rtppayload: set standard payload type as default

Initialize the PT to the default value of the codec and check if
it is still the default before declaring the pt to be dynamic or
not when setting the caps.

Also use the PT constants from the rtp lib when possible

https://bugzilla.gnome.org/show_bug.cgi?id=747965

gst/rtp/gstrtpL16pay.c

@@ -208,16 +208,16 @@ gst_rtp_L16_pay_getcaps (GstRTPBasePayload * rtppayload, GstPad * pad,
       if (gst_structure_get_int (structure, "channels", &channels)) {
         gst_caps_set_simple (caps, "channels", G_TYPE_INT, channels, NULL);
       } else if (gst_structure_get_int (structure, "payload", &pt)) {
-        if (pt == 10)
+        if (pt == GST_RTP_PAYLOAD_L16_STEREO)
           gst_caps_set_simple (caps, "channels", G_TYPE_INT, 2, NULL);
-        else if (pt == 11)
+        else if (pt == GST_RTP_PAYLOAD_L16_MONO)
           gst_caps_set_simple (caps, "channels", G_TYPE_INT, 1, NULL);
       }
 
       if (gst_structure_get_int (structure, "clock-rate", &rate)) {
         gst_caps_set_simple (caps, "rate", G_TYPE_INT, rate, NULL);
       } else if (gst_structure_get_int (structure, "payload", &pt)) {
-        if (pt == 10 || pt == 11)
+        if (pt == GST_RTP_PAYLOAD_L16_STEREO || pt == GST_RTP_PAYLOAD_L16_MONO)
           gst_caps_set_simple (caps, "rate", G_TYPE_INT, 44100, NULL);
       }
 

gst/rtp/gstrtpg722pay.c

@@ -40,13 +40,18 @@ GST_STATIC_PAD_TEMPLATE ("sink",
     );
 
 static GstStaticPadTemplate gst_rtp_g722_pay_src_template =
-GST_STATIC_PAD_TEMPLATE ("src",
+    GST_STATIC_PAD_TEMPLATE ("src",
     GST_PAD_SRC,
     GST_PAD_ALWAYS,
     GST_STATIC_CAPS ("application/x-rtp, "
         "media = (string) \"audio\", "
         "encoding-name = (string) \"G722\", "
         "payload = (int) " GST_RTP_PAYLOAD_G722_STRING ", "
+        "clock-rate = (int) 8000; "
+        "application/x-rtp, "
+        "media = (string) \"audio\", "
+        "encoding-name = (string) \"G722\", "
+        "payload = (int) " GST_RTP_PAYLOAD_DYNAMIC_STRING ", "
         "clock-rate = (int) 8000")
     );
 
@@ -92,6 +97,8 @@ gst_rtp_g722_pay_init (GstRtpG722Pay * rtpg722pay)
 
   rtpbaseaudiopayload = GST_RTP_BASE_AUDIO_PAYLOAD (rtpg722pay);
 
+  GST_RTP_BASE_PAYLOAD (rtpg722pay)->pt = GST_RTP_PAYLOAD_G722;
+
   /* tell rtpbaseaudiopayload that this is a sample based codec */
   gst_rtp_base_audio_payload_set_sample_based (rtpbaseaudiopayload);
 }
@@ -136,8 +143,8 @@ gst_rtp_g722_pay_setcaps (GstRTPBasePayload * basepayload, GstCaps * caps)
    * RFC 3551 although the sampling rate is 16000 Hz */
   clock_rate = 8000;
 
-  gst_rtp_base_payload_set_options (basepayload, "audio", TRUE, "G722",
+  gst_rtp_base_payload_set_options (basepayload, "audio",
-      clock_rate);
+      basepayload->pt != GST_RTP_PAYLOAD_G722, "G722", clock_rate);
   params = g_strdup_printf ("%d", channels);
 
 #if 0

gst/rtp/gstrtpg723pay.c

@@ -29,9 +29,6 @@
 
 #include "gstrtpg723pay.h"
 
