gstreamer/gst/rtp/gstrtpg722pay.c

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/* GStreamer
* Copyright (C) <2010> 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 <gst/audio/audio.h>
#include <gst/rtp/gstrtpbuffer.h>
#include "gstrtpg722pay.h"
#include "gstrtpchannels.h"
GST_DEBUG_CATEGORY_STATIC (rtpg722pay_debug);
#define GST_CAT_DEFAULT (rtpg722pay_debug)
static GstStaticPadTemplate gst_rtp_g722_pay_sink_template =
GST_STATIC_PAD_TEMPLATE ("sink",
GST_PAD_SINK,
GST_PAD_ALWAYS,
GST_STATIC_CAPS ("audio/G722, " "rate = (int) 16000, " "channels = (int) 1")
);
static GstStaticPadTemplate gst_rtp_g722_pay_src_template =
GST_STATIC_PAD_TEMPLATE ("src",
GST_PAD_SRC,
GST_PAD_ALWAYS,
GST_STATIC_CAPS ("application/x-rtp, "
"media = (string) \"audio\", "
"encoding-name = (string) \"G722\", "
"payload = (int) " GST_RTP_PAYLOAD_G722_STRING ", "
"clock-rate = (int) 8000")
);
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static gboolean gst_rtp_g722_pay_setcaps (GstRTPBasePayload * basepayload,
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GstCaps * caps);
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static GstCaps *gst_rtp_g722_pay_getcaps (GstRTPBasePayload * rtppayload,
GstPad * pad, GstCaps * filter);
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#define gst_rtp_g722_pay_parent_class parent_class
G_DEFINE_TYPE (GstRtpG722Pay, gst_rtp_g722_pay,
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GST_TYPE_RTP_BASE_AUDIO_PAYLOAD);
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static void
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gst_rtp_g722_pay_class_init (GstRtpG722PayClass * klass)
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{
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GstElementClass *gstelement_class;
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GstRTPBasePayloadClass *gstrtpbasepayload_class;
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GST_DEBUG_CATEGORY_INIT (rtpg722pay_debug, "rtpg722pay", 0,
"G722 RTP Payloader");
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gstelement_class = (GstElementClass *) klass;
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gstrtpbasepayload_class = (GstRTPBasePayloadClass *) klass;
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gst_element_class_add_pad_template (gstelement_class,
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gst_static_pad_template_get (&gst_rtp_g722_pay_src_template));
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gst_element_class_add_pad_template (gstelement_class,
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gst_static_pad_template_get (&gst_rtp_g722_pay_sink_template));
gst_element_class_set_static_metadata (gstelement_class,
"RTP audio payloader", "Codec/Payloader/Network/RTP",
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"Payload-encode Raw audio into RTP packets (RFC 3551)",
"Wim Taymans <wim.taymans@gmail.com>");
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gstrtpbasepayload_class->set_caps = gst_rtp_g722_pay_setcaps;
gstrtpbasepayload_class->get_caps = gst_rtp_g722_pay_getcaps;
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}
static void
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gst_rtp_g722_pay_init (GstRtpG722Pay * rtpg722pay)
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{
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GstRTPBaseAudioPayload *rtpbaseaudiopayload;
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rtpbaseaudiopayload = GST_RTP_BASE_AUDIO_PAYLOAD (rtpg722pay);
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/* tell rtpbaseaudiopayload that this is a sample based codec */
gst_rtp_base_audio_payload_set_sample_based (rtpbaseaudiopayload);
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}
static gboolean
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gst_rtp_g722_pay_setcaps (GstRTPBasePayload * basepayload, GstCaps * caps)
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{
GstRtpG722Pay *rtpg722pay;
GstStructure *structure;
gint rate, channels, clock_rate;
gboolean res;
gchar *params;
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#if 0
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GstAudioChannelPosition *pos;
const GstRTPChannelOrder *order;
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#endif
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GstRTPBaseAudioPayload *rtpbaseaudiopayload;
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rtpbaseaudiopayload = GST_RTP_BASE_AUDIO_PAYLOAD (basepayload);
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rtpg722pay = GST_RTP_G722_PAY (basepayload);
structure = gst_caps_get_structure (caps, 0);
/* first parse input caps */
if (!gst_structure_get_int (structure, "rate", &rate))
goto no_rate;
if (!gst_structure_get_int (structure, "channels", &channels))
goto no_channels;
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/* FIXME: Do something with the channel positions */
#if 0
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/* get the channel order */
pos = gst_audio_get_channel_positions (structure);
if (pos)
order = gst_rtp_channels_get_by_pos (channels, pos);
else
order = NULL;
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#endif
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/* Clock rate is always 8000 Hz for G722 according to
* RFC 3551 although the sampling rate is 16000 Hz */
clock_rate = 8000;
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gst_rtp_base_payload_set_options (basepayload, "audio", TRUE, "G722",
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clock_rate);
params = g_strdup_printf ("%d", channels);
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#if 0
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if (!order && channels > 2) {
GST_ELEMENT_WARNING (rtpg722pay, STREAM, DECODE,
(NULL), ("Unknown channel order for %d channels", channels));
}
if (order && order->name) {
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res = gst_rtp_base_payload_set_outcaps (basepayload,
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"encoding-params", G_TYPE_STRING, params, "channels", G_TYPE_INT,
channels, "channel-order", G_TYPE_STRING, order->name, NULL);
} else {
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#endif
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res = gst_rtp_base_payload_set_outcaps (basepayload,
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"encoding-params", G_TYPE_STRING, params, "channels", G_TYPE_INT,
channels, NULL);
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#if 0
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}
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#endif
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g_free (params);
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#if 0
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g_free (pos);
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#endif
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rtpg722pay->rate = rate;
rtpg722pay->channels = channels;
/* bits-per-sample is 4 * channels for G722, but as the RTP clock runs at
* half speed (8 instead of 16 khz), pretend it's 8 bits per sample
* channels. */
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gst_rtp_base_audio_payload_set_samplebits_options (rtpbaseaudiopayload,
8 * rtpg722pay->channels);
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return res;
/* ERRORS */
no_rate:
{
GST_DEBUG_OBJECT (rtpg722pay, "no rate given");
return FALSE;
}
no_channels:
{
GST_DEBUG_OBJECT (rtpg722pay, "no channels given");
return FALSE;
}
}
static GstCaps *
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gst_rtp_g722_pay_getcaps (GstRTPBasePayload * rtppayload, GstPad * pad,
GstCaps * filter)
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{
GstCaps *otherpadcaps;
GstCaps *caps;
otherpadcaps = gst_pad_get_allowed_caps (rtppayload->srcpad);
caps = gst_pad_get_pad_template_caps (pad);
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if (otherpadcaps) {
if (!gst_caps_is_empty (otherpadcaps)) {
caps = gst_caps_make_writable (caps);
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gst_caps_set_simple (caps, "channels", G_TYPE_INT, 1, NULL);
gst_caps_set_simple (caps, "rate", G_TYPE_INT, 16000, NULL);
}
gst_caps_unref (otherpadcaps);
}
return caps;
}
gboolean
gst_rtp_g722_pay_plugin_init (GstPlugin * plugin)
{
return gst_element_register (plugin, "rtpg722pay",
GST_RANK_SECONDARY, GST_TYPE_RTP_G722_PAY);
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}