gstreamer/gst/rtp/gstrtpspeexdepay.c
Tim-Philipp Müller 4a28e649c3 rtp: cache meta tag quarks and add more utility functions for metas
Every g_quark_from_static_string() is a hash table lookup serialised
on the global quark lock in GLib. Let's just look up the two quarks
we need once and cache them locally for future use. While we're at it,
add new utility functions for the two most commonly used tags
(audio + video). Make first argument a gpointer so we don't have to
cast and make the code ugly. These are used for logging purposes
only anyway.
2017-05-24 13:32:10 +01:00

226 lines
6.4 KiB
C

/* GStreamer
* Copyright (C) <2005> Edgard Lima <edgard.lima@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., 51 Franklin St, Fifth Floor,
* Boston, MA 02110-1301, USA.
*/
#ifdef HAVE_CONFIG_H
# include "config.h"
#endif
#include <string.h>
#include <stdlib.h>
#include <gst/rtp/gstrtpbuffer.h>
#include <gst/audio/audio.h>
#include "gstrtpspeexdepay.h"
#include "gstrtputils.h"
/* RtpSPEEXDepay signals and args */
enum
{
/* FILL ME */
LAST_SIGNAL
};
enum
{
PROP_0
};
static GstStaticPadTemplate gst_rtp_speex_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) [6000, 48000], "
"encoding-name = (string) \"SPEEX\"")
/* "encoding-params = (string) \"1\"" */
);
static GstStaticPadTemplate gst_rtp_speex_depay_src_template =
GST_STATIC_PAD_TEMPLATE ("src",
GST_PAD_SRC,
GST_PAD_ALWAYS,
GST_STATIC_CAPS ("audio/x-speex")
);
static GstBuffer *gst_rtp_speex_depay_process (GstRTPBaseDepayload * depayload,
GstRTPBuffer * rtp);
static gboolean gst_rtp_speex_depay_setcaps (GstRTPBaseDepayload * depayload,
GstCaps * caps);
G_DEFINE_TYPE (GstRtpSPEEXDepay, gst_rtp_speex_depay,
GST_TYPE_RTP_BASE_DEPAYLOAD);
static void
gst_rtp_speex_depay_class_init (GstRtpSPEEXDepayClass * klass)
{
GstElementClass *gstelement_class;
GstRTPBaseDepayloadClass *gstrtpbasedepayload_class;
gstelement_class = (GstElementClass *) klass;
gstrtpbasedepayload_class = (GstRTPBaseDepayloadClass *) klass;
gstrtpbasedepayload_class->process_rtp_packet = gst_rtp_speex_depay_process;
gstrtpbasedepayload_class->set_caps = gst_rtp_speex_depay_setcaps;
gst_element_class_add_static_pad_template (gstelement_class,
&gst_rtp_speex_depay_src_template);
gst_element_class_add_static_pad_template (gstelement_class,
&gst_rtp_speex_depay_sink_template);
gst_element_class_set_static_metadata (gstelement_class,
"RTP Speex depayloader", "Codec/Depayloader/Network/RTP",
"Extracts Speex audio from RTP packets",
"Edgard Lima <edgard.lima@gmail.com>");
}
static void
gst_rtp_speex_depay_init (GstRtpSPEEXDepay * rtpspeexdepay)
{
}
static gint
gst_rtp_speex_depay_get_mode (gint rate)
{
if (rate > 25000)
return 2;
else if (rate > 12500)
return 1;
else
return 0;
}
/* len 4 bytes LE,
* vendor string (len bytes),
* user_len 4 (0) bytes LE
*/
static const gchar gst_rtp_speex_comment[] =
"\045\0\0\0Depayloaded with GStreamer speexdepay\0\0\0\0";
static gboolean
