gst-libs/gst/audio/gstaudioclock.c: Fix clock name.

Original commit message from CVS:
* gst-libs/gst/audio/gstaudioclock.c: (gst_audio_clock_init),
(gst_audio_clock_new):
Fix clock name.
* gst-libs/gst/audio/gstbaseaudiosink.c:
(gst_base_audio_sink_init), (gst_base_audio_sink_query):
* gst-libs/gst/audio/gstbaseaudiosrc.c: (gst_base_audio_src_init),
(gst_base_audio_src_query), (gst_base_audio_src_get_offset),
(gst_base_audio_src_create):
Improve latency query code.
Use proper clock names.

ChangeLog

@@ -1,3 +1,17 @@
+2007-02-28  Wim Taymans  <wim@fluendo.com>
+
+	* gst-libs/gst/audio/gstaudioclock.c: (gst_audio_clock_init),
+	(gst_audio_clock_new):
+	Fix clock name.
+
+	* gst-libs/gst/audio/gstbaseaudiosink.c:
+	(gst_base_audio_sink_init), (gst_base_audio_sink_query):
+	* gst-libs/gst/audio/gstbaseaudiosrc.c: (gst_base_audio_src_init),
+	(gst_base_audio_src_query), (gst_base_audio_src_get_offset),
+	(gst_base_audio_src_create):
+	Improve latency query code.
+	Use proper clock names.
+
 2007-02-28  Thomas Vander Stichele  <thomas at apestaart dot org>
 
 	* tests/check/generic/states.c: (GST_START_TEST):

gst-libs/gst/audio/gstaudioclock.c

@@ -93,8 +93,6 @@ gst_audio_clock_class_init (GstAudioClockClass * klass)
 static void
 gst_audio_clock_init (GstAudioClock * clock)
 {
-  gst_object_set_name (GST_OBJECT (clock), "GstAudioClock");
-
   clock->last_time = 0;
   GST_OBJECT_FLAG_SET (clock, GST_CLOCK_FLAG_CAN_SET_MASTER);
 }
@@ -116,7 +114,7 @@ gst_audio_clock_new (gchar * name, GstAudioClockGetTimeFunc func,
     gpointer user_data)
 {
   GstAudioClock *aclock =
-      GST_AUDIO_CLOCK (g_object_new (GST_TYPE_AUDIO_CLOCK, NULL));
+      GST_AUDIO_CLOCK (g_object_new (GST_TYPE_AUDIO_CLOCK, "name", name, NULL));
 
   aclock->func = func;
   aclock->user_data = user_data;

gst-libs/gst/audio/gstbaseaudiosink.c

@@ -174,7 +174,7 @@ gst_base_audio_sink_init (GstBaseAudioSink * baseaudiosink,
   baseaudiosink->latency_time = DEFAULT_LATENCY_TIME;
   baseaudiosink->provide_clock = DEFAULT_PROVIDE_CLOCK;
 
-  baseaudiosink->provided_clock = gst_audio_clock_new ("clock",
+  baseaudiosink->provided_clock = gst_audio_clock_new ("GstAudioSinkClock",
       (GstAudioClockGetTimeFunc) gst_base_audio_sink_get_time, baseaudiosink);
 
   GST_BASE_SINK (baseaudiosink)->can_activate_push = TRUE;
@@ -257,6 +257,13 @@ gst_base_audio_sink_query (GstElement * element, GstQuery * query)
 
       GST_DEBUG_OBJECT (basesink, "latency query");
 
+      if (!basesink->ringbuffer || !basesink->ringbuffer->spec.rate) {
+        GST_DEBUG_OBJECT (basesink,
+            "we are not yet negotiated, can't report latency yet");
+        res = FALSE;
+        goto done;
+      }
+
       /* ask parent first, it will do an upstream query for us. */
       if ((res =
               gst_base_sink_query_latency (GST_BASE_SINK_CAST (basesink), &live,
@@ -264,16 +271,19 @@ gst_base_audio_sink_query (GstElement * element, GstQuery * query)
         GstClockTime min_latency, max_latency;
 
