gstreamer/gst/rtp/gstrtpilbcpay.c

216 lines
6.4 KiB
C

/* 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 <stdlib.h>
#include <gst/rtp/gstrtpbuffer.h>
#include "gstrtpilbcpay.h"
GST_DEBUG_CATEGORY_STATIC (rtpilbcpay_debug);
#define GST_CAT_DEFAULT (rtpilbcpay_debug)
static GstStaticPadTemplate gst_rtp_ilbc_pay_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_rtp_ilbc_pay_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 = (string) { \"20\", \"30\" }")
);
static GstCaps *gst_rtp_ilbc_pay_sink_getcaps (GstRTPBasePayload * payload,
GstPad * pad, GstCaps * filter);
static gboolean gst_rtp_ilbc_pay_sink_setcaps (GstRTPBasePayload * payload,
GstCaps * caps);
#define gst_rtp_ilbc_pay_parent_class parent_class
G_DEFINE_TYPE (GstRTPILBCPay, gst_rtp_ilbc_pay,
GST_TYPE_RTP_BASE_AUDIO_PAYLOAD);
static void
gst_rtp_ilbc_pay_class_init (GstRTPILBCPayClass * klass)
{
GstElementClass *gstelement_class;
GstRTPBasePayloadClass *gstrtpbasepayload_class;
GST_DEBUG_CATEGORY_INIT (rtpilbcpay_debug, "rtpilbcpay", 0,
"iLBC audio RTP payloader");
gstelement_class = (GstElementClass *) klass;
gstrtpbasepayload_class = (GstRTPBasePayloadClass *) klass;
gst_element_class_add_pad_template (gstelement_class,
gst_static_pad_template_get (&gst_rtp_ilbc_pay_sink_template));
gst_element_class_add_pad_template (gstelement_class,
gst_static_pad_template_get (&gst_rtp_ilbc_pay_src_template));
gst_element_class_set_details_simple (gstelement_class, "RTP iLBC Payloader",
"Codec/Payloader/Network/RTP",
"Packetize iLBC audio streams into RTP packets",
"Philippe Kalaf <philippe.kalaf@collabora.co.uk>");
gstrtpbasepayload_class->set_caps = gst_rtp_ilbc_pay_sink_setcaps;
gstrtpbasepayload_class->get_caps = gst_rtp_ilbc_pay_sink_getcaps;
}
static void
gst_rtp_ilbc_pay_init (GstRTPILBCPay * rtpilbcpay)
{
GstRTPBasePayload *rtpbasepayload;
GstRTPBaseAudioPayload *rtpbaseaudiopayload;
rtpbasepayload = GST_RTP_BASE_PAYLOAD (rtpilbcpay);
rtpbaseaudiopayload = GST_RTP_BASE_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 */
rtpbasepayload->clock_rate = 8000;
rtpilbcpay->mode = -1;
/* tell rtpbaseaudiopayload that this is a frame based codec */
gst_rtp_base_audio_payload_set_frame_based (rtpbaseaudiopayload);
}
static gboolean
gst_rtp_ilbc_pay_sink_setcaps (GstRTPBasePayload * rtpbasepayload,
GstCaps * caps)
{
GstRTPILBCPay *rtpilbcpay;
GstRTPBaseAudioPayload *rtpbaseaudiopayload;
gboolean ret;
gint mode;
gchar *mode_str;
GstStructure *structure;
const char *payload_name;
rtpilbcpay = GST_RTP_ILBC_PAY (rtpbasepayload);
rtpbaseaudiopayload = GST_RTP_BASE_AUDIO_PAYLOAD (rtpbasepayload);
structure = gst_caps_get_structure (caps, 0);
payload_name = gst_structure_get_name (structure);
if (g_ascii_strcasecmp ("audio/x-iLBC", payload_name))
goto wrong_caps;
if (!gst_structure_get_int (structure, "mode", &mode))
goto no_mode;
if (mode != 20 && mode != 30)
goto wrong_mode;
gst_rtp_base_payload_set_options (rtpbasepayload, "audio", TRUE, "ILBC",
8000);
/* set options for this frame based audio codec */
gst_rtp_base_audio_payload_set_frame_options (rtpbaseaudiopayload,
mode, mode == 30 ? 50 : 38);
mode_str = g_strdup_printf ("%d", mode);
ret =
gst_rtp_base_payload_set_outcaps (rtpbasepayload, "mode", G_TYPE_STRING,
mode_str, NULL);
g_free (mode_str);
if (mode != rtpilbcpay->mode && rtpilbcpay->mode != -1)
goto mode_changed;
rtpilbcpay->mode = mode;
return ret;
/* ERRORS */
wrong_caps:
{
GST_ERROR_OBJECT (rtpilbcpay, "expected audio/x-iLBC, received %s",
payload_name);
return FALSE;
}
no_mode:
{
GST_ERROR_OBJECT (rtpilbcpay, "did not receive a mode");
return FALSE;
}
wrong_mode:
{
GST_ERROR_OBJECT (rtpilbcpay, "mode must be 20 or 30, received %d", mode);
return FALSE;
}
mode_changed:
{
GST_ERROR_OBJECT (rtpilbcpay, "Mode has changed from %d to %d! "
"Mode cannot change while streaming", rtpilbcpay->mode, mode);
return FALSE;
}
}
/* we return the padtemplate caps with the mode field fixated to a value if we
* can */
static GstCaps *
gst_rtp_ilbc_pay_sink_getcaps (GstRTPBasePayload * rtppayload, GstPad * pad,
GstCaps * filter)
{
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);
/* parse mode, if we can */
mode_str = gst_structure_get_string (structure, "mode");
if (mode_str) {
mode = strtol (mode_str, NULL, 10);
if (mode == 20 || mode == 30) {
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_ilbc_pay_plugin_init (GstPlugin * plugin)
{
return gst_element_register (plugin, "rtpilbcpay",
GST_RANK_SECONDARY, GST_TYPE_RTP_ILBC_PAY);
}