gstreamer/subprojects/gst-plugins-good/gst/rtp/gstrtpbvpay.c

237 lines
6.9 KiB
C
Raw Normal View History

2009-11-11 10:37:07 +00:00
/* GStreamer
* Copyright (C) <2009> 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.
2009-11-11 10:37:07 +00:00
*/
/**
* SECTION:element-rtpbvpay
* @title: rtpbvpay
* @see_also: rtpbvdepay
*
* Payload BroadcomVoice audio into RTP packets according to RFC 4298.
* For detailed information see: http://www.rfc-editor.org/rfc/rfc4298.txt
*/
2009-11-11 10:37:07 +00:00
#ifdef HAVE_CONFIG_H
# include "config.h"
#endif
#include <stdlib.h>
#include <string.h>
#include <gst/rtp/gstrtpbuffer.h>
#include "gstrtpelements.h"
2009-11-11 10:37:07 +00:00
#include "gstrtpbvpay.h"
GST_DEBUG_CATEGORY_STATIC (rtpbvpay_debug);
#define GST_CAT_DEFAULT (rtpbvpay_debug)
static GstStaticPadTemplate gst_rtp_bv_pay_sink_template =
2009-11-11 10:37:07 +00:00
GST_STATIC_PAD_TEMPLATE ("sink",
GST_PAD_SINK,
GST_PAD_ALWAYS,
GST_STATIC_CAPS ("audio/x-bv, " "mode = (int) {16, 32}")
);
static GstStaticPadTemplate gst_rtp_bv_pay_src_template =
2009-11-11 10:37:07 +00:00
GST_STATIC_PAD_TEMPLATE ("src",
GST_PAD_SRC,
GST_PAD_ALWAYS,
GST_STATIC_CAPS ("application/x-rtp, "
"media = (string) \"audio\", "
"payload = (int) " GST_RTP_PAYLOAD_DYNAMIC_STRING ", "
"clock-rate = (int) 8000, "
"encoding-name = (string) \"BV16\";"
"application/x-rtp, "
"media = (string) \"audio\", "
"payload = (int) " GST_RTP_PAYLOAD_DYNAMIC_STRING ", "
"clock-rate = (int) 16000, " "encoding-name = (string) \"BV32\"")
);
2011-11-11 11:25:01 +00:00
static GstCaps *gst_rtp_bv_pay_sink_getcaps (GstRTPBasePayload * payload,
GstPad * pad, GstCaps * filter);
2011-11-11 11:25:01 +00:00
static gboolean gst_rtp_bv_pay_sink_setcaps (GstRTPBasePayload * payload,
2009-11-11 10:37:07 +00:00
GstCaps * caps);
2011-04-25 11:16:58 +00:00
#define gst_rtp_bv_pay_parent_class parent_class
2011-11-11 11:25:01 +00:00
G_DEFINE_TYPE (GstRTPBVPay, gst_rtp_bv_pay, GST_TYPE_RTP_BASE_AUDIO_PAYLOAD);
GST_ELEMENT_REGISTER_DEFINE_WITH_CODE (rtpbvpay, "rtpbvpay", GST_RANK_SECONDARY,
GST_TYPE_RTP_BV_PAY, rtp_element_init (plugin));
2009-11-11 10:37:07 +00:00
static void
2011-04-25 11:16:58 +00:00
gst_rtp_bv_pay_class_init (GstRTPBVPayClass * klass)
2009-11-11 10:37:07 +00:00
{
2011-04-25 11:16:58 +00:00
GstElementClass *gstelement_class;
2011-11-11 11:25:01 +00:00
GstRTPBasePayloadClass *gstrtpbasepayload_class;
2011-04-25 11:16:58 +00:00
GST_DEBUG_CATEGORY_INIT (rtpbvpay_debug, "rtpbvpay", 0,
"BroadcomVoice audio RTP payloader");
2009-11-11 10:37:07 +00:00
2011-04-25 11:16:58 +00:00
gstelement_class = (GstElementClass *) klass;
2011-11-11 11:25:01 +00:00
gstrtpbasepayload_class = (GstRTPBasePayloadClass *) klass;
2011-04-25 11:16:58 +00:00
gst_element_class_add_static_pad_template (gstelement_class,
&gst_rtp_bv_pay_sink_template);
gst_element_class_add_static_pad_template (gstelement_class,
&gst_rtp_bv_pay_src_template);
2011-04-25 11:16:58 +00:00
gst_element_class_set_static_metadata (gstelement_class, "RTP BV Payloader",
2010-12-21 15:58:47 +00:00
"Codec/Payloader/Network/RTP",
"Packetize BroadcomVoice audio streams into RTP packets (RFC 4298)",
"Wim Taymans <wim.taymans@collabora.co.uk>");
2009-11-11 10:37:07 +00:00
2011-11-11 11:25:01 +00:00
gstrtpbasepayload_class->set_caps = gst_rtp_bv_pay_sink_setcaps;
gstrtpbasepayload_class->get_caps = gst_rtp_bv_pay_sink_getcaps;
2009-11-11 10:37:07 +00:00
}
static void
2011-04-25 11:16:58 +00:00
gst_rtp_bv_pay_init (GstRTPBVPay * rtpbvpay)
2009-11-11 10:37:07 +00:00
{
2011-11-11 11:25:01 +00:00
GstRTPBaseAudioPayload *rtpbaseaudiopayload;
2009-11-11 10:37:07 +00:00
2011-11-11 11:25:01 +00:00
rtpbaseaudiopayload = GST_RTP_BASE_AUDIO_PAYLOAD (rtpbvpay);
