Merge remote-tracking branch 'origin/master' into 0.11

Conflicts:
	ext/pulse/pulseaudiosink.c
	ext/pulse/pulsesrc.c
	gst/audioparsers/gstaacparse.c
	gst/audioparsers/gstamrparse.c
	gst/audioparsers/gstdcaparse.c
	gst/audioparsers/gstflacparse.c
	gst/effectv/gstradioac.c
	gst/effectv/gstradioac.h
	gst/effectv/gstripple.c

Some possible FIXMEs remaining in the audio parser getcaps functions.

ext/pulse/pulseaudiosink.c

@@ -535,7 +535,7 @@ dbin_event_probe (GstPad * pad, GstPadProbeInfo * info, gpointer data)
 static void
 pad_added_cb (GstElement * dbin, GstPad * pad, gpointer * data)
 {
-  GstPulseAudioSink *pbin = GST_PULSE_AUDIO_SINK (data);
+  GstPulseAudioSink *pbin;
   GstPad *sinkpad = NULL;
 
   pbin = GST_PULSE_AUDIO_SINK (data);
@@ -584,11 +584,13 @@ gst_pulse_audio_sink_add_dbin (GstPulseAudioSink * pbin)
 
   /* Trap the newsegment events that we feed the decodebin and discard them */
   sinkpad = gst_element_get_static_pad (GST_ELEMENT (pbin->psink), "sink");
+
   if (pbin->event_probe_id == 0)
     pbin->event_probe_id =
         gst_pad_add_probe (sinkpad, GST_PAD_PROBE_TYPE_EVENT_DOWNSTREAM,
         dbin_event_probe, gst_object_ref (pbin),
         (GDestroyNotify) gst_object_unref);
+
   gst_object_unref (sinkpad);
   sinkpad = NULL;
 

ext/pulse/pulsesrc.c

@@ -43,6 +43,9 @@
 
 #include <gst/base/gstbasesrc.h>
 #include <gst/gsttaglist.h>
+#ifdef HAVE_PULSE_1_0
+#include <gst/interfaces/streamvolume.h>
+#endif
 
 #include "pulsesrc.h"
 #include "pulseutil.h"
@@ -55,6 +58,12 @@ GST_DEBUG_CATEGORY_EXTERN (pulse_debug);
 #define DEFAULT_DEVICE            NULL
 #define DEFAULT_DEVICE_NAME       NULL
 
+#ifdef HAVE_PULSE_1_0
+#define DEFAULT_VOLUME          1.0
+#define DEFAULT_MUTE            FALSE
+#define MAX_VOLUME              10.0
+#endif
+
 enum
 {
   PROP_0,
@@ -64,6 +73,10 @@ enum
   PROP_CLIENT,
   PROP_STREAM_PROPERTIES,
   PROP_SOURCE_OUTPUT_INDEX,
+#ifdef HAVE_PULSE_1_0
+  PROP_VOLUME,
+  PROP_MUTE,
+#endif
   PROP_LAST
 };
 
@@ -126,7 +139,8 @@ GST_IMPLEMENT_PULSEPROBE_METHODS (GstPulseSrc, gst_pulsesrc);
 G_DEFINE_TYPE_WITH_CODE (GstPulseSrc, gst_pulsesrc, GST_TYPE_AUDIO_SRC,
     G_IMPLEMENT_INTERFACE (GST_TYPE_MIXER, gst_pulsesrc_mixer_interface_init);
     G_IMPLEMENT_INTERFACE (GST_TYPE_PROPERTY_PROBE,
-        gst_pulsesrc_property_probe_interface_init));
+        gst_pulsesrc_property_probe_interface_init);
+    G_IMPLEMENT_INTERFACE (GST_TYPE_STREAM_VOLUME, NULL));
 
 static void
 gst_pulsesrc_class_init (GstPulseSrcClass * klass)
@@ -174,7 +188,7 @@ gst_pulsesrc_class_init (GstPulseSrcClass * klass)
 
   clientname = gst_pulse_client_name ();
   /**
-   * GstPulseSink:client
+   * GstPulseSrc:client
    *
    * The PulseAudio client name to use.
    *
@@ -230,6 +244,35 @@ gst_pulsesrc_class_init (GstPulseSrcClass * klass)
       "Captures audio from a PulseAudio server", "Lennart Poettering");
   gst_element_class_add_pad_template (gstelement_class,
       gst_static_pad_template_get (&pad_template));
+
+#ifdef HAVE_PULSE_1_0
+  /**
+   * GstPulseSrc:volume
+   *
+   * The volume of the record stream. Only works when using PulseAudio 1.0 or
+   * later.
+   *
+   * Since: 0.10.36
+   */
+  g_object_class_install_property (gobject_class,
+      PROP_VOLUME, g_param_spec_double ("volume", "Volume",
+          "Linear volume of this stream, 1.0=100%",
+          0.0, MAX_VOLUME, DEFAULT_VOLUME,
+          G_PARAM_READWRITE | G_PARAM_STATIC_STRINGS));
+
+  /**
+   * GstPulseSrc:mute
+   *
+   * Whether the stream is muted or not. Only works when using PulseAudio 1.0
+   * or later.
+   *
+   * Since: 0.10.36
+   */
+  g_object_class_install_property (gobject_class,
+      PROP_MUTE, g_param_spec_boolean ("mute", "Mute",
+          "Mute state of this stream",
+          DEFAULT_MUTE, G_PARAM_READWRITE | G_PARAM_STATIC_STRINGS));
+#endif
 }
 
 static void
@@ -254,6 +297,16 @@ gst_pulsesrc_init (GstPulseSrc * pulsesrc)
   pulsesrc->paused = TRUE;
   pulsesrc->in_read = FALSE;
 
+#ifdef HAVE_PULSE_1_0
+  pulsesrc->volume = DEFAULT_VOLUME;
+  pulsesrc->volume_set = FALSE;
+
+  pulsesrc->mute = DEFAULT_MUTE;
+  pulsesrc->mute_set = FALSE;
+
+  pulsesrc->notify = 0;
+#endif
+
   pulsesrc->mixer = NULL;
 
   pulsesrc->properties = NULL;
@@ -290,7 +343,15 @@ gst_pulsesrc_destroy_context (GstPulseSrc * pulsesrc)
 
   if (pulsesrc->context) {
     pa_context_disconnect (pulsesrc->context);
+
+    /* Make sure we don't get any further callbacks */
+    pa_context_set_state_callback (pulsesrc->context, NULL, NULL);
+#ifdef HAVE_PULSE_1_0
+    pa_context_set_subscribe_callback (pulsesrc->context, NULL, NULL);
+#endif
+
     pa_context_unref (pulsesrc->context);
+
     pulsesrc->context = NULL;
   }
 }
@@ -408,6 +469,260 @@ no_mainloop:
   }
 }
 