-#define GST_RTP_PAYLOAD_G723 4
-#define GST_RTP_PAYLOAD_G723_STRING "4"
-
 #define G723_FRAME_DURATION (30 * GST_MSECOND)
 
 static gboolean gst_rtp_g723_pay_set_caps (GstRTPBasePayload * payload,
@@ -107,7 +104,6 @@ gst_rtp_g723_pay_init (GstRTPG723Pay * pay)
   pay->adapter = gst_adapter_new ();
 
   payload->pt = GST_RTP_PAYLOAD_G723;
-  gst_rtp_base_payload_set_options (payload, "audio", FALSE, "G723", 8000);
 }
 
 static void
@@ -128,16 +124,9 @@ static gboolean
 gst_rtp_g723_pay_set_caps (GstRTPBasePayload * payload, GstCaps * caps)
 {
   gboolean res;
-  GstStructure *structure;
-  gint pt;
-
-  structure = gst_caps_get_structure (caps, 0);
-  if (!gst_structure_get_int (structure, "payload", &pt))
-    pt = GST_RTP_PAYLOAD_G723;
-
-  payload->pt = pt;
-  payload->dynamic = pt != GST_RTP_PAYLOAD_G723;
 
+  gst_rtp_base_payload_set_options (payload, "audio",
+      payload->pt != GST_RTP_PAYLOAD_G723, "G723", 8000);
   res = gst_rtp_base_payload_set_outcaps (payload, NULL);
 
   return res;

gst/rtp/gstrtpg729pay.c

@@ -121,7 +121,6 @@ gst_rtp_g729_pay_init (GstRTPG729Pay * pay)
   GstRTPBasePayload *payload = GST_RTP_BASE_PAYLOAD (pay);
 
   payload->pt = GST_RTP_PAYLOAD_G729;
-  gst_rtp_base_payload_set_options (payload, "audio", FALSE, "G729", 8000);
 
   pay->adapter = gst_adapter_new ();
 }
@@ -140,15 +139,9 @@ static gboolean
 gst_rtp_g729_pay_set_caps (GstRTPBasePayload * payload, GstCaps * caps)
 {
   gboolean res;
-  GstStructure *structure;
-  gint pt;
 
-  structure = gst_caps_get_structure (caps, 0);
+  gst_rtp_base_payload_set_options (payload, "audio",
-  if (!gst_structure_get_int (structure, "payload", &pt))
+      payload->pt != GST_RTP_PAYLOAD_G729, "G729", 8000);
-    pt = GST_RTP_PAYLOAD_G729;
-
-  payload->pt = pt;
-  payload->dynamic = pt != GST_RTP_PAYLOAD_G729;
 
   res = gst_rtp_base_payload_set_outcaps (payload, NULL);
 

gst/rtp/gstrtpgsmpay.c

@@ -107,7 +107,8 @@ gst_rtp_gsm_pay_setcaps (GstRTPBasePayload * payload, GstCaps * caps)
   if (strcmp ("audio/x-gsm", stname))
     goto invalid_type;
 
-  gst_rtp_base_payload_set_options (payload, "audio", FALSE, "GSM", 8000);
+  gst_rtp_base_payload_set_options (payload, "audio",
+      payload->pt != GST_RTP_PAYLOAD_GSM, "GSM", 8000);
   res = gst_rtp_base_payload_set_outcaps (payload, NULL);
 
   return res;

gst/rtp/gstrtph261pay.c

@@ -1018,7 +1018,8 @@ gst_rtp_h261_pay_setcaps (GstRTPBasePayload * payload, GstCaps * caps)
 {
   gboolean res;
 
-  gst_rtp_base_payload_set_options (payload, "video", TRUE, "H261", 90000);
+  gst_rtp_base_payload_set_options (payload, "video",
+      payload->pt != GST_RTP_PAYLOAD_H261, "H261", 90000);
   res = gst_rtp_base_payload_set_outcaps (payload, NULL);
 
   return res;

gst/rtp/gstrtph263pay.c

@@ -440,6 +440,7 @@ gst_rtp_h263_pay_class_init (GstRtpH263PayClass * klass)
 static void
 gst_rtp_h263_pay_init (GstRtpH263Pay * rtph263pay)
 {
+  GST_RTP_BASE_PAYLOAD_PT (rtph263pay) = GST_RTP_PAYLOAD_H263;
   rtph263pay->prop_payload_mode = DEFAULT_MODE_A;
 }
 
