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authorOlivier CrĂȘte <olivier.crete@collabora.co.uk>2009-02-10 17:20:57 +0000
committerOlivier CrĂȘte <olivier.crete@collabora.co.uk>2009-03-04 13:24:34 -0500
commite4b8c514ccdb072881a7378b46d4325a368a88d6 (patch)
tree173d87a0dfcb8bf9cd31454d8044460cb665ca52
parent5a5ba49c9b5e20dcbf85c578abcc67c7c9de9377 (diff)
Move siren rtp pay/depay from gst-plugins-farsight
-rw-r--r--gst/rtp/Makefile.am6
-rw-r--r--gst/rtp/gstrtp.c8
-rw-r--r--gst/rtp/gstrtpsirendepay.c133
-rw-r--r--gst/rtp/gstrtpsirendepay.h57
-rw-r--r--gst/rtp/gstrtpsirenpay.c164
-rw-r--r--gst/rtp/gstrtpsirenpay.h55
6 files changed, 422 insertions, 1 deletions
diff --git a/gst/rtp/Makefile.am b/gst/rtp/Makefile.am
index 534f6108b..679f011ab 100644
--- a/gst/rtp/Makefile.am
+++ b/gst/rtp/Makefile.am
@@ -46,6 +46,8 @@ libgstrtp_la_SOURCES = \
gstrtpmp4gpay.c \
gstrtpmp4adepay.c \
gstrtpmp4apay.c \
+ gstrtpsirenpay.c \
+ gstrtpsirendepay.c \
gstrtpspeexdepay.c \
gstrtpspeexpay.c \
gstrtpsv3vdepay.c \
@@ -55,7 +57,7 @@ libgstrtp_la_SOURCES = \
gstrtpvorbispay.c \
gstrtpvrawdepay.c \
gstrtpvrawpay.c
-
+
if HAVE_WINSOCK2_H
WINSOCK2_LIBS = -lws2_32
@@ -116,6 +118,8 @@ noinst_HEADERS = \
gstrtpmp4apay.h \
gstrtpdepay.h \
gstasteriskh263.h \
+ gstrtpsirenpay.h \
+ gstrtpsirendepay.h \
gstrtpspeexdepay.h \
gstrtpspeexpay.h \
gstrtpsv3vdepay.h \
diff --git a/gst/rtp/gstrtp.c b/gst/rtp/gstrtp.c
index 22022c602..fe81cc4ec 100644
--- a/gst/rtp/gstrtp.c
+++ b/gst/rtp/gstrtp.c
@@ -63,6 +63,8 @@
#include "gstrtpmp4apay.h"
#include "gstrtpmp4gdepay.h"
#include "gstrtpmp4gpay.h"
+#include "gstrtpsirenpay.h"
+#include "gstrtpsirendepay.h"
#include "gstrtpspeexpay.h"
#include "gstrtpspeexdepay.h"
#include "gstrtpsv3vdepay.h"
@@ -202,6 +204,12 @@ plugin_init (GstPlugin * plugin)
if (!gst_rtp_mp4g_pay_plugin_init (plugin))
return FALSE;
+ if (!gst_rtp_siren_pay_plugin_init (plugin))
+ return FALSE;
+
+ if (!gst_rtp_siren_depay_plugin_init (plugin))
+ return FALSE;
+
if (!gst_rtp_speex_pay_plugin_init (plugin))
return FALSE;
diff --git a/gst/rtp/gstrtpsirendepay.c b/gst/rtp/gstrtpsirendepay.c
new file mode 100644
index 000000000..445181495
--- /dev/null
+++ b/gst/rtp/gstrtpsirendepay.c
@@ -0,0 +1,133 @@
+/*
+ * Siren Depayloader Gst Element
+ *
+ * @author: Youness Alaoui <kakaroto@kakaroto.homelinux.net>
+ *
+ * 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 <string.h>
+#include <stdlib.h>
+#include <gst/rtp/gstrtpbuffer.h>
+#include "gstrtpsirendepay.h"
+
+/* elementfactory information */
+static const GstElementDetails gst_rtp_siren_depay_details =
+GST_ELEMENT_DETAILS ("RTP Siren packet depayloader",
+ "Codec/Depayloader/Network",
+ "Extracts Siren audio from RTP packets",
+ "Philippe Kalaf <philippe.kalaf@collabora.co.uk>");
+
+
+static GstStaticPadTemplate gst_rtp_siren_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_DYNAMIC_STRING ", "
+ "clock-rate = (int) 16000, "
+ "encoding-name = (string) \"SIREN\", " "dct-length = (int) 320")
+ );
+
