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gst/rtp/: Added MP4A-LATM depayloader. Fixes #417792.

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Original commit message from CVS:
Based on patch by: Stefan Kost  <ensonic@users.sf.net>
* gst/rtp/Makefile.am:
* gst/rtp/gstrtp.c: (plugin_init):
* gst/rtp/gstrtpmp4adepay.c: (gst_rtp_mp4a_depay_base_init),
(gst_rtp_mp4a_depay_class_init), (gst_rtp_mp4a_depay_init),
(gst_rtp_mp4a_depay_finalize), (gst_rtp_mp4a_depay_setcaps),
(gst_rtp_mp4a_depay_process), (gst_rtp_mp4a_depay_set_property),
(gst_rtp_mp4a_depay_get_property),
(gst_rtp_mp4a_depay_change_state),
(gst_rtp_mp4a_depay_plugin_init):
* gst/rtp/gstrtpmp4adepay.h:
Added MP4A-LATM depayloader. Fixes #417792.
* gst/rtp/gstrtpmp4vdepay.c: (gst_rtp_mp4v_depay_setcaps),
(gst_rtp_mp4v_depay_process):
Fixup depayloader, setting codec_data, using more efficient adaptor and
rtpbuffer handling.
* gst/rtsp/URLS:
Add url to test above.
This commit is contained in:
Stefan Kost 2007-03-28 18:40:12 +00:00 committed by Wim Taymans
parent ab589bff3e
commit c0cdcae569
7 changed files with 484 additions and 44 deletions
Show stats Download patch file Download diff file Expand all files Collapse all files

24
ChangeLog
View file

@ -1,3 +1,27 @@
2007-03-28 Wim Taymans <wim@fluendo.com>
Based on patch by: Stefan Kost <ensonic@users.sf.net>
* gst/rtp/Makefile.am:
* gst/rtp/gstrtp.c: (plugin_init):
* gst/rtp/gstrtpmp4adepay.c: (gst_rtp_mp4a_depay_base_init),
(gst_rtp_mp4a_depay_class_init), (gst_rtp_mp4a_depay_init),
(gst_rtp_mp4a_depay_finalize), (gst_rtp_mp4a_depay_setcaps),
(gst_rtp_mp4a_depay_process), (gst_rtp_mp4a_depay_set_property),
(gst_rtp_mp4a_depay_get_property),
(gst_rtp_mp4a_depay_change_state),
(gst_rtp_mp4a_depay_plugin_init):
* gst/rtp/gstrtpmp4adepay.h:
Added MP4A-LATM depayloader. Fixes #417792.
* gst/rtp/gstrtpmp4vdepay.c: (gst_rtp_mp4v_depay_setcaps),
(gst_rtp_mp4v_depay_process):
Fixup depayloader, setting codec_data, using more efficient adaptor and
rtpbuffer handling.
* gst/rtsp/URLS:
Add url to test above.
2007-03-25 Wim Taymans <wim@fluendo.com>
* gst/rtsp/gstrtspsrc.c: (find_stream_by_setup),

10
gst/rtp/Makefile.am
View file

@ -21,8 +21,8 @@ libgstrtp_la_SOURCES = \
gstrtph263ppay.c \
gstrtph263pay.c \
gstrtph264depay.c \
gstrtpL16depay.c \
gstrtpL16pay.c \
gstrtpL16depay.c \
gstrtpL16pay.c \
gstasteriskh263.c \
gstrtpmp2tdepay.c \
gstrtpmp2tpay.c \
@ -30,6 +30,7 @@ libgstrtp_la_SOURCES = \
gstrtpmp4vpay.c \
gstrtpmp4gdepay.c \
gstrtpmp4gpay.c \
gstrtpmp4adepay.c \
gstrtpspeexdepay.c \
gstrtpspeexpay.c \
gstrtpsv3vdepay.c \
@ -54,11 +55,11 @@ libgstrtp_la_LDFLAGS = $(GST_PLUGIN_LDFLAGS)
noinst_HEADERS = \
gstrtpL16depay.h \
gstrtpL16pay.h \
gstrtpac3depay.h \
gstrtpac3depay.h \
gstrtpamrdepay.h \
gstrtpamrpay.h \
gstrtpilbcdepay.h \
gstrtpilbcpay.h \
gstrtpilbcpay.h \
gstrtppcmadepay.h \
gstrtppcmudepay.h \
gstrtppcmupay.h \
@ -78,6 +79,7 @@ noinst_HEADERS = \
gstrtpmp4vpay.h \
gstrtpmp4gdepay.h \
gstrtpmp4gpay.h \
gstrtpmp4adepay.h \
gstrtpdepay.h \
gstasteriskh263.h \
gstrtpspeexdepay.h \

