gstreamer/gst/rtp/gstrtpvorbisdepay.c
Wim Taymans af6e4da92e gst/rtp/: Fix klass typos.
Original commit message from CVS:
* gst/rtp/gstasteriskh263.c: (gst_asteriskh263_plugin_init):
* gst/rtp/gstrtpL16depay.c: (gst_rtp_L16depay_plugin_init):
* gst/rtp/gstrtpamrdepay.c: (gst_rtp_amr_depay_plugin_init):
* gst/rtp/gstrtpdepay.c:
* gst/rtp/gstrtpgsmdepay.c: (gst_rtp_gsm_depay_plugin_init):
* gst/rtp/gstrtph263pdepay.c: (gst_rtp_h263p_depay_plugin_init):
* gst/rtp/gstrtph264depay.c: (gst_rtp_h264_depay_plugin_init):
* gst/rtp/gstrtpilbcdepay.c: (gst_rtp_ilbc_depay_plugin_init):
* gst/rtp/gstrtpmp2tdepay.c: (gst_rtp_mp2t_depay_setcaps),
(gst_rtp_mp2t_depay_plugin_init):
* gst/rtp/gstrtpmp4gdepay.c: (gst_rtp_mp4g_depay_plugin_init):
* gst/rtp/gstrtpmp4vdepay.c: (gst_rtp_mp4v_depay_plugin_init):
* gst/rtp/gstrtpmpadepay.c: (gst_rtp_mpa_depay_plugin_init):
* gst/rtp/gstrtppcmadepay.c: (gst_rtp_pcma_depay_plugin_init):
* gst/rtp/gstrtppcmudepay.c: (gst_rtp_pcmu_depay_plugin_init):
* gst/rtp/gstrtpspeexdepay.c: (gst_rtp_speex_depay_plugin_init):
* gst/rtp/gstrtpsv3vdepay.c: (gst_rtp_sv3v_depay_plugin_init):
* gst/rtp/gstrtpvorbisdepay.c: (gst_rtp_vorbis_depay_plugin_init):
Fix klass typos.
Mark RANK_MARGINAL, decodebin can handle the depayloaders fine.
2006-09-23 15:30:40 +00:00

441 lines
12 KiB
C

/* GStreamer
* Copyright (C) <2006> Wim Taymans <wim@fluendo.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 <gst/rtp/gstrtpbuffer.h>
#include <string.h>
#include "gstrtpvorbisdepay.h"
GST_DEBUG_CATEGORY_STATIC (rtpvorbisdepay_debug);
#define GST_CAT_DEFAULT (rtpvorbisdepay_debug)
/* elementfactory information */
static const GstElementDetails gst_rtp_vorbis_depay_details =
GST_ELEMENT_DETAILS ("RTP packet parser",
"Codec/Depayloader/Network",
"Extracts Vorbis Audio from RTP packets (draft-01 of RFC XXXX)",
"Wim Taymans <wim@fluendo.com>");
/* RtpVorbisDepay signals and args */
enum
{
/* FILL ME */
LAST_SIGNAL
};
enum
{
ARG_0,
};
static GstStaticPadTemplate gst_rtp_vorbis_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) [1, MAX ], " "encoding-name = (string) \"vorbis\""
/* All required parameters
*
* "encoding-params = (string) <num channels>"
* "delivery-method = (string) { inline, in_band, out_band/<specific_name> } "
* "configuration = (string) ANY"
*/
/* All optional parameters
*
* "configuration-uri ="
*/
)
);
static GstStaticPadTemplate gst_rtp_vorbis_depay_src_template =
GST_STATIC_PAD_TEMPLATE ("src",
GST_PAD_SRC,
GST_PAD_ALWAYS,
GST_STATIC_CAPS ("audio/x-vorbis")
);
GST_BOILERPLATE (GstRtpVorbisDepay, gst_rtp_vorbis_depay, GstBaseRTPDepayload,
GST_TYPE_BASE_RTP_DEPAYLOAD);
static gboolean gst_rtp_vorbis_depay_setcaps (GstBaseRTPDepayload * depayload,
GstCaps * caps);
static GstBuffer *gst_rtp_vorbis_depay_process (GstBaseRTPDepayload * depayload,
GstBuffer * buf);
static void gst_rtp_vorbis_depay_set_property (GObject * object, guint prop_id,
