ext/soundtouch/gstpitch.*: Implement LATENCY query and notify about latency changes.

Original commit message from CVS:
* ext/soundtouch/gstpitch.cc:
* ext/soundtouch/gstpitch.hh:
Implement LATENCY query and notify about latency changes.
Unfortunately we don't have a fixed latency but it changes
a bit with each buffer so we only send an LATENCY event with
the maximum latency if it changes.
Always calculate the timestamp, duration, etc from the sample
rate instead of using a pre-calculated duration for one sample
to prevent large rounding errors.

2 files changed, 93 insertions, 15 deletions

diff --git a/ext/soundtouch/gstpitch.cc b/ext/soundtouch/gstpitch.cc
index 7ee726f03..9e68fda4c 100644
--- a/ext/soundtouch/gstpitch.cc
+++ b/ext/soundtouch/gstpitch.cc
@@ -194,6 +194,7 @@ gst_pitch_init (GstPitch * pitch, GstPitchClass * pitch_class)
   pitch->priv->st->setPitch (pitch->pitch);
 
   pitch->priv->stream_time_ratio = 1.0;
+  pitch->min_latency = pitch->max_latency = 0;
 }
 
 
@@ -312,7 +313,6 @@ gst_pitch_sink_setcaps (GstPad * pad, GstCaps * caps)
 
   /* calculate sample size */
   pitch->sample_size = (sizeof (gfloat) * channels);
-  pitch->sample_duration = gst_util_uint64_scale_int (GST_SECOND, 1, rate);
 
   GST_OBJECT_UNLOCK (pitch);
 
@@ -371,7 +371,8 @@ gst_pitch_prepare_buffer (GstPitch * pitch)
     return NULL;
   }
 
-  GST_BUFFER_DURATION (buffer) = samples * pitch->sample_duration;
+  GST_BUFFER_DURATION (buffer) =
+      gst_util_uint64_scale (samples, GST_SECOND, pitch->samplerate);
   /* temporary store samples here, to avoid having to recalculate this */
   GST_BUFFER_OFFSET (buffer) = (gint64) samples;
 
@@ -466,18 +467,17 @@ gst_pitch_convert (GstPitch * pitch,
     GstFormat * dst_format, gint64 * dst_value)
 {
   gboolean res = TRUE;
-  GstClockTime sample_duration;
   guint sample_size;
+  gint samplerate;
 
   g_return_val_if_fail (dst_format && dst_value, FALSE);
 
   GST_OBJECT_LOCK (pitch);
-  sample_duration = pitch->sample_duration;
   sample_size = pitch->sample_size;
+  samplerate = pitch->samplerate;
   GST_OBJECT_UNLOCK (pitch);
 
-  if (sample_size == 0 || sample_duration == 0 ||
-      sample_duration == GST_CLOCK_TIME_NONE) {
+  if (sample_size == 0 || samplerate == 0) {
     return FALSE;
   }
 
@@ -490,11 +490,12 @@ gst_pitch_convert (GstPitch * pitch,
     case GST_FORMAT_BYTES:
       switch (*dst_format) {
         case GST_FORMAT_TIME:
-          *dst_value = src_value / sample_size;
-          *dst_value *= sample_duration;
+          *dst_value =
+              gst_util_uint64_scale_int (src_value, GST_SECOND,
+              sample_size * samplerate);
           break;
         case GST_FORMAT_DEFAULT:
-          *dst_value = src_value / sample_size;
+          *dst_value = gst_util_uint64_scale_int (src_value, 1, sample_size);
           break;
         default:
           res = FALSE;
@@ -504,11 +505,13 @@ gst_pitch_convert (GstPitch * pitch,
     case GST_FORMAT_TIME:
       switch (*dst_format) {
         case GST_FORMAT_BYTES:
-          *dst_value = src_value / sample_duration;
-          *dst_value *= sample_size;
+          *dst_value =
+              gst_util_uint64_scale_int (src_value, samplerate * sample_size,
+              GST_SECOND);
           break;
         case GST_FORMAT_DEFAULT:
-          *dst_value = src_value / sample_duration;
+          *dst_value =
+              gst_util_uint64_scale_int (src_value, samplerate, GST_SECOND);
           break;
         default:
           res = FALSE;
@@ -518,10 +521,11 @@ gst_pitch_convert (GstPitch * pitch,
     case GST_FORMAT_DEFAULT:
       switch (*dst_format) {
         case GST_FORMAT_BYTES:
-          *dst_value = src_value * sample_size;
+          *dst_value = gst_util_uint64_scale_int (src_value, sample_size, 1);
           break;
         case GST_FORMAT_TIME:
-          *dst_value = src_value * sample_duration;
+          *dst_value =
+              gst_util_uint64_scale_int (src_value, GST_SECOND, samplerate);
           break;
         default:
           res = FALSE;
@@ -543,6 +547,7 @@ gst_pitch_get_query_types (GstPad * pad)
     GST_QUERY_POSITION,
     GST_QUERY_DURATION,
     GST_QUERY_CONVERT,
+    GST_QUERY_LATENCY,
     GST_QUERY_NONE
   };
 
