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gst/rtp/: Small cleanups.

Browse source 
Original commit message from CVS:
* gst/rtp/gstrtpL16depay.c: (gst_rtp_L16depay_change_state):
* gst/rtp/gstrtpmp4gdepay.c: (gst_rtp_mp4g_depay_class_init):
Small cleanups.
* gst/rtp/Makefile.am:
* gst/rtp/gstrtp.c: (plugin_init):
* gst/rtp/gstrtpvorbisdepay.c: (gst_rtp_vorbis_depay_base_init),
(gst_rtp_vorbis_depay_class_init), (gst_rtp_vorbis_depay_init),
(gst_rtp_vorbis_depay_finalize), (gst_rtp_vorbis_depay_setcaps),
(gst_rtp_vorbis_depay_process),
(gst_rtp_vorbis_depay_set_property),
(gst_rtp_vorbis_depay_get_property),
(gst_rtp_vorbis_depay_change_state),
(gst_rtp_vorbis_depay_plugin_init):
* gst/rtp/gstrtpvorbisdepay.h:
* gst/rtp/gstrtpvorbispay.c: (gst_rtp_vorbis_pay_base_init),
(gst_rtp_vorbis_pay_class_init), (gst_rtp_vorbis_pay_init),
(gst_rtp_vorbis_pay_setcaps), (gst_rtp_vorbis_pay_init_packet),
(gst_rtp_vorbis_pay_flush_packet),
(gst_rtp_vorbis_pay_append_buffer),
(gst_rtp_vorbis_pay_handle_buffer),
(gst_rtp_vorbis_pay_plugin_init):
* gst/rtp/gstrtpvorbispay.h:
Add experimental vorbis pay and depayloaders.
This commit is contained in:
Wim Taymans 2006-09-22 12:08:14 +00:00
parent 3b5584f8d1
commit 8dbf033420
9 changed files with 951 additions and 11 deletions
Show stats Download patch file Download diff file Expand all files Collapse all files

27
ChangeLog
View file

@ -1,3 +1,30 @@
2006-09-22 Wim Taymans <wim@fluendo.com>
* gst/rtp/gstrtpL16depay.c: (gst_rtp_L16depay_change_state):
* gst/rtp/gstrtpmp4gdepay.c: (gst_rtp_mp4g_depay_class_init):
Small cleanups.
* gst/rtp/Makefile.am:
* gst/rtp/gstrtp.c: (plugin_init):
* gst/rtp/gstrtpvorbisdepay.c: (gst_rtp_vorbis_depay_base_init),
(gst_rtp_vorbis_depay_class_init), (gst_rtp_vorbis_depay_init),
(gst_rtp_vorbis_depay_finalize), (gst_rtp_vorbis_depay_setcaps),
(gst_rtp_vorbis_depay_process),
(gst_rtp_vorbis_depay_set_property),
(gst_rtp_vorbis_depay_get_property),
(gst_rtp_vorbis_depay_change_state),
(gst_rtp_vorbis_depay_plugin_init):
* gst/rtp/gstrtpvorbisdepay.h:
* gst/rtp/gstrtpvorbispay.c: (gst_rtp_vorbis_pay_base_init),
(gst_rtp_vorbis_pay_class_init), (gst_rtp_vorbis_pay_init),
(gst_rtp_vorbis_pay_setcaps), (gst_rtp_vorbis_pay_init_packet),
(gst_rtp_vorbis_pay_flush_packet),
(gst_rtp_vorbis_pay_append_buffer),
(gst_rtp_vorbis_pay_handle_buffer),
(gst_rtp_vorbis_pay_plugin_init):
* gst/rtp/gstrtpvorbispay.h:
Add experimental vorbis pay and depayloaders.
2006-09-21 Wim Taymans <wim@fluendo.com>
* gst/rtp/gstrtpmp4gpay.c: (gst_rtp_mp4g_pay_parse_audio_config):

8
gst/rtp/Makefile.am
View file

@ -26,7 +26,9 @@ libgstrtp_la_SOURCES = \
gstrtpmp4gpay.c \
gstrtpspeexdepay.c \
gstrtpspeexpay.c \
gstrtpsv3vdepay.c
gstrtpsv3vdepay.c \
gstrtpvorbisdepay.c \
gstrtpvorbispay.c
#gstrtpL16pay.c gstrtpL16depay.c
@ -68,4 +70,6 @@ noinst_HEADERS = \
gstasteriskh263.h \
gstrtpspeexdepay.h \
gstrtpspeexpay.h \
gstrtpsv3vdepay.h
gstrtpsv3vdepay.h \
gstrtpvorbisdepay.h \
gstrtpvorbispay.h

