Merge branch 'master' into 0.11

Conflicts:
	ext/flac/gstflacenc.c
	ext/jack/gstjackaudioclient.c
	ext/jack/gstjackaudiosink.c
	ext/jack/gstjackaudiosrc.c
	ext/pulse/plugin.c
	ext/shout2/gstshout2.c
	gst/matroska/matroska-mux.c
	gst/rtp/gstrtph264pay.c

diff --git a/ext/cairo/Makefile.am b/ext/cairo/Makefile.am
index 5a8d3be..7fa8fdc 100644 (file)
--- a/ext/cairo/Makefile.am
+++ b/ext/cairo/Makefile.am
@@ -36,8 +36,8 @@ libgstcairo_la_CFLAGS = \
        $(GST_BASE_CFLAGS) \
        $(GST_CFLAGS) $(CAIRO_CFLAGS) $(CAIRO_GOBJECT_CFLAGS)
 libgstcairo_la_LIBADD = \
-       $(GST_BASE_LIBS) -lgstvideo-$(GST_MAJORMINOR) \
-       $(GST_LIBS) $(CAIRO_LIBS) $(CAIRO_GOBJECT_LIBS) $(LIBM)
+       $(GST_PLUGINS_BASE_LIBS) -lgstvideo-$(GST_MAJORMINOR) \
+       $(GST_BASE_LIBS) $(GST_LIBS) $(CAIRO_LIBS) $(CAIRO_GOBJECT_LIBS) $(LIBM)
 libgstcairo_la_LDFLAGS = $(GST_PLUGIN_LDFLAGS)
 libgstcairo_la_LIBTOOLFLAGS = --tag=disable-static
 

diff --git a/ext/flac/gstflacenc.c b/ext/flac/gstflacenc.c
index cbd8f2d..e9cbc7c 100644 (file)
--- a/ext/flac/gstflacenc.c
+++ b/ext/flac/gstflacenc.c
@@ -839,7 +839,6 @@ gst_flac_enc_seek_callback (const FLAC__StreamEncoder * encoder,
     FLAC__uint64 absolute_byte_offset, void *client_data)
 {
   GstFlacEnc *flacenc;
-  GstEvent *event;
   GstPad *peerpad;
   GstSegment seg;
 
@@ -848,15 +847,17 @@ gst_flac_enc_seek_callback (const FLAC__StreamEncoder * encoder,
   if (flacenc->stopped)
     return FLAC__STREAM_ENCODER_SEEK_STATUS_OK;
 
-  gst_segment_init (&seg, GST_FORMAT_BYTES);
-  seg.start = absolute_byte_offset;
-  seg.stop = GST_BUFFER_OFFSET_NONE;
-  seg.time = 0;
-  event = gst_event_new_segment (&seg);
-
   if ((peerpad = gst_pad_get_peer (GST_AUDIO_ENCODER_SRC_PAD (flacenc)))) {
-    gboolean ret = gst_pad_send_event (peerpad, event);
+    GstEvent *event;
+    gboolean ret;
+
+    gst_segment_init (&seg, GST_FORMAT_BYTES);
+    seg.start = absolute_byte_offset;
+    seg.stop = GST_BUFFER_OFFSET_NONE;
+    seg.time = 0;
+    event = gst_event_new_segment (&seg);
 
+    ret = gst_pad_send_event (peerpad, event);
     gst_object_unref (peerpad);
 
     if (ret) {

diff --git a/ext/jack/gstjackaudioclient.c b/ext/jack/gstjackaudioclient.c
index 243b7ff..79a24c3 100644 (file)
--- a/ext/jack/gstjackaudioclient.c
+++ b/ext/jack/gstjackaudioclient.c
@@ -22,6 +22,7 @@
 #include <string.h>
 
 #include "gstjackaudioclient.h"
+#include "gstjack.h"
 
 #include <gst/glib-compat-private.h>
 
@@ -77,14 +78,29 @@ struct _GstJackAudioClient
   gpointer user_data;
 };
 
-typedef jack_default_audio_sample_t sample_t;
-
 typedef struct
 {
   jack_nframes_t nframes;
   gpointer user_data;
 } JackCB;
 
+static gboolean
+jack_handle_transport_change (GstJackAudioClient * client, GstState state)
+{
+  GstObject *obj = GST_OBJECT_PARENT (client->user_data);
+  GstJackTransport mode;
+
+  g_object_get (obj, "transport", &mode, NULL);
+  if ((mode == GST_JACK_TRANSPORT_SLAVE) && (GST_STATE (obj) != state)) {
+    GST_INFO_OBJECT (obj, "requesting state change: %s",
+        gst_element_state_get_name (state));
+    gst_element_post_message (GST_ELEMENT (obj),
+        gst_message_new_request_state (obj, state));
+    return TRUE;
+  }
+  return FALSE;
+}
+
 static int
 jack_process_cb (jack_nframes_t nframes, void *arg)
 {
@@ -111,6 +127,8 @@ jack_process_cb (jack_nframes_t nframes, void *arg)
       default:
         break;
     }
+    GST_DEBUG ("num of clients: src=%d, sink=%d",
+        g_list_length (conn->src_clients), g_list_length (conn->sink_clients));
   }
 
   g_mutex_lock (&conn->lock);
@@ -141,7 +159,29 @@ jack_process_cb (jack_nframes_t nframes, void *arg)
       }
     }
   }
+
+  /* handle transport state requisition, do sinks first, stop after the first
+   * element that handled it */
+  if (conn->transport_state != GST_STATE_VOID_PENDING) {
+    for (walk = conn->sink_clients; walk; walk = g_list_next (walk)) {
+      if (jack_handle_transport_change ((GstJackAudioClient *) walk->data,
+              conn->transport_state)) {
+        conn->transport_state = GST_STATE_VOID_PENDING;
+        break;
+      }
+    }
+  }
+  if (conn->transport_state != GST_STATE_VOID_PENDING) {
+    for (walk = conn->src_clients; walk; walk = g_list_next (walk)) {
+      if (jack_handle_transport_change ((GstJackAudioClient *) walk->data,
+              conn->transport_state)) {
+        conn->transport_state = GST_STATE_VOID_PENDING;
+        break;
+      }
+    }
+  }
   g_mutex_unlock (&conn->lock);
+
   return res;
 }
 
@@ -259,6 +299,7 @@ gst_jack_audio_make_connection (const gchar * id, const gchar * server,
   jack_on_shutdown (jclient, jack_shutdown_cb, conn);
 
   /* all callbacks are set, activate the client */
+  GST_INFO ("activate jack_client %p", jclient);
   if ((res = jack_activate (jclient)))
     goto could_not_activate;
 
