Commit 78e4a260 authored by Wim Taymans's avatar Wim Taymans

rtp: add G722 pay and depayloader

parent 2c2c90a7
......@@ -23,6 +23,8 @@ libgstrtp_la_SOURCES = \
gstrtppcmudepay.c \
gstrtppcmupay.c \
gstrtppcmapay.c \
gstrtpg722depay.c \
gstrtpg722pay.c \
gstrtpg723depay.c \
gstrtpg723pay.c \
gstrtpg726pay.c \
......@@ -106,6 +108,8 @@ noinst_HEADERS = \
gstrtppcmudepay.h \
gstrtppcmupay.h \
gstrtppcmapay.h \
gstrtpg722depay.h \
gstrtpg722pay.h \
gstrtpg723depay.h \
gstrtpg723pay.h \
gstrtpg726depay.h \
......
......@@ -35,6 +35,8 @@
#include "gstrtppcmapay.h"
#include "gstrtppcmadepay.h"
#include "gstrtppcmudepay.h"
#include "gstrtpg722depay.h"
#include "gstrtpg722pay.h"
#include "gstrtpg723depay.h"
#include "gstrtpg723pay.h"
#include "gstrtpg726depay.h"
......@@ -119,6 +121,12 @@ plugin_init (GstPlugin * plugin)
if (!gst_rtp_ilbc_depay_plugin_init (plugin))
return FALSE;
if (!gst_rtp_g722_depay_plugin_init (plugin))
return FALSE;
if (!gst_rtp_g722_pay_plugin_init (plugin))
return FALSE;
if (!gst_rtp_g723_depay_plugin_init (plugin))
return FALSE;
......
/* GStreamer
* Copyright (C) <2007> 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 <string.h>
#include <stdlib.h>
#include <gst/audio/audio.h>
#include <gst/audio/multichannel.h>
#include "gstrtpg722depay.h"
#include "gstrtpchannels.h"
GST_DEBUG_CATEGORY_STATIC (rtpg722depay_debug);
#define GST_CAT_DEFAULT (rtpg722depay_debug)
static GstStaticPadTemplate gst_rtp_g722_depay_src_template =
GST_STATIC_PAD_TEMPLATE ("src",
GST_PAD_SRC,
GST_PAD_ALWAYS,
GST_STATIC_CAPS ("audio/G722, "
"rate = (int) [ 1, MAX ], " "channels = (int) [ 1, MAX ]")
);
static GstStaticPadTemplate gst_rtp_g722_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) 8000, "
/* "channels = (int) [1, MAX]" */
/* "channel-order = (string) ANY" */
"encoding-name = (string) \"G722\";"
"application/x-rtp, "
"media = (string) \"audio\", "
"payload = (int) " GST_RTP_PAYLOAD_G722_STRING ", "
"clock-rate = (int) [ 1, MAX ]"
/* "channels = (int) [1, MAX]" */
/* "emphasis = (string) ANY" */
/* "channel-order = (string) ANY" */
)
);
GST_BOILERPLATE (GstRtpG722Depay, gst_rtp_g722_depay, GstBaseRTPDepayload,
GST_TYPE_BASE_RTP_DEPAYLOAD);
static gboolean gst_rtp_g722_depay_setcaps (GstBaseRTPDepayload * depayload,
GstCaps * caps);
static GstBuffer *gst_rtp_g722_depay_process (GstBaseRTPDepayload * depayload,
GstBuffer * buf);
static void
gst_rtp_g722_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_g722_depay_src_template));
gst_element_class_add_pad_template (element_class,
gst_static_pad_template_get (&gst_rtp_g722_depay_sink_template));
gst_element_class_set_details_simple (element_class, "RTP audio depayloader",
"Codec/Depayloader/Network",
"Extracts G722 audio from RTP packets",
"Wim Taymans <wim.taymans@gmail.com>");
}
static void
gst_rtp_g722_depay_class_init (GstRtpG722DepayClass * klass)
{
GstBaseRTPDepayloadClass *gstbasertpdepayload_class;
gstbasertpdepayload_class = (GstBaseRTPDepayloadClass *) klass;
gstbasertpdepayload_class->set_caps = gst_rtp_g722_depay_setcaps;
gstbasertpdepayload_class->process = gst_rtp_g722_depay_process;
GST_DEBUG_CATEGORY_INIT (rtpg722depay_debug, "rtpg722depay", 0,
"G722 RTP Depayloader");
}
static void
gst_rtp_g722_depay_init (GstRtpG722Depay * rtpg722depay,
GstRtpG722DepayClass * klass)
{
/* needed because of GST_BOILERPLATE */
}
static gint
gst_rtp_g722_depay_parse_int (GstStructure * structure, const gchar * field,
gint def)
{
const gchar *str;