8
gst/rtp/gstrtp.c
View file

@ -46,6 +46,8 @@
#include "gstrtpspeexpay.h"
#include "gstrtpspeexdepay.h"
#include "gstrtpsv3vdepay.h"
#include "gstrtpvorbisdepay.h"
#include "gstrtpvorbispay.h"
static gboolean
plugin_init (GstPlugin * plugin)
@ -125,6 +127,12 @@ plugin_init (GstPlugin * plugin)
if (!gst_rtp_sv3v_depay_plugin_init (plugin))
return FALSE;
if (!gst_rtp_vorbis_depay_plugin_init (plugin))
return FALSE;
if (!gst_rtp_vorbis_pay_plugin_init (plugin))
return FALSE;
return TRUE;
}

18
gst/rtp/gstrtpL16depay.c
View file

@ -318,28 +318,30 @@ static GstStateChangeReturn
gst_rtp_L16depay_change_state (GstElement * element, GstStateChange transition)
{
GstRtpL16Depay *rtpL16depay;
g_return_val_if_fail (GST_IS_RTP_L16_DEPAY (element),
GST_STATE_CHANGE_FAILURE);
GstStateChangeReturn ret;
rtpL16depay = GST_RTP_L16_DEPAY (element);
GST_DEBUG ("state pending %d\n", GST_STATE_PENDING (element));
switch (transition) {
case GST_STATE_CHANGE_NULL_TO_READY:
break;
default:
break;
}
/* if we haven't failed already, give the parent class a chance to ;-) */
ret = GST_ELEMENT_CLASS (parent_class)->change_state (element, transition);
switch (transition) {
case GST_STATE_CHANGE_READY_TO_NULL:
break;
default:
break;
}
/* if we haven't failed already, give the parent class a chance to ;-) */
if (GST_ELEMENT_CLASS (parent_class)->change_state)
return GST_ELEMENT_CLASS (parent_class)->change_state (element, transition);
return GST_STATE_CHANGE_SUCCESS;
return ret;
}
gboolean

1
gst/rtp/gstrtpmp4gdepay.c
View file

@ -128,7 +128,6 @@ gst_rtp_mp4g_depay_class_init (GstRtpMP4GDepayClass * klass)
gobject_class = (GObjectClass *) klass;
gstelement_class = (GstElementClass *) klass;
gstbasertpdepayload_class = (GstBaseRTPDepayloadClass *) klass;
parent_class = g_type_class_peek_parent (klass);

440
gst/rtp/gstrtpvorbisdepay.c Normal file
View file

@ -0,0 +1,440 @@
/* 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/Depay/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 (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 (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, packets: %d", ident, F,
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);
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 (length > payload_len)
goto length_short;
/* create buffer for packet */
if (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_NONE, GST_TYPE_RTP_VORBIS_DEPAY);
}

60
gst/rtp/gstrtpvorbisdepay.h Normal file
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@ -0,0 +1,60 @@
/* 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.
*/
#ifndef __GST_RTP_VORBIS_DEPAY_H__
#define __GST_RTP_VORBIS_DEPAY_H__
#include <gst/gst.h>
#include <gst/base/gstadapter.h>
#include <gst/rtp/gstbasertpdepayload.h>
G_BEGIN_DECLS
#define GST_TYPE_RTP_VORBIS_DEPAY \
(gst_rtp_vorbis_depay_get_type())
#define GST_RTP_VORBIS_DEPAY(obj) \
(G_TYPE_CHECK_INSTANCE_CAST((obj),GST_TYPE_RTP_VORBIS_DEPAY,GstRtpVorbisDepay))
#define GST_RTP_VORBIS_DEPAY_CLASS(klass) \
(G_TYPE_CHECK_CLASS_CAST((klass),GST_TYPE_RTP_VORBIS_DEPAY,GstRtpVorbisDepayClass))
#define GST_IS_RTP_VORBIS_DEPAY(obj) \
(G_TYPE_CHECK_INSTANCE_TYPE((obj),GST_TYPE_RTP_VORBIS_DEPAY))
#define GST_IS_RTP_VORBIS_DEPAY_CLASS(klass) \
(G_TYPE_CHECK_CLASS_TYPE((klass),GST_TYPE_RTP_VORBIS_DEPAY))
typedef struct _GstRtpVorbisDepay GstRtpVorbisDepay;
typedef struct _GstRtpVorbisDepayClass GstRtpVorbisDepayClass;
struct _GstRtpVorbisDepay
{
GstBaseRTPDepayload parent;
GstAdapter *adapter;
gboolean assembling;
};
struct _GstRtpVorbisDepayClass
{
GstBaseRTPDepayloadClass parent_class;
};
gboolean gst_rtp_vorbis_depay_plugin_init (GstPlugin * plugin);
G_END_DECLS
#endif /* __GST_RTP_VORBIS_DEPAY_H__ */