@@ -353,6 +394,7 @@ gst_jack_audio_unref_connection (GstJackAudioConnection * conn)
      *      waiting for the JACK thread, and can thus cause deadlock in 
      *      jack_process_cb()
      */
+    GST_INFO ("deactivate jack_client %p", conn->client);
     if ((res = jack_deactivate (conn->client))) {
       /* we only warn, this means the server is probably shut down and the client
        * is gone anyway. */

diff --git a/ext/jack/gstjackaudiosink.c b/ext/jack/gstjackaudiosink.c
index 8bdb664..c4b3afd 100644 (file)
--- a/ext/jack/gstjackaudiosink.c
+++ b/ext/jack/gstjackaudiosink.c
@@ -197,18 +197,6 @@ jack_process_cb (jack_nframes_t nframes, void *arg)
 
   channels = GST_AUDIO_INFO_CHANNELS (&buf->spec.info);
 
-  /* handle transport state requisitions */
-  if (sink->transport == GST_JACK_TRANSPORT_SLAVE) {
-    GstState state = gst_jack_audio_client_get_transport_state (sink->client);
-
-    if ((state != GST_STATE_VOID_PENDING) && (GST_STATE (sink) != state)) {
-      GST_DEBUG_OBJECT (sink, "requesting state change: %s",
-          gst_element_state_get_name (state));
-      gst_element_post_message (GST_ELEMENT (sink),
-          gst_message_new_request_state (GST_OBJECT (sink), state));
-    }
-  }
-
   /* get target buffers */
   for (i = 0; i < channels; i++) {
     sink->buffers[i] =

diff --git a/ext/jack/gstjackaudiosrc.c b/ext/jack/gstjackaudiosrc.c
index 417028c..d0e7646 100644 (file)
--- a/ext/jack/gstjackaudiosrc.c
+++ b/ext/jack/gstjackaudiosrc.c
@@ -217,18 +217,6 @@ jack_process_cb (jack_nframes_t nframes, void *arg)
 
   channels = GST_AUDIO_INFO_CHANNELS (&buf->spec.info);
 
-  /* handle transport state requisitions */
-  if (src->transport == GST_JACK_TRANSPORT_SLAVE) {
-    GstState state = gst_jack_audio_client_get_transport_state (src->client);
-
-    if ((state != GST_STATE_VOID_PENDING) && (GST_STATE (src) != state)) {
-      GST_DEBUG_OBJECT (src, "requesting state change: %s",
-          gst_element_state_get_name (state));
-      gst_element_post_message (GST_ELEMENT (src),
-          gst_message_new_request_state (GST_OBJECT (src), state));
-    }
-  }
-
   /* get input buffers */
   for (i = 0; i < channels; i++)
     src->buffers[i] =

diff --git a/ext/shout2/gstshout2.c b/ext/shout2/gstshout2.c
index 9a6363b..a0d9281 100644 (file)
--- a/ext/shout2/gstshout2.c
+++ b/ext/shout2/gstshout2.c
@@ -1,6 +1,7 @@
 /* GStreamer
  * Copyright (C) <1999> Erik Walthinsen <omega@cse.ogi.edu>
  * Copyright (C) <2006> Tim-Philipp Müller <tim centricular net>
+ * Copyright (C) <2012> Ralph Giles <giles@mozilla.com>
  *
  * This library is free software; you can redistribute it and/or
  * modify it under the terms of the GNU Library General Public
@@ -70,12 +71,17 @@ enum
 
 static GstElementClass *parent_class = NULL;
 
+#ifdef SHOUT_FORMAT_WEBM
+#define WEBM_CAPS "; video/webm"
+#else
+#define WEBM_CAPS ""
+#endif
 static GstStaticPadTemplate sink_template = GST_STATIC_PAD_TEMPLATE ("sink",
     GST_PAD_SINK,
     GST_PAD_ALWAYS,
     GST_STATIC_CAPS ("application/ogg; "
-        "audio/mpeg, mpegversion = (int) 1, layer = (int) [ 1, 3 ]")
-    );
+        "audio/mpeg, mpegversion = (int) 1, layer = (int) [ 1, 3 ]" WEBM_CAPS));
+
 static void gst_shout2send_class_init (GstShout2sendClass * klass);
 static void gst_shout2send_base_init (GstShout2sendClass * klass);
 static void gst_shout2send_init (GstShout2send * shout2send);
@@ -162,6 +168,7 @@ gst_shout2send_base_init (GstShout2sendClass * klass)
 
   gst_element_class_add_pad_template (element_class,
       gst_static_pad_template_get (&sink_template));
+
   gst_element_class_set_details_simple (element_class, "Icecast network sink",
       "Sink/Network", "Sends data to an icecast server",
       "Wim Taymans <wim.taymans@chello.be>, "
@@ -538,9 +545,14 @@ set_failed:
 static gboolean
 gst_shout2send_connect (GstShout2send * sink)
 {
-  GST_DEBUG_OBJECT (sink, "Connection format is: %s",
+  const char *format =
       (sink->audio_format == SHOUT_FORMAT_VORBIS) ? "vorbis" :
-      ((sink->audio_format == SHOUT_FORMAT_MP3) ? "mp3" : "unknown"));
+      ((sink->audio_format == SHOUT_FORMAT_MP3) ? "mp3" : "unknown");
+#ifdef SHOUT_FORMAT_WEBM
+  if (sink->audio_format == SHOUT_FORMAT_WEBM)
+    format = "webm";
+#endif
+  GST_DEBUG_OBJECT (sink, "Connection format is: %s", format);
 
   if (shout_set_format (sink->conn, sink->audio_format) != SHOUTERR_SUCCESS)
     goto could_not_set_format;
@@ -810,6 +822,10 @@ gst_shout2send_setcaps (GstPad * pad, GstCaps * caps)
     shout2send->audio_format = SHOUT_FORMAT_MP3;
   } else if (!strcmp (mimetype, "application/ogg")) {
     shout2send->audio_format = SHOUT_FORMAT_VORBIS;
+#ifdef SHOUT_FORMAT_WEBM
+  } else if (!strcmp (mimetype, "video/webm")) {
+    shout2send->audio_format = SHOUT_FORMAT_WEBM;
+#endif
   } else {
     ret = FALSE;
   }

diff --git a/gst/flv/gstflvdemux.c b/gst/flv/gstflvdemux.c
index 3d0ebf0..4c108a6 100644 (file)
--- a/gst/flv/gstflvdemux.c
+++ b/gst/flv/gstflvdemux.c
@@ -63,7 +63,7 @@ static GstStaticPadTemplate audio_src_template =
     GST_STATIC_CAPS
     ("audio/x-adpcm, layout = (string) swf, channels = (int) { 1, 2 }, rate = (int) { 5512, 11025, 22050, 44100 }; "
         "audio/mpeg, mpegversion = (int) 1, layer = (int) 3, channels = (int) { 1, 2 }, rate = (int) { 5512, 8000, 11025, 22050, 44100 }, parsed = (boolean) TRUE; "
-        "audio/mpeg, mpegversion = (int) 4, framed = (boolean) TRUE; "
+        "audio/mpeg, mpegversion = (int) 4, stream-format = (string) raw, framed = (boolean) TRUE; "
         "audio/x-nellymoser, channels = (int) { 1, 2 }, rate = (int) { 5512, 8000, 11025, 16000, 22050, 44100 }; "
         "audio/x-raw, format = (string) { U8, S16LE }, layout = (string) interleaved, channels = (int) { 1, 2 }, rate = (int) { 5512, 11025, 22050, 44100 }; "
         "audio/x-alaw, channels = (int) { 1, 2 }, rate = (int) { 5512, 11025, 22050, 44100 }; "

