--- /dev/null
+/* 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);
+}
--- /dev/null
+/* 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__ */