gint res;
if ((str = gst_structure_get_string (structure, field)))
return atoi (str);
if (gst_structure_get_int (structure, field, &res))
return res;
return def;
}
static gboolean
gst_rtp_g722_depay_setcaps (GstBaseRTPDepayload * depayload, GstCaps * caps)
{
GstStructure *structure;
GstRtpG722Depay *rtpg722depay;
gint clock_rate, payload, samplerate;
gint channels;
GstCaps *srccaps;
gboolean res;
const gchar *channel_order;
const GstRTPChannelOrder *order;
rtpg722depay = GST_RTP_G722_DEPAY (depayload);
structure = gst_caps_get_structure (caps, 0);
payload = 96;
gst_structure_get_int (structure, "payload", &payload);
switch (payload) {
case GST_RTP_PAYLOAD_G722:
channels = 1;
clock_rate = 8000;
samplerate = 16000;
break;
default:
/* no fixed mapping, we need clock-rate */
channels = 0;
clock_rate = 0;
samplerate = 0;
break;
}
/* caps can overwrite defaults */
clock_rate =
gst_rtp_g722_depay_parse_int (structure, "clock-rate", clock_rate);
if (clock_rate == 0)
goto no_clockrate;
if (clock_rate == 8000)
samplerate = 16000;
if (samplerate == 0)
samplerate = clock_rate;
channels =
gst_rtp_g722_depay_parse_int (structure, "encoding-params", channels);
if (channels == 0) {
channels = gst_rtp_g722_depay_parse_int (structure, "channels", channels);
if (channels == 0) {
/* channels defaults to 1 otherwise */
channels = 1;
}
}
depayload->clock_rate = clock_rate;
rtpg722depay->rate = samplerate;
rtpg722depay->channels = channels;
srccaps = gst_caps_new_simple ("audio/G722",
"rate", G_TYPE_INT, samplerate, "channels", G_TYPE_INT, channels, NULL);
/* add channel positions */
channel_order = gst_structure_get_string (structure, "channel-order");
order = gst_rtp_channels_get_by_order (channels, channel_order);
if (order) {
gst_audio_set_channel_positions (gst_caps_get_structure (srccaps, 0),
order->pos);
} else {
GstAudioChannelPosition *pos;
GST_ELEMENT_WARNING (rtpg722depay, STREAM, DECODE,
(NULL), ("Unknown channel order '%s' for %d channels",
GST_STR_NULL (channel_order), channels));
/* create default NONE layout */
pos = gst_rtp_channels_create_default (channels);
gst_audio_set_channel_positions (gst_caps_get_structure (srccaps, 0), pos);
g_free (pos);
}
res = gst_pad_set_caps (depayload->srcpad, srccaps);
gst_caps_unref (srccaps);
return res;
/* ERRORS */
no_clockrate:
{
GST_ERROR_OBJECT (depayload, "no clock-rate specified");
return FALSE;
}
}
static GstBuffer *
gst_rtp_g722_depay_process (GstBaseRTPDepayload * depayload, GstBuffer * buf)
{
GstRtpG722Depay *rtpg722depay;
GstBuffer *outbuf;
gint payload_len;
gboolean marker;
rtpg722depay = GST_RTP_G722_DEPAY (depayload);
payload_len = gst_rtp_buffer_get_payload_len (buf);
if (payload_len <= 0)
goto empty_packet;
GST_DEBUG_OBJECT (rtpg722depay, "got payload of %d bytes", payload_len);
outbuf = gst_rtp_buffer_get_payload_buffer (buf);
marker = gst_rtp_buffer_get_marker (buf);
if (marker) {
/* mark talk spurt with DISCONT */
GST_BUFFER_FLAG_SET (outbuf, GST_BUFFER_FLAG_DISCONT);
}
return outbuf;
/* ERRORS */
empty_packet:
{
GST_ELEMENT_WARNING (rtpg722depay, STREAM, DECODE,
("Empty Payload."), (NULL));
return NULL;
}
}
gboolean
gst_rtp_g722_depay_plugin_init (GstPlugin * plugin)
{
return gst_element_register (plugin, "rtpg722depay",
GST_RANK_MARGINAL, GST_TYPE_RTP_G722_DEPAY);
}
/* GStreamer
* Copyright (C) <2010> 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_G722_DEPAY_H__
#define __GST_RTP_G722_DEPAY_H__
#include <gst/gst.h>
#include <gst/rtp/gstbasertpdepayload.h>