+#ifdef HAVE_PULSE_1_0
+static void
+gst_pulsesrc_source_output_info_cb (pa_context * c,
+    const pa_source_output_info * i, int eol, void *userdata)
+{
+  GstPulseSrc *psrc;
+
+  psrc = GST_PULSESRC_CAST (userdata);
+
+  if (!i)
+    goto done;
+
+  /* If the index doesn't match our current stream,
+   * it implies we just recreated the stream (caps change)
+   */
+  if (i->index == psrc->source_output_idx) {
+    psrc->volume = pa_sw_volume_to_linear (pa_cvolume_max (&i->volume));
+    psrc->mute = i->mute;
+  }
+
+done:
+  pa_threaded_mainloop_signal (psrc->mainloop, 0);
+}
+
+static gdouble
+gst_pulsesrc_get_stream_volume (GstPulseSrc * pulsesrc)
+{
+  pa_operation *o = NULL;
+  gdouble v;
+
+  if (!pulsesrc->mainloop)
+    goto no_mainloop;
+
+  if (pulsesrc->source_output_idx == PA_INVALID_INDEX)
+    goto no_index;
+
+  pa_threaded_mainloop_lock (pulsesrc->mainloop);
+
+  if (!(o = pa_context_get_source_output_info (pulsesrc->context,
+              pulsesrc->source_output_idx, gst_pulsesrc_source_output_info_cb,
+              pulsesrc)))
+    goto info_failed;
+
+  while (pa_operation_get_state (o) == PA_OPERATION_RUNNING) {
+    pa_threaded_mainloop_wait (pulsesrc->mainloop);
+    if (gst_pulsesrc_is_dead (pulsesrc, TRUE))
+      goto unlock;
+  }
+
+unlock:
+  v = pulsesrc->volume;
+
+  if (o)
+    pa_operation_unref (o);
+
+  pa_threaded_mainloop_unlock (pulsesrc->mainloop);
+
+  if (v > MAX_VOLUME) {
+    GST_WARNING_OBJECT (pulsesrc, "Clipped volume from %f to %f", v,
+        MAX_VOLUME);
+    v = MAX_VOLUME;
+  }
+
+  return v;
+
+  /* ERRORS */
+no_mainloop:
+  {
+    v = pulsesrc->volume;
+    GST_DEBUG_OBJECT (pulsesrc, "we have no mainloop");
+    return v;
+  }
+no_index:
+  {
+    v = pulsesrc->volume;
+    GST_DEBUG_OBJECT (pulsesrc, "we don't have a stream index");
+    return v;
+  }
+info_failed:
+  {
+    GST_ELEMENT_ERROR (pulsesrc, RESOURCE, FAILED,
+        ("pa_context_get_source_output_info() failed: %s",
+            pa_strerror (pa_context_errno (pulsesrc->context))), (NULL));
+    goto unlock;
+  }
+}
+
+static gboolean
+gst_pulsesrc_get_stream_mute (GstPulseSrc * pulsesrc)
+{
+  pa_operation *o = NULL;
+  gboolean mute;
+
+  if (!pulsesrc->mainloop)
+    goto no_mainloop;
+
+  if (pulsesrc->source_output_idx == PA_INVALID_INDEX)
+    goto no_index;
+
+  pa_threaded_mainloop_lock (pulsesrc->mainloop);
+
+  if (!(o = pa_context_get_source_output_info (pulsesrc->context,
+              pulsesrc->source_output_idx, gst_pulsesrc_source_output_info_cb,
+              pulsesrc)))
+    goto info_failed;
+
+  while (pa_operation_get_state (o) == PA_OPERATION_RUNNING) {
+    pa_threaded_mainloop_wait (pulsesrc->mainloop);
+    if (gst_pulsesrc_is_dead (pulsesrc, TRUE))
+      goto unlock;
+  }
+
+unlock:
+  mute = pulsesrc->mute;
+
+  if (o)
+    pa_operation_unref (o);
+
+  pa_threaded_mainloop_unlock (pulsesrc->mainloop);
+
+  return mute;
+
+  /* ERRORS */
+no_mainloop:
+  {
+    mute = pulsesrc->mute;
+    GST_DEBUG_OBJECT (pulsesrc, "we have no mainloop");
+    return mute;
+  }
+no_index:
+  {
+    mute = pulsesrc->mute;
+    GST_DEBUG_OBJECT (pulsesrc, "we don't have a stream index");
+    return mute;
+  }
+info_failed:
+  {
+    GST_ELEMENT_ERROR (pulsesrc, RESOURCE, FAILED,
+        ("pa_context_get_source_output_info() failed: %s",
+            pa_strerror (pa_context_errno (pulsesrc->context))), (NULL));
+    goto unlock;
+  }
+}
+
+static void
+gst_pulsesrc_set_stream_volume (GstPulseSrc * pulsesrc, gdouble volume)
+{
+  pa_cvolume v;
+  pa_operation *o = NULL;
+
+  if (!pulsesrc->mainloop)
+    goto no_mainloop;
+
+  if (!pulsesrc->source_output_idx)
+    goto no_index;
+
+  pa_threaded_mainloop_lock (pulsesrc->mainloop);
+
+  GST_DEBUG_OBJECT (pulsesrc, "setting volume to %f", volume);
+
+  gst_pulse_cvolume_from_linear (&v, pulsesrc->sample_spec.channels, volume);
+
+  if (!(o = pa_context_set_source_output_volume (pulsesrc->context,
+              pulsesrc->source_output_idx, &v, NULL, NULL)))
+    goto volume_failed;
+
+  /* We don't really care about the result of this call */
+unlock:
+
+  if (o)
+    pa_operation_unref (o);
+
+  pa_threaded_mainloop_unlock (pulsesrc->mainloop);
+
+  return;
+
+  /* ERRORS */
+no_mainloop:
+  {
+    pulsesrc->volume = volume;
+    pulsesrc->volume_set = TRUE;
+    GST_DEBUG_OBJECT (pulsesrc, "we have no mainloop");
+    return;
+  }
+no_index:
+  {
+    pulsesrc->volume = volume;
+    pulsesrc->volume_set = TRUE;
+    GST_DEBUG_OBJECT (pulsesrc, "we don't have a stream index");
+    return;
+  }
+volume_failed:
+  {
+    GST_ELEMENT_ERROR (pulsesrc, RESOURCE, FAILED,
+        ("pa_stream_set_source_output_volume() failed: %s",
+            pa_strerror (pa_context_errno (pulsesrc->context))), (NULL));
+    goto unlock;
+  }
+}
+
+static void
+gst_pulsesrc_set_stream_mute (GstPulseSrc * pulsesrc, gboolean mute)
+{
+  pa_operation *o = NULL;
+
+  if (!pulsesrc->mainloop)
+    goto no_mainloop;
+
+  if (!pulsesrc->source_output_idx)
+    goto no_index;
+
+  pa_threaded_mainloop_lock (pulsesrc->mainloop);
+
+  GST_DEBUG_OBJECT (pulsesrc, "setting mute state to %d", mute);
+
+  if (!(o = pa_context_set_source_output_mute (pulsesrc->context,
+              pulsesrc->source_output_idx, mute, NULL, NULL)))
+    goto mute_failed;
+
+  /* We don't really care about the result of this call */
+unlock:
+
+  if (o)
+    pa_operation_unref (o);
+
+  pa_threaded_mainloop_unlock (pulsesrc->mainloop);
+
+  return;
+
+  /* ERRORS */
+no_mainloop:
+  {
+    pulsesrc->mute = mute;
+    pulsesrc->mute_set = TRUE;
+    GST_DEBUG_OBJECT (pulsesrc, "we have no mainloop");
+    return;
+  }
+no_index:
+  {
+    pulsesrc->mute = mute;
+    pulsesrc->mute_set = TRUE;
+    GST_DEBUG_OBJECT (pulsesrc, "we don't have a stream index");
+    return;
+  }
+mute_failed:
+  {
+    GST_ELEMENT_ERROR (pulsesrc, RESOURCE, FAILED,
+        ("pa_stream_set_source_output_mute() failed: %s",
+            pa_strerror (pa_context_errno (pulsesrc->context))), (NULL));
+    goto unlock;
+  }
+}
+#endif
+
 static void
 gst_pulsesrc_set_property (GObject * object,
     guint prop_id, const GValue * value, GParamSpec * pspec)
@@ -444,6 +759,14 @@ gst_pulsesrc_set_property (GObject * object,
         pa_proplist_free (pulsesrc->proplist);
       pulsesrc->proplist = gst_pulse_make_proplist (pulsesrc->properties);
       break;
+#ifdef HAVE_PULSE_1_0
+    case PROP_VOLUME:
+      gst_pulsesrc_set_stream_volume (pulsesrc, g_value_get_double (value));
+      break;
+    case PROP_MUTE:
+      gst_pulsesrc_set_stream_mute (pulsesrc, g_value_get_boolean (value));
+      break;
+#endif
     default:
       G_OBJECT_WARN_INVALID_PROPERTY_ID (object, prop_id, pspec);
       break;
@@ -476,6 +799,14 @@ gst_pulsesrc_get_property (GObject * object,
     case PROP_SOURCE_OUTPUT_INDEX:
       g_value_set_uint (value, pulsesrc->source_output_idx);
       break;
+#ifdef HAVE_PULSE_1_0
+    case PROP_VOLUME:
+      g_value_set_double (value, gst_pulsesrc_get_stream_volume (pulsesrc));
+      break;
+    case PROP_MUTE:
+      g_value_set_boolean (value, gst_pulsesrc_get_stream_mute (pulsesrc));
+      break;
+#endif
     default:
       G_OBJECT_WARN_INVALID_PROPERTY_ID (object, prop_id, pspec);
       break;
@@ -569,6 +900,30 @@ gst_pulsesrc_stream_overflow_cb (pa_stream * s, void *userdata)
   GST_WARNING_OBJECT (GST_PULSESRC_CAST (userdata), "Got overflow");
 }
 