const GValue * value, GParamSpec * pspec);
static void gst_rtp_vorbis_depay_get_property (GObject * object, guint prop_id,
GValue * value, GParamSpec * pspec);
static void gst_rtp_vorbis_depay_finalize (GObject * object);
static GstStateChangeReturn gst_rtp_vorbis_depay_change_state (GstElement *
element, GstStateChange transition);
static void
gst_rtp_vorbis_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_vorbis_depay_sink_template));
gst_element_class_add_pad_template (element_class,
gst_static_pad_template_get (&gst_rtp_vorbis_depay_src_template));
gst_element_class_set_details (element_class, &gst_rtp_vorbis_depay_details);
}
static void
gst_rtp_vorbis_depay_class_init (GstRtpVorbisDepayClass * klass)
{
GObjectClass *gobject_class;
GstElementClass *gstelement_class;
GstBaseRTPDepayloadClass *gstbasertpdepayload_class;
gobject_class = (GObjectClass *) klass;
gstelement_class = (GstElementClass *) klass;
gstbasertpdepayload_class = (GstBaseRTPDepayloadClass *) klass;
gobject_class->set_property = gst_rtp_vorbis_depay_set_property;
gobject_class->get_property = gst_rtp_vorbis_depay_get_property;
gobject_class->finalize = gst_rtp_vorbis_depay_finalize;
gstelement_class->change_state = gst_rtp_vorbis_depay_change_state;
gstbasertpdepayload_class->process = gst_rtp_vorbis_depay_process;
gstbasertpdepayload_class->set_caps = gst_rtp_vorbis_depay_setcaps;
GST_DEBUG_CATEGORY_INIT (rtpvorbisdepay_debug, "rtpvorbisdepay", 0,
"Vorbis RTP Depayloader");
}
static void
gst_rtp_vorbis_depay_init (GstRtpVorbisDepay * rtpvorbisdepay,
GstRtpVorbisDepayClass * klass)
{
rtpvorbisdepay->adapter = gst_adapter_new ();
}
static void
gst_rtp_vorbis_depay_finalize (GObject * object)
{
GstRtpVorbisDepay *rtpvorbisdepay = GST_RTP_VORBIS_DEPAY (object);
g_object_unref (rtpvorbisdepay->adapter);
G_OBJECT_CLASS (parent_class)->finalize (object);
}
static gboolean
gst_rtp_vorbis_depay_setcaps (GstBaseRTPDepayload * depayload, GstCaps * caps)
{
GstStructure *structure;
GstRtpVorbisDepay *rtpvorbisdepay;
GstCaps *srccaps;
gint clock_rate;
rtpvorbisdepay = GST_RTP_VORBIS_DEPAY (depayload);
structure = gst_caps_get_structure (caps, 0);
if (!gst_structure_get_int (structure, "clock-rate", &clock_rate))
goto no_rate;
/* caps seem good, configure element */
depayload->clock_rate = clock_rate;
/* set caps on pad and on header */
srccaps = gst_caps_new_simple ("audio/x-vorbis", NULL);
gst_pad_set_caps (depayload->srcpad, srccaps);
gst_caps_unref (srccaps);
return TRUE;
no_rate:
{
GST_ERROR_OBJECT (rtpvorbisdepay, "no clock-rate specified");
return FALSE;
}
}
static GstBuffer *
gst_rtp_vorbis_depay_process (GstBaseRTPDepayload * depayload, GstBuffer * buf)
{
GstRtpVorbisDepay *rtpvorbisdepay;
GstBuffer *outbuf;
GstFlowReturn ret;
gint payload_len;
guint8 *payload, *to_free = NULL;
guint32 timestamp;
guint32 header, ident;
