baseparse: use _set_duration to configure duration update interval

... as it logically belongs there as one or the other; either subclass
can provide a duration, or an estimate must be made (reguarly updated).

diff --git a/gst/audioparsers/gstaacparse.c b/gst/audioparsers/gstaacparse.c
index 50cb9e4..62e79ba 100644 (file)
--- a/gst/audioparsers/gstaacparse.c
+++ b/gst/audioparsers/gstaacparse.c
@@ -465,7 +465,7 @@ gst_aacparse_detect_stream (GstAacParse * aacparse,
     aacparse->channels = ((data[2] & 0x01) << 2) | ((data[3] & 0xc0) >> 6);
 
     gst_base_parse_set_frame_props (GST_BASE_PARSE (aacparse),
-        aacparse->sample_rate, 1024, 50);
+        aacparse->sample_rate, 1024);
 
     GST_DEBUG ("ADTS: samplerate %d, channels %d, objtype %d",
         aacparse->sample_rate, aacparse->channels, aacparse->object_type);

diff --git a/gst/audioparsers/gstac3parse.c b/gst/audioparsers/gstac3parse.c
index ddfe9f5..5a3913e 100644 (file)
--- a/gst/audioparsers/gstac3parse.c
+++ b/gst/audioparsers/gstac3parse.c
@@ -465,7 +465,7 @@ gst_ac3_parse_parse_frame (GstBaseParse * parse, GstBuffer * buf)
     ac3parse->sample_rate = rate;
     ac3parse->channels = chans;
 
-    gst_base_parse_set_frame_props (parse, rate, 256 * blocks, 50);
+    gst_base_parse_set_frame_props (parse, rate, 256 * blocks);
   }
 
   return GST_FLOW_OK;

diff --git a/gst/audioparsers/gstamrparse.c b/gst/audioparsers/gstamrparse.c
index 313c74d..1090708 100644 (file)
--- a/gst/audioparsers/gstamrparse.c
+++ b/gst/audioparsers/gstamrparse.c
@@ -237,7 +237,7 @@ gst_amrparse_sink_setcaps (GstBaseParse * parse, GstCaps * caps)
   }
 
   amrparse->need_header = FALSE;
-  gst_base_parse_set_frame_props (GST_BASE_PARSE (amrparse), 50, 1, 50);
+  gst_base_parse_set_frame_props (GST_BASE_PARSE (amrparse), 50, 1);
   gst_amrparse_set_src_caps (amrparse);
   return TRUE;
 }
@@ -306,7 +306,7 @@ gst_amrparse_check_valid_frame (GstBaseParse * parse,
     if (dsize >= AMR_MIME_HEADER_SIZE &&
         gst_amrparse_parse_header (amrparse, data, skipsize)) {
       amrparse->need_header = FALSE;
-      gst_base_parse_set_frame_props (GST_BASE_PARSE (amrparse), 50, 1, 50);
+      gst_base_parse_set_frame_props (GST_BASE_PARSE (amrparse), 50, 1);
     } else {
       GST_WARNING ("media doesn't look like a AMR format");
     }

diff --git a/gst/audioparsers/gstbaseparse.c b/gst/audioparsers/gstbaseparse.c
index 73da287..c73d096 100644 (file)
--- a/gst/audioparsers/gstbaseparse.c
+++ b/gst/audioparsers/gstbaseparse.c
@@ -2106,11 +2106,13 @@ gst_base_parse_sink_activate_pull (GstPad * sinkpad, gboolean active)
  * @duration: duration value.
  *
  * Sets the duration of the currently playing media. Subclass can use this
- * when it notices a change in the media duration.
- */
+ * when it is able to determine duration and/or notices a change in the media
+ * duration.  Alternatively, if @interval is non-zero (default), then stream
+ * duration is determined based on estimated bitrate, and updated every @interval
+ * frames. */
 void
 gst_base_parse_set_duration (GstBaseParse * parse,
-    GstFormat fmt, gint64 duration)
+    GstFormat fmt, gint64 duration, gint interval)
 {
   g_return_if_fail (parse != NULL);
 
