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gst/rtp/: Ported mulaw and alaw payloaders to use new base class

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Original commit message from CVS:
2006-04-12 Philippe Kalaf <philippe.kalaf@collabora.co.uk>

* gst/rtp/gstrtppcmapay.c:
* gst/rtp/gstrtppcmapay.h:
* gst/rtp/gstrtppcmupay.c:
* gst/rtp/gstrtppcmupay.h:
Ported mulaw and alaw payloaders to use new base class
* gst/rtp/Makefile.am:
* gst/rtp/gstrtp.c:
* gst/rtp/gstrtpilbcpay.c:
* gst/rtp/gstrtpilbcpay.h:
* gst/rtp/gstrtpilbcdepay.c:
* gst/rtp/gstrtpilbcdepay.h:
Added new iLBC payloader/depayloader. Payloader uses new audio payload base
class.
This commit is contained in:
Philippe Kalaf 2006-04-13 03:42:51 +00:00
parent 5f89255beb
commit 07f9b4f658
11 changed files with 558 additions and 260 deletions
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16
ChangeLog
View file

@ -1,3 +1,19 @@
2006-04-12 Philippe Kalaf <philippe.kalaf@collabora.co.uk>
* gst/rtp/gstrtppcmapay.c:
* gst/rtp/gstrtppcmapay.h:
* gst/rtp/gstrtppcmupay.c:
* gst/rtp/gstrtppcmupay.h:
Ported mulaw and alaw payloaders to use new base class
* gst/rtp/Makefile.am:
* gst/rtp/gstrtp.c:
* gst/rtp/gstrtpilbcpay.c:
* gst/rtp/gstrtpilbcpay.h:
* gst/rtp/gstrtpilbcdepay.c:
* gst/rtp/gstrtpilbcdepay.h:
Added new iLBC payloader/depayloader. Payloader uses new audio payload base
class.
2006-04-12 Wim Taymans <wim@fluendo.com>
* ext/gdk_pixbuf/gstgdkpixbuf.c: (gst_gdk_pixbuf_sink_setcaps),

4
gst/rtp/Makefile.am
View file

@ -16,6 +16,8 @@ libgstrtp_la_SOURCES = \
gstrtph263pdepay.c \
gstrtph263ppay.c \
gstrtph263pay.c \
gstrtpilbcpay.c \
gstrtpilbcdepay.c \
gstasteriskh263.c \
gstrtpmp4vdepay.c \
gstrtpmp4vpay.c \
@ -52,6 +54,8 @@ noinst_HEADERS = \
gstrtph263pdepay.h \
gstrtph263ppay.h \
gstrtph263pay.h \
gstrtpilbcpay.h \
gstrtpilbcdepay.h \
gstrtpmp4vdepay.h \
gstrtpmp4vpay.h \
gstrtpmp4gpay.h \

8
gst/rtp/gstrtp.c
View file

@ -35,6 +35,8 @@
#include "gstrtph263pdepay.h"
#include "gstrtph263ppay.h"
#include "gstrtph263pay.h"
#include "gstrtpilbcpay.h"
#include "gstrtpilbcdepay.h"
#include "gstasteriskh263.h"
#include "gstrtpmp4vpay.h"
#include "gstrtpmp4gpay.h"
@ -87,6 +89,12 @@ plugin_init (GstPlugin * plugin)
if (!gst_rtp_h263_pay_plugin_init (plugin))
return FALSE;
if (!gst_rtp_ilbc_pay_plugin_init (plugin))
return FALSE;
if (!gst_rtp_ilbc_depay_plugin_init (plugin))
return FALSE;
if (!gst_asteriskh263_plugin_init (plugin))
return FALSE;

