gstreamer/gst/rtp/gstrtpg722depay.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

262 lines
7.2 KiB
C

/* 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., 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/audio/audio.h>
#include "gstrtpg722depay.h"
#include "gstrtpchannels.h"
#include "gstrtputils.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\", " "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" */
)
);
#define gst_rtp_g722_depay_parent_class parent_class
G_DEFINE_TYPE (GstRtpG722Depay, gst_rtp_g722_depay,
GST_TYPE_RTP_BASE_DEPAYLOAD);
static gboolean gst_rtp_g722_depay_setcaps (GstRTPBaseDepayload * depayload,
GstCaps * caps);
static GstBuffer *gst_rtp_g722_depay_process (GstRTPBaseDepayload * depayload,
GstRTPBuffer * rtp);
static void
gst_rtp_g722_depay_class_init (GstRtpG722DepayClass * klass)
{
GstElementClass *gstelement_class;
GstRTPBaseDepayloadClass *gstrtpbasedepayload_class;
GST_DEBUG_CATEGORY_INIT (rtpg722depay_debug, "rtpg722depay", 0,
"G722 RTP Depayloader");
gstelement_class = (GstElementClass *) klass;
gstrtpbasedepayload_class = (GstRTPBaseDepayloadClass *) klass;
gst_element_class_add_static_pad_template (gstelement_class,
&gst_rtp_g722_depay_src_template);
gst_element_class_add_static_pad_template (gstelement_class,
&gst_rtp_g722_depay_sink_template);
gst_element_class_set_static_metadata (gstelement_class,
"RTP audio depayloader", "Codec/Depayloader/Network/RTP",
"Extracts G722 audio from RTP packets",
"Wim Taymans <wim.taymans@gmail.com>");
gstrtpbasedepayload_class->set_caps = gst_rtp_g722_depay_setcaps;
gstrtpbasedepayload_class->process_rtp_packet = gst_rtp_g722_depay_process;
}
static void
gst_rtp_g722_depay_init (GstRtpG722Depay * rtpg722depay)
{
}
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 (GstRTPBaseDepayload * depayload, GstCaps * caps)
{
GstStructure *structure;
GstRtpG722Depay *rtpg722depay;
gint clock_rate, payload, samplerate;
gint channels;
GstCaps *srccaps;
gboolean res;
#if 0
const gchar *channel_order;
const GstRTPChannelOrder *order;
#endif
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);
/* FIXME: Do something with the channel order */
#if 0
/* 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);
}
#endif
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 (GstRTPBaseDepayload * depayload, GstRTPBuffer * rtp)
{
GstRtpG722Depay *rtpg722depay;
GstBuffer *outbuf;
gint payload_len;
gboolean marker;
rtpg722depay = GST_RTP_G722_DEPAY (depayload);
payload_len = gst_rtp_buffer_get_payload_len (rtp);
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 (rtp);
marker = gst_rtp_buffer_get_marker (rtp);
if (marker && outbuf) {
/* mark talk spurt with RESYNC */
GST_BUFFER_FLAG_SET (outbuf, GST_BUFFER_FLAG_RESYNC);
}
if (outbuf) {
gst_rtp_drop_non_audio_meta (rtpg722depay, outbuf);
}
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_SECONDARY, GST_TYPE_RTP_G722_DEPAY);
}