Merge branch 'master' into 0.11

gst/audioparsers/gstac3parse.c

@@ -200,6 +200,7 @@ gst_ac3_parse_reset (GstAc3Parse * ac3parse)
 {
   ac3parse->channels = -1;
   ac3parse->sample_rate = -1;
+  ac3parse->blocks = -1;
   ac3parse->eac = FALSE;
 }
 
@@ -499,13 +500,14 @@ gst_ac3_parse_parse_frame (GstBaseParse * parse, GstBaseParseFrame * frame)
   GstAc3Parse *ac3parse = GST_AC3_PARSE (parse);
   GstBuffer *buf = frame->buffer;
   guint fsize, rate, chans, blocks, sid;
-  gboolean eac;
+  gboolean eac, update_rate = FALSE;
 
   if (!gst_ac3_parse_frame_header (ac3parse, buf, &fsize, &rate, &chans,
           &blocks, &sid, &eac))
     goto broken_header;
 
-  GST_LOG_OBJECT (parse, "size: %u, rate: %u, chans: %u", fsize, rate, chans);
+  GST_LOG_OBJECT (parse, "size: %u, blocks: %u, rate: %u, chans: %u", fsize,
+      blocks, rate, chans);
 
   if (G_UNLIKELY (sid)) {
     /* dependent frame, no need to (ac)count for or consider further */
@@ -533,9 +535,18 @@ gst_ac3_parse_parse_frame (GstBaseParse * parse, GstBaseParseFrame * frame)
     ac3parse->channels = chans;
     ac3parse->eac = eac;
 
-    gst_base_parse_set_frame_rate (parse, rate, 256 * blocks, 2, 2);
+    update_rate = TRUE;
   }
 
+  if (G_UNLIKELY (ac3parse->blocks != blocks)) {
+    ac3parse->blocks = blocks;
+
+    update_rate = TRUE;
+  }
+
+  if (G_UNLIKELY (update_rate))
+    gst_base_parse_set_frame_rate (parse, rate, 256 * blocks, 2, 2);
+
   return GST_FLOW_OK;
 
 /* ERRORS */

gst/audioparsers/gstac3parse.h

@@ -53,6 +53,7 @@ struct _GstAc3Parse {
   /*< private >*/
   gint         sample_rate;
   gint         channels;
+  gint         blocks;
   gboolean     eac;
 };
 

gst/rtp/Makefile.am

@@ -86,6 +86,7 @@ libgstrtp_la_LIBADD = $(GST_PLUGINS_BASE_LIBS) \
 		-lgstvideo-@GST_MAJORMINOR@ \
                 -lgsttag-@GST_MAJORMINOR@ \
                 -lgstrtp-@GST_MAJORMINOR@ \
+                -lgstpbutils-@GST_MAJORMINOR@ \
 		$(GST_BASE_LIBS) $(GST_LIBS) \
 		$(WINSOCK2_LIBS) $(LIBM)
 libgstrtp_la_LDFLAGS = $(GST_PLUGIN_LDFLAGS) 

gst/rtp/gstrtph264pay.c

@@ -22,8 +22,10 @@
 #endif
 
 #include <string.h>
+#include <stdlib.h>
 
 #include <gst/rtp/gstrtpbuffer.h>
+#include <gst/pbutils/pbutils.h>
 
 #include "gstrtph264pay.h"
 
@@ -108,6 +110,8 @@ static void gst_rtp_h264_pay_set_property (GObject * object, guint prop_id,
 static void gst_rtp_h264_pay_get_property (GObject * object, guint prop_id,
     GValue * value, GParamSpec * pspec);
 
+static GstCaps *gst_rtp_h264_pay_getcaps (GstBaseRTPPayload * payload,
+    GstPad * pad);
 static gboolean gst_rtp_h264_pay_setcaps (GstBaseRTPPayload * basepayload,
     GstCaps * caps);
 static GstFlowReturn gst_rtp_h264_pay_handle_buffer (GstBaseRTPPayload * pad,
@@ -186,6 +190,7 @@ gst_rtp_h264_pay_class_init (GstRtpH264PayClass * klass)
   gstelement_class->change_state =
       GST_DEBUG_FUNCPTR (gst_basertppayload_change_state);
 
