gst-libs/gst/audio/gstbaseaudiosink.c: Answer latency query.

Original commit message from CVS:
* gst-libs/gst/audio/gstbaseaudiosink.c:
(gst_base_audio_sink_class_init), (gst_base_audio_sink_query),
(gst_base_audio_sink_render), (gst_base_audio_sink_callback),
(gst_base_audio_sink_async_play),
(gst_base_audio_sink_change_state):
Answer latency query.
Use configured latency when syncing.
Fix clock slaving.
* gst-libs/gst/audio/gstbaseaudiosrc.c:
(gst_base_audio_src_class_init), (gst_base_audio_src_dispose),
(gst_base_audio_src_query), (gst_base_audio_src_change_state):
Fix possible memleak.
Implement latency query.
Small cleanups.

3 files changed, 215 insertions, 26 deletions

diff --git a/ChangeLog b/ChangeLog
index 00646251..d892a2c6 100644
--- a/ChangeLog
+++ b/ChangeLog
@@ -1,5 +1,23 @@
 2007-02-15  Wim Taymans  <wim@fluendo.com>
 
+	* gst-libs/gst/audio/gstbaseaudiosink.c:
+	(gst_base_audio_sink_class_init), (gst_base_audio_sink_query),
+	(gst_base_audio_sink_render), (gst_base_audio_sink_callback),
+	(gst_base_audio_sink_async_play),
+	(gst_base_audio_sink_change_state):
+	Answer latency query.
+	Use configured latency when syncing.
+	Fix clock slaving.
+
+	* gst-libs/gst/audio/gstbaseaudiosrc.c:
+	(gst_base_audio_src_class_init), (gst_base_audio_src_dispose),
+	(gst_base_audio_src_query), (gst_base_audio_src_change_state):
+	Fix possible memleak.
+	Implement latency query.
+	Small cleanups.
+
+2007-02-15  Wim Taymans  <wim@fluendo.com>
+
 	* ext/alsa/gstalsasink.c: (gst_alsasink_reset):
 	Ignore errors in reset, these are not fatal. They also grab the element
 	lock which is already taking when this function is called. Fixes
diff --git a/gst-libs/gst/audio/gstbaseaudiosink.c b/gst-libs/gst/audio/gstbaseaudiosink.c
index baff3c38..2167a31d 100644
--- a/gst-libs/gst/audio/gstbaseaudiosink.c
+++ b/gst-libs/gst/audio/gstbaseaudiosink.c
@@ -87,6 +87,8 @@ static GstStateChangeReturn gst_base_audio_sink_change_state (GstElement *
     element, GstStateChange transition);
 static gboolean gst_base_audio_sink_activate_pull (GstBaseSink * basesink,
     gboolean active);
+static gboolean gst_base_audio_sink_query (GstElement * element, GstQuery *
+    query);
 
 static GstClock *gst_base_audio_sink_provide_clock (GstElement * elem);
 static GstClockTime gst_base_audio_sink_get_time (GstClock * clock,
@@ -149,6 +151,7 @@ gst_base_audio_sink_class_init (GstBaseAudioSinkClass * klass)
       GST_DEBUG_FUNCPTR (gst_base_audio_sink_change_state);
   gstelement_class->provide_clock =
       GST_DEBUG_FUNCPTR (gst_base_audio_sink_provide_clock);
+  gstelement_class->query = GST_DEBUG_FUNCPTR (gst_base_audio_sink_query);
 
   gstbasesink_class->event = GST_DEBUG_FUNCPTR (gst_base_audio_sink_event);
   gstbasesink_class->preroll = GST_DEBUG_FUNCPTR (gst_base_audio_sink_preroll);
@@ -235,6 +238,62 @@ clock_disabled:
   }
 }
 
