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List:       gstreamer-cvs
Subject:    gst-plugins-good: Revert "rtph264pay: Restructuring to allow for adding optional caps"
From:       wtay () kemper ! freedesktop ! org (Wim Taymans)
Date:       2013-05-31 13:32:11
Message-ID: 20130531133211.BFDEED4005 () kemper ! freedesktop ! org
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Module: gst-plugins-good
Branch: master
Commit: 1f0600ee6f3492e0e33ce6b6601fc53d775a5c4d
URL:    http://cgit.freedesktop.org/gstreamer/gst-plugins-good/commit/?id=1f0600ee6f3492e0e33ce6b6601fc53d775a5c4d


Author: Wim Taymans <wim.taymans@collabora.co.uk>
Date:   Fri May 31 15:12:08 2013 +0200

Revert "rtph264pay: Restructuring to allow for adding optional caps"

This reverts commit 61666898cfe89a1b21d3e6850ab44f5b1633ed79.

This commit changes what the set_sps_pps() function does, not it doesn't
set caps anymore (and should have been renamed). The main problem is that
not all call sites are updated and thus leak the string.

---

 gst/rtp/gstrtph264pay.c |   46 +++++++++++++++++++---------------------------
 1 files changed, 19 insertions(+), 27 deletions(-)

diff --git a/gst/rtp/gstrtph264pay.c b/gst/rtp/gstrtph264pay.c
index 4e4621c..1929117 100644
--- a/gst/rtp/gstrtph264pay.c
+++ b/gst/rtp/gstrtph264pay.c
@@ -344,14 +344,16 @@ done:
 
 /* take the currently configured SPS and PPS lists and set them on the caps as
  * sprop-parameter-sets */
-static gchar *
+static gboolean
 gst_rtp_h264_pay_set_sps_pps (GstRTPBasePayload * basepayload)
 {
   GstRtpH264Pay *payloader = GST_RTP_H264_PAY (basepayload);
+  gchar *profile;
   gchar *set;
   GList *walk;
   GString *sprops;
   guint count;
+  gboolean res;
   GstMapInfo map;
 
   sprops = g_string_new ("");
@@ -381,12 +383,19 @@ gst_rtp_h264_pay_set_sps_pps (GstRTPBasePayload * basepayload)
     count++;
   }
 
-  if (G_UNLIKELY (count == 0)) {
-    g_string_free (sprops, TRUE);
-    return NULL;
+  if (G_LIKELY (count)) {
+    /* profile is 24 bit. Force it to respect the limit */
+    profile = g_strdup_printf ("%06x", payloader->profile & 0xffffff);
+    /* combine into output caps */
+    res = gst_rtp_base_payload_set_outcaps (basepayload,
+        "sprop-parameter-sets", G_TYPE_STRING, sprops->str, NULL);
+    g_free (profile);
+  } else {
+    res = gst_rtp_base_payload_set_outcaps (basepayload, NULL);
   }
+  g_string_free (sprops, TRUE);
 
-  return g_string_free (sprops, FALSE);
+  return res;
 }
 
 static GList *
@@ -424,8 +433,6 @@ gst_rtp_h264_pay_setcaps (GstRTPBasePayload * basepayload, \
GstCaps * caps)  gsize size;
   GstBuffer *buffer;
   const gchar *alignment, *stream_format;
-  gchar *sprops;
-  gboolean caps_set;
 
   rtph264pay = GST_RTP_H264_PAY (basepayload);
 
@@ -551,25 +558,10 @@ gst_rtp_h264_pay_setcaps (GstRTPBasePayload * basepayload, \
GstCaps * caps)  }
     gst_buffer_unmap (buffer, &map);
     /* and update the caps with the collected data */
-    sprops = gst_rtp_h264_pay_set_sps_pps (basepayload);
+    if (!gst_rtp_h264_pay_set_sps_pps (basepayload))
+      goto set_sps_pps_failed;
   } else {
     GST_DEBUG_OBJECT (rtph264pay, "have bytestream h264");
-    sprops = NULL;
-  }
-
-  if (sprops != NULL) {
-    caps_set = gst_rtp_base_payload_set_outcaps (basepayload,
-        "sprop-parameter-sets", G_TYPE_STRING, sprops, NULL);
-  } else {
-    caps_set = gst_rtp_base_payload_set_outcaps (basepayload, NULL);
-  }
-
-  if (sprops != NULL) {
-    g_free (sprops);
-  }
-
-  if (!caps_set) {
-    goto set_caps_failed;
   }
 
   return TRUE;
@@ -589,10 +581,10 @@ avcc_error:
     GST_ERROR_OBJECT (rtph264pay, "avcC too small ");
     goto error;
   }
-set_caps_failed:
+set_sps_pps_failed:
   {
-    GST_ERROR_OBJECT (rtph264pay, "failed to set caps");
-    return FALSE;
+    GST_ERROR_OBJECT (rtph264pay, "failed to set sps/pps");
+    goto error;
   }
 error:
   {

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