+#ifdef HAVE_PULSE_1_0
+static void
+gst_pulsesrc_context_subscribe_cb (pa_context * c,
+    pa_subscription_event_type_t t, uint32_t idx, void *userdata)
+{
+  GstPulseSrc *psrc = GST_PULSESRC (userdata);
+
+  if (t != (PA_SUBSCRIPTION_EVENT_SOURCE_OUTPUT | PA_SUBSCRIPTION_EVENT_CHANGE)
+      && t != (PA_SUBSCRIPTION_EVENT_SOURCE_OUTPUT | PA_SUBSCRIPTION_EVENT_NEW))
+    return;
+
+  if (idx != psrc->source_output_idx)
+    return;
+
+  /* Actually this event is also triggered when other properties of the stream
+   * change that are unrelated to the volume. However it is probably cheaper to
+   * signal the change here and check for the volume when the GObject property
+   * is read instead of querying it always. */
+
+  /* inform streaming thread to notify */
+  g_atomic_int_compare_and_exchange (&psrc->notify, 0, 1);
+}
+#endif
+
 static gboolean
 gst_pulsesrc_open (GstAudioSrc * asrc)
 {
@@ -591,6 +946,10 @@ gst_pulsesrc_open (GstAudioSrc * asrc)
 
   pa_context_set_state_callback (pulsesrc->context,
       gst_pulsesrc_context_state_cb, pulsesrc);
+#ifdef HAVE_PULSE_1_0
+  pa_context_set_subscribe_callback (pulsesrc->context,
+      gst_pulsesrc_context_subscribe_cb, pulsesrc);
+#endif
 
   GST_DEBUG_OBJECT (pulsesrc, "connect to server %s",
       GST_STR_NULL (pulsesrc->server));
@@ -672,6 +1031,13 @@ gst_pulsesrc_read (GstAudioSrc * asrc, gpointer data, guint length)
   pa_threaded_mainloop_lock (pulsesrc->mainloop);
   pulsesrc->in_read = TRUE;
 
+#ifdef HAVE_PULSE_1_0
+  if (g_atomic_int_compare_and_exchange (&pulsesrc->notify, 1, 0)) {
+    g_object_notify (G_OBJECT (pulsesrc), "volume");
+    g_object_notify (G_OBJECT (pulsesrc), "mute");
+  }
+#endif
+
   if (pulsesrc->paused)
     goto was_paused;
 
@@ -985,9 +1351,27 @@ gst_pulsesrc_prepare (GstAudioSrc * asrc, GstAudioRingBufferSpec * spec)
   pa_buffer_attr wanted;
   const pa_buffer_attr *actual;
   GstPulseSrc *pulsesrc = GST_PULSESRC_CAST (asrc);
+  pa_stream_flags_t flags;
+#ifdef HAVE_PULSE_1_0
+  pa_operation *o;
+#endif
 
   pa_threaded_mainloop_lock (pulsesrc->mainloop);
 
+#ifdef HAVE_PULSE_1_0
+  /* enable event notifications */
+  GST_LOG_OBJECT (pulsesrc, "subscribing to context events");
+  if (!(o = pa_context_subscribe (pulsesrc->context,
+              PA_SUBSCRIPTION_MASK_SINK_INPUT, NULL, NULL))) {
+    GST_ELEMENT_ERROR (pulsesrc, RESOURCE, FAILED,
+        ("pa_context_subscribe() failed: %s",
+            pa_strerror (pa_context_errno (pulsesrc->context))), (NULL));
+    goto unlock_and_fail;
+  }
+
+  pa_operation_unref (o);
+#endif
+
   wanted.maxlength = -1;
   wanted.tlength = -1;
   wanted.prebuf = 0;
@@ -1000,11 +1384,19 @@ gst_pulsesrc_prepare (GstAudioSrc * asrc, GstAudioRingBufferSpec * spec)
   GST_INFO_OBJECT (pulsesrc, "minreq:    %d", wanted.minreq);
   GST_INFO_OBJECT (pulsesrc, "fragsize:  %d", wanted.fragsize);
 
+  flags = PA_STREAM_INTERPOLATE_TIMING | PA_STREAM_AUTO_TIMING_UPDATE |
+      PA_STREAM_NOT_MONOTONIC | PA_STREAM_ADJUST_LATENCY |
+      PA_STREAM_START_CORKED;
+
+#ifdef HAVE_PULSE_1_0
+  if (pulsesrc->mute_set && pulsesrc->mute)
+    flags |= PA_STREAM_START_MUTED;
+#endif
+
   if (pa_stream_connect_record (pulsesrc->stream, pulsesrc->device, &wanted,
-          PA_STREAM_INTERPOLATE_TIMING | PA_STREAM_AUTO_TIMING_UPDATE |
+          flags) < 0) {
-          PA_STREAM_NOT_MONOTONIC | PA_STREAM_ADJUST_LATENCY |
-          PA_STREAM_START_CORKED) < 0)
     goto connect_failed;
+  }
 
   pulsesrc->corked = TRUE;
 
@@ -1028,6 +1420,13 @@ gst_pulsesrc_prepare (GstAudioSrc * asrc, GstAudioRingBufferSpec * spec)
   pulsesrc->source_output_idx = pa_stream_get_index (pulsesrc->stream);
   g_object_notify (G_OBJECT (pulsesrc), "source-output-index");
 
+#ifdef HAVE_PULSE_1_0
+  if (pulsesrc->volume_set) {
+    gst_pulsesrc_set_stream_volume (pulsesrc, pulsesrc->volume);
+    pulsesrc->volume_set = FALSE;
+  }
+#endif
+
   /* get the actual buffering properties now */
   actual = pa_stream_get_buffer_attr (pulsesrc->stream);
 

ext/pulse/pulsesrc.h

@@ -72,6 +72,15 @@ struct _GstPulseSrc
   GstPulseMixerCtrl *mixer;
   GstPulseProbe *probe;
 
+#ifdef HAVE_PULSE_1_0
+  gdouble volume;
+  gboolean volume_set:1;
+  gboolean mute:1;
+  gboolean mute_set:1;
+
+  gint notify; /* atomic */
+#endif
+
   gboolean corked:1;
   gboolean stream_connected:1;
   gboolean operation_success:1;

gst/audioparsers/gstaacparse.c

@@ -69,16 +69,18 @@ GST_DEBUG_CATEGORY_STATIC (aacparse_debug);
 
 #define AAC_FRAME_DURATION(parse) (GST_SECOND/parse->frames_per_sec)
 
-gboolean gst_aac_parse_start (GstBaseParse * parse);
+static gboolean gst_aac_parse_start (GstBaseParse * parse);
-gboolean gst_aac_parse_stop (GstBaseParse * parse);
+static gboolean gst_aac_parse_stop (GstBaseParse * parse);
 
 static gboolean gst_aac_parse_sink_setcaps (GstBaseParse * parse,
     GstCaps * caps);
+static GstCaps *gst_aac_parse_sink_getcaps (GstBaseParse * parse,
+    GstCaps * filter);
 
-gboolean gst_aac_parse_check_valid_frame (GstBaseParse * parse,
+static gboolean gst_aac_parse_check_valid_frame (GstBaseParse * parse,
     GstBaseParseFrame * frame, guint * size, gint * skipsize);
 