222
gst/rtp/gstrtpilbcdepay.c Normal file
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@ -0,0 +1,222 @@
/* GStreamer
* Copyright (C) <2006> Philippe Khalaf <burger@speedy.org>
*
* 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 <gst/rtp/gstrtpbuffer.h>
#include "gstrtpilbcdepay.h"
/* elementfactory information */
static GstElementDetails gst_rtp_ilbc_depay_details =
GST_ELEMENT_DETAILS ("RTP iLBC packet depayloader",
"Codec/Depayr/Network",
"Extracts iLBC audio from RTP packets",
"Philippe Kalaf <philippe.kalaf@collabora.co.uk>");
/* RtpiLBCDepay signals and args */
enum
{
/* FILL ME */
LAST_SIGNAL
};
enum
{
ARG_0,
ARG_MODE
};
static GstStaticPadTemplate gst_rtp_ilbc_depay_sink_template =
GST_STATIC_PAD_TEMPLATE ("sink",
GST_PAD_SINK,
GST_PAD_ALWAYS,
GST_STATIC_CAPS ("application/x-rtp, "
"media = (string) \"audio\", "
"clock-rate = (int) 8000, "
"encoding-name = (string) \"iLBC\", " "mode = (int) { 20, 30 }")
);
static GstStaticPadTemplate gst_rtp_ilbc_depay_src_template =
GST_STATIC_PAD_TEMPLATE ("src",
GST_PAD_SRC,
GST_PAD_ALWAYS,
GST_STATIC_CAPS ("audio/x-iLBC, " "mode = (int) { 20, 30 }")
);
static void gst_ilbc_depay_set_property (GObject * object,
guint prop_id, const GValue * value, GParamSpec * pspec);
static void gst_ilbc_depay_get_property (GObject * object,
guint prop_id, GValue * value, GParamSpec * pspec);
static GstBuffer *gst_rtp_ilbc_depay_process (GstBaseRTPDepayload * depayload,
GstBuffer * buf);
static gboolean gst_rtp_ilbc_depay_setcaps (GstBaseRTPDepayload * depayload,
GstCaps * caps);
GST_BOILERPLATE (GstRTPiLBCDepay, gst_rtp_ilbc_depay, GstBaseRTPDepayload,
GST_TYPE_BASE_RTP_DEPAYLOAD);
#define GST_TYPE_ILBC_MODE (gst_ilbc_mode_get_type())
static GType
gst_ilbc_mode_get_type (void)
{
static GType ilbc_mode_type = 0;
static GEnumValue ilbc_modes[] = {
{GST_ILBC_MODE_20, "20ms frames", "20ms"},
{GST_ILBC_MODE_30, "30ms frames", "30ms"},
{0, NULL, NULL},
};
if (!ilbc_mode_type) {
ilbc_mode_type = g_enum_register_static ("iLBCMode", ilbc_modes);
}
return ilbc_mode_type;
}
static void
gst_rtp_ilbc_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_ilbc_depay_src_template));
gst_element_class_add_pad_template (element_class,
gst_static_pad_template_get (&gst_rtp_ilbc_depay_sink_template));
gst_element_class_set_details (element_class, &gst_rtp_ilbc_depay_details);
}
static void
gst_rtp_ilbc_depay_class_init (GstRTPiLBCDepayClass * klass)
{
GObjectClass *gobject_class;
GstElementClass *gstelement_class;
GstBaseRTPDepayloadClass *gstbasertpdepayload_class;
gobject_class = (GObjectClass *) klass;
gstelement_class = (GstElementClass *) klass;
gstbasertpdepayload_class = (GstBaseRTPDepayloadClass *) klass;
gobject_class->set_property = gst_ilbc_depay_set_property;
gobject_class->get_property = gst_ilbc_depay_get_property;
g_object_class_install_property (gobject_class, ARG_MODE, g_param_spec_enum ("mode", "Mode", "iLBC frame mode", GST_TYPE_ILBC_MODE, /* enum type */
GST_ILBC_MODE_30, /* default value */
G_PARAM_READWRITE));
parent_class = g_type_class_peek_parent (klass);
gstbasertpdepayload_class->process = gst_rtp_ilbc_depay_process;
gstbasertpdepayload_class->set_caps = gst_rtp_ilbc_depay_setcaps;
}
static void
gst_rtp_ilbc_depay_init (GstRTPiLBCDepay * rtpilbcdepay,
GstRTPiLBCDepayClass * klass)
{
GstBaseRTPDepayload *depayload;
depayload = GST_BASE_RTP_DEPAYLOAD (rtpilbcdepay);
depayload->clock_rate = 8000;
/* Set default mode to 30 */
rtpilbcdepay->mode = GST_ILBC_MODE_30;
}
static gboolean
gst_rtp_ilbc_depay_setcaps (GstBaseRTPDepayload * depayload, GstCaps * caps)
{
GstRTPiLBCDepay *rtpilbcdepay = GST_RTP_ILBC_DEPAY (depayload);
GstCaps *srccaps;
GstStructure *structure;
gboolean ret;
srccaps = gst_caps_copy (gst_static_pad_template_get_caps
(&gst_rtp_ilbc_depay_src_template));
structure = gst_caps_get_structure (srccaps, 0);
gst_structure_set (structure, "mode", G_TYPE_INT,
rtpilbcdepay->mode == GST_ILBC_MODE_30 ? 30 : 20, NULL);
ret = gst_pad_set_caps (GST_BASE_RTP_DEPAYLOAD_SRCPAD (depayload), srccaps);
GST_DEBUG ("caps set on source are %s", gst_caps_to_string (srccaps));
gst_caps_unref (srccaps);
return ret;
}
static GstBuffer *
gst_rtp_ilbc_depay_process (GstBaseRTPDepayload * depayload, GstBuffer * buf)
{
GstBuffer *outbuf = NULL;
gint payload_len;
gint header_len;
GST_DEBUG ("process : got %d bytes, mark %d ts %u seqn %d",
GST_BUFFER_SIZE (buf),
gst_rtp_buffer_get_marker (buf),
gst_rtp_buffer_get_timestamp (buf), gst_rtp_buffer_get_seq (buf));
payload_len = gst_rtp_buffer_get_payload_len (buf);
header_len = gst_rtp_buffer_calc_header_len (0);
outbuf = gst_buffer_create_sub (buf, header_len, payload_len);
return outbuf;
}
static void
gst_ilbc_depay_set_property (GObject * object,
guint prop_id, const GValue * value, GParamSpec * pspec)
{
GstRTPiLBCDepay *rtpilbcdepay = GST_RTP_ILBC_DEPAY (object);
switch (prop_id) {
case ARG_MODE:
rtpilbcdepay->mode = g_value_get_enum (value);
break;
default:
G_OBJECT_WARN_INVALID_PROPERTY_ID (object, prop_id, pspec);
break;
}
}
static void
gst_ilbc_depay_get_property (GObject * object,
guint prop_id, GValue * value, GParamSpec * pspec)
{
GstRTPiLBCDepay *rtpilbcdepay = GST_RTP_ILBC_DEPAY (object);
switch (prop_id) {
case ARG_MODE:
g_value_set_enum (value, rtpilbcdepay->mode);
break;
default:
G_OBJECT_WARN_INVALID_PROPERTY_ID (object, prop_id, pspec);
break;
}
}
gboolean
gst_rtp_ilbc_depay_plugin_init (GstPlugin * plugin)
{
return gst_element_register (plugin, "rtpilbcdepay",
GST_RANK_NONE, GST_TYPE_RTP_ILBC_DEPAY);
}