diff --git a/gst/flv/gstflvmux.c b/gst/flv/gstflvmux.c
index a4d7eae..6f47c9b 100644 (file)
--- a/gst/flv/gstflvmux.c
+++ b/gst/flv/gstflvmux.c
@@ -76,7 +76,8 @@ static GstStaticPadTemplate audiosink_templ = GST_STATIC_PAD_TEMPLATE ("audio",
     GST_STATIC_CAPS
     ("audio/x-adpcm, layout = (string) swf, channels = (int) { 1, 2 }, rate = (int) { 5512, 11025, 22050, 44100 }; "
         "audio/mpeg, mpegversion = (int) 1, layer = (int) 3, channels = (int) { 1, 2 }, rate = (int) { 5512, 8000, 11025, 22050, 44100 }, parsed = (boolean) TRUE; "
-        "audio/mpeg, mpegversion = (int) { 2, 4 }, framed = (boolean) TRUE; "
+        "audio/mpeg, mpegversion = (int) 2, framed = (boolean) TRUE; "
+        "audio/mpeg, mpegversion = (int) 4, stream-format = (string) raw, framed = (boolean) TRUE; "
         "audio/x-nellymoser, channels = (int) { 1, 2 }, rate = (int) { 5512, 8000, 11025, 16000, 22050, 44100 }; "
         "audio/x-raw, format = (string) { U8, S16LE}, layout = (string) interleaved, channels = (int) { 1, 2 }, rate = (int) { 5512, 11025, 22050, 44100 }; "
         "audio/x-alaw, channels = (int) { 1, 2 }, rate = (int) { 5512, 11025, 22050, 44100 }; "

diff --git a/gst/matroska/matroska-demux.c b/gst/matroska/matroska-demux.c
index 584d30d..96504fc 100644 (file)
--- a/gst/matroska/matroska-demux.c
+++ b/gst/matroska/matroska-demux.c
@@ -3549,10 +3549,12 @@ gst_matroska_demux_parse_blockgroup_or_simpleblock (GstMatroskaDemux * demux,
           demux->common.segment.duration =
               last_stop_end - demux->stream_start_time;
           GST_OBJECT_UNLOCK (demux);
-          gst_element_post_message (GST_ELEMENT_CAST (demux),
-              gst_message_new_duration (GST_OBJECT_CAST (demux),
-                  GST_FORMAT_TIME, GST_CLOCK_TIME_NONE));
-          demux->invalid_duration = TRUE;
+          if (!demux->invalid_duration) {
+            gst_element_post_message (GST_ELEMENT_CAST (demux),
+                gst_message_new_duration (GST_OBJECT_CAST (demux),
+                    GST_FORMAT_TIME, GST_CLOCK_TIME_NONE));
+            demux->invalid_duration = TRUE;
+          }
         } else {
           GST_OBJECT_UNLOCK (demux);
         }

diff --git a/gst/matroska/matroska-mux.c b/gst/matroska/matroska-mux.c
index 841941f..f2d3d54 100644 (file)
--- a/gst/matroska/matroska-mux.c
+++ b/gst/matroska/matroska-mux.c
@@ -1955,7 +1955,6 @@ gst_matroska_mux_subtitle_pad_setcaps (GstPad * pad, GstCaps * caps)
   GstStructure *structure;
   const GValue *value = NULL;
   GstBuffer *buf = NULL;
-  gchar *id = NULL;
   gboolean ret = TRUE;
 
   mux = GST_MATROSKA_MUX (GST_PAD_PARENT (pad));
@@ -1971,9 +1970,6 @@ gst_matroska_mux_subtitle_pad_setcaps (GstPad * pad, GstCaps * caps)
   structure = gst_caps_get_structure (caps, 0);
   mimetype = gst_structure_get_name (structure);
 
-  /* keep track of default set in request_pad */
-  id = context->codec_id;
-
   /* general setup */
   scontext->check_utf8 = 1;
   scontext->invalid_utf8 = 0;
@@ -2007,7 +2003,6 @@ gst_matroska_mux_subtitle_pad_setcaps (GstPad * pad, GstCaps * caps)
     gst_matroska_mux_set_codec_id (context,
         GST_MATROSKA_CODEC_ID_SUBTITLE_VOBSUB);
   } else {
-    id = NULL;
     ret = FALSE;
     goto exit;
   }
@@ -2043,9 +2038,6 @@ gst_matroska_mux_subtitle_pad_setcaps (GstPad * pad, GstCaps * caps)
       GST_STR_NULL (context->codec_id), context->codec_priv_size);
 
 exit:
-  /* free default if modified */
-  if (id)
-    g_free (id);
 
   return ret;
 }

diff --git a/gst/rtp/gstrtph264depay.c b/gst/rtp/gstrtph264depay.c
index c239613..3e39606 100644 (file)
--- a/gst/rtp/gstrtph264depay.c
+++ b/gst/rtp/gstrtph264depay.c
@@ -46,10 +46,13 @@ enum
 static const guint8 sync_bytes[] = { 0, 0, 0, 1 };
 
 static GstStaticPadTemplate gst_rtp_h264_depay_src_template =
-GST_STATIC_PAD_TEMPLATE ("src",
+    GST_STATIC_PAD_TEMPLATE ("src",
     GST_PAD_SRC,
     GST_PAD_ALWAYS,
-    GST_STATIC_CAPS ("video/x-h264")
+    GST_STATIC_CAPS ("video/x-h264, "
+        "stream-format = (string) avc, alignment = (string) au; "
+        "video/x-h264, "
+        "stream-format = (string) byte-stream, alignment = (string) { nal, au }")
     );
 
 static GstStaticPadTemplate gst_rtp_h264_depay_sink_template =
@@ -735,8 +738,9 @@ gst_rtp_h264_depay_process (GstRTPBaseDepayload * depayload, GstBuffer * buf)
            */
           nalu_size = (payload[0] << 8) | payload[1];
 