333
gst/rtp/gstrtpvorbispay.c Normal file
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@ -0,0 +1,333 @@
/* 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 <string.h>
#include <gst/rtp/gstrtpbuffer.h>
#include "gstrtpvorbispay.h"
GST_DEBUG_CATEGORY_STATIC (rtpvorbispay_debug);
#define GST_CAT_DEFAULT (rtpvorbispay_debug)
/* references:
* http://svn.xiph.org/trunk/vorbis/doc/draft-ietf-avt-rtp-vorbis-01.txt
*/
/* elementfactory information */
static const GstElementDetails gst_rtp_vorbispay_details =
GST_ELEMENT_DETAILS ("RTP packet parser",
"Codec/Payloader/Network",
"Payload-encode Vorbis audio into RTP packets (draft-01 RFC XXXX)",
"Wim Taymans <wim@fluendo.com>");
static GstStaticPadTemplate gst_rtp_vorbis_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) [ 96, 127 ], "
"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_pay_sink_template =
GST_STATIC_PAD_TEMPLATE ("sink",
GST_PAD_SINK,
GST_PAD_ALWAYS,
GST_STATIC_CAPS ("audio/x-vorbis")
);
GST_BOILERPLATE (GstRtpVorbisPay, gst_rtp_vorbis_pay, GstBaseRTPPayload,
GST_TYPE_BASE_RTP_PAYLOAD);
static gboolean gst_rtp_vorbis_pay_setcaps (GstBaseRTPPayload * basepayload,
GstCaps * caps);
static GstFlowReturn gst_rtp_vorbis_pay_handle_buffer (GstBaseRTPPayload * pad,
GstBuffer * buffer);
static void
gst_rtp_vorbis_pay_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_pay_src_template));
gst_element_class_add_pad_template (element_class,
gst_static_pad_template_get (&gst_rtp_vorbis_pay_sink_template));
gst_element_class_set_details (element_class, &gst_rtp_vorbispay_details);
}
static void
gst_rtp_vorbis_pay_class_init (GstRtpVorbisPayClass * klass)
{
GObjectClass *gobject_class;
GstElementClass *gstelement_class;
GstBaseRTPPayloadClass *gstbasertppayload_class;
gobject_class = (GObjectClass *) klass;
gstelement_class = (GstElementClass *) klass;
gstbasertppayload_class = (GstBaseRTPPayloadClass *) klass;
parent_class = g_type_class_peek_parent (klass);
gstbasertppayload_class->set_caps = gst_rtp_vorbis_pay_setcaps;
gstbasertppayload_class->handle_buffer = gst_rtp_vorbis_pay_handle_buffer;
GST_DEBUG_CATEGORY_INIT (rtpvorbispay_debug, "rtpvorbispay", 0,
"Vorbis RTP Payloader");
}
static void
gst_rtp_vorbis_pay_init (GstRtpVorbisPay * rtpvorbispay,
GstRtpVorbisPayClass * klass)
{
}
static gboolean
gst_rtp_vorbis_pay_setcaps (GstBaseRTPPayload * basepayload, GstCaps * caps)
{
GstRtpVorbisPay *rtpvorbispay;
rtpvorbispay = GST_RTP_VORBIS_PAY (basepayload);
gst_basertppayload_set_options (basepayload, "audio", TRUE, "vorbis", 8000);
gst_basertppayload_set_outcaps (basepayload,
"encoding-params", G_TYPE_STRING, "1",
/* don't set the defaults
*/
NULL);
return TRUE;
}
static void
gst_rtp_vorbis_pay_init_packet (GstRtpVorbisPay * rtpvorbispay)
{
guint payload_len;
if (rtpvorbispay->packet)
gst_buffer_unref (rtpvorbispay->packet);
GST_DEBUG_OBJECT (rtpvorbispay, "starting new packet");
/* new packet allocate max packet size */
rtpvorbispay->packet =
gst_rtp_buffer_new_allocate_len (GST_BASE_RTP_PAYLOAD_MTU
(rtpvorbispay), 0, 0);
rtpvorbispay->payload_pos = 4;
payload_len = gst_rtp_buffer_get_payload_len (rtpvorbispay->packet);
rtpvorbispay->payload_left = payload_len - 4;
rtpvorbispay->payload_duration = 0;
rtpvorbispay->payload_ident = 0;
rtpvorbispay->payload_F = 0;
rtpvorbispay->payload_VDT = 0;
rtpvorbispay->payload_pkts = 0;
}
static GstFlowReturn
gst_rtp_vorbis_pay_flush_packet (GstRtpVorbisPay * rtpvorbispay)
{
GstFlowReturn ret;
guint8 *payload;
guint hlen;
/* check for empty packet */
if (!rtpvorbispay || rtpvorbispay->payload_pos <= 4)
return GST_FLOW_OK;
GST_DEBUG_OBJECT (rtpvorbispay, "flushing packet");
/* fix header */
payload = gst_rtp_buffer_get_payload (rtpvorbispay->packet);
/*
* 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.
*/
payload[0] = (rtpvorbispay->payload_ident >> 16) & 0xff;