gst_rtp_speex_depay_setcaps (GstRTPBaseDepayload * depayload, GstCaps * caps)
{
GstStructure *structure;
GstRtpSPEEXDepay *rtpspeexdepay;
gint clock_rate, nb_channels;
GstBuffer *buf;
GstMapInfo map;
guint8 *data;
const gchar *params;
GstCaps *srccaps;
gboolean res;
rtpspeexdepay = GST_RTP_SPEEX_DEPAY (depayload);
structure = gst_caps_get_structure (caps, 0);
if (!gst_structure_get_int (structure, "clock-rate", &clock_rate))
goto no_clockrate;
depayload->clock_rate = clock_rate;
if (!(params = gst_structure_get_string (structure, "encoding-params")))
nb_channels = 1;
else {
nb_channels = atoi (params);
}
/* construct minimal header and comment packet for the decoder */
buf = gst_buffer_new_and_alloc (80);
gst_buffer_map (buf, &map, GST_MAP_WRITE);
data = map.data;
memcpy (data, "Speex ", 8);
data += 8;
memcpy (data, "1.1.12", 7);
data += 20;
GST_WRITE_UINT32_LE (data, 1); /* version */
data += 4;
GST_WRITE_UINT32_LE (data, 80); /* header_size */
data += 4;
GST_WRITE_UINT32_LE (data, clock_rate); /* rate */
data += 4;
GST_WRITE_UINT32_LE (data, gst_rtp_speex_depay_get_mode (clock_rate)); /* mode */
data += 4;
GST_WRITE_UINT32_LE (data, 4); /* mode_bitstream_version */
data += 4;
GST_WRITE_UINT32_LE (data, nb_channels); /* nb_channels */
data += 4;
GST_WRITE_UINT32_LE (data, -1); /* bitrate */
data += 4;
GST_WRITE_UINT32_LE (data, 0xa0); /* frame_size */
data += 4;
GST_WRITE_UINT32_LE (data, 0); /* VBR */
data += 4;
GST_WRITE_UINT32_LE (data, 1); /* frames_per_packet */
data += 4;
GST_WRITE_UINT32_LE (data, 0); /* extra_headers */
data += 4;
GST_WRITE_UINT32_LE (data, 0); /* reserved1 */
data += 4;
GST_WRITE_UINT32_LE (data, 0); /* reserved2 */
gst_buffer_unmap (buf, &map);
srccaps = gst_caps_new_empty_simple ("audio/x-speex");
res = gst_pad_set_caps (depayload->srcpad, srccaps);
gst_caps_unref (srccaps);
gst_rtp_base_depayload_push (GST_RTP_BASE_DEPAYLOAD (rtpspeexdepay), buf);
buf = gst_buffer_new_and_alloc (sizeof (gst_rtp_speex_comment));
gst_buffer_fill (buf, 0, gst_rtp_speex_comment,
sizeof (gst_rtp_speex_comment));
gst_rtp_base_depayload_push (GST_RTP_BASE_DEPAYLOAD (rtpspeexdepay), buf);
return res;
/* ERRORS */
no_clockrate:
{
GST_DEBUG_OBJECT (depayload, "no clock-rate specified");
return FALSE;
}
}
static GstBuffer *
gst_rtp_speex_depay_process (GstRTPBaseDepayload * depayload,
GstRTPBuffer * rtp)
{
GstBuffer *outbuf = NULL;
GST_DEBUG ("process : got %" G_GSIZE_FORMAT " bytes, mark %d ts %u seqn %d",
gst_buffer_get_size (rtp->buffer),
gst_rtp_buffer_get_marker (rtp),
gst_rtp_buffer_get_timestamp (rtp), gst_rtp_buffer_get_seq (rtp));
/* nothing special to be done */
outbuf = gst_rtp_buffer_get_payload_buffer (rtp);
if (outbuf) {
GST_BUFFER_DURATION (outbuf) = 20 * GST_MSECOND;
gst_rtp_drop_non_audio_meta (depayload, outbuf);
}
return outbuf;
}
gboolean
gst_rtp_speex_depay_plugin_init (GstPlugin * plugin)
{
return gst_element_register (plugin, "rtpspeexdepay",
GST_RANK_SECONDARY, GST_TYPE_RTP_SPEEX_DEPAY);
}