-          if (nalu_size > payload_len)
-            nalu_size = payload_len;
+          /* dont include nalu_size */
+          if (nalu_size > (payload_len - 2))
+            nalu_size = payload_len - 2;
 
           outsize = nalu_size + sizeof (sync_bytes);
           outbuf = gst_buffer_new_and_alloc (outsize);

diff --git a/gst/rtp/gstrtph264pay.c b/gst/rtp/gstrtph264pay.c
index 1338ba1..2be9e00 100644 (file)
--- a/gst/rtp/gstrtph264pay.c
+++ b/gst/rtp/gstrtph264pay.c
@@ -45,10 +45,13 @@ GST_DEBUG_CATEGORY_STATIC (rtph264pay_debug);
  */
 
 static GstStaticPadTemplate gst_rtp_h264_pay_sink_template =
-GST_STATIC_PAD_TEMPLATE ("sink",
+    GST_STATIC_PAD_TEMPLATE ("sink",
     GST_PAD_SINK,
     GST_PAD_ALWAYS,
-    GST_STATIC_CAPS ("video/x-h264")
+    GST_STATIC_CAPS ("video/x-h264, "
+        "stream-format = (string) byte-stream, alignment = (string) { nal, au };"
+        "video/x-h264, "
+        "stream-format = (string) avc, alignment = (string) au")
     );
 
 static GstStaticPadTemplate gst_rtp_h264_pay_src_template =
@@ -267,98 +270,110 @@ static GstCaps *
 gst_rtp_h264_pay_getcaps (GstRTPBasePayload * payload, GstPad * pad,
     GstCaps * filter)
 {
+  GstCaps *template_caps;
   GstCaps *allowed_caps;
+  GstCaps *caps, *icaps;
+  guint i;
 
   allowed_caps =
       gst_pad_peer_query_caps (GST_RTP_BASE_PAYLOAD_SRCPAD (payload), filter);
 
-  if (allowed_caps) {
-    GstCaps *caps = NULL;
-    guint i;
+  if (allowed_caps == NULL)
+    return NULL;
 
-    if (gst_caps_is_any (allowed_caps)) {
-      gst_caps_unref (allowed_caps);
-      goto any;
-    }
+  template_caps =
+      gst_static_pad_template_get_caps (&gst_rtp_h264_pay_sink_template);
 
-    if (gst_caps_is_empty (allowed_caps))
-      return allowed_caps;
+  if (gst_caps_is_any (allowed_caps)) {
+    caps = gst_caps_ref (template_caps);
+    goto done;
+  }
 
-    caps = gst_caps_new_empty ();
+  if (gst_caps_is_empty (allowed_caps)) {
+    caps = gst_caps_ref (allowed_caps);
+    goto done;
+  }
 
-    for (i = 0; i < gst_caps_get_size (allowed_caps); i++) {
-      GstStructure *s = gst_caps_get_structure (allowed_caps, i);
-      GstStructure *new_s = gst_structure_new_empty ("video/x-h264");
-      const gchar *profile_level_id;
+  caps = gst_caps_new_empty ();
 
-      profile_level_id = gst_structure_get_string (s, "profile-level-id");
+  for (i = 0; i < gst_caps_get_size (allowed_caps); i++) {
+    GstStructure *s = gst_caps_get_structure (allowed_caps, i);
+    GstStructure *new_s = gst_structure_new_empty ("video/x-h264");
+    const gchar *profile_level_id;
 
-      if (profile_level_id && strlen (profile_level_id) == 6) {
-        const gchar *profile;
-        const gchar *level;
-        long int spsint;
-        guint8 sps[3];
+    profile_level_id = gst_structure_get_string (s, "profile-level-id");
 
-        spsint = strtol (profile_level_id, NULL, 16);
-        sps[0] = spsint >> 16;
-        sps[1] = spsint >> 8;
-        sps[2] = spsint;
+    if (profile_level_id && strlen (profile_level_id) == 6) {
+      const gchar *profile;
+      const gchar *level;
+      long int spsint;
+      guint8 sps[3];
 
-        profile = gst_codec_utils_h264_get_profile (sps, 3);
-        level = gst_codec_utils_h264_get_level (sps, 3);
+      spsint = strtol (profile_level_id, NULL, 16);
+      sps[0] = spsint >> 16;
+      sps[1] = spsint >> 8;
+      sps[2] = spsint;
 
-        if (profile && level) {
-          GST_LOG_OBJECT (payload, "In caps, have profile %s and level %s",
-              profile, level);
+      profile = gst_codec_utils_h264_get_profile (sps, 3);
+      level = gst_codec_utils_h264_get_level (sps, 3);
 
-          if (!strcmp (profile, "constrained-baseline"))
-            gst_structure_set (new_s, "profile", G_TYPE_STRING, profile, NULL);
-          else {
-            GValue val = { 0, };
-            GValue profiles = { 0, };
+      if (profile && level) {
+        GST_LOG_OBJECT (payload, "In caps, have profile %s and level %s",
+            profile, level);
 
-            g_value_init (&profiles, GST_TYPE_LIST);
-            g_value_init (&val, G_TYPE_STRING);
+        if (!strcmp (profile, "constrained-baseline"))
+          gst_structure_set (new_s, "profile", G_TYPE_STRING, profile, NULL);
+        else {
+          GValue val = { 0, };
+          GValue profiles = { 0, };
 
-            g_value_set_static_string (&val, profile);
-            gst_value_list_append_value (&profiles, &val);
+          g_value_init (&profiles, GST_TYPE_LIST);
+          g_value_init (&val, G_TYPE_STRING);
 
-            g_value_set_static_string (&val, "constrained-baseline");
-            gst_value_list_append_value (&profiles, &val);
+          g_value_set_static_string (&val, profile);
+          gst_value_list_append_value (&profiles, &val);
 
-            gst_structure_take_value (new_s, "profile", &profiles);
-          }
+          g_value_set_static_string (&val, "constrained-baseline");
+          gst_value_list_append_value (&profiles, &val);
 
-          if (!strcmp (level, "1"))
-            gst_structure_set (new_s, "level", G_TYPE_STRING, level, NULL);
-          else {
-            GValue levels = { 0, };
-            GValue val = { 0, };
-            int j;
-
-            g_value_init (&levels, GST_TYPE_LIST);
-            g_value_init (&val, G_TYPE_STRING);
-
-            for (j = 0; all_levels[j]; j++) {
-              g_value_set_static_string (&val, all_levels[j]);
-              gst_value_list_prepend_value (&levels, &val);
-              if (!strcmp (level, all_levels[j]))
-                break;
-            }
-            gst_structure_take_value (new_s, "level", &levels);
+          gst_structure_take_value (new_s, "profile", &profiles);
+        }
+
+        if (!strcmp (level, "1"))
+          gst_structure_set (new_s, "level", G_TYPE_STRING, level, NULL);
+        else {
+          GValue levels = { 0, };
+          GValue val = { 0, };
+          int j;
+
+          g_value_init (&levels, GST_TYPE_LIST);
+          g_value_init (&val, G_TYPE_STRING);
+
+          for (j = 0; all_levels[j]; j++) {
+            g_value_set_static_string (&val, all_levels[j]);
+            gst_value_list_prepend_value (&levels, &val);
+            if (!strcmp (level, all_levels[j]))
+              break;
           }
+          gst_structure_take_value (new_s, "level", &levels);
         }
       }
-
-      gst_caps_merge_structure (caps, new_s);
     }
 
