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/* 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., 59 Temple Place - Suite 330,
* Boston, MA 02111-1307, USA.
*/
#ifdef HAVE_CONFIG_H
# include "config.h"
#endif
#include <stdlib.h>
#include <string.h>
#include <gst/rtp/gstrtpbuffer.h>
#include "gstrtpbvpay.h"
GST_DEBUG_CATEGORY_STATIC (rtpbvpay_debug);
#define GST_CAT_DEFAULT (rtpbvpay_debug)
static GstStaticPadTemplate gst_rtp_bv_pay_sink_template =
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 =
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\"")
);
static GstCaps *gst_rtp_bv_pay_sink_getcaps (GstBaseRTPPayload * payload,
GstPad * pad);
static gboolean gst_rtp_bv_pay_sink_setcaps (GstBaseRTPPayload * payload,
GstCaps * caps);
GST_BOILERPLATE (GstRTPBVPay, gst_rtp_bv_pay, GstBaseRTPAudioPayload,
GST_TYPE_BASE_RTP_AUDIO_PAYLOAD);
static void
gst_rtp_bv_pay_base_init (gpointer klass)
{
GstElementClass *element_class = GST_ELEMENT_CLASS (klass);
gst_element_class_add_static_pad_template (element_class,
&gst_rtp_bv_pay_sink_template);
gst_element_class_add_static_pad_template (element_class,
&gst_rtp_bv_pay_src_template);
gst_element_class_set_details_simple (element_class, "RTP BV Payloader",
"Codec/Payloader/Network/RTP",
"Packetize BroadcomVoice audio streams into RTP packets (RFC 4298)",
"Wim Taymans <wim.taymans@collabora.co.uk>");
}
static void
gst_rtp_bv_pay_class_init (GstRTPBVPayClass * klass)
{
GstBaseRTPPayloadClass *gstbasertppayload_class;
gstbasertppayload_class = (GstBaseRTPPayloadClass *) klass;
gstbasertppayload_class->set_caps = gst_rtp_bv_pay_sink_setcaps;
gstbasertppayload_class->get_caps = gst_rtp_bv_pay_sink_getcaps;
GST_DEBUG_CATEGORY_INIT (rtpbvpay_debug, "rtpbvpay", 0,
"BroadcomVoice audio RTP payloader");
}
static void
gst_rtp_bv_pay_init (GstRTPBVPay * rtpbvpay, GstRTPBVPayClass * klass)
{
GstBaseRTPAudioPayload *basertpaudiopayload;
basertpaudiopayload = GST_BASE_RTP_AUDIO_PAYLOAD (rtpbvpay);
rtpbvpay->mode = -1;
/* tell basertpaudiopayload that this is a frame based codec */
gst_base_rtp_audio_payload_set_frame_based (basertpaudiopayload);
}
static gboolean
gst_rtp_bv_pay_sink_setcaps (GstBaseRTPPayload * basertppayload, GstCaps * caps)
{
GstRTPBVPay *rtpbvpay;
GstBaseRTPAudioPayload *basertpaudiopayload;
gint mode;
GstStructure *structure;
const char *payload_name;
rtpbvpay = GST_RTP_BV_PAY (basertppayload);
basertpaudiopayload = GST_BASE_RTP_AUDIO_PAYLOAD (basertppayload);
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) {
gst_basertppayload_set_options (basertppayload, "audio", TRUE, "BV16",
8000);
basertppayload->clock_rate = 8000;
} else {
gst_basertppayload_set_options (basertppayload, "audio", TRUE, "BV32",
16000);
basertppayload->clock_rate = 16000;
}
/* set options for this frame based audio codec */
gst_base_rtp_audio_payload_set_frame_options (basertpaudiopayload,
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 *
gst_rtp_bv_pay_sink_getcaps (GstBaseRTPPayload * rtppayload, GstPad * pad)
{
GstCaps *otherpadcaps;
GstCaps *caps;
otherpadcaps = gst_pad_get_allowed_caps (rtppayload->srcpad);
caps = gst_caps_copy (gst_pad_get_pad_template_caps (pad));
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) {
structure = gst_caps_get_structure (caps, 0);
gst_structure_set (structure, "mode", G_TYPE_INT, mode, NULL);
}
}
}
gst_caps_unref (otherpadcaps);
}
return caps;
}
gboolean
gst_rtp_bv_pay_plugin_init (GstPlugin * plugin)
{
return gst_element_register (plugin, "rtpbvpay",
GST_RANK_SECONDARY, GST_TYPE_RTP_BV_PAY);
}
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