63
gst/rtp/gstrtpilbcdepay.h Normal file
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@ -0,0 +1,63 @@
/* GStreamer
* Copyright (C) <2006> Philippe Khalaf <burger@speedy.org>
*
* 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_ILBC_DEPAY_H__
#define __GST_RTP_ILBC_DEPAY_H__
#include <gst/gst.h>
#include <gst/rtp/gstbasertpdepayload.h>
G_BEGIN_DECLS
typedef struct _GstRTPiLBCDepay GstRTPiLBCDepay;
typedef struct _GstRTPiLBCDepayClass GstRTPiLBCDepayClass;
#define GST_TYPE_RTP_ILBC_DEPAY \
(gst_rtp_ilbc_depay_get_type())
#define GST_RTP_ILBC_DEPAY(obj) \
(G_TYPE_CHECK_INSTANCE_CAST((obj),GST_TYPE_RTP_ILBC_DEPAY,GstRTPiLBCDepay))
#define GST_RTP_ILBC_DEPAY_CLASS(klass) \
(G_TYPE_CHECK_CLASS_CAST((klass),GST_TYPE_RTP_ILBC_DEPAY,GstRTPiLBCDepay))
#define GST_IS_RTP_ILBC_DEPAY(obj) \
(G_TYPE_CHECK_INSTANCE_TYPE((obj),GST_TYPE_RTP_ILBC_DEPAY))
#define GST_IS_RTP_ILBC_DEPAY_CLASS(obj) \
(G_TYPE_CHECK_CLASS_TYPE((klass),GST_TYPE_RTP_ILBC_DEPAY))
typedef enum {
GST_ILBC_MODE_20,
GST_ILBC_MODE_30
} GstiLBCMode;
struct _GstRTPiLBCDepay
{
GstBaseRTPDepayload depayload;
GstiLBCMode mode;
};
struct _GstRTPiLBCDepayClass
{
GstBaseRTPDepayloadClass parent_class;
};
gboolean gst_rtp_ilbc_depay_plugin_init (GstPlugin * plugin);
G_END_DECLS
#endif /* __GST_RTP_ILBC_DEPAY_H__ */