-    gst_caps_unref (allowed_caps);
-    return caps;
+    gst_caps_merge_structure (caps, new_s);
   }
 
-any:
-  return gst_caps_new_empty_simple ("video/x-h264");
+  icaps = gst_caps_intersect (caps, template_caps);
+  gst_caps_unref (caps);
+  caps = icaps;
+
+done:
+
+  gst_caps_unref (template_caps);
+  gst_caps_unref (allowed_caps);
+
+  GST_LOG_OBJECT (payload, "returning caps %" GST_PTR_FORMAT, caps);
+  return caps;
 }
 
 /* take the currently configured SPS and PPS lists and set them on the caps as

diff --git a/gst/rtp/gstrtpmp2tpay.c b/gst/rtp/gstrtpmp2tpay.c
index 35461dc..8cb7abf 100644 (file)
--- a/gst/rtp/gstrtpmp2tpay.c
+++ b/gst/rtp/gstrtpmp2tpay.c
@@ -124,6 +124,8 @@ gst_rtp_mp2t_pay_flush (GstRTPMP2TPay * rtpmp2tpay)
   GstRTPBuffer rtp = { NULL };
 
   avail = gst_adapter_available (rtpmp2tpay->adapter);
+  if (avail == 0)
+    return GST_FLOW_OK;
   outbuf = gst_rtp_buffer_new_allocate (avail, 0, 0);
 
   /* get payload */

diff --git a/gst/rtpmanager/gstrtpbin.c b/gst/rtpmanager/gstrtpbin.c
index 64fea5b..d5d6ca6 100644 (file)
--- a/gst/rtpmanager/gstrtpbin.c
+++ b/gst/rtpmanager/gstrtpbin.c
@@ -1901,7 +1901,7 @@ gst_rtp_bin_init (GstRtpBin * rtpbin)
   rtpbin->use_pipeline_clock = DEFAULT_USE_PIPELINE_CLOCK;
 
   /* some default SDES entries */
-  cname = g_strdup_printf ("user%u@x-%u.net", g_random_int (), g_random_int ());
+  cname = g_strdup_printf ("user%u@host-%x", g_random_int (), g_random_int ());
   rtpbin->sdes = gst_structure_new ("application/x-rtp-source-sdes",
       "cname", G_TYPE_STRING, cname, "tool", G_TYPE_STRING, "GStreamer", NULL);
   g_free (cname);

diff --git a/gst/rtpmanager/rtpsession.c b/gst/rtpmanager/rtpsession.c
index 3c9bc49..6b90619 100644 (file)
--- a/gst/rtpmanager/rtpsession.c
+++ b/gst/rtpmanager/rtpsession.c
@@ -488,7 +488,7 @@ rtp_session_init (RTPSession * sess)
   /* some default SDES entries */
 
   /* we do not want to leak details like the username or hostname here */
-  str = g_strdup_printf ("user%u@x-%u.net", g_random_int (), g_random_int ());
+  str = g_strdup_printf ("user%u@host-%x", g_random_int (), g_random_int ());
   rtp_source_set_sdes_string (sess->source, GST_RTCP_SDES_CNAME, str);
   g_free (str);
 

diff --git a/gst/udp/gstmultiudpsink.c b/gst/udp/gstmultiudpsink.c
index ae1d6b5..a13b360 100644 (file)
--- a/gst/udp/gstmultiudpsink.c
+++ b/gst/udp/gstmultiudpsink.c
@@ -240,7 +240,7 @@ gst_multiudpsink_class_init (GstMultiUDPSinkClass * klass)
           G_PARAM_READABLE | G_PARAM_STATIC_STRINGS));
   g_object_class_install_property (G_OBJECT_CLASS (klass), PROP_BYTES_SERVED,
       g_param_spec_uint64 ("bytes-served", "Bytes served",
-          "Total number of bytes send to all clients", 0, G_MAXUINT64, 0,
+          "Total number of bytes sent to all clients", 0, G_MAXUINT64, 0,
           G_PARAM_READABLE | G_PARAM_STATIC_STRINGS));
   g_object_class_install_property (gobject_class, PROP_SOCKET,
       g_param_spec_object ("socket", "Socket Handle",

diff --git a/gst/videobox/gstvideobox.c b/gst/videobox/gstvideobox.c
index 2294d26..966f94a 100644 (file)
--- a/gst/videobox/gstvideobox.c
+++ b/gst/videobox/gstvideobox.c
@@ -3270,8 +3270,8 @@ static void
 gst_video_box_process (GstVideoBox * video_box, const guint8 * src,
     guint8 * dest)
 {
-  guint b_alpha = CLAMP (video_box->border_alpha * 256, 0, 256);
-  guint i_alpha = CLAMP (video_box->alpha * 256, 0, 256);
+  guint b_alpha = CLAMP (video_box->border_alpha * 256, 0, 255);
+  guint i_alpha = CLAMP (video_box->alpha * 256, 0, 255);
   GstVideoBoxFill fill_type = video_box->fill_type;
   gint br, bl, bt, bb, crop_w, crop_h;
 

diff --git a/tests/check/elements/mpegaudioparse.c b/tests/check/elements/mpegaudioparse.c
index 69a0864..2bb2699 100644 (file)
--- a/tests/check/elements/mpegaudioparse.c
+++ b/tests/check/elements/mpegaudioparse.c
@@ -41,8 +41,6 @@ GstStaticPadTemplate srctemplate = GST_STATIC_PAD_TEMPLATE ("src",
     GST_STATIC_CAPS (SRC_CAPS_TMPL)
     );
 
-const gchar *factory = "aacparse";
-
 /* some data */
 static guint8 mp3_frame[384] = {
   0xff, 0xfb, 0x94, 0xc4, 0xff, 0x83, 0xc0, 0x00,

diff --git a/tests/check/elements/rtp-payloading.c b/tests/check/elements/rtp-payloading.c
index 5c06ac8..474f4b3 100644 (file)
--- a/tests/check/elements/rtp-payloading.c
+++ b/tests/check/elements/rtp-payloading.c
@@ -31,13 +31,11 @@
 typedef struct
 {
   GstElement *pipeline;
-  GstElement *fdsrc;
-  GstElement *capsfilter;
+  GstElement *appsrc;
   GstElement *rtppay;
   GstElement *rtpdepay;
   GstElement *fakesink;
-  int fd[2];
-  const char *frame_data;
+  const guint8 *frame_data;
   int frame_data_size;
   int frame_count;
 } rtp_pipeline;
@@ -134,13 +132,11 @@ rtp_bus_callback (GstBus * bus, GstMessage * message, gpointer data)
  * The user must free the RTP pipeline when it's not used anymore.
  */
 static rtp_pipeline *
-rtp_pipeline_create (const char *frame_data, int frame_data_size,
+rtp_pipeline_create (const guint8 * frame_data, int frame_data_size,
     int frame_count, const char *filtercaps, const char *pay, const char *depay)
 {
   gchar *pipeline_name;
-
   rtp_pipeline *p;
-
   GstCaps *caps;
 
