blob: 6b7bb18b5b55ca8ed8c75df1d04b0ab17bf9c6d1 [file] [log] [blame]
/* 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.
*/
/**
* SECTION:element-rtpbvdepay
* @see_also: rtpbvpay
*
* Extract BroadcomVoice audio from RTP packets according to RFC 4298.
* For detailed information see: http://www.rfc-editor.org/rfc/rfc4298.txt
*/
#ifdef HAVE_CONFIG_H
# include "config.h"
#endif
#include <string.h>
#include <stdlib.h>
#include <gst/rtp/gstrtpbuffer.h>
#include <gst/audio/audio.h>
#include "gstrtpbvdepay.h"
#include "gstrtputils.h"
static GstStaticPadTemplate gst_rtp_bv_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) 8000, "
"encoding-name = (string) \"BV16\"; "
"application/x-rtp, "
"media = (string) \"audio\", "
"clock-rate = (int) 16000, " "encoding-name = (string) \"BV32\"")
);
static GstStaticPadTemplate gst_rtp_bv_depay_src_template =
GST_STATIC_PAD_TEMPLATE ("src",
GST_PAD_SRC,
GST_PAD_ALWAYS,
GST_STATIC_CAPS ("audio/x-bv, " "mode = (int) { 16, 32 }")
);
static GstBuffer *gst_rtp_bv_depay_process (GstRTPBaseDepayload * depayload,
GstRTPBuffer * rtp);
static gboolean gst_rtp_bv_depay_setcaps (GstRTPBaseDepayload * depayload,
GstCaps * caps);
#define gst_rtp_bv_depay_parent_class parent_class
G_DEFINE_TYPE (GstRTPBVDepay, gst_rtp_bv_depay, GST_TYPE_RTP_BASE_DEPAYLOAD);
static void
gst_rtp_bv_depay_class_init (GstRTPBVDepayClass * klass)
{
GstElementClass *gstelement_class;
GstRTPBaseDepayloadClass *gstrtpbasedepayload_class;
gstelement_class = (GstElementClass *) klass;
gstrtpbasedepayload_class = (GstRTPBaseDepayloadClass *) klass;
gst_element_class_add_pad_template (gstelement_class,
gst_static_pad_template_get (&gst_rtp_bv_depay_src_template));
gst_element_class_add_pad_template (gstelement_class,
gst_static_pad_template_get (&gst_rtp_bv_depay_sink_template));
gst_element_class_set_static_metadata (gstelement_class,
"RTP BroadcomVoice depayloader", "Codec/Depayloader/Network/RTP",
"Extracts BroadcomVoice audio from RTP packets (RFC 4298)",
"Wim Taymans <wim.taymans@collabora.co.uk>");
gstrtpbasedepayload_class->process_rtp_packet = gst_rtp_bv_depay_process;
gstrtpbasedepayload_class->set_caps = gst_rtp_bv_depay_setcaps;
}
static void
gst_rtp_bv_depay_init (GstRTPBVDepay * rtpbvdepay)
{
rtpbvdepay->mode = -1;
}
static gboolean
gst_rtp_bv_depay_setcaps (GstRTPBaseDepayload * depayload, GstCaps * caps)
{
GstRTPBVDepay *rtpbvdepay = GST_RTP_BV_DEPAY (depayload);
GstCaps *srccaps;
GstStructure *structure;
const gchar *mode_str = NULL;
gint mode, clock_rate, expected_rate;
gboolean ret;
structure = gst_caps_get_structure (caps, 0);
mode_str = gst_structure_get_string (structure, "encoding-name");
if (!mode_str)
goto no_mode;
if (!strcmp (mode_str, "BV16")) {
mode = 16;
expected_rate = 8000;
} else if (!strcmp (mode_str, "BV32")) {
mode = 32;
expected_rate = 16000;
} else
goto invalid_mode;
if (!gst_structure_get_int (structure, "clock-rate", &clock_rate))
clock_rate = expected_rate;
else if (clock_rate != expected_rate)
goto wrong_rate;
depayload->clock_rate = clock_rate;
rtpbvdepay->mode = mode;
srccaps = gst_caps_new_simple ("audio/x-bv",
"mode", G_TYPE_INT, rtpbvdepay->mode, NULL);
ret = gst_pad_set_caps (GST_RTP_BASE_DEPAYLOAD_SRCPAD (depayload), srccaps);
GST_DEBUG ("set caps on source: %" GST_PTR_FORMAT " (ret=%d)", srccaps, ret);
gst_caps_unref (srccaps);
return ret;
/* ERRORS */
no_mode:
{
GST_ERROR_OBJECT (rtpbvdepay, "did not receive an encoding-name");
return FALSE;
}
invalid_mode:
{
GST_ERROR_OBJECT (rtpbvdepay,
"invalid encoding-name, expected BV16 or BV32, got %s", mode_str);
return FALSE;
}
wrong_rate:
{
GST_ERROR_OBJECT (rtpbvdepay, "invalid clock-rate, expected %d, got %d",
expected_rate, clock_rate);
return FALSE;
}
}
static GstBuffer *
gst_rtp_bv_depay_process (GstRTPBaseDepayload * depayload, GstRTPBuffer * rtp)
{
GstBuffer *outbuf;
gboolean marker;
marker = gst_rtp_buffer_get_marker (rtp);
GST_DEBUG ("process : got %" G_GSIZE_FORMAT " bytes, mark %d ts %u seqn %d",
gst_buffer_get_size (rtp->buffer), marker,
gst_rtp_buffer_get_timestamp (rtp), gst_rtp_buffer_get_seq (rtp));
outbuf = gst_rtp_buffer_get_payload_buffer (rtp);
if (marker && outbuf) {
/* mark start of talkspurt with RESYNC */
GST_BUFFER_FLAG_SET (outbuf, GST_BUFFER_FLAG_RESYNC);
}
if (outbuf) {
gst_rtp_drop_meta (GST_ELEMENT_CAST (depayload), outbuf,
g_quark_from_static_string (GST_META_TAG_AUDIO_STR));
}
return outbuf;
}
gboolean
gst_rtp_bv_depay_plugin_init (GstPlugin * plugin)
{
return gst_element_register (plugin, "rtpbvdepay",
GST_RANK_SECONDARY, GST_TYPE_RTP_BV_DEPAY);
}