-GstFlowReturn gst_aac_parse_parse_frame (GstBaseParse * parse,
+static GstFlowReturn gst_aac_parse_parse_frame (GstBaseParse * parse,
     GstBaseParseFrame * frame);
 
 gboolean gst_aac_parse_convert (GstBaseParse * parse,
@@ -131,6 +133,7 @@ gst_aac_parse_class_init (GstAacParseClass * klass)
   parse_class->start = GST_DEBUG_FUNCPTR (gst_aac_parse_start);
   parse_class->stop = GST_DEBUG_FUNCPTR (gst_aac_parse_stop);
   parse_class->set_sink_caps = GST_DEBUG_FUNCPTR (gst_aac_parse_sink_setcaps);
+  parse_class->get_sink_caps = GST_DEBUG_FUNCPTR (gst_aac_parse_sink_getcaps);
   parse_class->parse_frame = GST_DEBUG_FUNCPTR (gst_aac_parse_parse_frame);
   parse_class->check_valid_frame =
       GST_DEBUG_FUNCPTR (gst_aac_parse_check_valid_frame);
@@ -568,7 +571,7 @@ gst_aac_parse_detect_stream (GstAacParse * aacparse,
  *
  * Returns: TRUE if buffer contains a valid frame.
  */
-gboolean
+static gboolean
 gst_aac_parse_check_valid_frame (GstBaseParse * parse,
     GstBaseParseFrame * frame, guint * framesize, gint * skipsize)
 {
@@ -643,7 +646,7 @@ gst_aac_parse_check_valid_frame (GstBaseParse * parse,
  * Returns: GST_FLOW_OK if frame was successfully parsed and can be pushed
  *          forward. Otherwise appropriate error is returned.
  */
-GstFlowReturn
+static GstFlowReturn
 gst_aac_parse_parse_frame (GstBaseParse * parse, GstBaseParseFrame * frame)
 {
   GstAacParse *aacparse;
@@ -700,7 +703,7 @@ gst_aac_parse_parse_frame (GstBaseParse * parse, GstBaseParseFrame * frame)
  *
  * Returns: TRUE if startup succeeded.
  */
-gboolean
+static gboolean
 gst_aac_parse_start (GstBaseParse * parse)
 {
   GstAacParse *aacparse;
@@ -721,9 +724,44 @@ gst_aac_parse_start (GstBaseParse * parse)
  *
  * Returns: TRUE is stopping succeeded.
  */
-gboolean
+static gboolean
 gst_aac_parse_stop (GstBaseParse * parse)
 {
   GST_DEBUG ("stop");
   return TRUE;
 }
+
+static GstCaps *
+gst_aac_parse_sink_getcaps (GstBaseParse * parse, GstCaps * filter)
+{
+  GstCaps *peercaps;
+  GstCaps *res;
+
+  /* FIXME: handle filter caps */
+
+  peercaps = gst_pad_get_allowed_caps (GST_BASE_PARSE_SRC_PAD (parse));
+  if (peercaps) {
+    guint i, n;
+
+    /* Remove the framed field */
+    peercaps = gst_caps_make_writable (peercaps);
+    n = gst_caps_get_size (peercaps);
+    for (i = 0; i < n; i++) {
+      GstStructure *s = gst_caps_get_structure (peercaps, i);
+
+      gst_structure_remove_field (s, "framed");
+    }
+
+    res =
+        gst_caps_intersect_full (peercaps,
+        gst_pad_get_pad_template_caps (GST_BASE_PARSE_SRC_PAD (parse)),
+        GST_CAPS_INTERSECT_FIRST);
+    gst_caps_unref (peercaps);
+  } else {
+    res =
+        gst_caps_copy (gst_pad_get_pad_template_caps (GST_BASE_PARSE_SINK_PAD
+            (parse)));
+  }
+
+  return res;
+}

gst/audioparsers/gstac3parse.c

@@ -165,6 +165,8 @@ static GstFlowReturn gst_ac3_parse_parse_frame (GstBaseParse * parse,
     GstBaseParseFrame * frame);
 static gboolean gst_ac3_parse_src_event (GstBaseParse * parse,
     GstEvent * event);
+static GstCaps *gst_ac3_parse_get_sink_caps (GstBaseParse * parse,
+    GstCaps * filter);
 
 #define gst_ac3_parse_parent_class parent_class
 G_DEFINE_TYPE (GstAc3Parse, gst_ac3_parse, GST_TYPE_BASE_PARSE);
@@ -195,8 +197,8 @@ gst_ac3_parse_class_init (GstAc3ParseClass * klass)
   parse_class->check_valid_frame =
       GST_DEBUG_FUNCPTR (gst_ac3_parse_check_valid_frame);
   parse_class->parse_frame = GST_DEBUG_FUNCPTR (gst_ac3_parse_parse_frame);
-
   parse_class->src_event = GST_DEBUG_FUNCPTR (gst_ac3_parse_src_event);
+  parse_class->get_sink_caps = GST_DEBUG_FUNCPTR (gst_ac3_parse_get_sink_caps);
 }
 
 static void
@@ -702,3 +704,40 @@ gst_ac3_parse_src_event (GstBaseParse * parse, GstEvent * event)
 
   return GST_BASE_PARSE_CLASS (parent_class)->src_event (parse, event);
 }
+
+static GstCaps *
+gst_ac3_parse_get_sink_caps (GstBaseParse * parse, GstCaps * filter)
+{
+  GstCaps *peercaps;
+  GstCaps *res;
+
+  /* FIXME: handle filter */
+
+  peercaps = gst_pad_get_allowed_caps (GST_BASE_PARSE_SRC_PAD (parse));
+  if (peercaps) {
+    guint i, n;
+
+    /* Remove the framed and alignment field. We can convert
+     * between different alignments. */
+    peercaps = gst_caps_make_writable (peercaps);
+    n = gst_caps_get_size (peercaps);
+    for (i = 0; i < n; i++) {
+      GstStructure *s = gst_caps_get_structure (peercaps, i);
+
+      gst_structure_remove_field (s, "framed");
+      gst_structure_remove_field (s, "alignment");
+    }
+
+    res =
+        gst_caps_intersect_full (peercaps,
+        gst_pad_get_pad_template_caps (GST_BASE_PARSE_SRC_PAD (parse)),
+        GST_CAPS_INTERSECT_FIRST);
+    gst_caps_unref (peercaps);
+  } else {
+    res =
+        gst_caps_copy (gst_pad_get_pad_template_caps (GST_BASE_PARSE_SINK_PAD
+            (parse)));
+  }
+
+  return res;
+}

gst/audioparsers/gstamrparse.c

@@ -71,16 +71,18 @@ static const gint block_size_wb[16] =
 #define AMR_FRAME_DURATION (GST_SECOND/AMR_FRAMES_PER_SECOND)
 #define AMR_MIME_HEADER_SIZE 9
 
-gboolean gst_amr_parse_start (GstBaseParse * parse);
+static gboolean gst_amr_parse_start (GstBaseParse * parse);
-gboolean gst_amr_parse_stop (GstBaseParse * parse);
+static gboolean gst_amr_parse_stop (GstBaseParse * parse);
 
 static gboolean gst_amr_parse_sink_setcaps (GstBaseParse * parse,
     GstCaps * caps);
+static GstCaps *gst_amr_parse_sink_getcaps (GstBaseParse * parse,
+    GstCaps * filter);
 
-gboolean gst_amr_parse_check_valid_frame (GstBaseParse * parse,
+static gboolean gst_amr_parse_check_valid_frame (GstBaseParse * parse,
     GstBaseParseFrame * frame, guint * framesize, gint * skipsize);
 
-GstFlowReturn gst_amr_parse_parse_frame (GstBaseParse * parse,
+static GstFlowReturn gst_amr_parse_parse_frame (GstBaseParse * parse,
     GstBaseParseFrame * frame);
 