   /* Check parameters. */
@@ -159,60 +155,39 @@ rtp_pipeline_create (const char *frame_data, int frame_data_size,
   pipeline_name = g_strdup_printf ("%s-%s-pipeline", pay, depay);
   p->pipeline = gst_pipeline_new (pipeline_name);
   g_free (pipeline_name);
-  p->fdsrc = gst_element_factory_make ("fdsrc", NULL);
-  p->capsfilter = gst_element_factory_make ("capsfilter", NULL);
+  p->appsrc = gst_element_factory_make ("appsrc", NULL);
   p->rtppay = gst_element_factory_make (pay, NULL);
   p->rtpdepay = gst_element_factory_make (depay, NULL);
   p->fakesink = gst_element_factory_make ("fakesink", NULL);
 
   /* One or more elements are not created successfully or failed to create p? */
-  if (!p->pipeline || !p->fdsrc || !p->capsfilter || !p->rtppay || !p->rtpdepay
-      || !p->fakesink || pipe (p->fd) == -1) {
+  if (!p->pipeline || !p->appsrc || !p->rtppay || !p->rtpdepay || !p->fakesink) {
     /* Release created elements. */
     RELEASE_ELEMENT (p->pipeline);
-    RELEASE_ELEMENT (p->fdsrc);
-    RELEASE_ELEMENT (p->capsfilter);
+    RELEASE_ELEMENT (p->appsrc);
     RELEASE_ELEMENT (p->rtppay);
     RELEASE_ELEMENT (p->rtpdepay);
     RELEASE_ELEMENT (p->fakesink);
 
-    /* Close pipe. */
-    if (p->fd[0]) {
-      close (p->fd[0]);
-    }
-
-    if (p->fd[1]) {
-      close (p->fd[1]);
-    }
-
     /* Release allocated memory. */
     free (p);
 
     return NULL;
   }
 
-  /* Set fdsrc properties. */
-  g_object_set (p->fdsrc, "fd", p->fd[0], NULL);
-  g_object_set (p->fdsrc, "do-timestamp", TRUE, NULL);
-  g_object_set (p->fdsrc, "blocksize", p->frame_data_size, NULL);
-  g_object_set (p->fdsrc, "num-buffers", p->frame_count * LOOP_COUNT, NULL);
-
-  /* Set caps filters. */
+  /* Set src properties. */
   caps = gst_caps_from_string (filtercaps);
-
-  g_object_set (p->capsfilter, "caps", caps, NULL);
+  g_object_set (p->appsrc, "do-timestamp", TRUE, "caps", caps, NULL);
   gst_caps_unref (caps);
 
   /* Add elements to the pipeline. */
-  gst_bin_add (GST_BIN (p->pipeline), p->fdsrc);
-  gst_bin_add (GST_BIN (p->pipeline), p->capsfilter);
+  gst_bin_add (GST_BIN (p->pipeline), p->appsrc);
   gst_bin_add (GST_BIN (p->pipeline), p->rtppay);
   gst_bin_add (GST_BIN (p->pipeline), p->rtpdepay);
   gst_bin_add (GST_BIN (p->pipeline), p->fakesink);
 
   /* Link elements. */
-  gst_element_link (p->fdsrc, p->capsfilter);
-  gst_element_link (p->capsfilter, p->rtppay);
+  gst_element_link (p->appsrc, p->rtppay);
   gst_element_link (p->rtppay, p->rtpdepay);
   gst_element_link (p->rtpdepay, p->fakesink);
 
@@ -234,15 +209,6 @@ rtp_pipeline_destroy (rtp_pipeline * p)
   /* Release pipeline. */
   RELEASE_ELEMENT (p->pipeline);
 
-  /* Close pipe. */
-  if (p->fd[0]) {
-    close (p->fd[0]);
-  }
-
-  if (p->fd[1]) {
-    close (p->fd[1]);
-  }
-
   /* Release allocated memory. */
   free (p);
 }
@@ -254,11 +220,10 @@ rtp_pipeline_destroy (rtp_pipeline * p)
 static void
 rtp_pipeline_run (rtp_pipeline * p)
 {
+  GstFlowReturn flow_ret;
   GMainLoop *mainloop = NULL;
-
   GstBus *bus;
-
-  gint i;
+  gint i, j;
 
   /* Check parameters. */
   if (p == NULL) {
@@ -280,22 +245,27 @@ rtp_pipeline_run (rtp_pipeline * p)
   /* Set pipeline to PLAYING. */
   gst_element_set_state (p->pipeline, GST_STATE_PLAYING);
 
-  /* TODO: Writing may need some changes... */
-
+  /* Push data into the pipeline */
   for (i = 0; i < LOOP_COUNT; i++) {
-    const char *frame_data_pointer = p->frame_data;
-    int res;
-    int frame_count = p->frame_count;
-
-    /* Write in to the pipe. */
-    while (frame_count > 0) {
-      res = write (p->fd[1], frame_data_pointer, p->frame_data_size);
-      fail_unless_equals_int (res, p->frame_data_size);
-      frame_data_pointer += p->frame_data_size;
-      frame_count--;
+    const guint8 *data = p->frame_data;
+
+    for (j = 0; j < p->frame_count; j++) {
+      GstBuffer *buf;
+
+      buf =
+          gst_buffer_new_wrapped_full ((guint8 *) data, NULL, 0,
+          p->frame_data_size);
+
+      g_signal_emit_by_name (p->appsrc, "push-buffer", buf, &flow_ret);
+      fail_unless_equals_int (flow_ret, GST_FLOW_OK);
+      data += p->frame_data_size;
+
+      gst_buffer_unref (buf);
     }
   }
 
+  g_signal_emit_by_name (p->appsrc, "end-of-stream", &flow_ret);
+
   /* Run mainloop. */
   g_main_loop_run (mainloop);
 
@@ -344,8 +314,8 @@ rtp_pipeline_enable_lists (rtp_pipeline * p, guint mtu_size)
  * @use_lists enable buffer lists
  */
 static void
-rtp_pipeline_test (const char *frame_data, int frame_data_size, int frame_count,
-    const char *filtercaps, const char *pay, const char *depay,
+rtp_pipeline_test (const guint8 * frame_data, int frame_data_size,
+    int frame_count, const char *filtercaps, const char *pay, const char *depay,
     guint bytes_sent, guint mtu_size, gboolean use_lists)
 {
   /* Create RTP pipeline. */
@@ -374,7 +344,7 @@ rtp_pipeline_test (const char *frame_data, int frame_data_size, int frame_count,
   }
 }
 