+static gboolean
+gst_base_audio_sink_query (GstElement * element, GstQuery * query)
+{
+  gboolean res = FALSE;
+
+  GstBaseAudioSink *basesink = GST_BASE_AUDIO_SINK (element);
+
+  switch (GST_QUERY_TYPE (query)) {
+    case GST_QUERY_LATENCY:
+    {
+      gboolean live, us_live;
+      GstClockTime min_l, max_l;
+
+      GST_DEBUG_OBJECT (basesink, "latency query");
+
+      /* ask parent first, it will do an upstream query for us. */
+      if ((res =
+              gst_base_sink_query_latency (GST_BASE_SINK_CAST (basesink), &live,
+                  &us_live, &min_l, &max_l))) {
+        GstClockTime min_latency, max_latency;
+
+        /* we and upstream are both live, adjust the min_latency */
+        if (live && us_live && basesink->ringbuffer
+            && basesink->ringbuffer->spec.rate) {
+          GstRingBufferSpec *spec;
+
+          spec = &basesink->ringbuffer->spec;
+
+          max_latency =
+              spec->segtotal * spec->segsize * GST_SECOND / (spec->rate *
+              spec->bytes_per_sample);
+          min_latency = MAX (max_latency, min_l);
+
+          GST_DEBUG_OBJECT (basesink,
+              "peer min %" GST_TIME_FORMAT ", our min latency: %"
+              GST_TIME_FORMAT, GST_TIME_ARGS (min_l),
+              GST_TIME_ARGS (min_latency));
+        } else {
+          GST_DEBUG_OBJECT (basesink,
+              "peer or we are not live, don't care about latency");
+          min_latency = 0;
+          max_latency = -1;
+        }
+        gst_query_set_latency (query, live, min_latency, max_latency);
+      }
+      break;
+    }
+    default:
+      res = GST_ELEMENT_CLASS (parent_class)->query (element, query);
+      break;
+  }
+
+  return res;
+}
+
+
 static GstClockTime
 gst_base_audio_sink_get_time (GstClock * clock, GstBaseAudioSink * sink)
 {
@@ -550,7 +609,7 @@ gst_base_audio_sink_get_offset (GstBaseAudioSink * sink)
 static GstFlowReturn
 gst_base_audio_sink_render (GstBaseSink * bsink, GstBuffer * buf)
 {
-  guint64 in_offset, clock_offset;
+  guint64 in_offset;
   GstClockTime time, stop, render_start, render_stop, sample_offset;
   GstBaseAudioSink *sink;
   GstRingBuffer *ringbuf;
@@ -563,9 +622,9 @@ gst_base_audio_sink_render (GstBaseSink * bsink, GstBuffer * buf)
   GstClockTime crate_num;
   GstClockTime crate_denom;
   gint out_samples;
-  GstClockTime cinternal, cexternal;
+  GstClockTime base_time, cinternal, cexternal, latency;
   GstClock *clock;
-  gboolean sync;
+  gboolean sync, slaved;
 
   sink = GST_BASE_AUDIO_SINK (bsink);
 
@@ -663,29 +722,62 @@ gst_base_audio_sink_render (GstBaseSink * bsink, GstBuffer * buf)
       "running: start %" GST_TIME_FORMAT " - stop %" GST_TIME_FORMAT,
       GST_TIME_ARGS (render_start), GST_TIME_ARGS (render_stop));
 
-  /* get calibration parameters to compensate for speed and offset differences
-   * when we are slaved */
-  gst_clock_get_calibration (sink->provided_clock, &cinternal, &cexternal,
-      &crate_num, &crate_denom);
+  base_time = gst_element_get_base_time (GST_ELEMENT_CAST (bsink));
+
+  GST_DEBUG_OBJECT (sink, "base_time %" GST_TIME_FORMAT,
+      GST_TIME_ARGS (base_time));
+
+  /* add base time to sync against the clock */
+  render_start += base_time;
+  render_stop += base_time;
+
+  slaved = clock != sink->provided_clock;
+  if (slaved) {
+    /* get calibration parameters to compensate for speed and offset differences
+     * when we are slaved */
+    gst_clock_get_calibration (sink->provided_clock, &cinternal, &cexternal,
+        &crate_num, &crate_denom);
+
+    GST_DEBUG_OBJECT (sink, "internal %" GST_TIME_FORMAT " external %"
+        GST_TIME_FORMAT " %" G_GUINT64_FORMAT "/%" G_GUINT64_FORMAT " = %f",
+        GST_TIME_ARGS (cinternal), GST_TIME_ARGS (cexternal), crate_num,
+        crate_denom, (gdouble) crate_num / crate_denom);
+
+    /* bring to our slaved clock time */
+    if (render_start >= cexternal)
+      render_start =
+          gst_util_uint64_scale (render_start - cexternal, crate_denom,
+          crate_num) + cinternal;
+    else
+      render_start =
+          cinternal - gst_util_uint64_scale (cexternal - render_start,
+          crate_denom, crate_num);
+
+    if (render_stop >= cexternal)
+      render_stop =
+          gst_util_uint64_scale (render_stop - cexternal, crate_denom,
+          crate_num) + cinternal;
+    else
+      render_stop =
+          cinternal - gst_util_uint64_scale (cexternal - render_stop,
+          crate_denom, crate_num);
+  }
 
