support config the audio overflow ratio.

winlin
Commit 0319e85f9912e712e7ffcb0aba2899f3801c1333 0319e85f 1 parent e319da33
trunk/conf/full.conf
trunk/src/app/srs_app_config.cpp
trunk/src/app/srs_app_config.hpp
trunk/src/app/srs_app_hls.cpp
trunk/src/app/srs_app_hls.hpp
--- a/trunk/conf/full.conf
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+++ b/trunk/conf/full.conf
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@@ -498,6 +498,12 @@ vhost with-hls.srs.com {
         # @see https://github.com/winlinvip/simple-rtmp-server/issues/304#issuecomment-74000081
         # default: 1.5
         hls_td_ratio    1.5;
+        # the audio overflow ratio.
+        # for pure audio, the duration to reap the segment.
+        # for example, the hls_fragment is 10s, hsl_aof_ratio is 2.0,
+        # the segemnt will reap to 20s for pure audio.
+        # default: 2.0
+        hls_aof_ratio   2.0;
         # the hls window in seconds, the number of ts in m3u8.
         # default: 60
         hls_window      60;
--- a/trunk/src/app/srs_app_config.cpp
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+++ b/trunk/src/app/srs_app_config.cpp
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@@ -1481,7 +1481,7 @@ int SrsConfig::check_config()
                 for (int j = 0; j < (int)conf->directives.size(); j++) {
                     string m = conf->at(j)->name.c_str();
                     if (m != "enabled" && m != "hls_entry_prefix" && m != "hls_path" && m != "hls_fragment" && m != "hls_window" && m != "hls_on_error"
-                        && m != "hls_storage" && m != "hls_mount" && m != "hls_td_ratio" && m != "hls_acodec" && m != "hls_vcodec"
+                        && m != "hls_storage" && m != "hls_mount" && m != "hls_td_ratio" && m != "hls_aof_ratio" && m != "hls_acodec" && m != "hls_vcodec"
                     ) {
                         ret = ERROR_SYSTEM_CONFIG_INVALID;
                         srs_error("unsupported vhost hls directive %s, ret=%d", m.c_str(), ret);
@@ -3202,7 +3202,24 @@ double SrsConfig::get_hls_td_ratio(string vhost)
     if (!conf) {
         return SRS_CONF_DEFAULT_HLS_TD_RATIO;
     }
+    
+    return ::atof(conf->arg0().c_str());
+}
+double SrsConfig::get_hls_aof_ratio(string vhost)
+{
+    SrsConfDirective* hls = get_hls(vhost);
+    
+    if (!hls) {
+        return SRS_CONF_DEFAULT_HLS_AOF_RATIO;
+    }
+    
+    SrsConfDirective* conf = hls->get("hls_aof_ratio");
+    
+    if (!conf) {
+        return SRS_CONF_DEFAULT_HLS_AOF_RATIO;
+    }
+    
     return ::atof(conf->arg0().c_str());
 }
--- a/trunk/src/app/srs_app_config.hpp
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+++ b/trunk/src/app/srs_app_config.hpp
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@@ -48,6 +48,7 @@ CONNECTION WITH THE SOFTWARE OR THE USE OR OTHER DEALINGS IN THE SOFTWARE.
 #define SRS_CONF_DEFAULT_HLS_PATH "./objs/nginx/html"
 #define SRS_CONF_DEFAULT_HLS_FRAGMENT 10
 #define SRS_CONF_DEFAULT_HLS_TD_RATIO 1.5
+#define SRS_CONF_DEFAULT_HLS_AOF_RATIO 2.0
 #define SRS_CONF_DEFAULT_HLS_WINDOW 60
 #define SRS_CONF_DEFAULT_HLS_ON_ERROR_IGNORE "ignore"
 #define SRS_CONF_DEFAULT_HLS_ON_ERROR_DISCONNECT "disconnect"
@@ -875,15 +876,17 @@ public:
     virtual std::string         get_hls_path(std::string vhost);
     /**
     * get the hls fragment time, in seconds.
-    * a fragment is a ts file.
     */
     virtual double              get_hls_fragment(std::string vhost);
     /**
     * get the hls td(target duration) ratio.
-    * a fragment is a ts file.
     */
     virtual double              get_hls_td_ratio(std::string vhost);
     /**
+     * get the hls aof(audio overflow) ratio.
+     */
+    virtual double              get_hls_aof_ratio(std::string vhost);
+    /**
     * get the hls window time, in seconds.
     * a window is a set of ts, the ts collection in m3u8.
     * @remark SRS will delete the ts exceed the window.