guint8 F, VDT, packets;
gboolean free_payload;
rtpvorbisdepay = GST_RTP_VORBIS_DEPAY (depayload);
if (!gst_rtp_buffer_validate (buf))
goto bad_packet;
payload_len = gst_rtp_buffer_get_payload_len (buf);
GST_DEBUG_OBJECT (depayload, "got RTP packet of size %d", payload_len);
/* we need at least 4 bytes for the packet header */
if (G_UNLIKELY (payload_len < 4))
goto packet_short;
payload = gst_rtp_buffer_get_payload (buf);
free_payload = FALSE;
header = GST_READ_UINT32_BE (payload);
/*
* 0 1 2 3
* 0 1 2 3 4 5 6 7 8 9 0 1 2 3 4 5 6 7 8 9 0 1 2 3 4 5 6 7 8 9 0 1
* +-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+
* | Ident | F |VDT|# pkts.|
* +-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+
*
* F: Fragment type (0=none, 1=start, 2=cont, 3=end)
* VDT: Vorbis data type (0=vorbis, 1=config, 2=comment, 3=reserved)
* pkts: number of packets.
*/
VDT = (header & 0x30) >> 4;
if (G_UNLIKELY (VDT == 3))
goto ignore_reserved;
ident = (header >> 8) & 0xffffff;
F = (header & 0xc0) >> 6;
packets = (header & 0xf);
if (VDT == 0) {
/* FIXME, if we have a raw payload, we need the codebook for the ident */
}
/* skip header */
payload += 4;
payload_len -= 4;
GST_DEBUG_OBJECT (depayload, "ident: %u, F: %d, VDT: %d, packets: %d", ident,
F, VDT, packets);
/* fragmented packets, assemble */
if (F != 0) {
GstBuffer *vdata;
guint headerskip;
if (F == 1) {
/* if we start a packet, clear adapter and start assembling. */
gst_adapter_clear (rtpvorbisdepay->adapter);
GST_DEBUG_OBJECT (depayload, "start assemble");
rtpvorbisdepay->assembling = TRUE;
}
if (!rtpvorbisdepay->assembling)
goto no_output;
/* first assembled packet, reuse 2 bytes to store the length */
headerskip = (F == 1 ? 4 : 6);
/* skip header and length. */
vdata = gst_rtp_buffer_get_payload_subbuffer (buf, headerskip, -1);
GST_DEBUG_OBJECT (depayload, "assemble vorbis packet");
gst_adapter_push (rtpvorbisdepay->adapter, vdata);
/* packet is not complete, we are done */
if (F != 3)
goto no_output;
/* construct assembled buffer */
payload_len = gst_adapter_available (rtpvorbisdepay->adapter);
payload = gst_adapter_take (rtpvorbisdepay->adapter, payload_len);
/* fix the length */
payload[0] = ((payload_len - 2) >> 8) & 0xff;
payload[1] = (payload_len - 2) & 0xff;
to_free = payload;
}
GST_DEBUG_OBJECT (depayload, "assemble done");
/* we not assembling anymore now */
rtpvorbisdepay->assembling = FALSE;
gst_adapter_clear (rtpvorbisdepay->adapter);
/* payload now points to a length with that many vorbis data bytes.
* Iterate over the packets and send them out.
*
* 0 1 2 3
* 0 1 2 3 4 5 6 7 8 9 0 1 2 3 4 5 6 7 8 9 0 1 2 3 4 5 6 7 8 9 0 1
* +-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+
* | length | vorbis data ..
* +-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+
* .. vorbis data |
* +-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+
* | length | next vorbis packet data ..