G_BEGIN_DECLS
/* Standard macros for defining types for this element. */
#define GST_TYPE_RTP_G722_DEPAY \
(gst_rtp_g722_depay_get_type())
#define GST_RTP_G722_DEPAY(obj) \
(G_TYPE_CHECK_INSTANCE_CAST((obj),GST_TYPE_RTP_G722_DEPAY,GstRtpG722Depay))
#define GST_RTP_G722_DEPAY_CLASS(klass) \
(G_TYPE_CHECK_CLASS_CAST((klass),GST_TYPE_RTP_G722_DEPAY,GstRtpG722DepayClass))
#define GST_IS_RTP_G722_DEPAY(obj) \
(G_TYPE_CHECK_INSTANCE_TYPE((obj),GST_TYPE_RTP_G722_DEPAY))
#define GST_IS_RTP_G722_DEPAY_CLASS(klass) \
(G_TYPE_CHECK_CLASS_TYPE((klass),GST_TYPE_RTP_G722_DEPAY))
typedef struct _GstRtpG722Depay GstRtpG722Depay;
typedef struct _GstRtpG722DepayClass GstRtpG722DepayClass;
/* Definition of structure storing data for this element. */
struct _GstRtpG722Depay
{
GstBaseRTPDepayload depayload;
guint rate;
guint channels;
};
/* Standard definition defining a class for this element. */
struct _GstRtpG722DepayClass
{
GstBaseRTPDepayloadClass parent_class;
};
GType gst_rtp_g722_depay_get_type (void);
gboolean gst_rtp_g722_depay_plugin_init (GstPlugin * plugin);
G_END_DECLS
#endif /* __GST_RTP_G722_DEPAY_H__ */
/* GStreamer
* Copyright (C) <2010> 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 <string.h>
#include <gst/audio/audio.h>
#include <gst/audio/multichannel.h>
#include <gst/rtp/gstrtpbuffer.h>
#include "gstrtpg722pay.h"
#include "gstrtpchannels.h"
GST_DEBUG_CATEGORY_STATIC (rtpg722pay_debug);
#define GST_CAT_DEFAULT (rtpg722pay_debug)
static GstStaticPadTemplate gst_rtp_g722_pay_sink_template =
GST_STATIC_PAD_TEMPLATE ("sink",
GST_PAD_SINK,
GST_PAD_ALWAYS,
GST_STATIC_CAPS ("audio/G722, " "rate = (int) 16000, " "channels = (int) 1")
);
static GstStaticPadTemplate gst_rtp_g722_pay_src_template =
GST_STATIC_PAD_TEMPLATE ("src",
GST_PAD_SRC,
GST_PAD_ALWAYS,
GST_STATIC_CAPS ("application/x-rtp, "
"media = (string) \"audio\", "
"encoding-name = (string) \"G722\", "
"payload = (int) " GST_RTP_PAYLOAD_G722_STRING ", "
"clock-rate = (int) 8000")
);
static gboolean gst_rtp_g722_pay_setcaps (GstBaseRTPPayload * basepayload,
GstCaps * caps);
static GstCaps *gst_rtp_g722_pay_getcaps (GstBaseRTPPayload * rtppayload,
GstPad * pad);
GST_BOILERPLATE (GstRtpG722Pay, gst_rtp_g722_pay, GstBaseRTPAudioPayload,
GST_TYPE_BASE_RTP_AUDIO_PAYLOAD);
static void
gst_rtp_g722_pay_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_g722_pay_src_template));
gst_element_class_add_pad_template (element_class,
gst_static_pad_template_get (&gst_rtp_g722_pay_sink_template));
gst_element_class_set_details_simple (element_class, "RTP audio payloader",
"Codec/Payloader/Network",
"Payload-encode Raw audio into RTP packets (RFC 3551)",
"Wim Taymans <wim.taymans@gmail.com>");
}
static void
gst_rtp_g722_pay_class_init (GstRtpG722PayClass * klass)
{
GstBaseRTPPayloadClass *gstbasertppayload_class;
gstbasertppayload_class = (GstBaseRTPPayloadClass *) klass;
gstbasertppayload_class->set_caps = gst_rtp_g722_pay_setcaps;
gstbasertppayload_class->get_caps = gst_rtp_g722_pay_getcaps;
GST_DEBUG_CATEGORY_INIT (rtpg722pay_debug, "rtpg722pay", 0,
"G722 RTP Payloader");
}
static void
gst_rtp_g722_pay_init (GstRtpG722Pay * rtpg722pay, GstRtpG722PayClass * klass)
{
GstBaseRTPAudioPayload *basertpaudiopayload;
basertpaudiopayload = GST_BASE_RTP_AUDIO_PAYLOAD (rtpg722pay);
/* tell basertpaudiopayload that this is a sample based codec */
gst_base_rtp_audio_payload_set_sample_based (basertpaudiopayload);
}
static gboolean