@@ -2126,6 +2128,14 @@ gst_base_parse_set_duration (GstBaseParse * parse,
   parse->priv->duration = duration;
   parse->priv->duration_fmt = fmt;
   GST_DEBUG_OBJECT (parse, "set duration: %" G_GINT64_FORMAT, duration);
+  if (fmt == GST_FORMAT_TIME && GST_CLOCK_TIME_IS_VALID (duration)) {
+    if (interval != 0) {
+      GST_DEBUG_OBJECT (parse, "valid duration provided, disabling estimate");
+      interval = 0;
+    }
+  }
+  GST_DEBUG_OBJECT (parse, "set update interval: %d", interval);
+  parse->priv->update_interval = interval;
   GST_BASE_PARSE_UNLOCK (parse);
 }
 
@@ -2195,28 +2205,23 @@ gst_base_parse_set_passthrough (GstBaseParse * parse, gboolean passthrough)
  * @parse: the #GstBaseParse to set
  * @fps_num: frames per second (numerator).
  * @fps_den: frames per second (denominator).
- * @interval: duration update interval in frames.
  *
  * If frames per second is configured, parser can take care of buffer duration
- * and timestamping. If #interval is non-zero (default), then stream duration
- * is determined based on frame and byte counts, and updated every #interval
- * frames.
+ * and timestamping.
  */
 void
 gst_base_parse_set_frame_props (GstBaseParse * parse, guint fps_num,
-    guint fps_den, gint interval)
+    guint fps_den)
 {
   g_return_if_fail (parse != NULL);
 
   GST_BASE_PARSE_LOCK (parse);
   parse->priv->fps_num = fps_num;
   parse->priv->fps_den = fps_den;
-  parse->priv->update_interval = interval;
   if (!fps_num || !fps_den) {
     GST_DEBUG_OBJECT (parse, "invalid fps (%d/%d), ignoring parameters",
         fps_num, fps_den);
     fps_num = fps_den = 0;
-    interval = 0;
     parse->priv->frame_duration = GST_CLOCK_TIME_NONE;
   } else {
     parse->priv->frame_duration =
@@ -2225,7 +2230,6 @@ gst_base_parse_set_frame_props (GstBaseParse * parse, guint fps_num,
   }
   GST_LOG_OBJECT (parse, "set fps: %d/%d => duration: %" G_GINT64_FORMAT " ms",
       fps_num, fps_den, parse->priv->frame_duration / GST_MSECOND);
-  GST_LOG_OBJECT (parse, "set update interval: %d", interval);
   GST_BASE_PARSE_UNLOCK (parse);
 }
 

diff --git a/gst/audioparsers/gstbaseparse.h b/gst/audioparsers/gstbaseparse.h
index 958e38c..8d913b1 100644 (file)
--- a/gst/audioparsers/gstbaseparse.h
+++ b/gst/audioparsers/gstbaseparse.h
@@ -284,8 +284,7 @@ GstFlowReturn gst_base_parse_push_buffer (GstBaseParse *parse,
                                           GstBuffer *buffer);
 
 void gst_base_parse_set_duration (GstBaseParse *parse,
-                                  GstFormat fmt,
-                                  gint64 duration);
+                                  GstFormat fmt, gint64 duration, gint interval);
 
 void gst_base_parse_set_seek (GstBaseParse * parse,
                               GstBaseParseSeekable seek, guint bitrate);
@@ -295,7 +294,7 @@ void gst_base_parse_set_min_frame_size (GstBaseParse *parse,
 void gst_base_parse_set_passthrough (GstBaseParse * parse, gboolean passthrough);
 
 void gst_base_parse_set_frame_props (GstBaseParse * parse, guint fps_num,
-                                     guint fps_den, gint interval);
+                                     guint fps_den);
 
 gboolean gst_base_parse_get_sync (GstBaseParse * parse);
 

diff --git a/gst/audioparsers/gstflacparse.c b/gst/audioparsers/gstflacparse.c
index 0670fc0..855c95e 100644 (file)
--- a/gst/audioparsers/gstflacparse.c
+++ b/gst/audioparsers/gstflacparse.c
@@ -995,7 +995,7 @@ gst_flac_parse_handle_streaminfo (GstFlacParse * flacparse, GstBuffer * buffer)
   if (flacparse->total_samples)
     gst_base_parse_set_duration (GST_BASE_PARSE (flacparse), GST_FORMAT_TIME,
         GST_FRAMES_TO_CLOCK_TIME (flacparse->total_samples,
-            flacparse->samplerate));
+            flacparse->samplerate), 0);
 
   GST_DEBUG_OBJECT (flacparse, "STREAMINFO:\n"
       "\tmin/max blocksize: %u/%u,\n"