-static char rtp_ilbc_frame_data[] =
+static const guint8 rtp_ilbc_frame_data[] =
     { 0x00, 0x00, 0x00, 0x00, 0x00, 0x00, 0x00, 0x00, 0x00, 0x00, 0x00, 0x00,
   0x00, 0x00, 0x00, 0x00, 0x00, 0x00, 0x00, 0x00, 0x00
 };
@@ -391,7 +361,7 @@ GST_START_TEST (rtp_ilbc)
 }
 
 GST_END_TEST;
-static char rtp_gsm_frame_data[] =
+static const guint8 rtp_gsm_frame_data[] =
     { 0x00, 0x00, 0x00, 0x00, 0x00, 0x00, 0x00, 0x00, 0x00, 0x00, 0x00, 0x00,
   0x00, 0x00, 0x00, 0x00, 0x00, 0x00, 0x00, 0x00, 0x00
 };
@@ -408,7 +378,7 @@ GST_START_TEST (rtp_gsm)
 }
 
 GST_END_TEST;
-static char rtp_amr_frame_data[] =
+static const guint8 rtp_amr_frame_data[] =
     { 0x3c, 0x24, 0x03, 0xb3, 0x48, 0x10, 0x68, 0x46, 0x6c, 0xec, 0x03,
   0x7a, 0x37, 0x16, 0x41, 0x41, 0xc0, 0x00, 0x0d, 0xcd, 0x12, 0xed,
   0xad, 0x80, 0x00, 0x00, 0x11, 0x31, 0x00, 0x00, 0x0d, 0xa0
@@ -426,7 +396,7 @@ GST_START_TEST (rtp_amr)
 }
 
 GST_END_TEST;
-static char rtp_pcma_frame_data[] =
+static const guint8 rtp_pcma_frame_data[] =
     { 0x00, 0x00, 0x00, 0x00, 0x00, 0x00, 0x00, 0x00, 0x00, 0x00, 0x00, 0x00,
   0x00, 0x00, 0x00, 0x00, 0x00, 0x00, 0x00, 0x00, 0x00
 };
@@ -443,7 +413,7 @@ GST_START_TEST (rtp_pcma)
 }
 
 GST_END_TEST;
-static char rtp_pcmu_frame_data[] =
+static const guint8 rtp_pcmu_frame_data[] =
     { 0x00, 0x00, 0x00, 0x00, 0x00, 0x00, 0x00, 0x00, 0x00, 0x00, 0x00, 0x00,
   0x00, 0x00, 0x00, 0x00, 0x00, 0x00, 0x00, 0x00, 0x00
 };
@@ -460,7 +430,7 @@ GST_START_TEST (rtp_pcmu)
 }
 
 GST_END_TEST;
-static char rtp_mpa_frame_data[] =
+static const guint8 rtp_mpa_frame_data[] =
     { 0x00, 0x00, 0x00, 0x00, 0x00, 0x00, 0x00, 0x00, 0x00, 0x00, 0x00, 0x00,
   0x00, 0x00, 0x00, 0x00, 0x00, 0x00, 0x00, 0x00, 0x00
 };
@@ -477,7 +447,7 @@ GST_START_TEST (rtp_mpa)
 }
 
 GST_END_TEST;
-static char rtp_h263_frame_data[] =
+static const guint8 rtp_h263_frame_data[] =
     { 0x00, 0x00, 0x00, 0x00, 0x00, 0x00, 0x00, 0x00, 0x00, 0x00, 0x00, 0x00,
   0x00, 0x00, 0x00, 0x00, 0x00, 0x00, 0x00, 0x00, 0x00
 };
@@ -489,12 +459,12 @@ static int rtp_h263_frame_count = 1;
 GST_START_TEST (rtp_h263)
 {
   rtp_pipeline_test (rtp_h263_frame_data, rtp_h263_frame_data_size,
-      rtp_h263_frame_count, "video/x-h263,variant=itu,h263version=h263",
+      rtp_h263_frame_count, "video/x-h263,variant=(string)itu,h263version=h263",
       "rtph263pay", "rtph263depay", 0, 0, FALSE);
 }
 
 GST_END_TEST;
-static char rtp_h263p_frame_data[] =
+static const guint8 rtp_h263p_frame_data[] =
     { 0x00, 0x00, 0x00, 0x00, 0x00, 0x00, 0x00, 0x00, 0x00, 0x00, 0x00, 0x00,
   0x00, 0x00, 0x00, 0x00, 0x00, 0x00, 0x00, 0x00, 0x00
 };
@@ -506,12 +476,12 @@ static int rtp_h263p_frame_count = 1;
 GST_START_TEST (rtp_h263p)
 {
   rtp_pipeline_test (rtp_h263p_frame_data, rtp_h263p_frame_data_size,
-      rtp_h263p_frame_count, "video/x-h263,variant=itu", "rtph263ppay",
+      rtp_h263p_frame_count, "video/x-h263,variant=(string)itu", "rtph263ppay",
       "rtph263pdepay", 0, 0, FALSE);
 }
 
 GST_END_TEST;
-static char rtp_h264_frame_data[] =
+static const guint8 rtp_h264_frame_data[] =
     { 0x00, 0x00, 0x00, 0x00, 0x00, 0x00, 0x00, 0x00, 0x00, 0x00, 0x00, 0x00,
   0x00, 0x00, 0x00, 0x00, 0x00, 0x00, 0x00, 0x00, 0x00
 };
@@ -522,13 +492,14 @@ static int rtp_h264_frame_count = 1;
 
 GST_START_TEST (rtp_h264)
 {
+  /* FIXME 0.11: fully specify h264 caps (and make payloader check) */
   rtp_pipeline_test (rtp_h264_frame_data, rtp_h264_frame_data_size,
       rtp_h264_frame_count, "video/x-h264", "rtph264pay", "rtph264depay",
       0, 0, FALSE);
 }
 
 GST_END_TEST;
-static char rtp_h264_list_lt_mtu_frame_data[] =
+static const guint8 rtp_h264_list_lt_mtu_frame_data[] =
     /* not packetized, next NAL starts with 0001 */
 { 0x00, 0x00, 0x00, 0x01, 0x00, 0x10, 0x00, 0x00, 0x00, 0x00, 0x00,
   0x00, 0x00, 0x00, 0x00, 0x00, 0x00, 0x00, 0x00, 0x01, 0x00, 0x00,
@@ -546,6 +517,7 @@ static int rtp_h264_list_lt_mtu_mtu_size = 1024;
 