@@ -630,6 +635,46 @@ gst_pitch_src_query (GstPad * pad, GstQuery * query)
       }
       break;
     }
+    case GST_QUERY_LATENCY:
+    {
+      GstClockTime min, max;
+      gboolean live;
+      GstPad *peer;
+
+      if ((peer = gst_pad_get_peer (pitch->sinkpad))) {
+        if ((res = gst_pad_query (peer, query))) {
+          gst_query_parse_latency (query, &live, &min, &max);
+
+          GST_DEBUG ("Peer latency: min %"
+              GST_TIME_FORMAT " max %" GST_TIME_FORMAT,
+              GST_TIME_ARGS (min), GST_TIME_ARGS (max));
+
+          /* add our own latency */
+
+          GST_DEBUG ("Our latency: min %" GST_TIME_FORMAT
+              ", max %" GST_TIME_FORMAT,
+              GST_TIME_ARGS (pitch->min_latency),
+              GST_TIME_ARGS (pitch->max_latency));
+
+          min += pitch->min_latency;
+          if (max != GST_CLOCK_TIME_NONE)
+            max += pitch->max_latency;
+          else
+            max = pitch->max_latency;
+
+          GST_DEBUG ("Calculated total latency : min %"
+              GST_TIME_FORMAT " max %" GST_TIME_FORMAT,
+              GST_TIME_ARGS (min), GST_TIME_ARGS (max));
+          g_print ("Calculated total latency : min %"
+              GST_TIME_FORMAT " max %" GST_TIME_FORMAT,
+              GST_TIME_ARGS (min), GST_TIME_ARGS (max));
+
+          gst_query_set_latency (query, live, min, max);
+        }
+        gst_object_unref (peer);
+      }
+      break;
+    }
     default:
       res = gst_pad_query_default (pad, query);
       break;
@@ -728,10 +773,12 @@ gst_pitch_sink_event (GstPad * pad, GstEvent * event)
     case GST_EVENT_FLUSH_STOP:
       gst_pitch_flush_buffer (pitch, FALSE);
       pitch->priv->st->clear ();
+      pitch->min_latency = pitch->max_latency = 0;
       break;
     case GST_EVENT_EOS:
       gst_pitch_flush_buffer (pitch, TRUE);
       pitch->priv->st->clear ();
+      pitch->min_latency = pitch->max_latency = 0;
       break;
     case GST_EVENT_NEWSEGMENT:
       if (!gst_pitch_process_segment (pitch, &event)) {
@@ -742,6 +789,7 @@ gst_pitch_sink_event (GstPad * pad, GstEvent * event)
         event = NULL;
       }
       pitch->priv->st->clear ();
+      pitch->min_latency = pitch->max_latency = 0;
       break;
     default:
       break;
@@ -755,17 +803,41 @@ gst_pitch_sink_event (GstPad * pad, GstEvent * event)
   return res;
 }
 
+static void
+gst_pitch_update_latency (GstPitch * pitch, GstClockTime timestamp)
+{
+  GstClockTimeDiff current_latency, min_latency, max_latency;
+
+  current_latency =
+      timestamp / pitch->priv->stream_time_ratio - pitch->next_buffer_time;
+
+  min_latency = MIN (pitch->min_latency, current_latency);
+  max_latency = MAX (pitch->max_latency, current_latency);
+
+  if (pitch->min_latency != min_latency || pitch->max_latency != max_latency) {
+    pitch->min_latency = min_latency;
+    pitch->max_latency = max_latency;
+
+    gst_pad_push_event (pitch->sinkpad, gst_event_new_latency (max_latency));
+    gst_element_post_message (GST_ELEMENT (pitch),
+        gst_message_new_latency (GST_OBJECT (pitch)));
+  }
+}
+
 static GstFlowReturn
 gst_pitch_chain (GstPad * pad, GstBuffer * buffer)
 {
   GstPitch *pitch;
   GstPitchPrivate *priv;
+  GstClockTime timestamp;
 
   pitch = GST_PITCH (GST_PAD_PARENT (pad));
   priv = GST_PITCH_GET_PRIVATE (pitch);
 
   gst_object_sync_values (G_OBJECT (pitch), pitch->next_buffer_time);
 
+  timestamp = GST_BUFFER_TIMESTAMP (buffer);
+
   /* push the received samples on the soundtouch buffer */
   GST_LOG_OBJECT (pitch, "incoming buffer (%d samples)",
       (gint) (GST_BUFFER_SIZE (buffer) / pitch->sample_size));
@@ -793,6 +865,10 @@ gst_pitch_chain (GstPad * pad, GstBuffer * buffer)
       GST_BUFFER_SIZE (buffer) / pitch->sample_size);
   gst_buffer_unref (buffer);
 
+  /* Calculate latency */
+
+  gst_pitch_update_latency (pitch, timestamp);
+
   /* and try to extract some samples from the soundtouch buffer */
   if (!priv->st->isEmpty ()) {
     GstBuffer *out_buffer;
@@ -818,6 +894,7 @@ gst_pitch_change_state (GstElement * element, GstStateChange transition)
       pitch->next_buffer_time = 0;
       pitch->next_buffer_offset = 0;
       pitch->priv->st->clear ();
+      pitch->min_latency = pitch->max_latency = 0;
       break;
     case GST_STATE_CHANGE_PAUSED_TO_PLAYING:
       break;
diff --git a/ext/soundtouch/gstpitch.hh b/ext/soundtouch/gstpitch.hh
index 24d42ba2f..32cb04dd1 100644
--- a/ext/soundtouch/gstpitch.hh
+++ b/ext/soundtouch/gstpitch.hh
@@ -67,12 +67,13 @@ struct _GstPitch
   gint          samplerate;      /* samplerate                           */
   gint          channels;        /* number of audio channels             */
   gsize         sample_size;     /* number of bytes for a single sample  */
-  GstClockTime  sample_duration; /* time for 1 sample                    */
 
   /* stream tracking */
   GstClockTime  next_buffer_time;
   gint64        next_buffer_offset;
 
+  GstClockTimeDiff  min_latency, max_latency;
+
   GstPitchPrivate *priv;
 };