gst/rtp/: Added G726 pay/depayloaders. Fixes #538891.

Original commit message from CVS:
Patch by: mersad <mersad at axis dot com>
* gst/rtp/Makefile.am:
* gst/rtp/gstrtp.c: (plugin_init):
* gst/rtp/gstrtpg726depay.c: (gst_rtp_g726_depay_base_init),
(gst_rtp_g726_depay_class_init), (gst_rtp_g726_depay_init),
(gst_rtp_g726_depay_setcaps), (gst_rtp_g726_depay_process),
(gst_rtp_g726_depay_plugin_init):
* gst/rtp/gstrtpg726depay.h:
* gst/rtp/gstrtpg726pay.c: (gst_rtp_g726_pay_base_init),
(gst_rtp_g726_pay_class_init), (gst_rtp_g726_pay_init),
(gst_rtp_g726_pay_setcaps), (gst_rtp_g726_pay_plugin_init):
* gst/rtp/gstrtpg726pay.h:
Added G726 pay/depayloaders. Fixes #538891.

1 files changed, 181 insertions, 0 deletions

diff --git a/gst/rtp/gstrtpg726pay.c b/gst/rtp/gstrtpg726pay.c
new file mode 100644
index 00000000..59af7ab3
--- /dev/null
+++ b/gst/rtp/gstrtpg726pay.c
@@ -0,0 +1,181 @@
+/* GStreamer
+ * Copyright (C) 1999 Erik Walthinsen <omega@cse.ogi.edu>
+ * Copyright (C) 2005 Edgard Lima <edgard.lima@indt.org.br>
+ * Copyright (C) 2005 Nokia Corporation <kai.vehmanen@nokia.com>
+ * Copyright (C) 2007,2008 Axis Communications <dev-gstreamer@axis.com>
+ *
+ * This library is free software; you can redistribute it and/or
+ * modify it under the terms of the GNU Library General Public
+ * License as published by the Free Software Foundation; either
+ * version 2 of the License, or (at your option) any later version.
+ *
+ * This library is distributed in the hope that it will be useful,
+ * but WITHOUT ANY WARRANTY; without even the implied warranty of
+ * MERCHANTABILITY or FITNESS FOR A PARTICULAR PURPOSE.  See the GNU
+ * Library General Public License for more details.
+ *
+ * You should have received a copy of the GNU Library General Public
+ * License along with this library; if not, write to the
+ * Free Software Foundation, Inc., 59 Temple Place - Suite 330,
+ * Boston, MA 02111-1307, USA.
+ */
+
+#ifdef HAVE_CONFIG_H
+#  include "config.h"
+#endif
+
+#include <stdlib.h>
+#include <string.h>
+#include <gst/rtp/gstrtpbuffer.h>
+
+#include "gstrtpg726pay.h"
+
+static const GstElementDetails gst_rtp_g726_pay_details =
+GST_ELEMENT_DETAILS ("RTP packet payloader",
+    "Codec/Payloader/Network",
+    "Payload-encodes G.726 audio into a RTP packet",
+    "Axis Communications <dev-gstreamer@axis.com>");
+
+static GstStaticPadTemplate gst_rtp_g726_pay_sink_template =
+    GST_STATIC_PAD_TEMPLATE ("sink",
+    GST_PAD_SINK,
+    GST_PAD_ALWAYS,
+    GST_STATIC_CAPS ("audio/x-adpcm, "
+        "channels = (int) 1, "
+        "rate = (int) 8000, "
+        "bitrate = (int) { 16000, 24000, 32000, 40000 }, "
+        "layout = (string) \"g726\"; "
+        "audio/G723, channels=(int)1, rate=(int)8000; "
+        "audio/32KADPCM, channels=(int)1, rate=(int)8000")
+    );
+
+static GstStaticPadTemplate gst_rtp_g726_pay_src_template =
+GST_STATIC_PAD_TEMPLATE ("src",
+    GST_PAD_SRC,
+    GST_PAD_ALWAYS,
+    GST_STATIC_CAPS ("application/x-rtp, "
+        "media = (string) \"audio\", "
+        "payload = (int) " GST_RTP_PAYLOAD_DYNAMIC_STRING ", "
+        "clock-rate = (int) 8000, "
+        "encoding-name = (string) { \"G726-16\", \"G726-24\", \"G726-32\", \"G726-40\" } ")
+    );
+
+static gboolean gst_rtp_g726_pay_setcaps (GstBaseRTPPayload * payload,
+    GstCaps * caps);
