1 files changed, 172 insertions, 17 deletions

diff --git a/gst/rtp/gstrtph264pay.c b/gst/rtp/gstrtph264pay.c
index a4830e6..1d0721e 100644
--- a/gst/rtp/gstrtph264pay.c
+++ b/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,
@@ -124,10 +128,10 @@ gst_rtp_h264_pay_base_init (gpointer klass)
 {
   GstElementClass *element_class = GST_ELEMENT_CLASS (klass);
 
-  gst_element_class_add_pad_template (element_class,
-      gst_static_pad_template_get (&gst_rtp_h264_pay_src_template));
-  gst_element_class_add_pad_template (element_class,
-      gst_static_pad_template_get (&gst_rtp_h264_pay_sink_template));
+  gst_element_class_add_static_pad_template (element_class,
+      &gst_rtp_h264_pay_src_template);
+  gst_element_class_add_static_pad_template (element_class,
+      &gst_rtp_h264_pay_sink_template);
 
   gst_element_class_set_details_simple (element_class, "RTP H264 payloader",
       "Codec/Payloader/Network/RTP",
@@ -191,6 +195,7 @@ 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;
@@ -243,6 +248,133 @@ 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);
+          }
+        } else {
+          /* Invalid profile-level-id means baseline */
+
+          gst_structure_set (new_s,
+              "profile", G_TYPE_STRING, "constrained-baseline", NULL);
+        }
+      } else {
+        /* No profile-level-id also means baseline */
+
+        gst_structure_set (new_s,
+            "profile", G_TYPE_STRING, "constrained-baseline", NULL);
+      }
+
+      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
@@ -298,6 +430,7 @@ gst_rtp_h264_pay_setcaps (GstBaseRTPPayload * basepayload, GstCaps * caps)
   const GValue *value;
   guint8 *data;
   guint size;
+  const gchar *alignment;
 
   rtph264pay = GST_RTP_H264_PAY (basepayload);
 
@@ -307,7 +440,12 @@ gst_rtp_h264_pay_setcaps (GstBaseRTPPayload * basepayload, GstCaps * caps)
    * NALs */
   gst_basertppayload_set_options (basepayload, "video", TRUE, "H264", 90000);
 
-#if 0 // yoserb yi - not to check sps/pps when savsenc_h264 and rtph264pay are linked
+  alignment = gst_structure_get_string (str, "alignment");
+  if (alignment && !strcmp (alignment, "au"))
+    rtph264pay->au_alignment = TRUE;
+  else
+    rtph264pay->au_alignment = FALSE;
+
   /* packetized AVC video has a codec_data */
   if ((value = gst_structure_get_value (str, "codec_data"))) {
     GstBuffer *buffer;
@@ -409,10 +547,6 @@ gst_rtp_h264_pay_setcaps (GstBaseRTPPayload * basepayload, GstCaps * caps)
     GST_DEBUG_OBJECT (rtph264pay, "have bytestream h264");
     rtph264pay->packetized = FALSE;
   }
-#else
-    GST_DEBUG_OBJECT (rtph264pay, "have bytestream h264");
-    rtph264pay->packetized = FALSE;
-#endif
 
   return TRUE;
 
@@ -646,7 +780,7 @@ gst_rtp_h264_pay_decode_nal (GstRtpH264Pay * payloader,
 static GstFlowReturn
 gst_rtp_h264_pay_payload_nal (GstBaseRTPPayload * basepayload,
     const guint8 * data, guint size, GstClockTime timestamp,
-    GstBuffer * buffer_orig);
+    GstBuffer * buffer_orig, gboolean end_of_au);
 