+  gstbasertppayload_class->get_caps = gst_rtp_h264_pay_getcaps;
   gstbasertppayload_class->set_caps = gst_rtp_h264_pay_setcaps;
   gstbasertppayload_class->handle_buffer = gst_rtp_h264_pay_handle_buffer;
   gstbasertppayload_class->handle_event = gst_rtp_h264_pay_handle_event;
@@ -238,6 +243,123 @@ gst_rtp_h264_pay_finalize (GObject * object)
   G_OBJECT_CLASS (parent_class)->finalize (object);
 }
 
+static const gchar *all_levels[] = {
+  "1",
+  "1b",
+  "1.1",
+  "1.2",
+  "1.3",
+  "2",
+  "2.1",
+  "2.2",
+  "3",
+  "3.1",
+  "3.2",
+  "4",
+  "4.1",
+  "4.2",
+  "5",
+  "5.1",
+  NULL
+};
+
+static GstCaps *
+gst_rtp_h264_pay_getcaps (GstBaseRTPPayload * payload, GstPad * pad)
+{
+  GstCaps *allowed_caps;
+
+  allowed_caps =
+      gst_pad_peer_get_caps_reffed (GST_BASE_RTP_PAYLOAD_SRCPAD (payload));
+
+  if (allowed_caps) {
+    GstCaps *caps = NULL;
+    guint i;
+
+    if (gst_caps_is_any (allowed_caps)) {
+      gst_caps_unref (allowed_caps);
+      goto any;
+    }
+
+    if (gst_caps_is_empty (allowed_caps))
+      return allowed_caps;
+
+    caps = gst_caps_new_empty ();
+
+    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 ("video/x-h264", NULL);
+      const gchar *profile_level_id;
+
+      profile_level_id = gst_structure_get_string (s, "profile-level-id");
+
+      if (profile_level_id && strlen (profile_level_id) == 6) {
+        const gchar *profile;
+        const gchar *level;
+        long int spsint;
+        guint8 sps[3];
+
+        spsint = strtol (profile_level_id, NULL, 16);
+        sps[0] = spsint >> 16;
+        sps[1] = spsint >> 8;
+        sps[2] = spsint;
+
+        profile = gst_codec_utils_h264_get_profile (sps, 3);
+        level = gst_codec_utils_h264_get_level (sps, 3);
+
+        if (profile && level) {
+          GST_LOG_OBJECT (payload, "In caps, have profile %s and level %s",
+              profile, level);
+
+          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_init (&profiles, GST_TYPE_LIST);
+            g_value_init (&val, G_TYPE_STRING);
+
+            g_value_set_static_string (&val, profile);
+            gst_value_list_append_value (&profiles, &val);
+
+            g_value_set_static_string (&val, "constrained-baseline");
+            gst_value_list_append_value (&profiles, &val);
+
+            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;
+  }
+
+any:
+  return gst_caps_new_simple ("video/x-h264", NULL);
+}
+
 /* take the currently configured SPS and PPS lists and set them on the caps as
  * sprop-parameter-sets */
 static gboolean