gst_rtp_g722_pay_setcaps (GstBaseRTPPayload * basepayload, GstCaps * caps)
{
GstRtpG722Pay *rtpg722pay;
GstStructure *structure;
gint rate, channels, clock_rate;
gboolean res;
gchar *params;
GstAudioChannelPosition *pos;
const GstRTPChannelOrder *order;
GstBaseRTPAudioPayload *basertpaudiopayload;
basertpaudiopayload = GST_BASE_RTP_AUDIO_PAYLOAD (basepayload);
rtpg722pay = GST_RTP_G722_PAY (basepayload);
structure = gst_caps_get_structure (caps, 0);
/* first parse input caps */
if (!gst_structure_get_int (structure, "rate", &rate))
goto no_rate;
if (!gst_structure_get_int (structure, "channels", &channels))
goto no_channels;
/* get the channel order */
pos = gst_audio_get_channel_positions (structure);
if (pos)
order = gst_rtp_channels_get_by_pos (channels, pos);
else
order = NULL;
if (rate == 16000)
clock_rate = 8000;
gst_basertppayload_set_options (basepayload, "audio", TRUE, "G722",
clock_rate);
params = g_strdup_printf ("%d", channels);
if (!order && channels > 2) {
GST_ELEMENT_WARNING (rtpg722pay, STREAM, DECODE,
(NULL), ("Unknown channel order for %d channels", channels));
}
if (order && order->name) {
res = gst_basertppayload_set_outcaps (basepayload,
"encoding-params", G_TYPE_STRING, params, "channels", G_TYPE_INT,
channels, "channel-order", G_TYPE_STRING, order->name, NULL);
} else {
res = gst_basertppayload_set_outcaps (basepayload,
"encoding-params", G_TYPE_STRING, params, "channels", G_TYPE_INT,
channels, NULL);
}
g_free (params);
g_free (pos);
rtpg722pay->rate = rate;
rtpg722pay->channels = channels;
/* octet-per-sample is 1 * channels for G722 */
gst_base_rtp_audio_payload_set_samplebits_options (basertpaudiopayload,
4 * rtpg722pay->channels);
return res;
/* ERRORS */
no_rate:
{
GST_DEBUG_OBJECT (rtpg722pay, "no rate given");
return FALSE;
}
no_channels:
{
GST_DEBUG_OBJECT (rtpg722pay, "no channels given");
return FALSE;
}
}
static GstCaps *
gst_rtp_g722_pay_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;
structure = gst_caps_get_structure (otherpadcaps, 0);
gst_caps_set_simple (caps, "channels", G_TYPE_INT, 1, NULL);
gst_caps_set_simple (caps, "rate", G_TYPE_INT, 16000, NULL);
}
gst_caps_unref (otherpadcaps);
}
return caps;
}
gboolean
gst_rtp_g722_pay_plugin_init (GstPlugin * plugin)
{
return gst_element_register (plugin, "rtpg722pay",
GST_RANK_NONE, GST_TYPE_RTP_G722_PAY);
}
/* GStreamer
* Copyright (C) <2010> 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_G722_PAY_H__
#define __GST_RTP_G722_PAY_H__
#include <gst/gst.h>
#include <gst/rtp/gstbasertpaudiopayload.h>
G_BEGIN_DECLS
#define GST_TYPE_RTP_G722_PAY \
(gst_rtp_g722_pay_get_type())
#define GST_RTP_G722_PAY(obj) \
(G_TYPE_CHECK_INSTANCE_CAST((obj),GST_TYPE_RTP_G722_PAY,GstRtpG722Pay))
#define GST_RTP_G722_PAY_CLASS(klass) \
(G_TYPE_CHECK_CLASS_CAST((klass),GST_TYPE_RTP_G722_PAY,GstRtpG722PayClass))
#define GST_IS_RTP_G722_PAY(obj) \
(G_TYPE_CHECK_INSTANCE_TYPE((obj),GST_TYPE_RTP_G722_PAY))
#define GST_IS_RTP_G722_PAY_CLASS(klass) \
(G_TYPE_CHECK_CLASS_TYPE((klass),GST_TYPE_RTP_G722_PAY))
typedef struct _GstRtpG722Pay GstRtpG722Pay;
typedef struct _GstRtpG722PayClass GstRtpG722PayClass;
struct _GstRtpG722Pay
{
GstBaseRTPAudioPayload payload;
gint rate;
gint channels;
};
struct _GstRtpG722PayClass
{
GstBaseRTPAudioPayloadClass parent_class;
};
GType gst_rtp_g722_pay_get_type (void);
gboolean gst_rtp_g722_pay_plugin_init (GstPlugin * plugin);
G_END_DECLS
#endif /* __GST_RTP_G722_PAY_H__ */
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