4
gst/rtp/gstrtp.c
View file

@ -47,6 +47,7 @@
#include "gstrtpmp2tpay.h"
#include "gstrtpmp4vdepay.h"
#include "gstrtpmp4vpay.h"
#include "gstrtpmp4adepay.h"
#include "gstrtpmp4gdepay.h"
#include "gstrtpmp4gpay.h"
#include "gstrtpspeexpay.h"
@ -138,6 +139,9 @@ plugin_init (GstPlugin * plugin)
if (!gst_rtp_mp4v_depay_plugin_init (plugin))
return FALSE;
if (!gst_rtp_mp4a_depay_plugin_init (plugin))
return FALSE;
if (!gst_rtp_mp4g_depay_plugin_init (plugin))
return FALSE;

367
gst/rtp/gstrtpmp4adepay.c Normal file
View file

@ -0,0 +1,367 @@
/* GStreamer
* Copyright (C) <2007> Nokia Corporation (contact <stefan.kost@nokia.com>)
* <2007> 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 version 2 as published by the Free Software Foundation.
*
* 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 "gstrtpmp4adepay.h"
GST_DEBUG_CATEGORY_STATIC (rtpmp4adepay_debug);
#define GST_CAT_DEFAULT (rtpmp4adepay_debug)
/* elementfactory information */
static const GstElementDetails gst_rtp_mp4adepay_details =
GST_ELEMENT_DETAILS ("RTP packet parser",
"Codec/Depayloader/Network",
"Extracts MPEG4 audio from RTP packets (RFC 3016)",
"Nokia Corporation (contact <stefan.kost@nokia.com>), "
"Wim Taymans <wim@fluendo.com>");
/* RtpMP4ADepay signals and args */
enum
{
LAST_SIGNAL
};
enum
{
PROP_0,
};
static GstStaticPadTemplate gst_rtp_mp4a_depay_src_template =
GST_STATIC_PAD_TEMPLATE ("src",
GST_PAD_SRC,
GST_PAD_ALWAYS,
GST_STATIC_CAPS ("audio/mpeg,"
"mpegversion = (int) 4," "framed = (boolean) false")
);
static GstStaticPadTemplate gst_rtp_mp4a_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) \"MP4A-LATM\""
/* All optional parameters
*
* "profile-level-id=[1,MAX]"
* "config="
*/
)
);
GST_BOILERPLATE (GstRtpMP4ADepay, gst_rtp_mp4a_depay, GstBaseRTPDepayload,
GST_TYPE_BASE_RTP_DEPAYLOAD);
static void gst_rtp_mp4a_depay_finalize (GObject * object);
static gboolean gst_rtp_mp4a_depay_setcaps (GstBaseRTPDepayload * depayload,
GstCaps * caps);
static GstBuffer *gst_rtp_mp4a_depay_process (GstBaseRTPDepayload * depayload,
GstBuffer * buf);
static void gst_rtp_mp4a_depay_set_property (GObject * object, guint prop_id,
const GValue * value, GParamSpec * pspec);
static void gst_rtp_mp4a_depay_get_property (GObject * object, guint prop_id,
GValue * value, GParamSpec * pspec);
static GstStateChangeReturn gst_rtp_mp4a_depay_change_state (GstElement *
element, GstStateChange transition);
static void
gst_rtp_mp4a_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_mp4a_depay_src_template));
gst_element_class_add_pad_template (element_class,
gst_static_pad_template_get (&gst_rtp_mp4a_depay_sink_template));
gst_element_class_set_details (element_class, &gst_rtp_mp4adepay_details);
}
static void
gst_rtp_mp4a_depay_class_init (GstRtpMP4ADepayClass * klass)
{
GObjectClass *gobject_class;
GstElementClass *gstelement_class;
GstBaseRTPDepayloadClass *gstbasertpdepayload_class;
gobject_class = (GObjectClass *) klass;
gstelement_class = (GstElementClass *) klass;
gstbasertpdepayload_class = (GstBaseRTPDepayloadClass *) klass;