+static GstStaticPadTemplate gst_rtp_siren_depay_src_template =
+GST_STATIC_PAD_TEMPLATE ("src",
+ GST_PAD_SRC,
+ GST_PAD_ALWAYS,
+ GST_STATIC_CAPS ("audio/x-siren, " "dct-length = (int) 320")
+ );
+
+static GstBuffer *gst_rtp_siren_depay_process (GstBaseRTPDepayload * depayload,
+ GstBuffer * buf);
+static gboolean gst_rtp_siren_depay_setcaps (GstBaseRTPDepayload * depayload,
+ GstCaps * caps);
+
+GST_BOILERPLATE (GstRTPSirenDepay, gst_rtp_siren_depay, GstBaseRTPDepayload,
+ GST_TYPE_BASE_RTP_DEPAYLOAD);
+
+static void
+gst_rtp_siren_depay_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_siren_depay_src_template));
+ gst_element_class_add_pad_template (element_class,
+ gst_static_pad_template_get (&gst_rtp_siren_depay_sink_template));
+ gst_element_class_set_details (element_class, &gst_rtp_siren_depay_details);
+}
+
+static void
+gst_rtp_siren_depay_class_init (GstRTPSirenDepayClass * klass)
+{
+ GObjectClass *gobject_class;
+ GstElementClass *gstelement_class;
+ GstBaseRTPDepayloadClass *gstbasertpdepayload_class;
+
+ gobject_class = (GObjectClass *) klass;
+ gstelement_class = (GstElementClass *) klass;
+ gstbasertpdepayload_class = (GstBaseRTPDepayloadClass *) klass;
+
+ gstbasertpdepayload_class->process = gst_rtp_siren_depay_process;
+ gstbasertpdepayload_class->set_caps = gst_rtp_siren_depay_setcaps;
+}
+
+static void
+gst_rtp_siren_depay_init (GstRTPSirenDepay * rtpsirendepay,
+ GstRTPSirenDepayClass * klass)
+{
+
+}
+
+static gboolean
+gst_rtp_siren_depay_setcaps (GstBaseRTPDepayload * depayload, GstCaps * caps)
+{
+ GstCaps *srccaps;
+ gboolean ret;
+
+ srccaps = gst_caps_new_simple ("audio/x-siren",
+ "dct-length", G_TYPE_INT, 320, NULL);
+ ret = gst_pad_set_caps (GST_BASE_RTP_DEPAYLOAD_SRCPAD (depayload), srccaps);
+
+ GST_DEBUG ("set caps on source: %" GST_PTR_FORMAT " (ret=%d)", srccaps, ret);
+ gst_caps_unref (srccaps);
+
+ /* always fixed clock rate of 16000 */
+ depayload->clock_rate = 16000;
+
+ return ret;
+}
+
+static GstBuffer *
+gst_rtp_siren_depay_process (GstBaseRTPDepayload * depayload, GstBuffer * buf)
+{
+ GstBuffer *outbuf;
+
+ outbuf = gst_rtp_buffer_get_payload_buffer (buf);
+
+ return outbuf;
+}
+
+gboolean
+gst_rtp_siren_depay_plugin_init (GstPlugin * plugin)
+{
+ return gst_element_register (plugin, "rtpsirendepay",
+ GST_RANK_MARGINAL, GST_TYPE_RTP_SIREN_DEPAY);
+}
diff --git a/gst/rtp/gstrtpsirendepay.h b/gst/rtp/gstrtpsirendepay.h
new file mode 100644
index 000000000..78de4a7d6
--- /dev/null
+++ b/gst/rtp/gstrtpsirendepay.h
@@ -0,0 +1,57 @@
+/*
+ * Siren Depayloader Gst Element
+ *
+ * @author: Youness Alaoui <kakaroto@kakaroto.homelinux.net>
+ *
+ * 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_SIREN_DEPAY_H__
+#define __GST_RTP_SIREN_DEPAY_H__
+
+#include <gst/gst.h>
+#include <gst/rtp/gstbasertpdepayload.h>
+
+G_BEGIN_DECLS typedef struct _GstRTPSirenDepay GstRTPSirenDepay;
+typedef struct _GstRTPSirenDepayClass GstRTPSirenDepayClass;
+
+#define GST_TYPE_RTP_SIREN_DEPAY \
+ (gst_rtp_siren_depay_get_type())
+#define GST_RTP_SIREN_DEPAY(obj) \