-  clock_offset =
-      (gst_element_get_base_time (GST_ELEMENT_CAST (bsink)) - cexternal) +
-      cinternal;
+  /* compensate for latency */
+  latency = gst_base_sink_get_latency (bsink);
+  render_start += latency;
+  render_stop += latency;
 
-  GST_DEBUG_OBJECT (sink, "clock offset %" GST_TIME_FORMAT " %" G_GUINT64_FORMAT
-      "/%" G_GUINT64_FORMAT, GST_TIME_ARGS (clock_offset), crate_num,
-      crate_denom);
+  GST_DEBUG_OBJECT (sink,
+      "render: start %" GST_TIME_FORMAT " - stop %" GST_TIME_FORMAT,
+      GST_TIME_ARGS (render_start), GST_TIME_ARGS (render_stop));
 
   /* and bring the time to the rate corrected offset in the buffer */
-  render_start = gst_util_uint64_scale_int (render_start + clock_offset,
+  render_start = gst_util_uint64_scale_int (render_start,
       ringbuf->spec.rate, GST_SECOND);
-  render_stop = gst_util_uint64_scale_int (render_stop + clock_offset,
+  render_stop = gst_util_uint64_scale_int (render_stop,
       ringbuf->spec.rate, GST_SECOND);
 
-  GST_DEBUG_OBJECT (sink,
-      "render: start %" GST_TIME_FORMAT " - stop %" GST_TIME_FORMAT,
-      GST_TIME_ARGS (render_start), GST_TIME_ARGS (render_stop));
-
   /* always resync after a discont */
   if (G_UNLIKELY (GST_BUFFER_FLAG_IS_SET (buf, GST_BUFFER_FLAG_DISCONT))) {
     GST_DEBUG_OBJECT (sink, "resync after discont");
@@ -736,7 +828,7 @@ gst_base_audio_sink_render (GstBaseSink * bsink, GstBuffer * buf)
   render_start += align;
 
   /* only align stop if we are not slaved */
-  if (clock != sink->provided_clock) {
+  if (slaved) {
     GST_DEBUG_OBJECT (sink, "no stop time align needed: we are slaved");
     goto no_align;
   }
@@ -878,7 +970,8 @@ gst_base_audio_sink_callback (GstRingBuffer * rbuf, guint8 * data, guint len,
 
   /* would be nice to arrange for pad_alloc_buffer to return data -- as it is we
      will copy twice, once into data, once into DMA */
-  GST_LOG_OBJECT (basesink, "pulling %d bytes to fill audio buffer", len);
+  GST_LOG_OBJECT (basesink, "pulling %d bytes offset %" G_GUINT64_FORMAT
+      " to fill audio buffer", len, basesink->offset);
   ret = gst_pad_pull_range (basesink->sinkpad, basesink->offset, len, &buf);
   if (ret != GST_FLOW_OK)
     goto error;
@@ -944,6 +1037,8 @@ gst_base_audio_sink_async_play (GstBaseSink * basesink)
   }
 
 no_clock:
+  gst_ring_buffer_start (sink->ringbuffer);
+
   return GST_STATE_CHANGE_SUCCESS;
 }
 