 G_DEFINE_TYPE (GstAmrParse, gst_amr_parse, GST_TYPE_BASE_PARSE);
@@ -112,6 +114,7 @@ gst_amr_parse_class_init (GstAmrParseClass * klass)
   parse_class->start = GST_DEBUG_FUNCPTR (gst_amr_parse_start);
   parse_class->stop = GST_DEBUG_FUNCPTR (gst_amr_parse_stop);
   parse_class->set_sink_caps = GST_DEBUG_FUNCPTR (gst_amr_parse_sink_setcaps);
+  parse_class->get_sink_caps = GST_DEBUG_FUNCPTR (gst_amr_parse_sink_getcaps);
   parse_class->parse_frame = GST_DEBUG_FUNCPTR (gst_amr_parse_parse_frame);
   parse_class->check_valid_frame =
       GST_DEBUG_FUNCPTR (gst_amr_parse_check_valid_frame);
@@ -251,7 +254,7 @@ gst_amr_parse_parse_header (GstAmrParse * amrparse,
  *
  * Returns: TRUE if the given data contains valid frame.
  */
-gboolean
+static gboolean
 gst_amr_parse_check_valid_frame (GstBaseParse * parse,
     GstBaseParseFrame * frame, guint * framesize, gint * skipsize)
 {
@@ -338,7 +341,7 @@ done:
  *
  * Returns: #GstFlowReturn defining the parsing status.
  */
-GstFlowReturn
+static GstFlowReturn
 gst_amr_parse_parse_frame (GstBaseParse * parse, GstBaseParseFrame * frame)
 {
   return GST_FLOW_OK;
@@ -353,7 +356,7 @@ gst_amr_parse_parse_frame (GstBaseParse * parse, GstBaseParseFrame * frame)
  *
  * Returns: TRUE on success.
  */
-gboolean
+static gboolean
 gst_amr_parse_start (GstBaseParse * parse)
 {
   GstAmrParse *amrparse;
@@ -374,7 +377,7 @@ gst_amr_parse_start (GstBaseParse * parse)
  *
  * Returns: TRUE on success.
  */
-gboolean
+static gboolean
 gst_amr_parse_stop (GstBaseParse * parse)
 {
   GstAmrParse *amrparse;
@@ -385,3 +388,41 @@ gst_amr_parse_stop (GstBaseParse * parse)
   amrparse->header = 0;
   return TRUE;
 }
+
+static GstCaps *
+gst_amr_parse_sink_getcaps (GstBaseParse * parse, GstCaps * filter)
+{
+  GstCaps *peercaps;
+  GstCaps *res;
+
+  /* FIXME: handle filter caps */
+
+  peercaps = gst_pad_get_allowed_caps (GST_BASE_PARSE_SRC_PAD (parse));
+  if (peercaps) {
+    guint i, n;
+
+    /* Rename structure names */
+    peercaps = gst_caps_make_writable (peercaps);
+    n = gst_caps_get_size (peercaps);
+    for (i = 0; i < n; i++) {
+      GstStructure *s = gst_caps_get_structure (peercaps, i);
+
+      if (gst_structure_has_name (s, "audio/AMR"))
+        gst_structure_set_name (s, "audio/x-amr-nb-sh");
+      else
+        gst_structure_set_name (s, "audio/x-amr-wb-sh");
+    }
+
+    res =
+        gst_caps_intersect_full (peercaps,
+        gst_pad_get_pad_template_caps (GST_BASE_PARSE_SRC_PAD (parse)),
+        GST_CAPS_INTERSECT_FIRST);
+    gst_caps_unref (peercaps);
+  } else {
+    res =
+        gst_caps_copy (gst_pad_get_pad_template_caps (GST_BASE_PARSE_SINK_PAD
+            (parse)));
+  }
+
+  return res;
+}

gst/audioparsers/gstdcaparse.c

@@ -78,6 +78,8 @@ static gboolean gst_dca_parse_check_valid_frame (GstBaseParse * parse,
     GstBaseParseFrame * frame, guint * size, gint * skipsize);
 static GstFlowReturn gst_dca_parse_parse_frame (GstBaseParse * parse,
     GstBaseParseFrame * frame);
+static GstCaps *gst_dca_parse_get_sink_caps (GstBaseParse * parse,
+    GstCaps * filter);
 
 #define gst_dca_parse_parent_class parent_class
 G_DEFINE_TYPE (GstDcaParse, gst_dca_parse, GST_TYPE_BASE_PARSE);
@@ -99,6 +101,7 @@ gst_dca_parse_class_init (GstDcaParseClass * klass)
   parse_class->check_valid_frame =
       GST_DEBUG_FUNCPTR (gst_dca_parse_check_valid_frame);
   parse_class->parse_frame = GST_DEBUG_FUNCPTR (gst_dca_parse_parse_frame);
+  parse_class->get_sink_caps = GST_DEBUG_FUNCPTR (gst_dca_parse_get_sink_caps);
 
   gst_element_class_add_pad_template (element_class,
       gst_static_pad_template_get (&sink_template));
@@ -464,3 +467,38 @@ broken_header:
     return GST_FLOW_ERROR;
   }
 }
+
+static GstCaps *
+gst_dca_parse_get_sink_caps (GstBaseParse * parse, GstCaps * filter)
+{
+  GstCaps *peercaps;
+  GstCaps *res;
+
+  /* FIXME: handle filter caps */
+
+  peercaps = gst_pad_get_allowed_caps (GST_BASE_PARSE_SRC_PAD (parse));
+  if (peercaps) {
+    guint i, n;
+
+    /* Remove the framed field */
+    peercaps = gst_caps_make_writable (peercaps);
+    n = gst_caps_get_size (peercaps);
+    for (i = 0; i < n; i++) {
+      GstStructure *s = gst_caps_get_structure (peercaps, i);
+
+      gst_structure_remove_field (s, "framed");
+    }
+
+    res =
+        gst_caps_intersect_full (peercaps,
+        gst_pad_get_pad_template_caps (GST_BASE_PARSE_SRC_PAD (parse)),
+        GST_CAPS_INTERSECT_FIRST);
+    gst_caps_unref (peercaps);
+  } else {
+    res =
+        gst_caps_copy (gst_pad_get_pad_template_caps (GST_BASE_PARSE_SINK_PAD
+            (parse)));
+  }
+
+  return res;
+}

gst/audioparsers/gstflacparse.c

@@ -201,6 +201,8 @@ static GstFlowReturn gst_flac_parse_pre_push_frame (GstBaseParse * parse,
 static gboolean gst_flac_parse_convert (GstBaseParse * parse,
     GstFormat src_format, gint64 src_value, GstFormat dest_format,
     gint64 * dest_value);
+static GstCaps *gst_flac_parse_get_sink_caps (GstBaseParse * parse,
+    GstCaps * filter);
 
 #define gst_flac_parse_parent_class parent_class
 G_DEFINE_TYPE (GstFlacParse, gst_flac_parse, GST_TYPE_BASE_PARSE);
@@ -233,6 +235,8 @@ gst_flac_parse_class_init (GstFlacParseClass * klass)
   baseparse_class->pre_push_frame =
       GST_DEBUG_FUNCPTR (gst_flac_parse_pre_push_frame);
   baseparse_class->convert = GST_DEBUG_FUNCPTR (gst_flac_parse_convert);
+  baseparse_class->get_sink_caps =
+      GST_DEBUG_FUNCPTR (gst_flac_parse_get_sink_caps);
 
   gst_element_class_add_pad_template (element_class,
       gst_static_pad_template_get (&src_factory));
@@ -1502,3 +1506,38 @@ gst_flac_parse_convert (GstBaseParse * parse,
   return GST_BASE_PARSE_CLASS (parent_class)->convert (parse, src_format,
       src_value, dest_format, dest_value);
 }
+
+static GstCaps *
+gst_flac_parse_get_sink_caps (GstBaseParse * parse, GstCaps * filter)
+{
+  GstCaps *peercaps;
+  GstCaps *res;
+
+  /* FIXME: handle filter caps */
+
+  peercaps = gst_pad_get_allowed_caps (GST_BASE_PARSE_SRC_PAD (parse));
+  if (peercaps) {
+    guint i, n;
+
+    /* Remove the framed field */
+    peercaps = gst_caps_make_writable (peercaps);
+    n = gst_caps_get_size (peercaps);
+    for (i = 0; i < n; i++) {
+      GstStructure *s = gst_caps_get_structure (peercaps, i);
+
+      gst_structure_remove_field (s, "framed");
+    }
+
+    res =
+        gst_caps_intersect_full (peercaps,
+        gst_pad_get_pad_template_caps (GST_BASE_PARSE_SRC_PAD (parse)),
+        GST_CAPS_INTERSECT_FIRST);
+    gst_caps_unref (peercaps);
+  } else {
+    res =
+        gst_caps_copy (gst_pad_get_pad_template_caps (GST_BASE_PARSE_SINK_PAD
+            (parse)));
+  }
+
+  return res;
+}

gst/audioparsers/gstmpegaudioparse.c

@@ -100,6 +100,8 @@ static GstFlowReturn gst_mpeg_audio_parse_pre_push_frame (GstBaseParse * parse,
 static gboolean gst_mpeg_audio_parse_convert (GstBaseParse * parse,
     GstFormat src_format, gint64 src_value,
     GstFormat dest_format, gint64 * dest_value);
+static GstCaps *gst_mpeg_audio_parse_get_sink_caps (GstBaseParse * parse,
+    GstCaps * filter);
 