gobject_class->finalize = gst_rtp_mp4a_depay_finalize;
gobject_class->set_property = gst_rtp_mp4a_depay_set_property;
gobject_class->get_property = gst_rtp_mp4a_depay_get_property;
gstelement_class->change_state = gst_rtp_mp4a_depay_change_state;
gstbasertpdepayload_class->process = gst_rtp_mp4a_depay_process;
gstbasertpdepayload_class->set_caps = gst_rtp_mp4a_depay_setcaps;
GST_DEBUG_CATEGORY_INIT (rtpmp4adepay_debug, "rtpmp4adepay", 0,
"MPEG4 audio RTP Depayloader");
}
static void
gst_rtp_mp4a_depay_init (GstRtpMP4ADepay * rtpmp4adepay,
GstRtpMP4ADepayClass * klass)
{
rtpmp4adepay->adapter = gst_adapter_new ();
}
static void
gst_rtp_mp4a_depay_finalize (GObject * object)
{
GstRtpMP4ADepay *rtpmp4adepay;
rtpmp4adepay = GST_RTP_MP4A_DEPAY (object);
g_object_unref (rtpmp4adepay->adapter);
rtpmp4adepay->adapter = NULL;
G_OBJECT_CLASS (parent_class)->finalize (object);
}
static gboolean
gst_rtp_mp4a_depay_setcaps (GstBaseRTPDepayload * depayload, GstCaps * caps)
{
GstStructure *structure;
GstRtpMP4ADepay *rtpmp4adepay;
GstCaps *srccaps;
const gchar *str;
gint clock_rate = 90000; /* default */
gint object_type = 2; /* AAC LC default */
gint channels = 2; /* default */
rtpmp4adepay = GST_RTP_MP4A_DEPAY (depayload);
structure = gst_caps_get_structure (caps, 0);
if (gst_structure_has_field (structure, "clock-rate"))
gst_structure_get_int (structure, "clock-rate", &clock_rate);
depayload->clock_rate = clock_rate;
if (gst_structure_has_field (structure, "object"))
gst_structure_get_int (structure, "object", &object_type);
srccaps = gst_caps_new_simple ("audio/mpeg",
"mpegversion", G_TYPE_INT, 4,
"framed", G_TYPE_BOOLEAN, FALSE, "channels", G_TYPE_INT, channels, NULL);
if ((str = gst_structure_get_string (structure, "config"))) {
GValue v = { 0 };
g_value_init (&v, GST_TYPE_BUFFER);
if (gst_value_deserialize (&v, str)) {
GstBuffer *buffer;
guint8 *data;
guint size;
gint i;
buffer = gst_value_get_buffer (&v);
gst_buffer_ref (buffer);
g_value_unset (&v);
data = GST_BUFFER_DATA (buffer);
size = GST_BUFFER_SIZE (buffer);
if (size < 2) {
GST_WARNING_OBJECT (depayload, "config too short (%d < 2)", size);
goto bad_config;
}
/* Parse StreamMuxConfig according to ISO/IEC 14496-3:
*
* audioMuxVersion == 0 (1 bit)
* allStreamsSameTimeFraming == 1 (1 bit)
* numSubFrames == 0 (6 bits)
* numProgram == 0 (4 bits)
* numLayer == 0 (3 bits)
*
* We only require audioMuxVersion == 0;
*
* The remaining bit of the second byte and the rest of the bits are used
* for audioSpecificConfig which we need to set in codec_info.
*/
if ((data[0] & 0x80) != 0x00) {
GST_WARNING_OBJECT (depayload, "unknown audioMuxVersion 1");
goto bad_config;
}
/* shift rest of string 15 bits down */
size -= 2;
for (i = 0; i < size; i++) {
data[i] = ((data[i + 1] & 1) << 7) | ((data[i + 2] & 0xfe) >> 1);
}
/* last bit, this is probably not needed. */
data[i] = ((data[i + 1] & 1) << 7);
GST_BUFFER_SIZE (buffer) = size + 1;
gst_caps_set_simple (srccaps,
"codec_data", GST_TYPE_BUFFER, buffer, NULL);
gst_buffer_unref (buffer);
} else {