161
gst/rtp/gstrtpilbcpay.c Normal file
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@ -0,0 +1,161 @@
/* GStreamer
* Copyright (C) <2006> Philippe Khalaf <burger@speedy.org>
*
* 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 "gstrtpilbcpay.h"
#include <gst/rtp/gstrtpbuffer.h>
/* elementfactory information */
static GstElementDetails gst_rtpilbcpay_details = {
"RTP Payloader for iLBC Audio",
"Codec/Payloader/Network",
"Packetize iLBC audio streams into RTP packets",
"Philippe Kalaf <philippe.kalaf@collabora.co.uk>"
};
GST_DEBUG_CATEGORY (rtpilbcpay_debug);
#define GST_CAT_DEFAULT (rtpilbcpay_debug)
static GstStaticPadTemplate gst_rtpilbcpay_sink_template =
GST_STATIC_PAD_TEMPLATE ("sink",
GST_PAD_SINK,
GST_PAD_ALWAYS,
GST_STATIC_CAPS ("audio/x-iLBC, " "mode = (int) {20, 30}")
);
static GstStaticPadTemplate gst_rtpilbcpay_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) \"iLBC\", " "mode = (int) {20, 30}")
);
static gboolean gst_rtpilbcpay_setcaps (GstBaseRTPPayload * payload,
GstCaps * caps);
GST_BOILERPLATE (GstRTPILBCPay, gst_rtpilbcpay, GstBaseRTPAudioPayload,
GST_TYPE_BASE_RTP_AUDIO_PAYLOAD);
static void
gst_rtpilbcpay_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_rtpilbcpay_sink_template));
gst_element_class_add_pad_template (element_class,
gst_static_pad_template_get (&gst_rtpilbcpay_src_template));
gst_element_class_set_details (element_class, &gst_rtpilbcpay_details);
}
static void
gst_rtpilbcpay_class_init (GstRTPILBCPayClass * 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_rtpilbcpay_setcaps;
GST_DEBUG_CATEGORY_INIT (rtpilbcpay_debug, "rtpilbcpay", 0,
"iLBC audio RTP payloader");
}
static void
gst_rtpilbcpay_init (GstRTPILBCPay * rtpilbcpay, GstRTPILBCPayClass * klass)
{
GstBaseRTPPayload *basertppayload;
GstBaseRTPAudioPayload *basertpaudiopayload;
basertppayload = GST_BASE_RTP_PAYLOAD (rtpilbcpay);
basertpaudiopayload = GST_BASE_RTP_AUDIO_PAYLOAD (rtpilbcpay);
/* 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 = 8000;
rtpilbcpay->mode = -1;
/* tell basertpaudiopayload that this is a frame based codec */
gst_basertpaudiopayload_set_frame_based (basertpaudiopayload);
}
static gboolean
gst_rtpilbcpay_setcaps (GstBaseRTPPayload * basertppayload, GstCaps * caps)
{
GstRTPILBCPay *rtpilbcpay;
GstBaseRTPAudioPayload *basertpaudiopayload;
gboolean ret;
gint mode;
GstStructure *structure;
const char *payload_name;
rtpilbcpay = GST_RTP_ILBC_PAY (basertppayload);
basertpaudiopayload = GST_BASE_RTP_AUDIO_PAYLOAD (basertppayload);
structure = gst_caps_get_structure (caps, 0);
gst_structure_get_int (structure, "mode", &mode);
if (mode != 20 && mode != 30) {
return FALSE;
}
payload_name = gst_structure_get_name (structure);
if (g_strcasecmp ("audio/x-iLBC", payload_name) == 0) {
gst_basertppayload_set_options (basertppayload, "audio", TRUE, "iLBC",
8000);
/* set options for this frame based audio codec */
gst_basertpaudiopayload_set_frame_options (basertpaudiopayload,
mode, mode == 30 ? 50 : 38);
} else {
return FALSE;
}
ret =
gst_basertppayload_set_outcaps (basertppayload, "mode", G_TYPE_INT, mode,
NULL);
if (mode != rtpilbcpay->mode && rtpilbcpay->mode != -1) {
GST_ERROR_OBJECT (rtpilbcpay, "Mode has changed from %d to %d! \
Mode cannot change while streaming", rtpilbcpay->mode, mode);
return FALSE;
}
rtpilbcpay->mode = mode;
return ret;
}
gboolean
gst_rtp_ilbc_pay_plugin_init (GstPlugin * plugin)
{
return gst_element_register (plugin, "rtpilbcpay",
GST_RANK_NONE, GST_TYPE_RTP_ILBC_PAY);
}

