gstreamer/gst/rtp/gstrtpL16depay.c

264 lines
7.5 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., 59 Temple Place - Suite 330,
* Boston, MA 02111-1307, USA.
*/
#ifdef HAVE_CONFIG_H
#include "config.h"
#endif
#include <string.h>
#include <stdlib.h>
#include <gst/audio/audio.h>
#include <gst/audio/multichannel.h>
#include "gstrtpL16depay.h"
#include "gstrtpchannels.h"
GST_DEBUG_CATEGORY_STATIC (rtpL16depay_debug);
#define GST_CAT_DEFAULT (rtpL16depay_debug)
static GstStaticPadTemplate gst_rtp_L16_depay_src_template =
GST_STATIC_PAD_TEMPLATE ("src",
GST_PAD_SRC,
GST_PAD_ALWAYS,
GST_STATIC_CAPS ("audio/x-raw-int, "
"endianness = (int) BIG_ENDIAN, "
"signed = (boolean) true, "
"width = (int) 16, "
"depth = (int) 16, "
"rate = (int) [ 1, MAX ], " "channels = (int) [ 1, MAX ]")
);
static GstStaticPadTemplate gst_rtp_L16_depay_sink_template =
GST_STATIC_PAD_TEMPLATE ("sink",
GST_PAD_SINK,
GST_PAD_ALWAYS,
GST_STATIC_CAPS ("application/x-rtp, "
"media = (string) \"audio\", "
"payload = (int) " GST_RTP_PAYLOAD_DYNAMIC_STRING ", "
"clock-rate = (int) [ 1, MAX ], "
/* "channels = (int) [1, MAX]" */
/* "emphasis = (string) ANY" */
/* "channel-order = (string) ANY" */
"encoding-name = (string) \"L16\";"
"application/x-rtp, "
"media = (string) \"audio\", "
"payload = (int) { " GST_RTP_PAYLOAD_L16_STEREO_STRING ", "
GST_RTP_PAYLOAD_L16_MONO_STRING " }," "clock-rate = (int) [ 1, MAX ]"
/* "channels = (int) [1, MAX]" */
/* "emphasis = (string) ANY" */
/* "channel-order = (string) ANY" */
)
);
GST_BOILERPLATE (GstRtpL16Depay, gst_rtp_L16_depay, GstBaseRTPDepayload,
GST_TYPE_BASE_RTP_DEPAYLOAD);
static gboolean gst_rtp_L16_depay_setcaps (GstBaseRTPDepayload * depayload,
GstCaps * caps);
static GstBuffer *gst_rtp_L16_depay_process (GstBaseRTPDepayload * depayload,
GstBuffer * buf);
static void
gst_rtp_L16_depay_base_init (gpointer klass)
{
GstElementClass *element_class = GST_ELEMENT_CLASS (klass);
gst_element_class_add_pad_template (element_class,
gst_static_pad_template_get (&gst_rtp_L16_depay_src_template));
gst_element_class_add_pad_template (element_class,
gst_static_pad_template_get (&gst_rtp_L16_depay_sink_template));
gst_element_class_set_details_simple (element_class, "RTP audio depayloader",
"Codec/Depayloader/Network",
"Extracts raw audio from RTP packets",
"Zeeshan Ali <zak147@yahoo.com>," "Wim Taymans <wim.taymans@gmail.com>");
}
static void
gst_rtp_L16_depay_class_init (GstRtpL16DepayClass * klass)
{
GstBaseRTPDepayloadClass *gstbasertpdepayload_class;
gstbasertpdepayload_class = (GstBaseRTPDepayloadClass *) klass;
gstbasertpdepayload_class->set_caps = gst_rtp_L16_depay_setcaps;
gstbasertpdepayload_class->process = gst_rtp_L16_depay_process;
GST_DEBUG_CATEGORY_INIT (rtpL16depay_debug, "rtpL16depay", 0,
"Raw Audio RTP Depayloader");
}
static void
gst_rtp_L16_depay_init (GstRtpL16Depay * rtpL16depay,
GstRtpL16DepayClass * klass)
{
/* needed because of GST_BOILERPLATE */
}
static gint
gst_rtp_L16_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_L16_depay_setcaps (GstBaseRTPDepayload * depayload, GstCaps * caps)
{
GstStructure *structure;
GstRtpL16Depay *rtpL16depay;
gint clock_rate, payload;
gint channels;
GstCaps *srccaps;
gboolean res;
const gchar *channel_order;
const GstRTPChannelOrder *order;
rtpL16depay = GST_RTP_L16_DEPAY (depayload);
structure = gst_caps_get_structure (caps, 0);
payload = 96;
gst_structure_get_int (structure, "payload", &payload);
switch (payload) {
case GST_RTP_PAYLOAD_L16_STEREO:
channels = 2;
clock_rate = 44100;
break;
case GST_RTP_PAYLOAD_L16_MONO:
channels = 1;
clock_rate = 44100;
break;
default:
/* no fixed mapping, we need channels and clock-rate */
channels = 0;
clock_rate = 0;
break;
}
/* caps can overwrite defaults */
clock_rate =
gst_rtp_L16_depay_parse_int (structure, "clock-rate", clock_rate);
if (clock_rate == 0)
goto no_clockrate;
channels = gst_rtp_L16_depay_parse_int (structure, "channels", channels);
if (channels == 0)
goto no_channels;
depayload->clock_rate = clock_rate;
rtpL16depay->rate = clock_rate;
rtpL16depay->channels = channels;
srccaps = gst_caps_new_simple ("audio/x-raw-int",
"endianness", G_TYPE_INT, G_BIG_ENDIAN,
"signed", G_TYPE_BOOLEAN, TRUE,
"width", G_TYPE_INT, 16,
"depth", G_TYPE_INT, 16,
"rate", G_TYPE_INT, clock_rate, "channels", G_TYPE_INT, channels, NULL);
/* 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 (rtpL16depay, 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);
}
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;
}
no_channels:
{
GST_ERROR_OBJECT (depayload, "no channels specified");
return FALSE;
}
}
static GstBuffer *
gst_rtp_L16_depay_process (GstBaseRTPDepayload * depayload, GstBuffer * buf)
{
GstRtpL16Depay *rtpL16depay;
GstBuffer *outbuf;
gint payload_len;
gboolean marker;
rtpL16depay = GST_RTP_L16_DEPAY (depayload);
payload_len = gst_rtp_buffer_get_payload_len (buf);
if (payload_len <= 0)
goto empty_packet;
GST_DEBUG_OBJECT (rtpL16depay, "got payload of %d bytes", payload_len);
outbuf = gst_rtp_buffer_get_payload_buffer (buf);
marker = gst_rtp_buffer_get_marker (buf);
if (marker) {
/* mark talk spurt with DISCONT */
GST_BUFFER_FLAG_SET (outbuf, GST_BUFFER_FLAG_DISCONT);
}
return outbuf;
/* ERRORS */
empty_packet:
{
GST_ELEMENT_WARNING (rtpL16depay, STREAM, DECODE,
("Empty Payload."), (NULL));
return NULL;
}
}
gboolean
gst_rtp_L16_depay_plugin_init (GstPlugin * plugin)
{
return gst_element_register (plugin, "rtpL16depay",
GST_RANK_MARGINAL, GST_TYPE_RTP_L16_DEPAY);
}