g_warning ("cannot convert config to buffer");
}
}
bad_config:
gst_pad_set_caps (depayload->srcpad, srccaps);
gst_caps_unref (srccaps);
return TRUE;
}
static GstBuffer *
gst_rtp_mp4a_depay_process (GstBaseRTPDepayload * depayload, GstBuffer * buf)
{
GstRtpMP4ADepay *rtpmp4adepay;
GstBuffer *outbuf;
rtpmp4adepay = GST_RTP_MP4A_DEPAY (depayload);
if (!gst_rtp_buffer_validate (buf))
goto bad_packet;
/* flush remaining data on discont */
if (GST_BUFFER_IS_DISCONT (buf))
gst_adapter_clear (rtpmp4adepay->adapter);
outbuf = gst_rtp_buffer_get_payload_buffer (buf);
gst_adapter_push (rtpmp4adepay->adapter, outbuf);
/* RTP marker bit indicates the last packet of the AudioMuxElement => create
* and push a buffer */
if (gst_rtp_buffer_get_marker (buf)) {
guint avail;
guint latm_header_len;
guint data_len;
guint8 *data;
avail = gst_adapter_available (rtpmp4adepay->adapter);
outbuf = gst_adapter_take_buffer (rtpmp4adepay->adapter, avail);
/* determine payload length and set buffer data pointer accordingly */
/* FIXME, check for overrun */
latm_header_len = 0;
data_len = 0;
data = GST_BUFFER_DATA (outbuf);
do {
data_len += data[latm_header_len];
} while (data[latm_header_len++] == 0xff);
/* just a check that lengths match, possibly there can be more than one
* audioMuxElement in the payload? */
if ((data_len + latm_header_len) != avail) {
GST_WARNING_OBJECT (depayload, "not all payload consumed");
}
GST_BUFFER_SIZE (outbuf) = avail - latm_header_len;
GST_BUFFER_DATA (outbuf) += latm_header_len;
gst_buffer_set_caps (outbuf, GST_PAD_CAPS (depayload->srcpad));
GST_DEBUG ("gst_rtp_mp4a_depay_process: pushing buffer of size %d",
GST_BUFFER_SIZE (outbuf));
return outbuf;
}
return NULL;
bad_packet:
{
GST_ELEMENT_WARNING (rtpmp4adepay, STREAM, DECODE,
("Packet did not validate"), (NULL));
return NULL;
}
}
static void
gst_rtp_mp4a_depay_set_property (GObject * object, guint prop_id,
const GValue * value, GParamSpec * pspec)
{
GstRtpMP4ADepay *rtpmp4adepay;
rtpmp4adepay = GST_RTP_MP4A_DEPAY (object);
switch (prop_id) {
default:
G_OBJECT_WARN_INVALID_PROPERTY_ID (object, prop_id, pspec);
break;
}
}
static void
gst_rtp_mp4a_depay_get_property (GObject * object, guint prop_id,
GValue * value, GParamSpec * pspec)
{
GstRtpMP4ADepay *rtpmp4adepay;
rtpmp4adepay = GST_RTP_MP4A_DEPAY (object);
switch (prop_id) {
default:
G_OBJECT_WARN_INVALID_PROPERTY_ID (object, prop_id, pspec);
break;
}
}
static GstStateChangeReturn
gst_rtp_mp4a_depay_change_state (GstElement * element,
GstStateChange transition)
{
GstRtpMP4ADepay *rtpmp4adepay;
GstStateChangeReturn ret;
rtpmp4adepay = GST_RTP_MP4A_DEPAY (element);
switch (transition) {
case GST_STATE_CHANGE_READY_TO_PAUSED:
gst_adapter_clear (rtpmp4adepay->adapter);
break;
default:
break;
}
ret = GST_ELEMENT_CLASS (parent_class)->change_state (element, transition);
switch (transition) {
default:
break;
}
return ret;
}
gboolean
gst_rtp_mp4a_depay_plugin_init (GstPlugin * plugin)
{
return gst_element_register (plugin, "rtpmp4adepay",
GST_RANK_NONE, GST_TYPE_RTP_MP4A_DEPAY);
}

