1 files changed, 165 insertions, 0 deletions
diff --git a/gst/rtp/gstrtppcmadepay.c b/gst/rtp/gstrtppcmadepay.c
new file mode 100755
index 0000000..8f96250
--- /dev/null
+++ b/gst/rtp/gstrtppcmadepay.c
@@ -0,0 +1,165 @@
+/* GStreamer
+ * Copyright (C) <1999> Erik Walthinsen <omega@cse.ogi.edu>
+ * Copyright (C) <2005> Edgard Lima <edgard.lima@indt.org.br>
+ * Copyright (C) <2005> Zeeshan Ali <zeenix@gmail.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., 51 Franklin St, Fifth Floor,
+ * Boston, MA 02110-1301, USA.
+ */
+
+#ifdef HAVE_CONFIG_H
+#  include "config.h"
+#endif
+
+#include <string.h>
+#include <gst/rtp/gstrtpbuffer.h>
+#include "gstrtppcmadepay.h"
+
+/* RtpPcmaDepay signals and args */
+enum
+{
+  /* FILL ME */
+  LAST_SIGNAL
+};
+
+enum
+{
+  PROP_0
+};
+
+static GstStaticPadTemplate gst_rtp_pcma_depay_sink_template =
+    GST_STATIC_PAD_TEMPLATE ("sink",
+    GST_PAD_SINK,
+    GST_PAD_ALWAYS,
+    GST_STATIC_CAPS ("application/x-rtp, "
+        "media = (string) \"audio\", "
+        "payload = (int) " GST_RTP_PAYLOAD_PCMA_STRING ", "
+        "clock-rate = (int) 8000;"
+        "application/x-rtp, "
+        "media = (string) \"audio\", "
+        "clock-rate = (int) [1, MAX ], encoding-name = (string) \"PCMA\"")
+    );
+
+static GstStaticPadTemplate gst_rtp_pcma_depay_src_template =
+GST_STATIC_PAD_TEMPLATE ("src",
+    GST_PAD_SRC,
+    GST_PAD_ALWAYS,
+    GST_STATIC_CAPS ("audio/x-alaw, channels = (int) 1, rate = (int) [1, MAX ]")
+    );
+
+static GstBuffer *gst_rtp_pcma_depay_process (GstRTPBaseDepayload * depayload,
+    GstBuffer * buf);
+static gboolean gst_rtp_pcma_depay_setcaps (GstRTPBaseDepayload * depayload,
+    GstCaps * caps);
+
+#define gst_rtp_pcma_depay_parent_class parent_class
+G_DEFINE_TYPE (GstRtpPcmaDepay, gst_rtp_pcma_depay,
+    GST_TYPE_RTP_BASE_DEPAYLOAD);
+
+static void
+gst_rtp_pcma_depay_class_init (GstRtpPcmaDepayClass * klass)
+{
+  GstElementClass *gstelement_class;
+  GstRTPBaseDepayloadClass *gstrtpbasedepayload_class;
+
+  gstelement_class = (GstElementClass *) klass;
+  gstrtpbasedepayload_class = (GstRTPBaseDepayloadClass *) klass;
+
+  gst_element_class_add_pad_template (gstelement_class,
+      gst_static_pad_template_get (&gst_rtp_pcma_depay_src_template));
+  gst_element_class_add_pad_template (gstelement_class,
+      gst_static_pad_template_get (&gst_rtp_pcma_depay_sink_template));
+
+  gst_element_class_set_static_metadata (gstelement_class,
+      "RTP PCMA depayloader", "Codec/Depayloader/Network/RTP",
+      "Extracts PCMA audio from RTP packets",
+      "Edgard Lima <edgard.lima@indt.org.br>, Zeeshan Ali <zeenix@gmail.com>");
+
+  gstrtpbasedepayload_class->process = gst_rtp_pcma_depay_process;
+  gstrtpbasedepayload_class->set_caps = gst_rtp_pcma_depay_setcaps;
+}
+
+static void
+gst_rtp_pcma_depay_init (GstRtpPcmaDepay * rtppcmadepay)
+{
+  GstRTPBaseDepayload *depayload;
+
+  depayload = GST_RTP_BASE_DEPAYLOAD (rtppcmadepay);
+
+  gst_pad_use_fixed_caps (GST_RTP_BASE_DEPAYLOAD_SRCPAD (depayload));
+}
+
+static gboolean
+gst_rtp_pcma_depay_setcaps (GstRTPBaseDepayload * depayload, GstCaps * caps)
+{
+  GstCaps *srccaps;
+  GstStructure *structure;
+  gboolean ret;
+  gint clock_rate;
+
+  structure = gst_caps_get_structure (caps, 0);
+
+  if (!gst_structure_get_int (structure, "clock-rate", &clock_rate))
+    clock_rate = 8000;          /* default */
+  depayload->clock_rate = clock_rate;
+
+  srccaps = gst_caps_new_simple ("audio/x-alaw",
+      "channels", G_TYPE_INT, 1, "rate", G_TYPE_INT, clock_rate, NULL);
+  ret = gst_pad_set_caps (GST_RTP_BASE_DEPAYLOAD_SRCPAD (depayload), srccaps);
+  gst_caps_unref (srccaps);
+
+  return ret;
+}
+
+static GstBuffer *
+gst_rtp_pcma_depay_process (GstRTPBaseDepayload * depayload, GstBuffer * buf)
+{
+  GstBuffer *outbuf = NULL;
+  gboolean marker;
+  guint len;
+  GstRTPBuffer rtp = { NULL };
+
+  gst_rtp_buffer_map (buf, GST_MAP_READ, &rtp);
+
+  marker = gst_rtp_buffer_get_marker (&rtp);
+
+  GST_DEBUG ("process : got %" G_GSIZE_FORMAT " bytes, mark %d ts %u seqn %d",
+      gst_buffer_get_size (buf), marker,
+      gst_rtp_buffer_get_timestamp (&rtp), gst_rtp_buffer_get_seq (&rtp));
+
+  len = gst_rtp_buffer_get_payload_len (&rtp);
+  outbuf = gst_rtp_buffer_get_payload_buffer (&rtp);
+  gst_rtp_buffer_unmap (&rtp);
+
+  if (outbuf) {
+    GST_BUFFER_DURATION (outbuf) =
+        gst_util_uint64_scale_int (len, GST_SECOND, depayload->clock_rate);
+
+    if (marker) {
+      /* mark start of talkspurt with RESYNC */
+      GST_BUFFER_FLAG_SET (outbuf, GST_BUFFER_FLAG_RESYNC);
+    }
+  }
+
+
+  return outbuf;
+}
+
+gboolean
+gst_rtp_pcma_depay_plugin_init (GstPlugin * plugin)
+{
+  return gst_element_register (plugin, "rtppcmadepay",
+      GST_RANK_SECONDARY, GST_TYPE_RTP_PCMA_DEPAY);
+}