blob: 9ec03499f3052d39cdcdf473126c6bd4e396217a [file] [log] [blame]
/* 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.
*/
/**
* SECTION:element-rtpL24depay
* @see_also: rtpL24pay
*
* Extract raw audio from RTP packets according to RFC 3190, section 4.
* For detailed information see: http://www.rfc-editor.org/rfc/rfc3190.txt
*
* <refsect2>
* <title>Example pipeline</title>
* |[
* gst-launch-1.0 udpsrc caps='application/x-rtp, media=(string)audio, clock-rate=(int)44100, encoding-name=(string)L24, encoding-params=(string)1, channels=(int)1, payload=(int)96' ! rtpL24depay ! pulsesink
* ]| This example pipeline will depayload an RTP raw audio stream. Refer to
* the rtpL24pay example to create the RTP stream.
* </refsect2>
*/
#ifdef HAVE_CONFIG_H
#include "config.h"
#endif
#include <string.h>
#include <stdlib.h>
#include <gst/audio/audio.h>
#include "gstrtpL24depay.h"
#include "gstrtpchannels.h"
#include "gstrtputils.h"
GST_DEBUG_CATEGORY_STATIC (rtpL24depay_debug);
#define GST_CAT_DEFAULT (rtpL24depay_debug)
static GstStaticPadTemplate gst_rtp_L24_depay_src_template =
GST_STATIC_PAD_TEMPLATE ("src",
GST_PAD_SRC,
GST_PAD_ALWAYS,
GST_STATIC_CAPS ("audio/x-raw, "
"format = (string) S24BE, "
"layout = (string) interleaved, "
"rate = (int) [ 1, MAX ], " "channels = (int) [ 1, MAX ]")
);
static GstStaticPadTemplate gst_rtp_L24_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) [ 1, MAX ], "
"encoding-name = (string) \"L24\"")
);
#define gst_rtp_L24_depay_parent_class parent_class
G_DEFINE_TYPE (GstRtpL24Depay, gst_rtp_L24_depay, GST_TYPE_RTP_BASE_DEPAYLOAD);
static gboolean gst_rtp_L24_depay_setcaps (GstRTPBaseDepayload * depayload,
GstCaps * caps);
static GstBuffer *gst_rtp_L24_depay_process (GstRTPBaseDepayload * depayload,
GstRTPBuffer * rtp);
static void
gst_rtp_L24_depay_class_init (GstRtpL24DepayClass * klass)
{
GstElementClass *gstelement_class;
GstRTPBaseDepayloadClass *gstrtpbasedepayload_class;
gstelement_class = (GstElementClass *) klass;
gstrtpbasedepayload_class = (GstRTPBaseDepayloadClass *) klass;
gstrtpbasedepayload_class->set_caps = gst_rtp_L24_depay_setcaps;
gstrtpbasedepayload_class->process_rtp_packet = gst_rtp_L24_depay_process;
gst_element_class_add_static_pad_template (gstelement_class,
&gst_rtp_L24_depay_src_template);
gst_element_class_add_static_pad_template (gstelement_class,
&gst_rtp_L24_depay_sink_template);
gst_element_class_set_static_metadata (gstelement_class,
"RTP audio depayloader", "Codec/Depayloader/Network/RTP",
"Extracts raw 24-bit audio from RTP packets",
"Zeeshan Ali <zak147@yahoo.com>," "Wim Taymans <wim.taymans@gmail.com>,"
"David Holroyd <dave@badgers-in-foil.co.uk>");
GST_DEBUG_CATEGORY_INIT (rtpL24depay_debug, "rtpL24depay", 0,
"Raw Audio RTP Depayloader");
}
static void
gst_rtp_L24_depay_init (GstRtpL24Depay * rtpL24depay)
{
}
static gint
gst_rtp_L24_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_L24_depay_setcaps (GstRTPBaseDepayload * depayload, GstCaps * caps)
{
GstStructure *structure;
GstRtpL24Depay *rtpL24depay;
gint clock_rate, payload;
gint channels;
GstCaps *srccaps;