59
gst/rtp/gstrtpmp4adepay.h Normal file
View file

@ -0,0 +1,59 @@
/* GStreamer
* Copyright (C) <2007> Nokia Corporation (contact <stefan.kost@nokia.com>)
*
* This library is free software; you can redistribute it and/or
* modify it under the terms of the GNU Library General Public
* License version 2 as published by the Free Software Foundation.
*
* 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_MP4A_DEPAY_H__
#define __GST_RTP_MP4A_DEPAY_H__
#include <gst/gst.h>
#include <gst/base/gstadapter.h>
#include <gst/rtp/gstbasertpdepayload.h>
G_BEGIN_DECLS
#define GST_TYPE_RTP_MP4A_DEPAY \
(gst_rtp_mp4a_depay_get_type())
#define GST_RTP_MP4A_DEPAY(obj) \
(G_TYPE_CHECK_INSTANCE_CAST((obj),GST_TYPE_RTP_MP4A_DEPAY,GstRtpMP4ADepay))
#define GST_RTP_MP4A_DEPAY_CLASS(klass) \
(G_TYPE_CHECK_CLASS_CAST((klass),GST_TYPE_RTP_MP4A_DEPAY,GstRtpMP4ADepayClass))
#define GST_IS_RTP_MP4A_DEPAY(obj) \
(G_TYPE_CHECK_INSTANCE_TYPE((obj),GST_TYPE_RTP_MP4A_DEPAY))
#define GST_IS_RTP_MP4A_DEPAY_CLASS(klass) \
(G_TYPE_CHECK_CLASS_TYPE((klass),GST_TYPE_RTP_MP4A_DEPAY))
typedef struct _GstRtpMP4ADepay GstRtpMP4ADepay;
typedef struct _GstRtpMP4ADepayClass GstRtpMP4ADepayClass;
struct _GstRtpMP4ADepay
{
GstBaseRTPDepayload depayload;
GstAdapter *adapter;
};
struct _GstRtpMP4ADepayClass
{
GstBaseRTPDepayloadClass parent_class;
};
gboolean gst_rtp_mp4a_depay_plugin_init (GstPlugin * plugin);
G_END_DECLS
#endif /* __GST_RTP_MP4A_DEPAY_H__ */