--- a/trunk/src/app/srs_app_hls.cpp
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+++ b/trunk/src/app/srs_app_hls.cpp
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@@ -168,6 +168,7 @@ SrsHlsMuxer::SrsHlsMuxer()
     req = NULL;
     handler = NULL;
     hls_fragment = hls_window = 0;
+    hls_aof_ratio = 1.0;
     target_duration = 0;
     _sequence_no = 0;
     current = NULL;
@@ -203,7 +204,7 @@ int SrsHlsMuxer::sequence_no()
     return _sequence_no;
 }
-int SrsHlsMuxer::update_config(SrsRequest* r, string entry_prefix, string path, int fragment, int window)
+int SrsHlsMuxer::update_config(SrsRequest* r, string entry_prefix, string path, int fragment, int window, double aof_ratio)
 {
     int ret = ERROR_SUCCESS;
@@ -213,11 +214,12 @@ int SrsHlsMuxer::update_config(SrsRequest* r, string entry_prefix, string path, 
     hls_entry_prefix = entry_prefix;
     hls_path = path;
     hls_fragment = fragment;
+    hls_aof_ratio = aof_ratio;
     hls_window = window;
     // we always keep the target duration increasing.
     int max_td = srs_max(target_duration, (int)(fragment * _srs_config->get_hls_td_ratio(r->vhost)));
-    srs_info("hls update target duration %d=>%d", target_duration, max_td);
+    srs_info("hls update target duration %d=>%d, aof=%.2f", target_duration, max_td, aof_ratio);
     target_duration = max_td;
     std::string storage = _srs_config->get_hls_storage(r->vhost);
@@ -345,7 +347,7 @@ bool SrsHlsMuxer::is_segment_overflow()
 bool SrsHlsMuxer::is_segment_absolutely_overflow()
 {
     srs_assert(current);
-    return current->duration >= 2 * hls_fragment;
+    return current->duration >= hls_aof_ratio * hls_fragment;
 }
 int SrsHlsMuxer::update_acodec(SrsCodecAudio ac)
@@ -676,12 +678,14 @@ int SrsHlsCache::on_publish(SrsHlsMuxer* muxer, SrsRequest* req, int64_t segment
     std::string entry_prefix = _srs_config->get_hls_entry_prefix(vhost);
     // get the hls path config
     std::string hls_path = _srs_config->get_hls_path(vhost);
+    // the audio overflow, for pure audio to reap segment.
+    double hls_aof_ratio = _srs_config->get_hls_aof_ratio(vhost);
     // TODO: FIXME: support load exists m3u8, to continue publish stream.
     // for the HLS donot requires the EXT-X-MEDIA-SEQUENCE be monotonically increase.
     // open muxer
-    if ((ret = muxer->update_config(req, entry_prefix, hls_path, hls_fragment, hls_window)) != ERROR_SUCCESS) {
+    if ((ret = muxer->update_config(req, entry_prefix, hls_path, hls_fragment, hls_window, hls_aof_ratio)) != ERROR_SUCCESS) {
         srs_error("m3u8 muxer update config failed. ret=%d", ret);
         return ret;
     }
@@ -737,17 +741,13 @@ int SrsHlsCache::write_audio(SrsAvcAacCodec* codec, SrsHlsMuxer* muxer, int64_t 
         }
     }
-    // cache->audio will be free in flush_audio
-    // so we must check whether it's null ptr.
-    if (!cache->audio) {
-        return ret;
-    }
-
     // TODO: config it.
     // in ms, audio delay to flush the audios.
     int64_t audio_delay = SRS_CONF_DEFAULT_AAC_DELAY;
     // flush if audio delay exceed
-    if (pts - cache->audio->start_pts > audio_delay * 90) {
+    // cache->audio will be free in flush_audio
+    // so we must check whether it's null ptr.
+    if (cache->audio && pts - cache->audio->start_pts > audio_delay * 90) {
         if ((ret = muxer->flush_audio(cache)) != ERROR_SUCCESS) {
             return ret;
         }
@@ -761,7 +761,7 @@ int SrsHlsCache::write_audio(SrsAvcAacCodec* codec, SrsHlsMuxer* muxer, int64_t 
     // @see https://github.com/winlinvip/simple-rtmp-server/issues/151
     // 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 (cache->audio && muxer->is_segment_absolutely_overflow()) {
         if ((ret = reap_segment("audio", muxer, cache->audio->pts)) != ERROR_SUCCESS) {
             return ret;
         }
--- a/trunk/src/app/srs_app_hls.hpp
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+++ b/trunk/src/app/srs_app_hls.hpp
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@@ -169,6 +169,7 @@ private:
 private:
     std::string hls_entry_prefix;
     std::string hls_path;
+    double hls_aof_ratio;
     int hls_fragment;
     int hls_window;
 private:
@@ -208,7 +209,7 @@ public:
     /**
     * when publish, update the config for muxer.
     */
-    virtual int update_config(SrsRequest* r, std::string entry_prefix, std::string path, int fragment, int window);
+    virtual int update_config(SrsRequest* r, std::string entry_prefix, std::string path, int fragment, int window, double aof_ratio);
     /**
     * open a new segment(a new ts file),
     * @param segment_start_dts use to calc the segment duration,