@@ -476,8 +477,8 @@ gst_rtp_h263_pay_setcaps (GstRTPBasePayload * payload, GstCaps * caps)
     framesize = g_strdup_printf ("%d-%d", width, height);
   }
 
-  payload->pt = GST_RTP_PAYLOAD_H263;
+  gst_rtp_base_payload_set_options (payload, "video",
-  gst_rtp_base_payload_set_options (payload, "video", TRUE, "H263", 90000);
+      payload->pt != GST_RTP_PAYLOAD_H263, "H263", 90000);
 
   if (framesize != NULL) {
     res = gst_rtp_base_payload_set_outcaps (payload,

gst/rtp/gstrtpjpegpay.c

@@ -49,12 +49,18 @@ static GstStaticPadTemplate gst_rtp_jpeg_pay_sink_template =
     );
 
 static GstStaticPadTemplate gst_rtp_jpeg_pay_src_template =
-GST_STATIC_PAD_TEMPLATE ("src",
+    GST_STATIC_PAD_TEMPLATE ("src",
     GST_PAD_SRC,
     GST_PAD_ALWAYS,
     GST_STATIC_CAPS ("application/x-rtp, "
         "  media = (string) \"video\", "
-        "  payload = (int) 26 ,        "
+        "  payload = (int) " GST_RTP_PAYLOAD_JPEG_STRING ", "
+        "  clock-rate = (int) 90000,   "
+        "  encoding-name = (string) \"JPEG\", "
+        "  width = (int) [ 1, 65536 ], " "  height = (int) [ 1, 65536 ]; "
+        " application/x-rtp, "
+        "  media = (string) \"video\", "
+        "  payload = (int) " GST_RTP_PAYLOAD_DYNAMIC_STRING ", "
         "  clock-rate = (int) 90000,   "
         "  encoding-name = (string) \"JPEG\", "
         "  width = (int) [ 1, 65536 ], " "  height = (int) [ 1, 65536 ]")
@@ -283,6 +289,8 @@ gst_rtp_jpeg_pay_init (GstRtpJPEGPay * pay)
   pay->type = DEFAULT_JPEG_TYPE;
   pay->width = -1;
   pay->height = -1;
+
+  GST_RTP_BASE_PAYLOAD_PT (pay) = GST_RTP_PAYLOAD_JPEG;
 }
 
 static gboolean
@@ -321,7 +329,8 @@ gst_rtp_jpeg_pay_setcaps (GstRTPBasePayload * basepayload, GstCaps * caps)
     pay->width = GST_ROUND_UP_8 (width) / 8;
   }
 
-  gst_rtp_base_payload_set_options (basepayload, "video", TRUE, "JPEG", 90000);
+  gst_rtp_base_payload_set_options (basepayload, "video",
+      basepayload->pt != GST_RTP_PAYLOAD_JPEG, "JPEG", 90000);
 
   if (num > 0) {
     gdouble framerate;

gst/rtp/gstrtpmp2tpay.c

@@ -112,7 +112,8 @@ gst_rtp_mp2t_pay_setcaps (GstRTPBasePayload * payload, GstCaps * caps)
 {
   gboolean res;
 
-  gst_rtp_base_payload_set_options (payload, "video", TRUE, "MP2T", 90000);
+  gst_rtp_base_payload_set_options (payload, "video",
+      payload->pt != GST_RTP_PAYLOAD_MP2T, "MP2T", 90000);
   res = gst_rtp_base_payload_set_outcaps (payload, NULL);
 
   return res;

gst/rtp/gstrtpmpapay.c

@@ -104,6 +104,8 @@ static void
 gst_rtp_mpa_pay_init (GstRtpMPAPay * rtpmpapay)
 {
   rtpmpapay->adapter = gst_adapter_new ();
+
+  GST_RTP_BASE_PAYLOAD (rtpmpapay)->pt = GST_RTP_PAYLOAD_MPA;
 }
 
 static void
@@ -133,7 +135,8 @@ gst_rtp_mpa_pay_setcaps (GstRTPBasePayload * payload, GstCaps * caps)
 {
   gboolean res;
 