+ (G_TYPE_CHECK_INSTANCE_CAST((obj),GST_TYPE_RTP_SIREN_DEPAY,GstRTPSirenDepay))
+#define GST_RTP_SIREN_DEPAY_CLASS(klass) \
+ (G_TYPE_CHECK_CLASS_CAST((klass),GST_TYPE_RTP_SIREN_DEPAY,GstRTPSirenDepayClass))
+#define GST_IS_RTP_SIREN_DEPAY(obj) \
+ (G_TYPE_CHECK_INSTANCE_TYPE((obj),GST_TYPE_RTP_SIREN_DEPAY))
+#define GST_IS_RTP_SIREN_DEPAY_CLASS(klass) \
+ (G_TYPE_CHECK_CLASS_TYPE((klass),GST_TYPE_RTP_SIREN_DEPAY))
+
+
+struct _GstRTPSirenDepay
+{
+ GstBaseRTPDepayload depayload;
+
+};
+
+struct _GstRTPSirenDepayClass
+{
+ GstBaseRTPDepayloadClass parent_class;
+};
+
+gboolean gst_rtp_siren_depay_plugin_init (GstPlugin * plugin);
+
+G_END_DECLS
+#endif /* __GST_RTP_SIREN_DEPAY_H__ */
diff --git a/gst/rtp/gstrtpsirenpay.c b/gst/rtp/gstrtpsirenpay.c
new file mode 100644
index 000000000..5dd34d2c6
--- /dev/null
+++ b/gst/rtp/gstrtpsirenpay.c
@@ -0,0 +1,164 @@
+/*
+ * Siren Payloader Gst Element
+ *
+ * @author: Youness Alaoui <kakaroto@kakaroto.homelinux.net>
+ *
+ * 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 "gstrtpsirenpay.h"
+#include <gst/rtp/gstrtpbuffer.h>
+
+/* elementfactory information */
+static GstElementDetails gst_rtpsirenpay_details = {
+ "RTP Payloader for Siren Audio",
+ "Codec/Payloader/Network",
+ "Packetize Siren audio streams into RTP packets",
+ "Youness Alaoui <kakaroto@kakaroto.homelinux.net>"
+};
+
+GST_DEBUG_CATEGORY_STATIC (rtpsirenpay_debug);
+#define GST_CAT_DEFAULT (rtpsirenpay_debug)
+
+static GstStaticPadTemplate gst_rtpsirenpay_sink_template =
+GST_STATIC_PAD_TEMPLATE ("sink",
+ GST_PAD_SINK,
+ GST_PAD_ALWAYS,
+ GST_STATIC_CAPS ("audio/x-siren, " "dct-length = (int) 320")
+ );
+
+static GstStaticPadTemplate gst_rtpsirenpay_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) 16000, "
+ "encoding-name = (string) \"SIREN\", " "dct-length = (int) 320")
+ );
+
+static gboolean gst_rtpsirenpay_setcaps (GstBaseRTPPayload * payload,
+ GstCaps * caps);
+
+GST_BOILERPLATE (GstRTPSirenPay, gst_rtpsirenpay, GstBaseRTPAudioPayload,
+ GST_TYPE_BASE_RTP_AUDIO_PAYLOAD);
+
+static void
+gst_rtpsirenpay_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_rtpsirenpay_sink_template));
+ gst_element_class_add_pad_template (element_class,
+ gst_static_pad_template_get (&gst_rtpsirenpay_src_template));
+ gst_element_class_set_details (element_class, &gst_rtpsirenpay_details);
+}
+
+static void
+gst_rtpsirenpay_class_init (GstRTPSirenPayClass * 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_ref (GST_TYPE_BASE_RTP_PAYLOAD);
+
+ gstbasertppayload_class->set_caps = gst_rtpsirenpay_setcaps;
+
+ GST_DEBUG_CATEGORY_INIT (rtpsirenpay_debug, "rtpsirenpay", 0,
+ "siren audio RTP payloader");
+}
+
+static void
+gst_rtpsirenpay_init (GstRTPSirenPay * rtpsirenpay, GstRTPSirenPayClass * klass)
+{
+ GstBaseRTPPayload *basertppayload;
+ GstBaseRTPAudioPayload *basertpaudiopayload;
+
+ basertppayload = GST_BASE_RTP_PAYLOAD (rtpsirenpay);
+ basertpaudiopayload = GST_BASE_RTP_AUDIO_PAYLOAD (rtpsirenpay);
+
+ /* we don't set the payload type, it should be set by the application using