58
gst/rtp/gstrtpilbcpay.h Normal file
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@ -0,0 +1,58 @@
/* GStreamer
* Copyright (C) <2006> Philippe Khalaf <burger@speedy.org>
*
* 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_ILBC_PAY_H__
#define __GST_RTP_ILBC_PAY_H__
#include <gst/gst.h>
#include <gst/rtp/gstbasertpaudiopayload.h>
G_BEGIN_DECLS
#define GST_TYPE_RTP_ILBC_PAY \
(gst_rtpilbcpay_get_type())
#define GST_RTP_ILBC_PAY(obj) \
(G_TYPE_CHECK_INSTANCE_CAST((obj),GST_TYPE_RTP_ILBC_PAY,GstRTPILBCPay))
#define GST_RTP_ILBC_PAY_CLASS(klass) \
(G_TYPE_CHECK_CLASS_CAST((klass),GST_TYPE_RTP_ILBC_PAY,GstRTPILBCPay))
#define GST_IS_RTP_ILBC_PAY(obj) \
(G_TYPE_CHECK_INSTANCE_TYPE((obj),GST_TYPE_RTP_ILBC_PAY))
#define GST_IS_RTP_ILBC_PAY_CLASS(obj) \
(G_TYPE_CHECK_CLASS_TYPE((klass),GST_TYPE_RTP_ILBC_PAY))
typedef struct _GstRTPILBCPay GstRTPILBCPay;
typedef struct _GstRTPILBCPayClass GstRTPILBCPayClass;
struct _GstRTPILBCPay
{
GstBaseRTPAudioPayload audiopayload;
gint mode;
};
struct _GstRTPILBCPayClass
{
GstBaseRTPAudioPayloadClass parent_class;
};
gboolean gst_rtp_ilbc_pay_plugin_init (GstPlugin * plugin);
G_END_DECLS
#endif /* __GST_RTP_ILBC_PAY_H__ */

