflow,
+    * that is whether the current segment duration>=(the segment in config)
     */
     virtual bool is_segment_overflow();
+    /**
+    * whether segment absolutely overflow, for pure audio to reap segment,
+    * that is whether the current segment duration>=2*(the segment in config)
+    * @see https://github.com/winlinvip/simple-rtmp-server/issues/151#issuecomment-71155184
+    */
+    virtual bool is_segment_absolutely_overflow();
     virtual int flush_audio(SrsMpegtsFrame* af, SrsBuffer* ab);
     virtual int flush_video(SrsMpegtsFrame* af, SrsBuffer* ab, SrsMpegtsFrame* vf, SrsBuffer* vb);
     /**

@@ -956,7 +956,7 @@ int SrsSource::on_meta_data(SrsMessage* msg, SrsOnMetaDataPacket* metadata)
     }
     
     // add server info to metadata
-    metadata->metadata->set("server", SrsAmf0Any::str(RTMP_SIG_SRS_KEY" "RTMP_SIG_SRS_VERSION" ("RTMP_SIG_SRS_URL_SHORT")"));
+    metadata->metadata->set("server", SrsAmf0Any::str(RTMP_SIG_SRS_SERVER));
     metadata->metadata->set("primary", SrsAmf0Any::str(RTMP_SIG_SRS_PRIMARY));
     metadata->metadata->set("authors", SrsAmf0Any::str(RTMP_SIG_SRS_AUTHROS));
     

@@ -31,9 +31,11 @@ CONNECTION WITH THE SOFTWARE OR THE USE OR OTHER DEALINGS IN THE SOFTWARE.
 // current release version
 #define VERSION_MAJOR       1
 #define VERSION_MINOR       0
-#define VERSION_REVISION    21
+#define VERSION_REVISION    27
+
 // server info.
 #define RTMP_SIG_SRS_KEY "SRS"
+#define RTMP_SIG_SRS_CODE "HuKaiqun"
 #define RTMP_SIG_SRS_ROLE "origin/edge server"
 #define RTMP_SIG_SRS_NAME RTMP_SIG_SRS_KEY"(Simple RTMP Server)"
 #define RTMP_SIG_SRS_URL_SHORT "github.com/winlinvip/simple-rtmp-server"
@@ -49,6 +51,7 @@ CONNECTION WITH THE SOFTWARE OR THE USE OR OTHER DEALINGS IN THE SOFTWARE.
 #define RTMP_SIG_SRS_RELEASE "https://github.com/winlinvip/simple-rtmp-server/tree/1.0release"
 #define RTMP_SIG_SRS_HTTP_SERVER "https://github.com/winlinvip/simple-rtmp-server/wiki/v1_CN_HTTPServer#feature"
 #define RTMP_SIG_SRS_VERSION __SRS_XSTR(VERSION_MAJOR)"."__SRS_XSTR(VERSION_MINOR)"."__SRS_XSTR(VERSION_REVISION)
+#define RTMP_SIG_SRS_SERVER RTMP_SIG_SRS_KEY"/"RTMP_SIG_SRS_VERSION"("RTMP_SIG_SRS_CODE")"
 
 // internal macros, covert macro values to str,
 // see: read https://gcc.gnu.org/onlinedocs/cpp/Stringification.html#Stringification

@@ -905,7 +905,7 @@ int SrsRtmpServer::response_connect_app(SrsRequest *req, const char* server_ip)
     
     data->set("version", SrsAmf0Any::str(RTMP_SIG_FMS_VER));
     data->set("srs_sig", SrsAmf0Any::str(RTMP_SIG_SRS_KEY));
-    data->set("srs_server", SrsAmf0Any::str(RTMP_SIG_SRS_KEY" "RTMP_SIG_SRS_VERSION" ("RTMP_SIG_SRS_URL_SHORT")"));
+    data->set("srs_server", SrsAmf0Any::str(RTMP_SIG_SRS_SERVER));
     data->set("srs_license", SrsAmf0Any::str(RTMP_SIG_SRS_LICENSE));
     data->set("srs_role", SrsAmf0Any::str(RTMP_SIG_SRS_ROLE));
     data->set("srs_url", SrsAmf0Any::str(RTMP_SIG_SRS_URL));