+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):
gstrtpmp4gpay.c \
gstrtpspeexdepay.c \
gstrtpspeexpay.c \
- gstrtpsv3vdepay.c
+ gstrtpsv3vdepay.c \
+ gstrtpvorbisdepay.c \
+ gstrtpvorbispay.c
#gstrtpL16pay.c gstrtpL16depay.c
gstasteriskh263.h \
gstrtpspeexdepay.h \
gstrtpspeexpay.h \
- gstrtpsv3vdepay.h
+ gstrtpsv3vdepay.h \
+ gstrtpvorbisdepay.h \
+ gstrtpvorbispay.h
#include "gstrtpspeexpay.h"
#include "gstrtpspeexdepay.h"
#include "gstrtpsv3vdepay.h"
+#include "gstrtpvorbisdepay.h"
+#include "gstrtpvorbispay.h"
static gboolean
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;
}
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
gobject_class = (GObjectClass *) klass;
gstelement_class = (GstElementClass *) klass;
-
gstbasertpdepayload_class = (GstBaseRTPDepayloadClass *) klass;
parent_class = g_type_class_peek_parent (klass);
--- /dev/null
+/* 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);
+}
--- /dev/null
+/* 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__ */
--- /dev/null
+/* 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);
+}
--- /dev/null
+/* 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__ */
projects / platform / upstream / gst-plugins-good.git / commitdiff