@@ -1009,7 +1104,6 @@ gst_base_audio_sink_change_state (GstElement * element,
       break;
     case GST_STATE_CHANGE_PAUSED_TO_READY:
       gst_ring_buffer_release (sink->ringbuffer);
-      gst_pad_set_caps (GST_BASE_SINK_PAD (sink), NULL);
       break;
     case GST_STATE_CHANGE_READY_TO_NULL:
       gst_ring_buffer_close_device (sink->ringbuffer);
diff --git a/gst-libs/gst/audio/gstbaseaudiosrc.c b/gst-libs/gst/audio/gstbaseaudiosrc.c
index 992c5d61..fa0c720f 100644
--- a/gst-libs/gst/audio/gstbaseaudiosrc.c
+++ b/gst-libs/gst/audio/gstbaseaudiosrc.c
@@ -66,6 +66,7 @@ static void gst_base_audio_src_set_property (GObject * object, guint prop_id,
     const GValue * value, GParamSpec * pspec);
 static void gst_base_audio_src_get_property (GObject * object, guint prop_id,
     GValue * value, GParamSpec * pspec);
+static void gst_base_audio_src_dispose (GObject * object);
 
 static GstStateChangeReturn gst_base_audio_src_change_state (GstElement *
     element, GstStateChange transition);
@@ -84,6 +85,7 @@ static gboolean gst_base_audio_src_event (GstBaseSrc * bsrc, GstEvent * event);
 static void gst_base_audio_src_get_times (GstBaseSrc * bsrc,
     GstBuffer * buffer, GstClockTime * start, GstClockTime * end);
 static gboolean gst_base_audio_src_setcaps (GstBaseSrc * bsrc, GstCaps * caps);
+static gboolean gst_base_audio_src_query (GstBaseSrc * bsrc, GstQuery * query);
 static void gst_base_audio_src_fixate (GstBaseSrc * bsrc, GstCaps * caps);
 
 /* static guint gst_base_audio_src_signals[LAST_SIGNAL] = { 0 }; */
@@ -110,6 +112,7 @@ gst_base_audio_src_class_init (GstBaseAudioSrcClass * klass)
       GST_DEBUG_FUNCPTR (gst_base_audio_src_set_property);
   gobject_class->get_property =
       GST_DEBUG_FUNCPTR (gst_base_audio_src_get_property);
+  gobject_class->dispose = GST_DEBUG_FUNCPTR (gst_base_audio_src_dispose);
 
   g_object_class_install_property (gobject_class, PROP_BUFFER_TIME,
       g_param_spec_int64 ("buffer-time", "Buffer Time",
@@ -130,6 +133,7 @@ gst_base_audio_src_class_init (GstBaseAudioSrcClass * klass)
 
   gstbasesrc_class->set_caps = GST_DEBUG_FUNCPTR (gst_base_audio_src_setcaps);
   gstbasesrc_class->event = GST_DEBUG_FUNCPTR (gst_base_audio_src_event);
+  gstbasesrc_class->query = GST_DEBUG_FUNCPTR (gst_base_audio_src_query);
   gstbasesrc_class->get_times =
       GST_DEBUG_FUNCPTR (gst_base_audio_src_get_times);
   gstbasesrc_class->create = GST_DEBUG_FUNCPTR (gst_base_audio_src_create);
@@ -157,6 +161,25 @@ gst_base_audio_src_init (GstBaseAudioSrc * baseaudiosrc,
   gst_base_src_set_format (GST_BASE_SRC (baseaudiosrc), GST_FORMAT_TIME);
 }
 