gst/rtp/gstrtppcmadepay.c

@@ -45,12 +45,12 @@ static GstStaticPadTemplate gst_rtp_pcma_depay_sink_template =
     GST_PAD_ALWAYS,
     GST_STATIC_CAPS ("application/x-rtp, "
         "media = (string) \"audio\", "
-        "payload = (int) " GST_RTP_PAYLOAD_DYNAMIC_STRING ", "
-        "clock-rate = (int) [1, MAX ], " "encoding-name = (string) \"PCMA\";"
+        "payload = (int) " GST_RTP_PAYLOAD_PCMA_STRING ", "
+        "clock-rate = (int) 8000, encoding-name = (string) \"PCMA\";"
         "application/x-rtp, "
         "media = (string) \"audio\", "
-        "payload = (int) " GST_RTP_PAYLOAD_PCMA_STRING ", "
-        "clock-rate = (int) [1, MAX ]")
+        "payload = (int) " GST_RTP_PAYLOAD_DYNAMIC_STRING ", "
+        "clock-rate = (int) [1, MAX ], encoding-name = (string) \"PCMA\"")
     );
 
 static GstStaticPadTemplate gst_rtp_pcma_depay_src_template =

gst/rtp/gstrtppcmapay.c

@@ -47,7 +47,7 @@ static GstStaticPadTemplate gst_rtp_pcma_pay_src_template =
         "application/x-rtp, "
         "media = (string) \"audio\", "
         "payload = (int) " GST_RTP_PAYLOAD_DYNAMIC_STRING ", "
-        "clock-rate = (int) 8000, " "encoding-name = (string) \"PCMA\"")
+        "clock-rate = (int) [1, MAX ], " "encoding-name = (string) \"PCMA\"")
     );
 
 static gboolean gst_rtp_pcma_pay_setcaps (GstBaseRTPPayload * payload,

gst/rtp/gstrtppcmudepay.c

@@ -45,12 +45,12 @@ static GstStaticPadTemplate gst_rtp_pcmu_depay_sink_template =
     GST_PAD_ALWAYS,
     GST_STATIC_CAPS ("application/x-rtp, "
         "media = (string) \"audio\", "
-        "payload = (int) " GST_RTP_PAYLOAD_DYNAMIC_STRING ", "
-        "clock-rate = (int) [1, MAX ], " "encoding-name = (string) \"PCMU\";"
+        "payload = (int) " GST_RTP_PAYLOAD_PCMU_STRING ", "
+        "encoding-name = (string) \"PCMU\", clock-rate = (int) 8000; "
         "application/x-rtp, "
         "media = (string) \"audio\", "
-        "payload = (int) " GST_RTP_PAYLOAD_PCMU_STRING ", "
-        "clock-rate = (int) [1, MAX ]")
+        "payload = (int) " GST_RTP_PAYLOAD_DYNAMIC_STRING ", "
+        "encoding-name = (string) \"PCMU\", clock-rate = (int) [1, MAX ]")
     );
 
 static GstStaticPadTemplate gst_rtp_pcmu_depay_src_template =

gst/rtp/gstrtppcmupay.c

@@ -47,7 +47,7 @@ static GstStaticPadTemplate gst_rtp_pcmu_pay_src_template =
         "application/x-rtp, "
         "media = (string) \"audio\", "
         "payload = (int) " GST_RTP_PAYLOAD_DYNAMIC_STRING ", "
-        "clock-rate = (int) 8000, " "encoding-name = (string) \"PCMU\"")
+        "clock-rate = (int) [1, MAX ], " "encoding-name = (string) \"PCMU\"")
     );
 
 static gboolean gst_rtp_pcmu_pay_setcaps (GstBaseRTPPayload * payload,

gst/rtsp/gstrtspsrc.c

@@ -5965,8 +5965,12 @@ gst_rtspsrc_play (GstRTSPSrc * src, GstSegment * segment, gboolean async)
    * udp sources */
   gst_rtspsrc_send_dummy_packets (src);
 
-  /* activate receive elements */
-  gst_element_set_state (GST_ELEMENT_CAST (src), GST_STATE_PLAYING);
+  /* activate receive elements;
+   * only in async case, since receive elements may not have been affected
+   * by overall state change (e.g. not around yet),
+   * do not mess with state in sync case (e.g. seeking) */
+  if (async)
+    gst_element_set_state (GST_ELEMENT_CAST (src), GST_STATE_PLAYING);
 
   /* construct a control url */
   if (src->control)