132
gst/rtp/gstrtppcmapay.c
View file

@ -50,17 +50,9 @@ GST_STATIC_PAD_TEMPLATE ("src",
static gboolean gst_rtp_pcma_pay_setcaps (GstBaseRTPPayload * payload,
GstCaps * caps);
static GstFlowReturn gst_rtp_pcma_pay_handle_buffer (GstBaseRTPPayload *
payload, GstBuffer * buffer);
static void gst_rtp_pcma_pay_finalize (GObject * object);
GST_BOILERPLATE (GstRtpPmcaPay, gst_rtp_pcma_pay, GstBaseRTPPayload,
GST_TYPE_BASE_RTP_PAYLOAD);
/* The lower limit for number of octet to put in one packet
* (clock-rate=8000, octet-per-sample=1). The default 80 is equal
* to to 10msec (see RFC3551) */
#define GST_RTP_PCMA_MIN_PTIME_OCTETS 80
GST_BOILERPLATE (GstRtpPmcaPay, gst_rtp_pcma_pay, GstBaseRTPAudioPayload,
GST_TYPE_BASE_RTP_AUDIO_PAYLOAD);
static void
gst_rtp_pcma_pay_base_init (gpointer klass)
@ -86,30 +78,24 @@ gst_rtp_pcma_pay_class_init (GstRtpPmcaPayClass * klass)
gstbasertppayload_class = (GstBaseRTPPayloadClass *) klass;
parent_class = g_type_class_peek_parent (klass);
gobject_class->finalize = gst_rtp_pcma_pay_finalize;
gstbasertppayload_class->set_caps = gst_rtp_pcma_pay_setcaps;
gstbasertppayload_class->handle_buffer = gst_rtp_pcma_pay_handle_buffer;
}
static void
gst_rtp_pcma_pay_init (GstRtpPmcaPay * rtppcmapay, GstRtpPmcaPayClass * klass)
{
rtppcmapay->adapter = gst_adapter_new ();
GstBaseRTPAudioPayload *basertpaudiopayload;
basertpaudiopayload = GST_BASE_RTP_AUDIO_PAYLOAD (rtppcmapay);
GST_BASE_RTP_PAYLOAD (rtppcmapay)->clock_rate = 8000;
}
static void
gst_rtp_pcma_pay_finalize (GObject * object)
{
GstRtpPmcaPay *rtppcmapay;
/* tell basertpaudiopayload that this is a sample based codec */
gst_basertpaudiopayload_set_sample_based (basertpaudiopayload);
rtppcmapay = GST_RTP_PCMA_PAY (object);
g_object_unref (rtppcmapay->adapter);
rtppcmapay->adapter = NULL;
G_OBJECT_CLASS (parent_class)->finalize (object);
/* octet-per-sample is 1 for PCM */
gst_basertpaudiopayload_set_sample_options (basertpaudiopayload, 1);
}
static gboolean
@ -123,104 +109,6 @@ gst_rtp_pcma_pay_setcaps (GstBaseRTPPayload * payload, GstCaps * caps)
return TRUE;
}
static GstFlowReturn
gst_rtp_pcma_pay_flush (GstRtpPmcaPay * rtppcmapay)
{
guint avail;
GstBuffer *outbuf;
GstFlowReturn ret;
guint maxptime_octets = G_MAXUINT;
guint minptime_octets = GST_RTP_PCMA_MIN_PTIME_OCTETS;
if (GST_BASE_RTP_PAYLOAD (rtppcmapay)->max_ptime > 0) {
/* calculate octet count with:
maxptime-nsec * samples-per-sec / nsecs-per-sec * octets-per-sample */
maxptime_octets =
GST_BASE_RTP_PAYLOAD (rtppcmapay)->max_ptime *
GST_BASE_RTP_PAYLOAD (rtppcmapay)->clock_rate / GST_SECOND;
}
/* the data available in the adapter is either smaller
* than the MTU or bigger. In the case it is smaller, the complete
* adapter contents can be put in one packet. */
avail = gst_adapter_available (rtppcmapay->adapter);
ret = GST_FLOW_OK;
while (avail >= minptime_octets) {
guint8 *payload;
guint8 *data;
guint payload_len;
guint packet_len;
/* fill one MTU or all available bytes */
payload_len =
MIN (MIN (GST_BASE_RTP_PAYLOAD_MTU (rtppcmapay), maxptime_octets),
avail);
/* this will be the total lenght of the packet */
packet_len = gst_rtp_buffer_calc_packet_len (payload_len, 0, 0);
/* create buffer to hold the payload */
outbuf = gst_rtp_buffer_new_allocate (payload_len, 0, 0);
/* copy payload */
gst_rtp_buffer_set_payload_type (outbuf,
GST_BASE_RTP_PAYLOAD_PT (rtppcmapay));
payload = gst_rtp_buffer_get_payload (outbuf);
data = (guint8 *) gst_adapter_peek (rtppcmapay->adapter, payload_len);
memcpy (payload, data, payload_len);
gst_adapter_flush (rtppcmapay->adapter, payload_len);
avail -= payload_len;
GST_BUFFER_TIMESTAMP (outbuf) = rtppcmapay->first_ts;
ret = gst_basertppayload_push (GST_BASE_RTP_PAYLOAD (rtppcmapay), outbuf);
}
return ret;
}
static GstFlowReturn
gst_rtp_pcma_pay_handle_buffer (GstBaseRTPPayload * basepayload,
GstBuffer * buffer)
{
GstRtpPmcaPay *rtppcmapay;
guint size, packet_len, avail;
GstFlowReturn ret;
GstClockTime duration;
rtppcmapay = GST_RTP_PCMA_PAY (basepayload);
size = GST_BUFFER_SIZE (buffer);
duration = GST_BUFFER_TIMESTAMP (buffer);
avail = gst_adapter_available (rtppcmapay->adapter);
if (avail == 0) {
rtppcmapay->first_ts = GST_BUFFER_TIMESTAMP (buffer);
rtppcmapay->duration = 0;
}
/* get packet length of data and see if we exceeded MTU. */
packet_len = gst_rtp_buffer_calc_packet_len (avail + size, 0, 0);
/* if this buffer is going to overflow the packet, flush what we
* have. */
if (gst_basertppayload_is_filled (basepayload,
packet_len, rtppcmapay->duration + duration)) {
ret = gst_rtp_pcma_pay_flush (rtppcmapay);
rtppcmapay->first_ts = GST_BUFFER_TIMESTAMP (buffer);
rtppcmapay->duration = 0;
} else {
ret = GST_FLOW_OK;
}
gst_adapter_push (rtppcmapay->adapter, buffer);
rtppcmapay->duration += duration;
return ret;
}
gboolean
gst_rtp_pcma_pay_plugin_init (GstPlugin * plugin)
{