+static void
+gst_base_audio_src_dispose (GObject * object)
+{
+  GstBaseAudioSrc *src;
+
+  src = GST_BASE_AUDIO_SRC (object);
+
+  if (src->clock)
+    gst_object_unref (src->clock);
+  src->clock = NULL;
+
+  if (src->ringbuffer) {
+    gst_object_unparent (GST_OBJECT_CAST (src->ringbuffer));
+    src->ringbuffer = NULL;
+  }
+
+  G_OBJECT_CLASS (parent_class)->dispose (object);
+}
+
 static gboolean
 gst_base_audio_src_set_clock (GstElement * elem, GstClock * clock)
 {
@@ -379,6 +402,50 @@ gst_base_audio_src_get_times (GstBaseSrc * bsrc, GstBuffer * buffer,
 }
 
 static gboolean
+gst_base_audio_src_query (GstBaseSrc * bsrc, GstQuery * query)
+{
+  GstBaseAudioSrc *src = GST_BASE_AUDIO_SRC (bsrc);
+  gboolean res = FALSE;
+
+  switch (GST_QUERY_TYPE (query)) {
+    case GST_QUERY_LATENCY:
+    {
+      GstClockTime min_latency, max_latency;
+      GstRingBufferSpec *spec;
+
+      if (G_UNLIKELY (src->ringbuffer == NULL
+              || src->ringbuffer->spec.rate == 0))
+        goto done;
+
+      spec = &src->ringbuffer->spec;
+
+      min_latency =
+          gst_util_uint64_scale_int (spec->segsize, GST_SECOND,
+          spec->rate * spec->bytes_per_sample);
+      max_latency =
+          gst_util_uint64_scale_int (spec->segtotal * spec->segsize, GST_SECOND,
+          spec->rate * spec->bytes_per_sample);
+
+      GST_DEBUG_OBJECT (src,
+          "report latency min %" GST_TIME_FORMAT " max %" GST_TIME_FORMAT,
+          GST_TIME_ARGS (min_latency), GST_TIME_ARGS (max_latency));
+
+      /* we are always live, the min latency is 1 segment and the max latency is
+       * the complete buffer of segments. */
+      gst_query_set_latency (query, TRUE, min_latency, max_latency);
+
+      res = TRUE;
+      break;
+    }
+    default:
+      res = GST_BASE_SRC_CLASS (parent_class)->query (bsrc, query);
+      break;
+  }
+done:
+  return res;
+}
+
+static gboolean
 gst_base_audio_src_event (GstBaseSrc * bsrc, GstEvent * event)
 {
   GstBaseAudioSrc *src = GST_BASE_AUDIO_SRC (bsrc);
@@ -588,15 +655,18 @@ gst_base_audio_src_change_state (GstElement * element,
         src->ringbuffer = gst_base_audio_src_create_ringbuffer (src);
       }
       if (!gst_ring_buffer_open_device (src->ringbuffer))
-        return GST_STATE_CHANGE_FAILURE;
-      src->next_sample = -1;
+        goto open_failed;
       break;
     case GST_STATE_CHANGE_READY_TO_PAUSED:
+      src->next_sample = -1;
       gst_ring_buffer_set_flushing (src->ringbuffer, FALSE);
       break;
     case GST_STATE_CHANGE_PAUSED_TO_PLAYING:
       gst_ring_buffer_may_start (src->ringbuffer, TRUE);
       break;
+    case GST_STATE_CHANGE_PAUSED_TO_READY:
+      gst_ring_buffer_set_flushing (src->ringbuffer, TRUE);
+      break;
     default:
       break;
   }
@@ -609,9 +679,7 @@ gst_base_audio_src_change_state (GstElement * element,
       gst_ring_buffer_pause (src->ringbuffer);
       break;
     case GST_STATE_CHANGE_PAUSED_TO_READY:
-      gst_ring_buffer_set_flushing (src->ringbuffer, TRUE);
       gst_ring_buffer_release (src->ringbuffer);
-      src->next_sample = -1;
       break;
     case GST_STATE_CHANGE_READY_TO_NULL:
       gst_ring_buffer_close_device (src->ringbuffer);
@@ -623,4 +691,13 @@ gst_base_audio_src_change_state (GstElement * element,
   }
 
   return ret;
+
+  /* ERRORS */
+open_failed:
+  {
+    /* subclass must post a meaningfull error message */
+    GST_DEBUG_OBJECT (src, "open failed");
+    return GST_STATE_CHANGE_FAILURE;
+  }
+
 }