gstreamer/gst/rtp/gstrtpg729depay.c
Tim-Philipp Müller 6717c86061 rtp: depayloaders: implement process_rtp_packet() vfunc
For more optimised RTP packet handling: means we don't
need to map the input buffer again but can just re-use
the mapping the base class has already done.

https://bugzilla.gnome.org/show_bug.cgi?id=750235
2015-07-12 14:28:29 +01:00

221 lines
5.8 KiB
C

/* GStreamer
*
* 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., 51 Franklin St, Fifth Floor,
* Boston, MA 02110-1301, USA.
*/
#ifdef HAVE_CONFIG_H
# include "config.h"
#endif
#include <gst/rtp/gstrtpbuffer.h>
#include <stdlib.h>
#include <string.h>
#include "gstrtpg729depay.h"
GST_DEBUG_CATEGORY_STATIC (rtpg729depay_debug);
#define GST_CAT_DEFAULT (rtpg729depay_debug)
/* references:
*
* RFC 3551 (4.5.6)
*/
enum
{
/* FILL ME */
LAST_SIGNAL
};
enum
{
PROP_0
};
/* input is an RTP packet
*
*/
static GstStaticPadTemplate gst_rtp_g729_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) \"G729\"; "
"application/x-rtp, "
"media = (string) \"audio\", "
"payload = (int) " GST_RTP_PAYLOAD_G729_STRING ", "
"clock-rate = (int) 8000")
);
static GstStaticPadTemplate gst_rtp_g729_depay_src_template =
GST_STATIC_PAD_TEMPLATE ("src",
GST_PAD_SRC,
GST_PAD_ALWAYS,
GST_STATIC_CAPS ("audio/G729, " "channels = (int) 1," "rate = (int) 8000")
);
static gboolean gst_rtp_g729_depay_setcaps (GstRTPBaseDepayload * depayload,
GstCaps * caps);
static GstBuffer *gst_rtp_g729_depay_process (GstRTPBaseDepayload * depayload,
GstRTPBuffer * rtp);
#define gst_rtp_g729_depay_parent_class parent_class
G_DEFINE_TYPE (GstRtpG729Depay, gst_rtp_g729_depay,
GST_TYPE_RTP_BASE_DEPAYLOAD);
static void
gst_rtp_g729_depay_class_init (GstRtpG729DepayClass * klass)
{
GstElementClass *gstelement_class;
GstRTPBaseDepayloadClass *gstrtpbasedepayload_class;
GST_DEBUG_CATEGORY_INIT (rtpg729depay_debug, "rtpg729depay", 0,
"G.729 RTP Depayloader");
gstelement_class = (GstElementClass *) klass;
gstrtpbasedepayload_class = (GstRTPBaseDepayloadClass *) klass;
gst_element_class_add_pad_template (gstelement_class,
gst_static_pad_template_get (&gst_rtp_g729_depay_src_template));
gst_element_class_add_pad_template (gstelement_class,
gst_static_pad_template_get (&gst_rtp_g729_depay_sink_template));
gst_element_class_set_static_metadata (gstelement_class,
"RTP G.729 depayloader", "Codec/Depayloader/Network/RTP",
"Extracts G.729 audio from RTP packets (RFC 3551)",
"Laurent Glayal <spglegle@yahoo.fr>");
gstrtpbasedepayload_class->process_rtp_packet = gst_rtp_g729_depay_process;
gstrtpbasedepayload_class->set_caps = gst_rtp_g729_depay_setcaps;
}
static void
gst_rtp_g729_depay_init (GstRtpG729Depay * rtpg729depay)
{
GstRTPBaseDepayload *depayload;
depayload = GST_RTP_BASE_DEPAYLOAD (rtpg729depay);
gst_pad_use_fixed_caps (GST_RTP_BASE_DEPAYLOAD_SRCPAD (depayload));
}
static gboolean
gst_rtp_g729_depay_setcaps (GstRTPBaseDepayload * depayload, GstCaps * caps)
{
GstStructure *structure;
GstCaps *srccaps;
GstRtpG729Depay *rtpg729depay;
const gchar *params;
gint clock_rate, channels;
gboolean ret;
rtpg729depay = GST_RTP_G729_DEPAY (depayload);
structure = gst_caps_get_structure (caps, 0);
if (!(params = gst_structure_get_string (structure, "encoding-params")))
channels = 1;
else {
channels = atoi (params);
}
if (!gst_structure_get_int (structure, "clock-rate", &clock_rate))
clock_rate = 8000;
if (channels != 1)
goto wrong_channels;
if (clock_rate != 8000)
goto wrong_clock_rate;
depayload->clock_rate = clock_rate;
srccaps = gst_caps_new_simple ("audio/G729",
"channels", G_TYPE_INT, channels, "rate", G_TYPE_INT, clock_rate, NULL);
ret = gst_pad_set_caps (GST_RTP_BASE_DEPAYLOAD_SRCPAD (depayload), srccaps);
gst_caps_unref (srccaps);
return ret;
/* ERRORS */
wrong_channels:
{
GST_DEBUG_OBJECT (rtpg729depay, "expected 1 channel, got %d", channels);
return FALSE;
}
wrong_clock_rate:
{
GST_DEBUG_OBJECT (rtpg729depay, "expected 8000 clock-rate, got %d",
clock_rate);
return FALSE;
}
}
static GstBuffer *
gst_rtp_g729_depay_process (GstRTPBaseDepayload * depayload, GstRTPBuffer * rtp)
{
GstRtpG729Depay *rtpg729depay;
GstBuffer *outbuf = NULL;
gint payload_len;
gboolean marker;
rtpg729depay = GST_RTP_G729_DEPAY (depayload);
payload_len = gst_rtp_buffer_get_payload_len (rtp);
/* At least 2 bytes (CNG from G729 Annex B) */
if (payload_len < 2) {
GST_ELEMENT_WARNING (rtpg729depay, STREAM, DECODE,
(NULL), ("G729 RTP payload too small (%d)", payload_len));
goto bad_packet;
}
GST_LOG_OBJECT (rtpg729depay, "payload len %d", payload_len);
if ((payload_len % 10) == 2) {
GST_LOG_OBJECT (rtpg729depay, "G729 payload contains CNG frame");
}
outbuf = gst_rtp_buffer_get_payload_buffer (rtp);
marker = gst_rtp_buffer_get_marker (rtp);
if (marker) {
/* marker bit starts talkspurt */
GST_BUFFER_FLAG_SET (outbuf, GST_BUFFER_FLAG_RESYNC);
}
GST_LOG_OBJECT (depayload, "pushing buffer of size %" G_GSIZE_FORMAT,
gst_buffer_get_size (outbuf));
return outbuf;
/* ERRORS */
bad_packet:
{
/* no fatal error */
return NULL;
}
}
gboolean
gst_rtp_g729_depay_plugin_init (GstPlugin * plugin)
{
return gst_element_register (plugin, "rtpg729depay",
GST_RANK_SECONDARY, GST_TYPE_RTP_G729_DEPAY);
}