 #define gst_mpeg_audio_parse_parent_class parent_class
 G_DEFINE_TYPE (GstMpegAudioParse, gst_mpeg_audio_parse, GST_TYPE_BASE_PARSE);
@@ -161,6 +163,8 @@ gst_mpeg_audio_parse_class_init (GstMpegAudioParseClass * klass)
   parse_class->pre_push_frame =
       GST_DEBUG_FUNCPTR (gst_mpeg_audio_parse_pre_push_frame);
   parse_class->convert = GST_DEBUG_FUNCPTR (gst_mpeg_audio_parse_convert);
+  parse_class->get_sink_caps =
+      GST_DEBUG_FUNCPTR (gst_mpeg_audio_parse_get_sink_caps);
 
   /* register tags */
 #define GST_TAG_CRC      "has-crc"
@@ -1288,3 +1292,38 @@ gst_mpeg_audio_parse_pre_push_frame (GstBaseParse * parse,
 
   return GST_FLOW_OK;
 }
+
+static GstCaps *
+gst_mpeg_audio_parse_get_sink_caps (GstBaseParse * parse, GstCaps * filter)
+{
+  GstCaps *peercaps;
+  GstCaps *res;
+
+  /* FIXME: handle filter caps */
+
+  peercaps = gst_pad_get_allowed_caps (GST_BASE_PARSE_SRC_PAD (parse));
+  if (peercaps) {
+    guint i, n;
+
+    /* Remove the parsed field */
+    peercaps = gst_caps_make_writable (peercaps);
+    n = gst_caps_get_size (peercaps);
+    for (i = 0; i < n; i++) {
+      GstStructure *s = gst_caps_get_structure (peercaps, i);
+
+      gst_structure_remove_field (s, "parsed");
+    }
+
+    res =
+        gst_caps_intersect_full (peercaps,
+        gst_pad_get_pad_template_caps (GST_BASE_PARSE_SRC_PAD (parse)),
+        GST_CAPS_INTERSECT_FIRST);
+    gst_caps_unref (peercaps);
+  } else {
+    res =
+        gst_caps_copy (gst_pad_get_pad_template_caps (GST_BASE_PARSE_SINK_PAD
+            (parse)));
+  }
+
+  return res;
+}

gst/effectv/gstedge.c

@@ -140,7 +140,7 @@ gst_edgetv_transform (GstBaseTransform * trans, GstBuffer * in, GstBuffer * out)
       p = *src;
       q = *(src - 4);
 
-      /* difference between the current pixel and right neighbor. */
+      /* difference between the current pixel and left neighbor. */
       r = ((p & 0xff0000) - (q & 0xff0000)) >> 16;
       g = ((p & 0xff00) - (q & 0xff00)) >> 8;
       b = (p & 0xff) - (q & 0xff);
@@ -199,8 +199,12 @@ gst_edgetv_transform (GstBaseTransform * trans, GstBuffer * in, GstBuffer * out)
       dest[width + 3] = v3;
       dest[width * 2] = v2;
       dest[width * 2 + 1] = v2;
+      dest[width * 2 + 2] = 0;
+      dest[width * 2 + 3] = 0;
       dest[width * 3] = v2;
       dest[width * 3 + 1] = v2;
+      dest[width * 3 + 2] = 0;
+      dest[width * 3 + 3] = 0;
 
       src += 4;
       dest += 4;

gst/effectv/gstquark.c

@@ -328,7 +328,4 @@ gst_quarktv_init (GstQuarkTV * filter)
 {
   filter->planes = PLANES;
   filter->current_plane = filter->planes - 1;
-
-  gst_pad_use_fixed_caps (GST_BASE_TRANSFORM_SRC_PAD (filter));
-  gst_pad_use_fixed_caps (GST_BASE_TRANSFORM_SINK_PAD (filter));
 }

gst/effectv/gstradioac.c

@@ -131,14 +131,15 @@ enum
 #define RATIO 0.95
 
 static guint32 palettes[COLORS * PATTERN];
+static gint swap_tab[] = { 2, 1, 0, 3 };
 
 #define gst_radioactv_parent_class parent_class
 G_DEFINE_TYPE (GstRadioacTV, gst_radioactv, GST_TYPE_VIDEO_FILTER);
 
 #if G_BYTE_ORDER == G_LITTLE_ENDIAN
-#define CAPS_STR GST_VIDEO_CAPS_MAKE ("RGBx")
+#define CAPS_STR GST_VIDEO_CAPS_MAKE ("{ RGBx, BGRx }")
 #else
-#define CAPS_STR GST_VIDEO_CAPS_MAKE ("xBGR")
+#define CAPS_STR GST_VIDEO_CAPS_MAKE ("{ xBGR, xRGB }")
 #endif
 
 static GstStaticPadTemplate gst_radioactv_src_template =
@@ -162,18 +163,20 @@ makePalette (void)
 
 #define DELTA (255/(COLORS/2-1))
 
+  /* red, gree, blue */
   for (i = 0; i < COLORS / 2; i++) {
     palettes[i] = i * DELTA;
     palettes[COLORS + i] = (i * DELTA) << 8;
     palettes[COLORS * 2 + i] = (i * DELTA) << 16;
   }
   for (i = 0; i < COLORS / 2; i++) {
-    palettes[+i + COLORS / 2] = 255 | (i * DELTA) << 16 | (i * DELTA) << 8;
+    palettes[i + COLORS / 2] = 255 | (i * DELTA) << 16 | (i * DELTA) << 8;
     palettes[COLORS + i + COLORS / 2] =
         (255 << 8) | (i * DELTA) << 16 | i * DELTA;
     palettes[COLORS * 2 + i + COLORS / 2] =
         (255 << 16) | (i * DELTA) << 8 | i * DELTA;
   }
+  /* white */
   for (i = 0; i < COLORS; i++) {
     palettes[COLORS * 3 + i] = (255 * i / COLORS) * 0x10101;
   }
@@ -347,7 +350,19 @@ gst_radioactv_transform (GstBaseTransform * trans, GstBuffer * in,
   height = GST_VIDEO_FRAME_HEIGHT (&in_frame);
 
