rtp: add BroadcomVoice payloader

4 files changed, 294 insertions, 0 deletions

diff --git a/gst/rtp/Makefile.am b/gst/rtp/Makefile.am
index 1cfaa7267..facc59ed5 100644
--- a/gst/rtp/Makefile.am
+++ b/gst/rtp/Makefile.am
@@ -6,6 +6,7 @@ libgstrtp_la_SOURCES = \
 	gstrtpchannels.c \
 	gstrtpdepay.c \
 	gstrtpac3depay.c \
+	gstrtpbvpay.c \
 	gstrtpceltdepay.c \
 	gstrtpceltpay.c \
 	gstrtpdvdepay.c \
@@ -85,6 +86,7 @@ noinst_HEADERS =			\
 		 gstrtpL16depay.h 	\
 		 gstrtpL16pay.h 	\
 		 gstrtpac3depay.h 	\
+		 gstrtpbvpay.h 	        \
 		 gstrtpceltpay.h	\
 		 gstrtpceltdepay.h	\
 		 gstrtpdvdepay.h 	\
diff --git a/gst/rtp/gstrtp.c b/gst/rtp/gstrtp.c
index a7cd1ca16..d21efccf8 100644
--- a/gst/rtp/gstrtp.c
+++ b/gst/rtp/gstrtp.c
@@ -23,6 +23,7 @@
 
 #include "gstrtpdepay.h"
 #include "gstrtpac3depay.h"
+#include "gstrtpbvpay.h"
 #include "gstrtpceltdepay.h"
 #include "gstrtpceltpay.h"
 #include "gstrtpdvdepay.h"
@@ -89,6 +90,9 @@ plugin_init (GstPlugin * plugin)
   if (!gst_rtp_ac3_depay_plugin_init (plugin))
     return FALSE;
 
+  if (!gst_rtp_bv_pay_plugin_init (plugin))
+    return FALSE;
+
   if (!gst_rtp_celt_depay_plugin_init (plugin))
     return FALSE;
 