2009-11-11 10:37:07 +00:00
rtpbvpay->mode = -1;
2011-11-11 11:25:01 +00:00
/* tell rtpbaseaudiopayload that this is a frame based codec */
gst_rtp_base_audio_payload_set_frame_based (rtpbaseaudiopayload);
2009-11-11 10:37:07 +00:00
}
static gboolean
2011-11-11 11:25:01 +00:00
gst_rtp_bv_pay_sink_setcaps (GstRTPBasePayload * rtpbasepayload, GstCaps * caps)
2009-11-11 10:37:07 +00:00
{
GstRTPBVPay *rtpbvpay;
2011-11-11 11:25:01 +00:00
GstRTPBaseAudioPayload *rtpbaseaudiopayload;
2009-11-11 10:37:07 +00:00
gint mode;
GstStructure *structure;
const char *payload_name;
2011-11-11 11:25:01 +00:00
rtpbvpay = GST_RTP_BV_PAY (rtpbasepayload);
rtpbaseaudiopayload = GST_RTP_BASE_AUDIO_PAYLOAD (rtpbasepayload);
2009-11-11 10:37:07 +00:00
structure = gst_caps_get_structure (caps, 0);
payload_name = gst_structure_get_name (structure);
if (g_ascii_strcasecmp ("audio/x-bv", payload_name))
goto wrong_caps;
if (!gst_structure_get_int (structure, "mode", &mode))
goto no_mode;
if (mode != 16 && mode != 32)
goto wrong_mode;
if (mode == 16) {
2011-11-11 11:25:01 +00:00
gst_rtp_base_payload_set_options (rtpbasepayload, "audio", TRUE, "BV16",
2009-11-11 10:37:07 +00:00
8000);
2011-11-11 11:25:01 +00:00
rtpbasepayload->clock_rate = 8000;
2009-11-11 10:37:07 +00:00
} else {
2011-11-11 11:25:01 +00:00
gst_rtp_base_payload_set_options (rtpbasepayload, "audio", TRUE, "BV32",
2009-11-11 10:37:07 +00:00
16000);
2011-11-11 11:25:01 +00:00
rtpbasepayload->clock_rate = 16000;
2009-11-11 10:37:07 +00:00
}
/* set options for this frame based audio codec */
2011-11-11 11:25:01 +00:00
gst_rtp_base_audio_payload_set_frame_options (rtpbaseaudiopayload,
2009-11-11 10:37:07 +00:00
mode, mode == 16 ? 10 : 20);
if (mode != rtpbvpay->mode && rtpbvpay->mode != -1)
goto mode_changed;
rtpbvpay->mode = mode;
return TRUE;
/* ERRORS */
wrong_caps:
{
GST_ERROR_OBJECT (rtpbvpay, "expected audio/x-bv, received %s",
payload_name);
return FALSE;
}
no_mode:
{
GST_ERROR_OBJECT (rtpbvpay, "did not receive a mode");
return FALSE;
}
wrong_mode:
{
GST_ERROR_OBJECT (rtpbvpay, "mode must be 16 or 32, received %d", mode);
return FALSE;
}
mode_changed:
{
GST_ERROR_OBJECT (rtpbvpay, "Mode has changed from %d to %d! "
"Mode cannot change while streaming", rtpbvpay->mode, mode);
return FALSE;
}
}
/* we return the padtemplate caps with the mode field fixated to a value if we
* can */
static GstCaps *
2011-11-11 11:25:01 +00:00
gst_rtp_bv_pay_sink_getcaps (GstRTPBasePayload * rtppayload, GstPad * pad,
GstCaps * filter)
2009-11-11 10:37:07 +00:00
{
GstCaps *otherpadcaps;
GstCaps *caps;
caps = gst_pad_get_pad_template_caps (pad);
2009-11-11 10:37:07 +00:00
otherpadcaps = gst_pad_get_allowed_caps (rtppayload->srcpad);
2009-11-11 10:37:07 +00:00
if (otherpadcaps) {
if (!gst_caps_is_empty (otherpadcaps)) {
GstStructure *structure;
const gchar *mode_str;
gint mode;
structure = gst_caps_get_structure (otherpadcaps, 0);
/* construct mode, if we can */
mode_str = gst_structure_get_string (structure, "encoding-name");
if (mode_str) {
if (!strcmp (mode_str, "BV16"))
mode = 16;
else if (!strcmp (mode_str, "BV32"))
mode = 32;
else
mode = -1;
if (mode == 16 || mode == 32) {
caps = gst_caps_make_writable (caps);
2009-11-11 10:37:07 +00:00
structure = gst_caps_get_structure (caps, 0);
gst_structure_set (structure, "mode", G_TYPE_INT, mode, NULL);
}
}
}
gst_caps_unref (otherpadcaps);
}
if (filter) {
GstCaps *tmp;
GST_DEBUG_OBJECT (rtppayload, "Intersect %" GST_PTR_FORMAT " and filter %"
GST_PTR_FORMAT, caps, filter);
tmp = gst_caps_intersect_full (filter, caps, GST_CAPS_INTERSECT_FIRST);
gst_caps_unref (caps);
caps = tmp;
}
2009-11-11 10:37:07 +00:00
return caps;
}