   GST_OBJECT_LOCK (filter);
-  palette = &palettes[COLORS * filter->color];
+#if G_BYTE_ORDER == G_LITTLE_ENDIAN
+  if (GST_VIDEO_FRAME_FORMAT (&in_frame) == GST_VIDEO_FORMAT_RGBx) {
+    palette = &palettes[COLORS * filter->color];
+  } else {
+    palette = &palettes[COLORS * swap_tab[filter->color]];
+  }
+#else
+  if (GST_VIDEO_FRAME_FORMAT (&in_frame) == GST_VIDEO_FORMAT_xBGR) {
+    palette = &palettes[COLORS * filter->color];
+  } else {
+    palette = &palettes[COLORS * swap_tab[filter->color]];
+  }
+#endif
   diff = filter->diff;
 
   if (filter->mode == 3 && filter->trigger)
@@ -643,7 +658,4 @@ gst_radioactv_init (GstRadioacTV * filter)
   filter->color = DEFAULT_COLOR;
   filter->interval = DEFAULT_INTERVAL;
   filter->trigger = DEFAULT_TRIGGER;
-
-  gst_pad_use_fixed_caps (GST_BASE_TRANSFORM_SRC_PAD (filter));
-  gst_pad_use_fixed_caps (GST_BASE_TRANSFORM_SINK_PAD (filter));
 }

gst/effectv/gstripple.c

@@ -311,9 +311,9 @@ gst_rippletv_transform (GstBaseTransform * trans, GstBuffer * in,
   guint32 *src, *dest;
   GstVideoFrame in_frame, out_frame;
   gint x, y, i;
-  gint dx, dy;
+  gint dx, dy, o_dx;
   gint h, v;
-  gint width, height;
+  gint m_w, m_h, v_w, v_h;
   gint *p, *q, *r;
   gint8 *vp;
   GstClockTime timestamp, stream_time;
@@ -344,22 +344,24 @@ gst_rippletv_transform (GstBaseTransform * trans, GstBuffer * in,
   else
     motiondetect (filter, src);
 
+  m_w = filter->map_w;
+  m_h = filter->map_h;
+  v_w = GST_VIDEO_FRAME_WIDTH (&in_frame);
+  v_h = GST_VIDEO_FRAME_HEIGHT (&in_frame);
+
   /* simulate surface wave */
-  width = filter->map_w;
-  height = filter->map_h;
 
   /* This function is called only 30 times per second. To increase a speed
    * of wave, iterates this loop several times. */
   for (i = loopnum; i > 0; i--) {
     /* wave simulation */
-    p = filter->map1 + width + 1;
+    p = filter->map1 + m_w + 1;
-    q = filter->map2 + width + 1;
+    q = filter->map2 + m_w + 1;
-    r = filter->map3 + width + 1;
+    r = filter->map3 + m_w + 1;
-    for (y = height - 2; y > 0; y--) {
+    for (y = m_h - 2; y > 0; y--) {
-      for (x = width - 2; x > 0; x--) {
+      for (x = m_w - 2; x > 0; x--) {
-        h = *(p - width - 1) + *(p - width + 1) + *(p + width - 1) + *(p +
+        h = *(p - m_w - 1) + *(p - m_w + 1) + *(p + m_w - 1) + *(p + m_w + 1)
-            width + 1)
+            + *(p - m_w) + *(p - 1) + *(p + 1) + *(p + m_w) - (*p) * 9;
-            + *(p - width) + *(p - 1) + *(p + 1) + *(p + width) - (*p) * 9;
         h = h >> 3;
         v = *p - *q;
         v += h - (v >> decay);
@@ -374,11 +376,11 @@ gst_rippletv_transform (GstBaseTransform * trans, GstBuffer * in,
     }
 
     /* low pass filter */
-    p = filter->map3 + width + 1;
+    p = filter->map3 + m_w + 1;
-    q = filter->map2 + width + 1;
+    q = filter->map2 + m_w + 1;
-    for (y = height - 2; y > 0; y--) {
+    for (y = m_h - 2; y > 0; y--) {
-      for (x = width - 2; x > 0; x--) {
+      for (x = m_w - 2; x > 0; x--) {
-        h = *(p - width) + *(p - 1) + *(p + 1) + *(p + width) + (*p) * 60;
+        h = *(p - m_w) + *(p - 1) + *(p + 1) + *(p + m_w) + (*p) * 60;
         *q = h >> 6;
         p++;
         q++;
@@ -394,12 +396,12 @@ gst_rippletv_transform (GstBaseTransform * trans, GstBuffer * in,
 
   vp = filter->vtable;
   p = filter->map1;
-  for (y = height - 1; y > 0; y--) {
+  for (y = m_h - 1; y > 0; y--) {
-    for (x = width - 1; x > 0; x--) {
+    for (x = m_w - 1; x > 0; x--) {
       /* difference of the height between two voxel. They are twiced to
        * emphasise the wave. */
       vp[0] = sqrtable[((p[0] - p[1]) >> (point - 1)) & 0xff];
-      vp[1] = sqrtable[((p[0] - p[width]) >> (point - 1)) & 0xff];
+      vp[1] = sqrtable[((p[0] - p[m_w]) >> (point - 1)) & 0xff];
       p++;
       vp += 2;
     }
@@ -407,48 +409,34 @@ gst_rippletv_transform (GstBaseTransform * trans, GstBuffer * in,
     vp += 2;
   }
 
-  width = GST_VIDEO_FRAME_WIDTH (&in_frame);
-  height = GST_VIDEO_FRAME_HEIGHT (&in_frame);
   vp = filter->vtable;
 
   /* draw refracted image. The vector table is stretched. */
-  for (y = 0; y < height; y += 2) {
+  for (y = 0; y < v_h; y += 2) {
-    for (x = 0; x < width; x += 2) {
+    for (x = 0; x < v_w; x += 2) {
       h = (gint) vp[0];
       v = (gint) vp[1];
       dx = x + h;
       dy = y + v;
-      if (dx < 0)
+      dx = CLAMP (dx, 0, (v_w - 1));
-        dx = 0;
+      dy = CLAMP (dy, 0, (v_h - 1));
-      if (dy < 0)
+      dest[0] = src[dy * v_w + dx];
-        dy = 0;
-      if (dx >= width)
-        dx = width - 1;
-      if (dy >= height)
-        dy = height - 1;
-      dest[0] = src[dy * width + dx];
 
-      i = dx;
+      o_dx = dx;
 
       dx = x + 1 + (h + (gint) vp[2]) / 2;
-      if (dx < 0)
+      dx = CLAMP (dx, 0, (v_w - 1));
-        dx = 0;
+      dest[1] = src[dy * v_w + dx];
-      if (dx >= width)
-        dx = width - 1;
-      dest[1] = src[dy * width + dx];
 
-      dy = y + 1 + (v + (gint) vp[filter->map_w * 2 + 1]) / 2;
+      dy = y + 1 + (v + (gint) vp[m_w * 2 + 1]) / 2;
-      if (dy < 0)
+      dy = CLAMP (dy, 0, (v_h - 1));
-        dy = 0;
+      dest[v_w] = src[dy * v_w + o_dx];
-      if (dy >= height)
-        dy = height - 1;
-      dest[width] = src[dy * width + i];
 
-      dest[width + 1] = src[dy * width + dx];
+      dest[v_w + 1] = src[dy * v_w + dx];
       dest += 2;
       vp += 2;
     }
-    dest += width;
+    dest += v_w;
     vp += 2;
   }
   GST_OBJECT_UNLOCK (filter);
@@ -493,9 +481,11 @@ gst_rippletv_set_caps (GstBaseTransform * btrans, GstCaps * incaps,
   filter->map_h = height / 2 + 1;
   filter->map_w = width / 2 + 1;
 
+  /* we over allocate the buffers, as the render code does not handle clipping
+   * very well */
   if (filter->map)
     g_free (filter->map);
-  filter->map = g_new0 (gint, filter->map_h * filter->map_w * 3);
+  filter->map = g_new0 (gint, (1 + filter->map_h) * filter->map_w * 3);
 
   filter->map1 = filter->map;
   filter->map2 = filter->map + filter->map_w * filter->map_h;
@@ -503,15 +493,15 @@ gst_rippletv_set_caps (GstBaseTransform * btrans, GstCaps * incaps,
 
   if (filter->vtable)
     g_free (filter->vtable);
-  filter->vtable = g_new0 (gint8, filter->map_h * filter->map_w * 2);
+  filter->vtable = g_new0 (gint8, (1 + filter->map_h) * filter->map_w * 2);
 
   if (filter->background)
     g_free (filter->background);
-  filter->background = g_new0 (gint16, width * height);
+  filter->background = g_new0 (gint16, width * (height + 1));
 
   if (filter->diff)
     g_free (filter->diff);
-  filter->diff = g_new0 (guint8, width * height);
+  filter->diff = g_new0 (guint8, width * (height + 1));
   GST_OBJECT_UNLOCK (filter);
 
   return TRUE;
@@ -650,6 +640,7 @@ gst_rippletv_init (GstRippleTV * filter)
 {
   filter->mode = DEFAULT_MODE;
 