* +-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+
* .. vorbis data |
* +-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+*
*/
timestamp = gst_rtp_buffer_get_timestamp (buf);
while (payload_len > 2) {
guint16 length;
length = GST_READ_UINT16_BE (payload);
payload += 2;
payload_len -= 2;
GST_DEBUG_OBJECT (depayload, "read length %u, avail: %d", length,
payload_len);
/* skip packet if something odd happens */
if (G_UNLIKELY (length > payload_len))
goto length_short;
/* create buffer for packet */
if (G_UNLIKELY (to_free)) {
outbuf = gst_buffer_new ();
GST_BUFFER_DATA (outbuf) = payload;
GST_BUFFER_MALLOCDATA (outbuf) = to_free;
GST_BUFFER_SIZE (outbuf) = length;
to_free = NULL;
} else {
outbuf = gst_buffer_new_and_alloc (length);
memcpy (GST_BUFFER_DATA (outbuf), payload, length);
}
payload += length;
payload_len -= length;
if (timestamp != -1)
/* push with timestamp of the last packet, which is the same timestamp that
* should apply to the first assembled packet. */
ret = gst_base_rtp_depayload_push_ts (depayload, timestamp, outbuf);
else
ret = gst_base_rtp_depayload_push (depayload, outbuf);
if (ret != GST_FLOW_OK)
break;
/* make sure we don't set a timestamp on next buffers */
timestamp = -1;
}
g_free (to_free);
return NULL;
no_output:
{
return NULL;
}
/* ERORRS */
bad_packet:
{
GST_ELEMENT_WARNING (rtpvorbisdepay, STREAM, DECODE,
("Packet did not validate"), (NULL));
return NULL;
}
packet_short:
{
GST_ELEMENT_WARNING (rtpvorbisdepay, STREAM, DECODE,
("Packet was too short (%d < 4)", payload_len), (NULL));
return NULL;
}
ignore_reserved:
{
GST_WARNING_OBJECT (rtpvorbisdepay, "reserved VDT ignored");
return NULL;
}
length_short:
{
GST_ELEMENT_WARNING (rtpvorbisdepay, STREAM, DECODE,
("Packet contains invalid data"), (NULL));
return NULL;
}
}
static void
gst_rtp_vorbis_depay_set_property (GObject * object, guint prop_id,
const GValue * value, GParamSpec * pspec)
{
GstRtpVorbisDepay *rtpvorbisdepay;
rtpvorbisdepay = GST_RTP_VORBIS_DEPAY (object);
switch (prop_id) {
default:
G_OBJECT_WARN_INVALID_PROPERTY_ID (object, prop_id, pspec);
break;
}
}
static void
gst_rtp_vorbis_depay_get_property (GObject * object, guint prop_id,
GValue * value, GParamSpec * pspec)
{
GstRtpVorbisDepay *rtpvorbisdepay;
rtpvorbisdepay = GST_RTP_VORBIS_DEPAY (object);
switch (prop_id) {
default:
G_OBJECT_WARN_INVALID_PROPERTY_ID (object, prop_id, pspec);
break;
}
}
static GstStateChangeReturn
gst_rtp_vorbis_depay_change_state (GstElement * element,
GstStateChange transition)
{
GstRtpVorbisDepay *rtpvorbisdepay;
GstStateChangeReturn ret;
rtpvorbisdepay = GST_RTP_VORBIS_DEPAY (element);
switch (transition) {
case GST_STATE_CHANGE_NULL_TO_READY:
break;
case GST_STATE_CHANGE_READY_TO_PAUSED:
break;
default:
break;
}
ret = GST_ELEMENT_CLASS (parent_class)->change_state (element, transition);
switch (transition) {
case GST_STATE_CHANGE_READY_TO_NULL:
break;
default:
break;
}
return ret;
}
gboolean
gst_rtp_vorbis_depay_plugin_init (GstPlugin * plugin)
{
return gst_element_register (plugin, "rtpvorbisdepay",
GST_RANK_MARGINAL, GST_TYPE_RTP_VORBIS_DEPAY);
}