payload[1] = (rtpvorbispay->payload_ident >> 8) & 0xff;
payload[2] = (rtpvorbispay->payload_ident) & 0xff;
payload[3] = (rtpvorbispay->payload_F & 0x3) << 6 |
(rtpvorbispay->payload_VDT & 0x3) << 4 |
(rtpvorbispay->payload_pkts & 0xf);
/* shrink the buffer size to the last written byte */
hlen = gst_rtp_buffer_calc_header_len (0);
GST_BUFFER_SIZE (rtpvorbispay->packet) = hlen + rtpvorbispay->payload_pos;
/* push, this gives away our ref to the packet, so clear it. */
ret =
gst_basertppayload_push (GST_BASE_RTP_PAYLOAD (rtpvorbispay),
rtpvorbispay->packet);
rtpvorbispay->packet = NULL;
/* prepare new packet */
gst_rtp_vorbis_pay_init_packet (rtpvorbispay);
return ret;
}
static GstFlowReturn
gst_rtp_vorbis_pay_append_buffer (GstRtpVorbisPay * rtpvorbispay,
GstBuffer * buffer)
{
GstFlowReturn res;
guint size;
GstClockTime duration;
guint plen;
guint8 *ppos, *payload, *data;
gboolean fragmented;
res = GST_FLOW_OK;
if (rtpvorbispay->payload_left < 2)
return res;
size = GST_BUFFER_SIZE (buffer);
/* skip packets that are too big */
if (size > 0xffff)
return res;
data = GST_BUFFER_DATA (buffer);
duration = GST_BUFFER_DURATION (buffer);
payload = gst_rtp_buffer_get_payload (rtpvorbispay->packet);
ppos = payload + rtpvorbispay->payload_pos;
fragmented = FALSE;
while (size) {
plen = MIN (rtpvorbispay->payload_left - 2, size);
GST_DEBUG_OBJECT (rtpvorbispay, "append %u bytes", plen);
ppos[0] = (plen >> 8) & 0xff;
ppos[1] = (plen & 0xff);
memcpy (&ppos[2], data, plen);
size -= plen;
data += plen;
rtpvorbispay->payload_pos += plen + 2;
rtpvorbispay->payload_left -= plen + 2;
if (fragmented) {
if (size == 0)
/* last fragment, set F to 0x3. */
rtpvorbispay->payload_F = 0x3;
else
/* fragment continues, set F to 0x2. */
rtpvorbispay->payload_F = 0x2;
} else {
if (size == 0) {
/* unfragmented packet, update stats for next packet */
rtpvorbispay->payload_pkts++;
if (duration != GST_CLOCK_TIME_NONE)
rtpvorbispay->payload_duration += duration;
} else {
/* fragmented packet starts, set F to 0x1, mark ourselves as
* fragmented. */
rtpvorbispay->payload_F = 0x1;
fragmented = TRUE;
}
}
if (fragmented) {
/* fragmented packets are always flushed and have ptks of 0 */
rtpvorbispay->payload_pkts = 0;
res = gst_rtp_vorbis_pay_flush_packet (rtpvorbispay);
/* get new pointers */
payload = gst_rtp_buffer_get_payload (rtpvorbispay->packet);
ppos = payload + rtpvorbispay->payload_pos;
}
}
return res;
}
static GstFlowReturn
gst_rtp_vorbis_pay_handle_buffer (GstBaseRTPPayload * basepayload,
GstBuffer * buffer)
{
GstRtpVorbisPay *rtpvorbispay;
GstFlowReturn ret;
guint size, newsize;
guint packet_len;
GstClockTime duration, newduration;
gboolean flush;
rtpvorbispay = GST_RTP_VORBIS_PAY (basepayload);
size = GST_BUFFER_SIZE (buffer);
duration = GST_BUFFER_DURATION (buffer);
GST_DEBUG_OBJECT (rtpvorbispay, "size %u, duration %" GST_TIME_FORMAT,
size, GST_TIME_ARGS (duration));
if (!rtpvorbispay->packet)
gst_rtp_vorbis_pay_init_packet (rtpvorbispay);
/* size increases with packet length and 2 bytes size eader. */
newduration = rtpvorbispay->payload_duration;
if (duration != GST_CLOCK_TIME_NONE)
newduration += duration;
newsize = rtpvorbispay->payload_pos + 2 + size;
packet_len = gst_rtp_buffer_calc_packet_len (newsize, 0, 0);
/* check buffer filled against length and max latency */
flush = gst_basertppayload_is_filled (basepayload, packet_len, newduration);
/* we can store up to 15 vorbis packets in one RTP packet. */
flush |= (rtpvorbispay->payload_pkts == 15);
if (flush)
ret = gst_rtp_vorbis_pay_flush_packet (rtpvorbispay);
/* put buffer in packet */
ret = gst_rtp_vorbis_pay_append_buffer (rtpvorbispay, buffer);
return ret;
}
gboolean
gst_rtp_vorbis_pay_plugin_init (GstPlugin * plugin)
{
return gst_element_register (plugin, "rtpvorbispay",
GST_RANK_NONE, GST_TYPE_RTP_VORBIS_PAY);
}