diff --git a/gst/rtp/gstrtpbvpay.c b/gst/rtp/gstrtpbvpay.c
new file mode 100644
index 000000000..b30efb1a9
--- /dev/null
+++ b/gst/rtp/gstrtpbvpay.c
@@ -0,0 +1,230 @@
+/* GStreamer
+ * Copyright (C) <2009> Wim Taymans <wim.taymans@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., 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 "gstrtpbvpay.h"
+
+/* elementfactory information */
+static GstElementDetails gst_rtpbvpay_details = {
+  "RTP BV Payloader",
+  "Codec/Payloader/Network",
+  "Packetize BroadcomVoice audio streams into RTP packets",
+  "Wim Taymans <wim.taymans@collabora.co.uk>"
+};
+
+GST_DEBUG_CATEGORY_STATIC (rtpbvpay_debug);
+#define GST_CAT_DEFAULT (rtpbvpay_debug)
+
+static GstStaticPadTemplate gst_rtpbvpay_sink_template =
+GST_STATIC_PAD_TEMPLATE ("sink",
+    GST_PAD_SINK,
+    GST_PAD_ALWAYS,
+    GST_STATIC_CAPS ("audio/x-bv, " "mode = (int) {16, 32}")
+    );
+
+static GstStaticPadTemplate gst_rtpbvpay_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) \"BV16\";"
+        "application/x-rtp, "
+        "media = (string) \"audio\", "
+        "payload = (int) " GST_RTP_PAYLOAD_DYNAMIC_STRING ", "
+        "clock-rate = (int) 16000, " "encoding-name = (string) \"BV32\"")
+    );
+
+
+static GstCaps *gst_rtpbvpay_sink_getcaps (GstBaseRTPPayload * payload,
+    GstPad * pad);
+static gboolean gst_rtpbvpay_sink_setcaps (GstBaseRTPPayload * payload,
+    GstCaps * caps);
+
+GST_BOILERPLATE (GstRTPBVPay, gst_rtpbvpay, GstBaseRTPAudioPayload,
+    GST_TYPE_BASE_RTP_AUDIO_PAYLOAD);
+
+static void
+gst_rtpbvpay_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_rtpbvpay_sink_template));
+  gst_element_class_add_pad_template (element_class,
+      gst_static_pad_template_get (&gst_rtpbvpay_src_template));
+  gst_element_class_set_details (element_class, &gst_rtpbvpay_details);
+}
+
+static void
+gst_rtpbvpay_class_init (GstRTPBVPayClass * klass)
+{
+  GstBaseRTPPayloadClass *gstbasertppayload_class;
+
+  gstbasertppayload_class = (GstBaseRTPPayloadClass *) klass;
+
+  parent_class = g_type_class_ref (GST_TYPE_BASE_RTP_PAYLOAD);
+
+  gstbasertppayload_class->set_caps = gst_rtpbvpay_sink_setcaps;
+  gstbasertppayload_class->get_caps = gst_rtpbvpay_sink_getcaps;
+
+  GST_DEBUG_CATEGORY_INIT (rtpbvpay_debug, "rtpbvpay", 0,
+      "BroadcomVoice audio RTP payloader");
+}
+
+static void
+gst_rtpbvpay_init (GstRTPBVPay * rtpbvpay, GstRTPBVPayClass * klass)
+{
+  GstBaseRTPPayload *basertppayload;
+  GstBaseRTPAudioPayload *basertpaudiopayload;
+
+  basertppayload = GST_BASE_RTP_PAYLOAD (rtpbvpay);
+  basertpaudiopayload = GST_BASE_RTP_AUDIO_PAYLOAD (rtpbvpay);
+
+  rtpbvpay->mode = -1;
+
+  /* tell basertpaudiopayload that this is a frame based codec */
+  gst_base_rtp_audio_payload_set_frame_based (basertpaudiopayload);
+}
+
+static gboolean
+gst_rtpbvpay_sink_setcaps (GstBaseRTPPayload * basertppayload, GstCaps * caps)
+{
+  GstRTPBVPay *rtpbvpay;
+  GstBaseRTPAudioPayload *basertpaudiopayload;
+  gint mode;
+  GstStructure *structure;
+  const char *payload_name;
+
+  rtpbvpay = GST_RTP_BV_PAY (basertppayload);
+  basertpaudiopayload = GST_BASE_RTP_AUDIO_PAYLOAD (basertppayload);
+
+  structure = gst_caps_get_structure (caps, 0);
+
+  payload_name = gst_structure_get_name (structure);
+  if (g_ascii_strcasecmp ("audio/x-bv", payload_name))
+    goto wrong_caps;
+
+  if (!gst_structure_get_int (structure, "mode", &mode))
+    goto no_mode;
+
+  if (mode != 16 && mode != 32)
+    goto wrong_mode;
+
+  if (mode == 16) {
+    gst_basertppayload_set_options (basertppayload, "audio", TRUE, "BV16",
+        8000);
+    basertppayload->clock_rate = 8000;