+  /* FIXME: remove this when memory corruption after resizes are fixed */
   gst_pad_use_fixed_caps (GST_BASE_TRANSFORM_SRC_PAD (filter));
   gst_pad_use_fixed_caps (GST_BASE_TRANSFORM_SINK_PAD (filter));
 }

gst/effectv/gstshagadelic.c

@@ -289,7 +289,4 @@ gst_shagadelictv_init (GstShagadelicTV * filter)
 {
   filter->ripple = NULL;
   filter->spiral = NULL;
-
-  gst_pad_use_fixed_caps (GST_BASE_TRANSFORM_SRC_PAD (filter));
-  gst_pad_use_fixed_caps (GST_BASE_TRANSFORM_SINK_PAD (filter));
 }

gst/effectv/gststreak.c

@@ -296,7 +296,4 @@ static void
 gst_streaktv_init (GstStreakTV * filter)
 {
   filter->feedback = DEFAULT_FEEDBACK;
-
-  gst_pad_use_fixed_caps (GST_BASE_TRANSFORM_SRC_PAD (filter));
-  gst_pad_use_fixed_caps (GST_BASE_TRANSFORM_SINK_PAD (filter));
 }

gst/effectv/gstvertigo.c

@@ -351,7 +351,4 @@ gst_vertigotv_init (GstVertigoTV * filter)
   filter->phase = 0.0;
   filter->phase_increment = 0.02;
   filter->zoomrate = 1.01;
-
-  gst_pad_use_fixed_caps (GST_BASE_TRANSFORM_SRC_PAD (filter));
-  gst_pad_use_fixed_caps (GST_BASE_TRANSFORM_SINK_PAD (filter));
 }

gst/effectv/gstwarp.c

@@ -133,11 +133,7 @@ initDistTable (GstWarpTV * filter, gint width, gint height)
 {
   gint32 halfw, halfh, *distptr;
   gint x, y;
-#ifdef PS2
   float m;
-#else
-  float m;
-#endif
 
   halfw = width >> 1;
   halfh = height >> 1;
@@ -304,6 +300,5 @@ gst_warptv_class_init (GstWarpTVClass * klass)
 static void
 gst_warptv_init (GstWarpTV * warptv)
 {
-  gst_pad_use_fixed_caps (GST_BASE_TRANSFORM_SRC_PAD (warptv));
+  /* nothing to do */
-  gst_pad_use_fixed_caps (GST_BASE_TRANSFORM_SINK_PAD (warptv));
 }

gst/flv/gstflvdemux.c

@@ -781,10 +781,10 @@ gst_flv_demux_push_tags (GstFlvDemux * demux)
 }
 
 static void
-gst_flv_demux_update_resync (GstFlvDemux * demux, guint32 pts, guint32 * last,
+gst_flv_demux_update_resync (GstFlvDemux * demux, guint32 pts, gboolean discont,
-    GstClockTime * offset)
+    guint32 * last, GstClockTime * offset)
 {
-  if (ABS (pts - *last) >= RESYNC_THRESHOLD) {
+  if (!discont && ABS (pts - *last) >= RESYNC_THRESHOLD) {
     /* Theoretically, we should use substract the duration of the last buffer,
        but this demuxer sends no durations on buffers, not sure if it cannot
        know, or just does not care to calculate. */
@@ -992,8 +992,8 @@ gst_flv_demux_parse_tag_audio (GstFlvDemux * demux, GstBuffer * buffer)
   }
 
   /* detect (and deem to be resyncs)  large pts gaps */
-  gst_flv_demux_update_resync (demux, pts, &demux->last_audio_pts,
+  gst_flv_demux_update_resync (demux, pts, demux->audio_need_discont,
-      &demux->audio_time_offset);
+      &demux->last_audio_pts, &demux->audio_time_offset);
 
   /* Fill buffer with data */
   GST_BUFFER_TIMESTAMP (outbuf) = pts * GST_MSECOND + demux->audio_time_offset;
@@ -1369,8 +1369,8 @@ gst_flv_demux_parse_tag_video (GstFlvDemux * demux, GstBuffer * buffer)
   }
 
   /* detect (and deem to be resyncs)  large pts gaps */
-  gst_flv_demux_update_resync (demux, pts, &demux->last_video_pts,
+  gst_flv_demux_update_resync (demux, pts, demux->video_need_discont,
-      &demux->video_time_offset);
+      &demux->last_video_pts, &demux->video_time_offset);
 
   /* Fill buffer with data */
   GST_BUFFER_TIMESTAMP (outbuf) = pts * GST_MSECOND + demux->video_time_offset;

gst/isomp4/gstqtmuxmap.c

@@ -199,7 +199,7 @@ GstQTMuxFormatProp gst_qt_mux_format_list[] = {
         "ismlmux",
         "ISML",
         "GstISMLMux",
-        GST_STATIC_CAPS ("video/quicktime, variant = (string) iso"),
+        GST_STATIC_CAPS ("video/quicktime, variant = (string) iso-fragmented"),
         GST_STATIC_CAPS (MPEG4V_CAPS "; " H264_CAPS),
         GST_STATIC_CAPS (MP3_CAPS "; " AAC_CAPS)
       }

gst/multifile/gstmultifilesink.c

@@ -594,6 +594,7 @@ gst_multi_file_sink_ensure_max_files (GstMultiFileSink * multifilesink)
       multifilesink->n_files >= multifilesink->max_files) {
     filename = multifilesink->files->data;
     g_remove (filename);
+    g_free (filename);
     multifilesink->files = g_slist_delete_link (multifilesink->files,
         multifilesink->files);
     multifilesink->n_files -= 1;

gst/wavparse/gstwavparse.c

@@ -1164,7 +1164,8 @@ gst_wavparse_stream_headers (GstWavParse * wav)
 
     if (tag == GST_RIFF_TAG_JUNK || tag == GST_RIFF_TAG_JUNQ ||
         tag == GST_RIFF_TAG_bext || tag == GST_RIFF_TAG_BEXT ||
-        tag == GST_RIFF_TAG_LIST) {
+        tag == GST_RIFF_TAG_LIST || tag == GST_RIFF_TAG_ID32 ||
+        tag == GST_RIFF_TAG_IDVX) {
       GST_DEBUG_OBJECT (wav, "skipping %" GST_FOURCC_FORMAT " chunk",
           GST_FOURCC_ARGS (tag));
       gst_buffer_unref (buf);

tests/check/elements/multifile.c

@@ -40,19 +40,19 @@ run_pipeline (GstElement * pipeline)
   gst_element_set_state (pipeline, GST_STATE_NULL);
 }
 
+#if !GLIB_CHECK_VERSION(2,26,0)
 static gchar *
-g_mkdtemp (const gchar * template)
+g_mkdtemp (gchar * template)
 {
-  gchar *s;
   gchar *tmpdir;
 
-  s = g_strdup (template);
+  tmpdir = mkdtemp (template);
-  tmpdir = mkdtemp (s);
   if (tmpdir == NULL) {
-    g_free (s);
+    g_free (template);
   }
   return tmpdir;
 }
+#endif
 
 GST_START_TEST (test_multifilesink_key_frame)
 {
@@ -93,7 +93,6 @@ GST_START_TEST (test_multifilesink_key_frame)
 
   g_free (mfs_pattern);
   g_free (my_tmpdir);
-  g_free (template);
 }
 
 GST_END_TEST;
@@ -144,7 +143,6 @@ GST_START_TEST (test_multifilesink_max_files)
 
   g_free (mfs_pattern);
   g_free (my_tmpdir);
-  g_free (template);
 }
 
 GST_END_TEST;
@@ -202,7 +200,6 @@ GST_START_TEST (test_multifilesrc)
 
   g_free (mfs_pattern);
   g_free (my_tmpdir);
-  g_free (template);
 }
 
 GST_END_TEST;