+
+GST_BOILERPLATE (GstRtpG726Pay, gst_rtp_g726_pay, GstBaseRTPAudioPayload,
+    GST_TYPE_BASE_RTP_AUDIO_PAYLOAD);
+
+static void
+gst_rtp_g726_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_g726_pay_sink_template));
+  gst_element_class_add_pad_template (element_class,
+      gst_static_pad_template_get (&gst_rtp_g726_pay_src_template));
+  gst_element_class_set_details (element_class, &gst_rtp_g726_pay_details);
+}
+
+static void
+gst_rtp_g726_pay_class_init (GstRtpG726PayClass * klass)
+{
+  GObjectClass *gobject_class;
+  GstElementClass *gstelement_class;
+  GstBaseRTPPayloadClass *gstbasertppayload_class;
+
+  gobject_class = (GObjectClass *) klass;
+  gstelement_class = (GstElementClass *) klass;
+  gstbasertppayload_class = (GstBaseRTPPayloadClass *) klass;
+
+  parent_class = g_type_class_peek_parent (klass);
+
+  gstbasertppayload_class->set_caps = gst_rtp_g726_pay_setcaps;
+}
+
+static void
+gst_rtp_g726_pay_init (GstRtpG726Pay * rtpg726pay, GstRtpG726PayClass * klass)
+{
+  GstBaseRTPAudioPayload *basertpaudiopayload;
+
+  basertpaudiopayload = GST_BASE_RTP_AUDIO_PAYLOAD (rtpg726pay);
+
+  GST_BASE_RTP_PAYLOAD (rtpg726pay)->clock_rate = 8000;
+
+  /* sample based codec */
+  gst_base_rtp_audio_payload_set_sample_based (basertpaudiopayload);
+}
+
+static gboolean
+gst_rtp_g726_pay_setcaps (GstBaseRTPPayload * payload, GstCaps * caps)
+{
+  gchar *encoding_name;
+  GstStructure *structure = gst_caps_get_structure (caps, 0);
+  const gchar *stname = gst_structure_get_name (structure);
+  GstBaseRTPAudioPayload *basertpaudiopayload;
+  gint bitrate;
+
+  basertpaudiopayload = GST_BASE_RTP_AUDIO_PAYLOAD (payload);
+
+  if (strcmp ("audio/x-adpcm", stname) == 0) {
+    if (!gst_structure_get_int (structure, "bitrate", &bitrate))
+      bitrate = 32000;
+  } else if (strcmp ("audio/G723", stname) == 0) {
+    bitrate = 24000;
+  } else if (strcmp ("audio/32KADPCM", stname) == 0) {
+    bitrate = 32000;
+  } else
+    goto invalid_caps;
+
+  switch (bitrate) {
+    case 16000:
+      encoding_name = g_strdup ("G726-16");
+      gst_base_rtp_audio_payload_set_samplebits_options (basertpaudiopayload,
+          2);
+      break;
+    case 24000:
+      encoding_name = g_strdup ("G726-24");
+      gst_base_rtp_audio_payload_set_samplebits_options (basertpaudiopayload,
+          3);
+      break;
+    case 32000:
+      encoding_name = g_strdup ("G726-32");
+      gst_base_rtp_audio_payload_set_samplebits_options (basertpaudiopayload,
+          4);
+      break;
+    case 40000:
+      encoding_name = g_strdup ("G726-40");
+      gst_base_rtp_audio_payload_set_samplebits_options (basertpaudiopayload,
+          5);
+      break;
+    default:
+      goto invalid_bitrate;
+  }
+
+  gst_basertppayload_set_options (payload, "audio", TRUE, encoding_name, 8000);
+  gst_basertppayload_set_outcaps (payload, NULL);
+
+  g_free (encoding_name);
+
+  return TRUE;
+
+  /* ERRORS */
+invalid_caps:
+  {
+    GST_ERROR_OBJECT (payload, "unknown caps specified");
+    return FALSE;
+  }
+invalid_bitrate:
+  {
+    GST_ERROR_OBJECT (payload, "invalid bitrate %d specified", bitrate);
+    return FALSE;
+  }
+}
+
+gboolean
+gst_rtp_g726_pay_plugin_init (GstPlugin * plugin)
+{
+  return gst_element_register (plugin, "rtpg726pay",
+      GST_RANK_NONE, GST_TYPE_RTP_G726_PAY);
+}