gboolean res;
const gchar *channel_order;
const GstRTPChannelOrder *order;
GstAudioInfo *info;
rtpL24depay = GST_RTP_L24_DEPAY (depayload);
structure = gst_caps_get_structure (caps, 0);
payload = 96;
gst_structure_get_int (structure, "payload", &payload);
/* no fixed mapping, we need clock-rate */
channels = 0;
clock_rate = 0;
/* caps can overwrite defaults */
clock_rate =
gst_rtp_L24_depay_parse_int (structure, "clock-rate", clock_rate);
if (clock_rate == 0)
goto no_clockrate;
channels =
gst_rtp_L24_depay_parse_int (structure, "encoding-params", channels);
if (channels == 0) {
channels = gst_rtp_L24_depay_parse_int (structure, "channels", channels);
if (channels == 0) {
/* channels defaults to 1 otherwise */
channels = 1;
}
}
depayload->clock_rate = clock_rate;
info = &rtpL24depay->info;
gst_audio_info_init (info);
info->finfo = gst_audio_format_get_info (GST_AUDIO_FORMAT_S24BE);
info->rate = clock_rate;
info->channels = channels;
info->bpf = (info->finfo->width / 8) * channels;
/* add channel positions */
channel_order = gst_structure_get_string (structure, "channel-order");
order = gst_rtp_channels_get_by_order (channels, channel_order);
rtpL24depay->order = order;
if (order) {
memcpy (info->position, order->pos,
sizeof (GstAudioChannelPosition) * channels);
gst_audio_channel_positions_to_valid_order (info->position, info->channels);
} else {
GST_ELEMENT_WARNING (rtpL24depay, STREAM, DECODE,
(NULL), ("Unknown channel order '%s' for %d channels",
GST_STR_NULL (channel_order), channels));
/* create default NONE layout */
gst_rtp_channels_create_default (channels, info->position);
}
srccaps = gst_audio_info_to_caps (info);
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_L24_depay_process (GstRTPBaseDepayload * depayload, GstRTPBuffer * rtp)
{
GstRtpL24Depay *rtpL24depay;
GstBuffer *outbuf;
gint payload_len;
gboolean marker;
rtpL24depay = GST_RTP_L24_DEPAY (depayload);
payload_len = gst_rtp_buffer_get_payload_len (rtp);
if (payload_len <= 0)
goto empty_packet;
GST_DEBUG_OBJECT (rtpL24depay, "got payload of %d bytes", payload_len);
outbuf = gst_rtp_buffer_get_payload_buffer (rtp);
marker = gst_rtp_buffer_get_marker (rtp);
if (marker) {
/* mark talk spurt with RESYNC */
GST_BUFFER_FLAG_SET (outbuf, GST_BUFFER_FLAG_RESYNC);
}
outbuf = gst_buffer_make_writable (outbuf);
if (outbuf) {
gst_rtp_drop_meta (GST_ELEMENT_CAST (rtpL24depay), outbuf,
g_quark_from_static_string (GST_META_TAG_AUDIO_STR));
}
if (rtpL24depay->order &&
!gst_audio_buffer_reorder_channels (outbuf,
rtpL24depay->info.finfo->format, rtpL24depay->info.channels,
rtpL24depay->info.position, rtpL24depay->order->pos)) {
goto reorder_failed;
}
return outbuf;
/* ERRORS */
empty_packet:
{
GST_ELEMENT_WARNING (rtpL24depay, STREAM, DECODE,
("Empty Payload."), (NULL));
return NULL;
}
reorder_failed:
{
GST_ELEMENT_ERROR (rtpL24depay, STREAM, DECODE,
("Channel reordering failed."), (NULL));
return NULL;
}
}
gboolean
gst_rtp_L24_depay_plugin_init (GstPlugin * plugin)
{
return gst_element_register (plugin, "rtpL24depay",
GST_RANK_SECONDARY, GST_TYPE_RTP_L24_DEPAY);
}