+  } else {
+    gst_basertppayload_set_options (basertppayload, "audio", TRUE, "BV32",
+        16000);
+    basertppayload->clock_rate = 16000;
+  }
+
+  /* set options for this frame based audio codec */
+  gst_base_rtp_audio_payload_set_frame_options (basertpaudiopayload,
+      mode, mode == 16 ? 10 : 20);
+
+  if (mode != rtpbvpay->mode && rtpbvpay->mode != -1)
+    goto mode_changed;
+
+  rtpbvpay->mode = mode;
+
+  return TRUE;
+
+  /* ERRORS */
+wrong_caps:
+  {
+    GST_ERROR_OBJECT (rtpbvpay, "expected audio/x-bv, received %s",
+        payload_name);
+    return FALSE;
+  }
+no_mode:
+  {
+    GST_ERROR_OBJECT (rtpbvpay, "did not receive a mode");
+    return FALSE;
+  }
+wrong_mode:
+  {
+    GST_ERROR_OBJECT (rtpbvpay, "mode must be 16 or 32, received %d", mode);
+    return FALSE;
+  }
+mode_changed:
+  {
+    GST_ERROR_OBJECT (rtpbvpay, "Mode has changed from %d to %d! "
+        "Mode cannot change while streaming", rtpbvpay->mode, mode);
+    return FALSE;
+  }
+}
+
+/* we return the padtemplate caps with the mode field fixated to a value if we
+ * can */
+static GstCaps *
+gst_rtpbvpay_sink_getcaps (GstBaseRTPPayload * rtppayload, GstPad * pad)
+{
+  GstCaps *otherpadcaps;
+  GstCaps *caps;
+
+  otherpadcaps = gst_pad_get_allowed_caps (rtppayload->srcpad);
+  caps = gst_caps_copy (gst_pad_get_pad_template_caps (pad));
+
+  if (otherpadcaps) {
+    if (!gst_caps_is_empty (otherpadcaps)) {
+      GstStructure *structure;
+      const gchar *mode_str;
+      gint mode;
+
+      structure = gst_caps_get_structure (otherpadcaps, 0);
+
+      /* construct mode, if we can */
+      mode_str = gst_structure_get_string (structure, "encoding-name");
+      if (mode_str) {
+        if (!strcmp (mode_str, "BV16"))
+          mode = 16;
+        else if (!strcmp (mode_str, "BV32"))
+          mode = 32;
+        else
+          mode = -1;
+
+        if (mode == 16 || mode == 32) {
+          structure = gst_caps_get_structure (caps, 0);
+          gst_structure_set (structure, "mode", G_TYPE_INT, mode, NULL);
+        }
+      }
+    }
+    gst_caps_unref (otherpadcaps);
+  }
+  return caps;
+}
+
+gboolean
+gst_rtp_bv_pay_plugin_init (GstPlugin * plugin)
+{
+  return gst_element_register (plugin, "rtpbvpay",
+      GST_RANK_NONE, GST_TYPE_RTP_BV_PAY);
+}
diff --git a/gst/rtp/gstrtpbvpay.h b/gst/rtp/gstrtpbvpay.h
new file mode 100644
index 000000000..ed7f2987a
--- /dev/null
+++ b/gst/rtp/gstrtpbvpay.h
@@ -0,0 +1,58 @@
+/* GStreamer
+ * Copyright (C) <2009> Wim Taymans <wim.taymans@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., 59 Temple Place - Suite 330,
+ * Boston, MA 02111-1307, USA.
+ */
+
+#ifndef __GST_RTP_BV_PAY_H__
+#define __GST_RTP_BV_PAY_H__
+
+#include <gst/gst.h>
+#include <gst/rtp/gstbasertpaudiopayload.h>
+
+G_BEGIN_DECLS
+
+#define GST_TYPE_RTP_BV_PAY \
+  (gst_rtpbvpay_get_type())
+#define GST_RTP_BV_PAY(obj) \
+  (G_TYPE_CHECK_INSTANCE_CAST((obj),GST_TYPE_RTP_BV_PAY,GstRTPBVPay))
+#define GST_RTP_BV_PAY_CLASS(klass) \
+  (G_TYPE_CHECK_CLASS_CAST((klass),GST_TYPE_RTP_BV_PAY,GstRTPBVPayClass))
+#define GST_IS_RTP_BV_PAY(obj) \
+  (G_TYPE_CHECK_INSTANCE_TYPE((obj),GST_TYPE_RTP_BV_PAY))
+#define GST_IS_RTP_BV_PAY_CLASS(klass) \
+  (G_TYPE_CHECK_CLASS_TYPE((klass),GST_TYPE_RTP_BV_PAY))
+
+typedef struct _GstRTPBVPay GstRTPBVPay;
+typedef struct _GstRTPBVPayClass GstRTPBVPayClass;
+
+struct _GstRTPBVPay
+{
+  GstBaseRTPAudioPayload audiopayload;
+
+  gint mode;
+};
+
+struct _GstRTPBVPayClass
+{
+  GstBaseRTPAudioPayloadClass parent_class;
+};
+
+gboolean gst_rtp_bv_pay_plugin_init (GstPlugin * plugin);
+
+G_END_DECLS
+
+#endif /* __GST_RTP_BV_PAY_H__ */