11
gst/rtp/gstrtppcmapay.h
View file

@ -17,8 +17,7 @@
#define __GST_RTP_PCMA_PAY_H__
#include <gst/gst.h>
#include <gst/rtp/gstbasertppayload.h>
#include <gst/base/gstadapter.h>
#include <gst/rtp/gstbasertpaudiopayload.h>
G_BEGIN_DECLS
@ -38,16 +37,12 @@ typedef struct _GstRtpPmcaPayClass GstRtpPmcaPayClass;
struct _GstRtpPmcaPay
{
GstBaseRTPPayload payload;
GstAdapter *adapter;
GstClockTime first_ts;
GstClockTime duration;
GstBaseRTPAudioPayload audiopayload;
};
struct _GstRtpPmcaPayClass
{
GstBaseRTPPayloadClass parent_class;
GstBaseRTPAudioPayloadClass parent_class;
};
gboolean gst_rtp_pcma_pay_plugin_init (GstPlugin * plugin);

132
gst/rtp/gstrtppcmupay.c
View file

@ -50,17 +50,9 @@ GST_STATIC_PAD_TEMPLATE ("src",
static gboolean gst_rtp_pcmu_pay_setcaps (GstBaseRTPPayload * payload,
GstCaps * caps);
static GstFlowReturn gst_rtp_pcmu_pay_handle_buffer (GstBaseRTPPayload *
payload, GstBuffer * buffer);
static void gst_rtp_pcmu_pay_finalize (GObject * object);
GST_BOILERPLATE (GstRtpPcmuPay, gst_rtp_pcmu_pay, GstBaseRTPPayload,
GST_TYPE_BASE_RTP_PAYLOAD);
/* The lower limit for number of octet to put in one packet
* (clock-rate=8000, octet-per-sample=1). The default 80 is equal
* to to 10msec (see RFC3551) */
#define GST_RTP_PCMU_MIN_PTIME_OCTETS 80
GST_BOILERPLATE (GstRtpPcmuPay, gst_rtp_pcmu_pay, GstBaseRTPAudioPayload,
GST_TYPE_BASE_RTP_AUDIO_PAYLOAD);
static void
gst_rtp_pcmu_pay_base_init (gpointer klass)
@ -86,30 +78,24 @@ gst_rtp_pcmu_pay_class_init (GstRtpPcmuPayClass * klass)
gstbasertppayload_class = (GstBaseRTPPayloadClass *) klass;
parent_class = g_type_class_peek_parent (klass);
gobject_class->finalize = gst_rtp_pcmu_pay_finalize;
gstbasertppayload_class->set_caps = gst_rtp_pcmu_pay_setcaps;
gstbasertppayload_class->handle_buffer = gst_rtp_pcmu_pay_handle_buffer;
}
static void
gst_rtp_pcmu_pay_init (GstRtpPcmuPay * rtppcmupay, GstRtpPcmuPayClass * klass)
{
rtppcmupay->adapter = gst_adapter_new ();
GstBaseRTPAudioPayload *basertpaudiopayload;
basertpaudiopayload = GST_BASE_RTP_AUDIO_PAYLOAD (rtppcmupay);
GST_BASE_RTP_PAYLOAD (rtppcmupay)->clock_rate = 8000;
}
static void
gst_rtp_pcmu_pay_finalize (GObject * object)
{
GstRtpPcmuPay *rtppcmupay;
/* tell basertpaudiopayload that this is a sample based codec */
gst_basertpaudiopayload_set_sample_based (basertpaudiopayload);
rtppcmupay = GST_RTP_PCMU_PAY (object);
g_object_unref (rtppcmupay->adapter);
rtppcmupay->adapter = NULL;
G_OBJECT_CLASS (parent_class)->finalize (object);
/* octet-per-sample is 1 for PCM */
gst_basertpaudiopayload_set_sample_options (basertpaudiopayload, 1);
}
static gboolean
@ -123,104 +109,6 @@ gst_rtp_pcmu_pay_setcaps (GstBaseRTPPayload * payload, GstCaps * caps)
return TRUE;
}
static GstFlowReturn
gst_rtp_pcmu_pay_flush (GstRtpPcmuPay * rtppcmupay)
{
guint avail;
GstBuffer *outbuf;
GstFlowReturn ret;
guint maxptime_octets = G_MAXUINT;
guint minptime_octets = GST_RTP_PCMU_MIN_PTIME_OCTETS;
if (GST_BASE_RTP_PAYLOAD (rtppcmupay)->max_ptime > 0) {
/* calculate octet count with:
maxptime-nsec * samples-per-sec / nsecs-per-sec * octets-per-sample */