61
gst/rtp/gstrtpmp4vdepay.c
View file

@ -39,14 +39,12 @@ GST_ELEMENT_DETAILS ("RTP packet depayloader",
/* RtpMP4VDepay signals and args */
enum
{
/* FILL ME */
LAST_SIGNAL
};
enum
{
ARG_0,
ARG_FREQUENCY
PROP_0,
};
static GstStaticPadTemplate gst_rtp_mp4v_depay_src_template =
@ -152,7 +150,6 @@ gst_rtp_mp4v_depay_finalize (GObject * object)
static gboolean
gst_rtp_mp4v_depay_setcaps (GstBaseRTPDepayload * depayload, GstCaps * caps)
{
GstStructure *structure;
GstRtpMP4VDepay *rtpmp4vdepay;
GstCaps *srccaps;
@ -163,7 +160,8 @@ gst_rtp_mp4v_depay_setcaps (GstBaseRTPDepayload * depayload, GstCaps * caps)
structure = gst_caps_get_structure (caps, 0);
gst_structure_get_int (structure, "clock-rate", &clock_rate);
if (gst_structure_has_field (structure, "clock-rate"))
gst_structure_get_int (structure, "clock-rate", &clock_rate);
depayload->clock_rate = clock_rate;
srccaps = gst_caps_new_simple ("video/mpeg",
@ -181,15 +179,14 @@ gst_rtp_mp4v_depay_setcaps (GstBaseRTPDepayload * depayload, GstCaps * caps)
gst_buffer_ref (buffer);
g_value_unset (&v);
gst_buffer_set_caps (buffer, srccaps);
gst_caps_unref (srccaps);
gst_pad_push (depayload->srcpad, buffer);
gst_caps_set_simple (srccaps,
"codec_data", GST_TYPE_BUFFER, buffer, NULL);
} else {
g_warning ("cannot convert config to buffer");
}
}
gst_pad_set_caps (depayload->srcpad, srccaps);
gst_caps_unref (srccaps);
return TRUE;
}
@ -205,45 +202,29 @@ gst_rtp_mp4v_depay_process (GstBaseRTPDepayload * depayload, GstBuffer * buf)
if (!gst_rtp_buffer_validate (buf))
goto bad_packet;
{
gint payload_len;
guint8 *payload;
guint32 timestamp;
/* flush remaining data on discont */
if (GST_BUFFER_IS_DISCONT (buf))
gst_adapter_clear (rtpmp4vdepay->adapter);
payload_len = gst_rtp_buffer_get_payload_len (buf);
payload = gst_rtp_buffer_get_payload (buf);
outbuf = gst_rtp_buffer_get_payload_buffer (buf);
timestamp = gst_rtp_buffer_get_timestamp (buf);
gst_adapter_push (rtpmp4vdepay->adapter, outbuf);
outbuf = gst_buffer_new_and_alloc (payload_len);
memcpy (GST_BUFFER_DATA (outbuf), payload, payload_len);
/* if this was the last packet of the VOP, create and push a buffer */
if (gst_rtp_buffer_get_marker (buf)) {
guint avail;
gst_adapter_push (rtpmp4vdepay->adapter, outbuf);
avail = gst_adapter_available (rtpmp4vdepay->adapter);
/* if this was the last packet of the VOP, create and push a buffer */
if (gst_rtp_buffer_get_marker (buf)) {
guint avail;
outbuf = gst_adapter_take_buffer (rtpmp4vdepay->adapter, avail);
avail = gst_adapter_available (rtpmp4vdepay->adapter);
gst_buffer_set_caps (outbuf, GST_PAD_CAPS (depayload->srcpad));
outbuf = gst_buffer_new ();
GST_BUFFER_SIZE (outbuf) = avail;
GST_BUFFER_MALLOCDATA (outbuf) =
gst_adapter_take (rtpmp4vdepay->adapter, avail);
GST_BUFFER_DATA (outbuf) = GST_BUFFER_MALLOCDATA (outbuf);
gst_buffer_set_caps (outbuf, GST_PAD_CAPS (depayload->srcpad));
GST_BUFFER_TIMESTAMP (outbuf) =
timestamp * GST_SECOND / depayload->clock_rate;
GST_DEBUG ("gst_rtp_mp4v_depay_chain: pushing buffer of size %d",
GST_BUFFER_SIZE (outbuf));
GST_DEBUG ("gst_rtp_mp4v_depay_chain: pushing buffer of size %d",
GST_BUFFER_SIZE (outbuf));
return outbuf;
} else {
return NULL;
}
return outbuf;
}
return NULL;
bad_packet:

3
gst/rtsp/URLS
View file

@ -16,6 +16,9 @@ MP4V-ES/mpeg4-generic(ACC):
rtsp://kmdi.utoronto.ca:555/osconf/2004_may9.1.mp4
rtsp://a2047.v1413b.c1413.g.vq.akamaistream.net/5/2047/1413/1_h264_110/1a1a1ae656c632970267e04ebd3196c428970e7ce857b81c4aab1677e445aedc3fae1b4a7bafe013/8848125_1_110.mov
MP4V-ES/MP4A-LATM
rtsp://68.251.168.13/thisislove.3gp
REAL:
rtsp://213.254.239.61/farm/*/encoder/tagesschau/live1high.rm
rtsp://64.192.137.105:554/real.amazon-de.eu2/phononet/B/0/0/0/H/W/Y/4/K/S/01.01.rm?cloakport=80,554,7070