 GST_START_TEST (rtp_h264_list_lt_mtu)
 {
+  /* FIXME 0.11: fully specify h264 caps (and make payloader check) */
   rtp_pipeline_test (rtp_h264_list_lt_mtu_frame_data,
       rtp_h264_list_lt_mtu_frame_data_size, rtp_h264_list_lt_mtu_frame_count,
       "video/x-h264", "rtph264pay", "rtph264depay",
@@ -553,7 +525,7 @@ GST_START_TEST (rtp_h264_list_lt_mtu)
 }
 
 GST_END_TEST;
-static char rtp_h264_list_gt_mtu_frame_data[] =
+static const guint8 rtp_h264_list_gt_mtu_frame_data[] =
     /* not packetized, next NAL starts with 0001 */
 { 0x00, 0x00, 0x00, 0x01, 0x00, 0x00, 0x00, 0x00, 0x00, 0x00, 0x00,
   0x00, 0x00, 0x00, 0x00, 0x00, 0x00, 0x00, 0x00, 0x00, 0x00, 0x00,
@@ -574,6 +546,7 @@ static int rtp_h264_list_gt_mtu_mty_size = 28;
 
 GST_START_TEST (rtp_h264_list_gt_mtu)
 {
+  /* FIXME 0.11: fully specify h264 caps (and make payloader check) */
   rtp_pipeline_test (rtp_h264_list_gt_mtu_frame_data,
       rtp_h264_list_gt_mtu_frame_data_size, rtp_h264_list_gt_mtu_frame_count,
       "video/x-h264", "rtph264pay", "rtph264depay",
@@ -581,7 +554,7 @@ GST_START_TEST (rtp_h264_list_gt_mtu)
 }
 
 GST_END_TEST;
-static char rtp_L16_frame_data[] =
+static const guint8 rtp_L16_frame_data[] =
     { 0x00, 0x00, 0x00, 0x00, 0x00, 0x00, 0x00, 0x00, 0x00, 0x00, 0x00, 0x00,
   0x00, 0x00, 0x00, 0x00, 0x00, 0x00, 0x00, 0x00, 0x00
 };
@@ -599,7 +572,7 @@ GST_START_TEST (rtp_L16)
 }
 
 GST_END_TEST;
-static char rtp_mp2t_frame_data[] =
+static const guint8 rtp_mp2t_frame_data[] =
     { 0x00, 0x00, 0x00, 0x00, 0x00, 0x00, 0x00, 0x00, 0x00, 0x00, 0x00, 0x00,
   0x00, 0x00, 0x00, 0x00, 0x00, 0x00, 0x00, 0x00, 0x00
 };
@@ -616,7 +589,7 @@ GST_START_TEST (rtp_mp2t)
 }
 
 GST_END_TEST;
-static char rtp_mp4v_frame_data[] =
+static const guint8 rtp_mp4v_frame_data[] =
     { 0x00, 0x00, 0x00, 0x00, 0x00, 0x00, 0x00, 0x00, 0x00, 0x00, 0x00, 0x00,
   0x00, 0x00, 0x00, 0x00, 0x00, 0x00, 0x00, 0x00, 0x00
 };
@@ -633,7 +606,7 @@ GST_START_TEST (rtp_mp4v)
 }
 
 GST_END_TEST;
-static char rtp_mp4v_list_frame_data[] =
+static const guint8 rtp_mp4v_list_frame_data[] =
     { 0x00, 0x00, 0x00, 0x00, 0x00, 0x00, 0x00, 0x00, 0x00, 0x00, 0x00, 0x00,
   0x00, 0x00, 0x00, 0x00, 0x00, 0x00, 0x00, 0x00, 0x00
 };
@@ -653,7 +626,7 @@ GST_START_TEST (rtp_mp4v_list)
 }
 
 GST_END_TEST;
-static char rtp_mp4g_frame_data[] =
+static const guint8 rtp_mp4g_frame_data[] =
     { 0x00, 0x00, 0x00, 0x00, 0x00, 0x00, 0x00, 0x00, 0x00, 0x00, 0x00, 0x00,
   0x00, 0x00, 0x00, 0x00, 0x00, 0x00, 0x00, 0x00, 0x00
 };
@@ -671,7 +644,7 @@ GST_START_TEST (rtp_mp4g)
 }
 
 GST_END_TEST;
-static char rtp_theora_frame_data[] =
+static const guint8 rtp_theora_frame_data[] =
     { 0x00, 0x00, 0x00, 0x00, 0x00, 0x00, 0x00, 0x00, 0x00, 0x00, 0x00, 0x00,
   0x00, 0x00, 0x00, 0x00, 0x00, 0x00, 0x00, 0x00, 0x00
 };
@@ -688,7 +661,7 @@ GST_START_TEST (rtp_theora)
 }
 
 GST_END_TEST;
-static char rtp_vorbis_frame_data[] =
+static const guint8 rtp_vorbis_frame_data[] =
     { 0x00, 0x00, 0x00, 0x00, 0x00, 0x00, 0x00, 0x00, 0x00, 0x00, 0x00, 0x00,
   0x00, 0x00, 0x00, 0x00, 0x00, 0x00, 0x00, 0x00, 0x00
 };
@@ -705,7 +678,7 @@ GST_START_TEST (rtp_vorbis)
 }
 
 GST_END_TEST;
-static char rtp_jpeg_frame_data[] =
+static const guint8 rtp_jpeg_frame_data[] =
     { /* SOF */ 0xFF, 0xC0, 0x00, 0x11, 0x08, 0x00, 0x08, 0x00, 0x08,
   0x03, 0x00, 0x21, 0x08, 0x01, 0x11, 0x08, 0x02, 0x11, 0x08,
   /* DQT */ 0xFF, 0xDB, 0x00, 0x43, 0x08,
@@ -732,7 +705,7 @@ GST_START_TEST (rtp_jpeg)
 }
 
 GST_END_TEST;
-static char rtp_jpeg_list_frame_data[] =
+static const guint8 rtp_jpeg_list_frame_data[] =
     { /* SOF */ 0xFF, 0xC0, 0x00, 0x11, 0x08, 0x00, 0x08, 0x00, 0x08,
   0x03, 0x00, 0x21, 0x08, 0x01, 0x11, 0x08, 0x02, 0x11, 0x08,
   /* DQT */ 0xFF, 0xDB, 0x00, 0x43, 0x08,
@@ -761,7 +734,7 @@ GST_START_TEST (rtp_jpeg_list)
 }
 
 GST_END_TEST;
-static char rtp_g729_frame_data[] =
+static const guint8 rtp_g729_frame_data[] =
     { 0x00, 0x00, 0x00, 0x00, 0x00, 0x00, 0x00, 0x00, 0x00, 0x00, 0x00, 0x00,
   0x00, 0x00, 0x00, 0x00, 0x00, 0x00, 0x00, 0x00, 0x00, 0x00, 0x00
 };