maxptime_octets =
GST_BASE_RTP_PAYLOAD (rtppcmupay)->max_ptime *
GST_BASE_RTP_PAYLOAD (rtppcmupay)->clock_rate / GST_SECOND;
}
/* the data available in the adapter is either smaller
* than the MTU or bigger. In the case it is smaller, the complete
* adapter contents can be put in one packet. */
avail = gst_adapter_available (rtppcmupay->adapter);
ret = GST_FLOW_OK;
while (avail >= minptime_octets) {
guint8 *payload;
guint8 *data;
guint payload_len;
guint packet_len;
/* fill one MTU or all available bytes */
payload_len =
MIN (MIN (GST_BASE_RTP_PAYLOAD_MTU (rtppcmupay), maxptime_octets),
avail);
/* this will be the total lenght of the packet */
packet_len = gst_rtp_buffer_calc_packet_len (payload_len, 0, 0);
/* create buffer to hold the payload */
outbuf = gst_rtp_buffer_new_allocate (payload_len, 0, 0);
/* copy payload */
gst_rtp_buffer_set_payload_type (outbuf,
GST_BASE_RTP_PAYLOAD_PT (rtppcmupay));
payload = gst_rtp_buffer_get_payload (outbuf);
data = (guint8 *) gst_adapter_peek (rtppcmupay->adapter, payload_len);
memcpy (payload, data, payload_len);
gst_adapter_flush (rtppcmupay->adapter, payload_len);
avail -= payload_len;
GST_BUFFER_TIMESTAMP (outbuf) = rtppcmupay->first_ts;
ret = gst_basertppayload_push (GST_BASE_RTP_PAYLOAD (rtppcmupay), outbuf);
}
return ret;
}
static GstFlowReturn
gst_rtp_pcmu_pay_handle_buffer (GstBaseRTPPayload * basepayload,
GstBuffer * buffer)
{
GstRtpPcmuPay *rtppcmupay;
guint size, packet_len, avail;
GstFlowReturn ret;
GstClockTime duration;
rtppcmupay = GST_RTP_PCMU_PAY (basepayload);
size = GST_BUFFER_SIZE (buffer);
duration = GST_BUFFER_TIMESTAMP (buffer);
avail = gst_adapter_available (rtppcmupay->adapter);
if (avail == 0) {
rtppcmupay->first_ts = GST_BUFFER_TIMESTAMP (buffer);
rtppcmupay->duration = 0;
}
/* get packet length of data and see if we exceeded MTU. */
packet_len = gst_rtp_buffer_calc_packet_len (avail + size, 0, 0);
/* if this buffer is going to overflow the packet, flush what we
* have. */
if (gst_basertppayload_is_filled (basepayload,
packet_len, rtppcmupay->duration + duration)) {
ret = gst_rtp_pcmu_pay_flush (rtppcmupay);
rtppcmupay->first_ts = GST_BUFFER_TIMESTAMP (buffer);
rtppcmupay->duration = 0;
} else {
ret = GST_FLOW_OK;
}
gst_adapter_push (rtppcmupay->adapter, buffer);
rtppcmupay->duration += duration;
return ret;
}
gboolean
gst_rtp_pcmu_pay_plugin_init (GstPlugin * plugin)
{

11
gst/rtp/gstrtppcmupay.h
View file

@ -17,8 +17,7 @@
#define __GST_RTP_PCMU_PAY_H__
#include <gst/gst.h>
#include <gst/rtp/gstbasertppayload.h>
#include <gst/base/gstadapter.h>
#include <gst/rtp/gstbasertpaudiopayload.h>
G_BEGIN_DECLS
@ -38,16 +37,12 @@ typedef struct _GstRtpPcmuPayClass GstRtpPcmuPayClass;
struct _GstRtpPcmuPay
{
GstBaseRTPPayload payload;
GstAdapter *adapter;
GstClockTime first_ts;
GstClockTime duration;
GstBaseRTPAudioPayload audiopayload;
};
struct _GstRtpPcmuPayClass
{
GstBaseRTPPayloadClass parent_class;
GstBaseRTPAudioPayloadClass parent_class;
};
gboolean gst_rtp_pcmu_pay_plugin_init (GstPlugin * plugin);