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/* GStreamer
* Copyright (C) <2005> 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.
*/
#ifndef __GST_RTP_VORBIS_PAY_H__
#define __GST_RTP_VORBIS_PAY_H__
#include <gst/gst.h>
#include <gst/rtp/gstbasertppayload.h>
#include <gst/base/gstadapter.h>
G_BEGIN_DECLS
#define GST_TYPE_RTP_VORBIS_PAY \
(gst_rtp_vorbis_pay_get_type())
#define GST_RTP_VORBIS_PAY(obj) \
(G_TYPE_CHECK_INSTANCE_CAST((obj),GST_TYPE_RTP_VORBIS_PAY,GstRtpVorbisPay))
#define GST_RTP_VORBIS_PAY_CLASS(klass) \
(G_TYPE_CHECK_CLASS_CAST((klass),GST_TYPE_RTP_VORBIS_PAY,GstRtpVorbisPayClass))
#define GST_IS_RTP_VORBIS_PAY(obj) \
(G_TYPE_CHECK_INSTANCE_TYPE((obj),GST_TYPE_RTP_VORBIS_PAY))
#define GST_IS_RTP_VORBIS_PAY_CLASS(klass) \
(G_TYPE_CHECK_CLASS_TYPE((klass),GST_TYPE_RTP_VORBIS_PAY))
typedef struct _GstRtpVorbisPay GstRtpVorbisPay;
typedef struct _GstRtpVorbisPayClass GstRtpVorbisPayClass;
struct _GstRtpVorbisPay
{
GstBaseRTPPayload payload;
/* queues of buffers along with some stats. */
GstBuffer *packet;
guint payload_pos;
guint payload_left;
guint32 payload_ident;
guint8 payload_F;
guint8 payload_VDT;
guint payload_pkts;
GstClockTime payload_duration;
};
struct _GstRtpVorbisPayClass
{
GstBaseRTPPayloadClass parent_class;
};
gboolean gst_rtp_vorbis_pay_plugin_init (GstPlugin * plugin);
G_END_DECLS
#endif /* __GST_RTP_VORBIS_PAY_H__ */