-  gst_rtp_base_payload_set_options (payload, "audio", TRUE, "MPA", 90000);
+  gst_rtp_base_payload_set_options (payload, "audio",
+      payload->pt != GST_RTP_PAYLOAD_MPA, "MPA", 90000);
   res = gst_rtp_base_payload_set_outcaps (payload, NULL);
 
   return res;

gst/rtp/gstrtpmpvpay.c

@@ -39,12 +39,16 @@ GST_STATIC_PAD_TEMPLATE ("sink",
     );
 
 static GstStaticPadTemplate gst_rtp_mpv_pay_src_template =
-GST_STATIC_PAD_TEMPLATE ("src",
+    GST_STATIC_PAD_TEMPLATE ("src",
     GST_PAD_SRC,
     GST_PAD_ALWAYS,
     GST_STATIC_CAPS ("application/x-rtp, "
         "media = (string) \"video\", "
         "payload = (int) " GST_RTP_PAYLOAD_MPV_STRING ", "
+        "clock-rate = (int) 90000, " "encoding-name = (string) \"MPV\"; "
+        "application/x-rtp, "
+        "media = (string) \"video\", "
+        "payload = (int) " GST_RTP_PAYLOAD_DYNAMIC_STRING ", "
         "clock-rate = (int) 90000, " "encoding-name = (string) \"MPV\"")
     );
 
@@ -131,7 +135,8 @@ gst_rtp_mpv_pay_reset (GstRTPMPVPay * pay)
 static gboolean
 gst_rtp_mpv_pay_setcaps (GstRTPBasePayload * payload, GstCaps * caps)
 {
-  gst_rtp_base_payload_set_options (payload, "video", FALSE, "MPV", 90000);
+  gst_rtp_base_payload_set_options (payload, "video",
+      payload->pt != GST_RTP_PAYLOAD_MPV, "MPV", 90000);
   return gst_rtp_base_payload_set_outcaps (payload, NULL);
 }
 

gst/rtp/gstrtppcmapay.c

@@ -86,6 +86,7 @@ gst_rtp_pcma_pay_init (GstRtpPcmaPay * rtppcmapay)
 
   rtpbaseaudiopayload = GST_RTP_BASE_AUDIO_PAYLOAD (rtppcmapay);
 
+  GST_RTP_BASE_PAYLOAD (rtppcmapay)->pt = GST_RTP_PAYLOAD_PCMA;
   GST_RTP_BASE_PAYLOAD (rtppcmapay)->clock_rate = 8000;
 
   /* tell rtpbaseaudiopayload that this is a sample based codec */
@@ -100,9 +101,8 @@ gst_rtp_pcma_pay_setcaps (GstRTPBasePayload * payload, GstCaps * caps)
 {
   gboolean res;
 
-  payload->pt = GST_RTP_PAYLOAD_PCMA;
+  gst_rtp_base_payload_set_options (payload, "audio",
-
+      payload->pt != GST_RTP_PAYLOAD_PCMA, "PCMA", 8000);
-  gst_rtp_base_payload_set_options (payload, "audio", FALSE, "PCMA", 8000);
   res = gst_rtp_base_payload_set_outcaps (payload, NULL);
 
   return res;

gst/rtp/gstrtppcmupay.c

@@ -86,6 +86,7 @@ gst_rtp_pcmu_pay_init (GstRtpPcmuPay * rtppcmupay)
 
   rtpbaseaudiopayload = GST_RTP_BASE_AUDIO_PAYLOAD (rtppcmupay);
 
+  GST_RTP_BASE_PAYLOAD (rtppcmupay)->pt = GST_RTP_PAYLOAD_PCMU;
   GST_RTP_BASE_PAYLOAD (rtppcmupay)->clock_rate = 8000;
 
   /* tell rtpbaseaudiopayload that this is a sample based codec */
@@ -100,9 +101,8 @@ gst_rtp_pcmu_pay_setcaps (GstRTPBasePayload * payload, GstCaps * caps)
 {
   gboolean res;
 
-  payload->pt = GST_RTP_PAYLOAD_PCMU;
+  gst_rtp_base_payload_set_options (payload, "audio",
-
+      payload->pt != GST_RTP_PAYLOAD_PCMU, "PCMU", 8000);
-  gst_rtp_base_payload_set_options (payload, "audio", FALSE, "PCMU", 8000);
   res = gst_rtp_base_payload_set_outcaps (payload, NULL);
 
   return res;