Move siren rtp pay/depay to good

This commit is contained in:
Olivier Crête 2009-02-10 17:23:38 +00:00 committed by Edward Hervey
parent 158e8692c5
commit ce558ab988
6 changed files with 1 additions and 426 deletions

View file

@ -3,8 +3,6 @@ plugin_LTLIBRARIES = libgstsiren.la
libgstsiren_la_SOURCES = gstsiren.c \
gstsirendec.c gstsirendec.h \
gstsirenenc.c gstsirenenc.h \
gstrtpsirenpay.c gstrtpsirenpay.h \
gstrtpsirendepay.c gstrtpsirendepay.h \
common.c dct4.c decoder.c encoder.c huffman.c \
huffman.h rmlt.h common.h dct4.h decoder.h \
encoder.h huffman_consts.h rmlt.c siren7.h

View file

@ -1,134 +0,0 @@
/*
* 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);
}

View file

@ -1,60 +0,0 @@
/*
* 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__ */

View file

@ -1,163 +0,0 @@
/*
* 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);
}

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@ -1,58 +0,0 @@
/*
* 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__ */

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@ -26,8 +26,6 @@
#include "gstsirendec.h"
#include "gstsirenenc.h"
#include "gstrtpsirenpay.h"
#include "gstrtpsirendepay.h"
static gboolean
@ -39,12 +37,6 @@ plugin_init (GstPlugin * plugin)
if (!gst_siren_enc_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;
return TRUE;
}
@ -52,4 +44,4 @@ GST_PLUGIN_DEFINE (GST_VERSION_MAJOR,
GST_VERSION_MINOR,
"gstsiren",
"Siren encoder/decoder/payloader/depayloader plugins",
plugin_init, "0.1" , "LGPL", "Siren", "");
plugin_init, VERSION, GST_LICENSE, GST_PACKAGE_NAME, GST_PACKAGE_ORIGIN)