+ * the pt property or the default 96 will be used */
+ basertppayload->clock_rate = 16000;
+
+ /* tell basertpaudiopayload that this is a frame based codec */
+ gst_base_rtp_audio_payload_set_frame_based (basertpaudiopayload);
+}
+
+static gboolean
+gst_rtpsirenpay_setcaps (GstBaseRTPPayload * basertppayload, GstCaps * caps)
+{
+ GstRTPSirenPay *rtpsirenpay;
+ GstBaseRTPAudioPayload *basertpaudiopayload;
+ gboolean ret;
+ gint dct_length;
+ GstStructure *structure;
+ const char *payload_name;
+
+ rtpsirenpay = GST_RTP_SIREN_PAY (basertppayload);
+ basertpaudiopayload = GST_BASE_RTP_AUDIO_PAYLOAD (basertppayload);
+
+ structure = gst_caps_get_structure (caps, 0);
+
+ gst_structure_get_int (structure, "dct-length", &dct_length);
+ if (dct_length != 320)
+ goto wrong_dct;
+
+ payload_name = gst_structure_get_name (structure);
+ if (g_strcasecmp ("audio/x-siren", payload_name))
+ goto wrong_caps;
+
+ gst_basertppayload_set_options (basertppayload, "audio", TRUE, "SIREN",
+ 16000);
+ /* set options for this frame based audio codec */
+ gst_base_rtp_audio_payload_set_frame_options (basertpaudiopayload, 20, 40);
+
+ ret = gst_basertppayload_set_outcaps (basertppayload, NULL);
+
+ return TRUE;
+
+ /* ERRORS */
+wrong_dct:
+ {
+ GST_ERROR_OBJECT (rtpsirenpay, "dct-length must be 320, received %d",
+ dct_length);
+ return FALSE;
+ }
+wrong_caps:
+ {
+ GST_ERROR_OBJECT (rtpsirenpay, "expected audio/x-siren, received %s",
+ payload_name);
+ return FALSE;
+ }
+}
+
+gboolean
+gst_rtp_siren_pay_plugin_init (GstPlugin * plugin)
+{
+ return gst_element_register (plugin, "rtpsirenpay",
+ GST_RANK_NONE, GST_TYPE_RTP_SIREN_PAY);
+}
diff --git a/gst/rtp/gstrtpsirenpay.h b/gst/rtp/gstrtpsirenpay.h
new file mode 100644
index 000000000..71eb83937
--- /dev/null
+++ b/gst/rtp/gstrtpsirenpay.h
@@ -0,0 +1,55 @@
+/*
+ * Siren Payloader Gst Element
+ *
+ * @author: Youness Alaoui <kakaroto@kakaroto.homelinux.net>
+ *
+ * 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_SIREN_PAY_H__
+#define __GST_RTP_SIREN_PAY_H__
+
+#include <gst/gst.h>
+#include <gst/rtp/gstbasertpaudiopayload.h>
+
+G_BEGIN_DECLS
+#define GST_TYPE_RTP_SIREN_PAY \
+ (gst_rtpsirenpay_get_type())
+#define GST_RTP_SIREN_PAY(obj) \
+ (G_TYPE_CHECK_INSTANCE_CAST((obj),GST_TYPE_RTP_SIREN_PAY,GstRTPSirenPay))
+#define GST_RTP_SIREN_PAY_CLASS(klass) \
+ (G_TYPE_CHECK_CLASS_CAST((klass),GST_TYPE_RTP_SIREN_PAY,GstRTPSirenPayClass))
+#define GST_IS_RTP_SIREN_PAY(obj) \
+ (G_TYPE_CHECK_INSTANCE_TYPE((obj),GST_TYPE_RTP_SIREN_PAY))
+#define GST_IS_RTP_SIREN_PAY_CLASS(klass) \
+ (G_TYPE_CHECK_CLASS_TYPE((klass),GST_TYPE_RTP_SIREN_PAY))
+typedef struct _GstRTPSirenPay GstRTPSirenPay;
+typedef struct _GstRTPSirenPayClass GstRTPSirenPayClass;
+
+struct _GstRTPSirenPay
+{
+ GstBaseRTPAudioPayload audiopayload;
+};
+
+struct _GstRTPSirenPayClass
+{
+ GstBaseRTPAudioPayloadClass parent_class;
+};
+
+gboolean gst_rtp_siren_pay_plugin_init (GstPlugin * plugin);
+
+G_END_DECLS
+#endif /* __GST_RTP_SIREN_PAY_H__ */