Fix #547, support HLS audio in TS. 3.0.22

ossrs · Apr 16, 2017 · a98c9e0 · a98c9e0
1 parent 6ee85ae
commit a98c9e0
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Showing 5 changed files with 30 additions and 4 deletions.
diff --git a/README.md b/README.md
@@ -174,7 +174,7 @@ Please select your language:
 - [x] [experiment] Support push POST FLV over HTTP, please read [wiki]([CN][v2_CN_Streamer2], [EN][v2_EN_Streamer2]).
 - [x] [experiment] Support multiple processes by [dolphin][srs-dolphin] or [oryx][oryx].
 - [x] [experiment] Support [mgmt console][console], please read [srs-ngb][srs-ngb].
-- [ ] Enhanced HLS audio-only use aac instead of ts.
+- [x] Enhanced HLS audio-only use ts, see [#547][bug #547].
 - [ ] Enhanced forward with vhost and url variables.
 - [ ] Support source or idle stream cleanup.
 - [ ] Support origin cluster, please read [#464][bug #464], [RTMP 302][bug #92].
@@ -186,6 +186,7 @@ Please select your language:
 
 ### V3 changes
 
+* v3.0, 2017-04-16, Fix [#547][bug #547], support HLS audio in TS. 3.0.22
 * v3.0, 2017-03-26, Fix [#820][bug #820], extract service for modules. 3.0.21
 * v3.0, 2017-03-02, Fix [#786][bug #786], simply don't reuse object. 3.0.20
 * v3.0, 2017-03-01, For [#110][bug #110], refine thread object. 3.0.19
@@ -1400,6 +1401,7 @@ Winlin
 [bug #738]: https://github.com/ossrs/srs/issues/738
 [bug #786]: https://github.com/ossrs/srs/issues/786
 [bug #820]: https://github.com/ossrs/srs/issues/820
+[bug #547]: https://github.com/ossrs/srs/issues/547
 [bug #xxxxxxxxxxxxx]: https://github.com/ossrs/srs/issues/xxxxxxxxxxxxx
 
 [exo #828]: https://github.com/google/ExoPlayer/pull/828

diff --git a/trunk/src/app/srs_app_hls.cpp b/trunk/src/app/srs_app_hls.cpp
@@ -969,6 +969,9 @@ SrsHls::SrsHls()
     disposable = false;
     last_update_time = 0;
 
+    previous_audio_dts = 0;
+    aac_samples = 0;
+
     jitter = new SrsRtmpJitter();
     controller = new SrsHlsController();
 
@@ -1115,8 +1118,22 @@ int SrsHls::on_audio(SrsSharedPtrMessage* shared_audio, SrsFormat* format)
         return ret;
     }
 
-    // the dts calc from rtmp/flv header.
-    int64_t dts = audio->timestamp * 90;
+    // Reset the aac samples counter when DTS jitter.
+    if (previous_audio_dts > audio->timestamp) {
+        previous_audio_dts = audio->timestamp;
+        aac_samples = 0;
+    }
+
+    // Use the diff to guess whether the samples is 1024 or 960.
+    int nb_samples_per_frame = 1024;
+    int diff = ::abs((int)(audio->timestamp - previous_audio_dts)) * srs_flv_srates[format->acodec->sound_rate];
+    if (diff > 100 && diff < 950) {
+        nb_samples_per_frame = 960;
+    }
+
+    // Recalc the DTS by the samples of AAC.
+    int64_t dts = 90000 * aac_samples / srs_flv_srates[format->acodec->sound_rate];
+    aac_samples += nb_samples_per_frame;
 
     if ((ret = controller->write_audio(format->audio, dts)) != ERROR_SUCCESS) {
         srs_error("hls cache write audio failed. ret=%d", ret);

diff --git a/trunk/src/app/srs_app_hls.hpp b/trunk/src/app/srs_app_hls.hpp
@@ -298,6 +298,12 @@ class SrsHls
     bool enabled;
     bool disposable;
     int64_t last_update_time;
+private:
+    // If the diff=dts-previous_audio_dts is about 23,
+    // that's the AAC samples is 1024, and we use the samples to calc the dts.
+    int64_t previous_audio_dts;
+    // The total aac samples.
+    uint64_t aac_samples;
 private:
     SrsOriginHub* hub;
     SrsRtmpJitter* jitter;

diff --git a/trunk/src/core/srs_core.hpp b/trunk/src/core/srs_core.hpp
@@ -27,7 +27,7 @@
 // current release version
 #define VERSION_MAJOR       3
 #define VERSION_MINOR       0
-#define VERSION_REVISION    21
+#define VERSION_REVISION    22
 
 // generated by configure, only macros.
 #include <srs_auto_headers.hpp>

diff --git a/trunk/src/kernel/srs_kernel_codec.hpp b/trunk/src/kernel/srs_kernel_codec.hpp
@@ -183,6 +183,7 @@ enum SrsAudioAacFrameTrait
  *      2 = 22 kHz = 22050 Hz
  *      3 = 44 kHz = 44100 Hz
  * However, we can extends this table.
+ * @remark Use srs_flv_srates to convert it.
  */
 enum SrsAudioSampleRate
 {