         /* we and upstream are both live, adjust the min_latency */
-        if (live && us_live && basesink->ringbuffer
+        if (live && us_live) {
-            && basesink->ringbuffer->spec.rate) {
           GstRingBufferSpec *spec;
 
           spec = &basesink->ringbuffer->spec;
 
-          max_latency =
+          min_latency =
-              spec->segtotal * spec->segsize * GST_SECOND / (spec->rate *
+              gst_util_uint64_scale_int (spec->segtotal * spec->segsize,
-              spec->bytes_per_sample);
+              GST_SECOND, spec->rate * spec->bytes_per_sample);
-          min_latency = MAX (max_latency, min_l);
+          /* we cannot go lower than the buffer size */
+          min_latency = MAX (min_latency, min_l);
+          /* the max latency is the max of the peer, we can delay an infinite
+           * amount of time. */
+          max_latency = max_l;
 
           GST_DEBUG_OBJECT (basesink,
               "peer min %" GST_TIME_FORMAT ", our min latency: %"
@@ -294,6 +304,7 @@ gst_base_audio_sink_query (GstElement * element, GstQuery * query)
       break;
   }
 
+done:
   return res;
 }
 

gst-libs/gst/audio/gstbaseaudiosrc.c

@@ -152,7 +152,7 @@ gst_base_audio_src_init (GstBaseAudioSrc * baseaudiosrc,
    * value based on negotiated format. */
   GST_BASE_SRC (baseaudiosrc)->blocksize = 0;
 
-  baseaudiosrc->clock = gst_audio_clock_new ("clock",
+  baseaudiosrc->clock = gst_audio_clock_new ("GstAudioSrcClock",
       (GstAudioClockGetTimeFunc) gst_base_audio_src_get_time, baseaudiosrc);
 
   /* we are always a live source */
@@ -419,9 +419,11 @@ gst_base_audio_src_query (GstBaseSrc * bsrc, GstQuery * query)
 
       spec = &src->ringbuffer->spec;
 
+      /* we have at least 1 segment of latency */
       min_latency =
           gst_util_uint64_scale_int (spec->segsize, GST_SECOND,
           spec->rate * spec->bytes_per_sample);
+      /* we cannot delay more than the buffersize else we lose data */
       max_latency =
           gst_util_uint64_scale_int (spec->segtotal * spec->segsize, GST_SECOND,
           spec->rate * spec->bytes_per_sample);
@@ -493,11 +495,14 @@ gst_base_audio_src_get_offset (GstBaseAudioSrc * src)
   segdone = g_atomic_int_get (&src->ringbuffer->segdone)
       - src->ringbuffer->segbase;
 
+  GST_DEBUG_OBJECT (src, "reading from %d, we are at %d", readseg, segdone);
+
   /* see how far away it is from the read segment, normally segdone (where new
    * data is written in the ringbuffer) is bigger than readseg (where we are
    * reading). */
   diff = segdone - readseg;
   if (diff >= segtotal) {
+    GST_DEBUG_OBJECT (src, "dropped, align to segment %d", segdone);
     /* sample would be dropped, position to next playable position */
     sample = (segdone - segtotal + 1) * sps;
   }
@@ -574,6 +579,8 @@ gst_base_audio_src_create (GstBaseSrc * bsrc, guint64 offset, guint length,
     GST_WARNING_OBJECT (src,
         "create DISCONT of %" G_GUINT64_FORMAT " samples at sample %"
         G_GUINT64_FORMAT, sample - src->next_sample, sample);
+    GST_ELEMENT_WARNING (src, CORE, CLOCK, (NULL),
+        ("dropped %" G_GUINT64_FORMAT " samples", sample - src->next_sample));
     GST_BUFFER_FLAG_SET (buf, GST_BUFFER_FLAG_DISCONT);
   }