 static GstFlowReturn
 gst_rtp_h264_pay_send_sps_pps (GstBaseRTPPayload * basepayload,
@@ -662,7 +796,7 @@ gst_rtp_h264_pay_send_sps_pps (GstBaseRTPPayload * basepayload,
     /* resend SPS */
     ret = gst_rtp_h264_pay_payload_nal (basepayload,
         GST_BUFFER_DATA (sps_buf), GST_BUFFER_SIZE (sps_buf), timestamp,
-        sps_buf);
+        sps_buf, FALSE);
     /* Not critical here; but throw a warning */
     if (ret != GST_FLOW_OK)
       GST_WARNING ("Problem pushing SPS");
@@ -674,7 +808,7 @@ gst_rtp_h264_pay_send_sps_pps (GstBaseRTPPayload * basepayload,
     /* resend PPS */
     ret = gst_rtp_h264_pay_payload_nal (basepayload,
         GST_BUFFER_DATA (pps_buf), GST_BUFFER_SIZE (pps_buf), timestamp,
-        pps_buf);
+        pps_buf, FALSE);
     /* Not critical here; but throw a warning */
     if (ret != GST_FLOW_OK)
       GST_WARNING ("Problem pushing PPS");
@@ -689,7 +823,7 @@ gst_rtp_h264_pay_send_sps_pps (GstBaseRTPPayload * basepayload,
 static GstFlowReturn
 gst_rtp_h264_pay_payload_nal (GstBaseRTPPayload * basepayload,
     const guint8 * data, guint size, GstClockTime timestamp,
-    GstBuffer * buffer_orig)
+    GstBuffer * buffer_orig, gboolean end_of_au)
 {
   GstRtpH264Pay *rtph264pay;
   GstFlowReturn ret;
@@ -768,7 +902,7 @@ gst_rtp_h264_pay_payload_nal (GstBaseRTPPayload * basepayload,
     }
 
     /* only set the marker bit on packets containing access units */
-    if (IS_ACCESS_UNIT (nalType)) {
+    if (IS_ACCESS_UNIT (nalType) && end_of_au) {
       gst_rtp_buffer_set_marker (outbuf, 1);
     }
 
@@ -858,7 +992,7 @@ gst_rtp_h264_pay_payload_nal (GstBaseRTPPayload * basepayload,
         end = 1;
       }
       if (IS_ACCESS_UNIT (nalType)) {
-        gst_rtp_buffer_set_marker (outbuf, end);
+        gst_rtp_buffer_set_marker (outbuf, end && end_of_au);
       }
 
       /* FU indicator */
@@ -958,6 +1092,7 @@ gst_rtp_h264_pay_handle_buffer (GstBaseRTPPayload * basepayload,
 
     while (size > nal_length_size) {
       gint i;
+      gboolean end_of_au = FALSE;
 
       nal_len = 0;
       for (i = 0; i < nal_length_size; i++) {
@@ -976,9 +1111,16 @@ gst_rtp_h264_pay_handle_buffer (GstBaseRTPPayload * basepayload,
             nal_len);
       }
 
+      /* If we're at the end of the buffer, then we're at the end of the
+       * access unit
+       */
+      if (rtph264pay->au_alignment && size - nal_len <= nal_length_size) {
+        end_of_au = TRUE;
+      }
+
       ret =
           gst_rtp_h264_pay_payload_nal (basepayload, data, nal_len, timestamp,
-          buffer);
+          buffer, end_of_au);
       if (ret != GST_FLOW_OK)
         break;
 
@@ -1080,6 +1222,7 @@ gst_rtp_h264_pay_handle_buffer (GstBaseRTPPayload * basepayload,
 
     for (i = 0; i < nal_queue->len; i++) {
       guint size;
+      gboolean end_of_au = FALSE;
 
       nal_len = g_array_index (nal_queue, guint, i);
       /* skip start code */
@@ -1095,10 +1238,22 @@ gst_rtp_h264_pay_handle_buffer (GstBaseRTPPayload * basepayload,
         for (; size > 1 && data[size - 1] == 0x0; size--)
           /* skip */ ;
 
+      /* If it's the last nal unit we have in non-bytestream mode, we can
+       * assume it's the end of an access-unit
+       *
+       * FIXME: We need to wait until the next packet or EOS to
+       * actually payload the NAL so we can know if the current NAL is
+       * the last one of an access unit or not if we are in bytestream mode
+       */
+      if (rtph264pay->au_alignment &&
+          rtph264pay->scan_mode != GST_H264_SCAN_MODE_BYTESTREAM &&
+          i == nal_queue->len - 1)
+        end_of_au = TRUE;
+
       /* put the data in one or more RTP packets */
       ret =
           gst_rtp_h264_pay_payload_nal (basepayload, data, size, timestamp,
-          buffer);
+          buffer, end_of_au);
       if (ret != GST_FLOW_OK) {
         break;
       }