mirrors/gstreamer
mirrors/gstreamer
1
1
Fork 0
mirror of https://gitlab.freedesktop.org/gstreamer/gstreamer.git synced 2024-12-30 12:10:37 +00:00
Code Issues 1 Projects Releases Wiki Activity

rtpbin: add support for buffer-list

Browse source 
Add support for sending buffer-lists.
Add unit test for testing that the buffer-list passed through rtpbin.

fixes #585839
This commit is contained in:
Branko Subasic 2009-06-19 19:09:19 +02:00 committed by Wim Taymans
parent 11dc33bea0
commit c70dbe94b5
7 changed files with 501 additions and 63 deletions
Show stats Download patch file Download diff file Expand all files Collapse all files

62
gst/rtpmanager/gstrtpsession.c
View file

@ -259,7 +259,7 @@ struct _GstRtpSessionPrivate
static GstFlowReturn gst_rtp_session_process_rtp (RTPSession * sess,
RTPSource * src, GstBuffer * buffer, gpointer user_data);
static GstFlowReturn gst_rtp_session_send_rtp (RTPSession * sess,
RTPSource * src, GstBuffer * buffer, gpointer user_data);
RTPSource * src, gpointer data, gpointer user_data);
static GstFlowReturn gst_rtp_session_send_rtcp (RTPSession * sess,
RTPSource * src, GstBuffer * buffer, gboolean eos, gpointer user_data);
static GstFlowReturn gst_rtp_session_sync_rtcp (RTPSession * sess,
@ -1032,8 +1032,8 @@ gst_rtp_session_clear_pt_map (GstRtpSession * rtpsession)
g_hash_table_foreach_remove (rtpsession->priv->ptmap, return_true, NULL);
}
/* called when the session manager has an RTP packet ready for further
* processing */
/* called when the session manager has an RTP packet or a list of packets
* ready for further processing */
static GstFlowReturn
gst_rtp_session_process_rtp (RTPSession * sess, RTPSource * src,
GstBuffer * buffer, gpointer user_data)
@ -1060,7 +1060,7 @@ gst_rtp_session_process_rtp (RTPSession * sess, RTPSource * src,
* sending */
static GstFlowReturn
gst_rtp_session_send_rtp (RTPSession * sess, RTPSource * src,
GstBuffer * buffer, gpointer user_data)
gpointer data, gpointer user_data)
{
GstFlowReturn result;
GstRtpSession *rtpsession;
@ -1069,12 +1069,17 @@ gst_rtp_session_send_rtp (RTPSession * sess, RTPSource * src,
rtpsession = GST_RTP_SESSION (user_data);
priv = rtpsession->priv;
GST_LOG_OBJECT (rtpsession, "sending RTP packet");
if (rtpsession->send_rtp_src) {
result = gst_pad_push (rtpsession->send_rtp_src, buffer);
if (GST_IS_BUFFER (data)) {
GST_LOG_OBJECT (rtpsession, "sending RTP packet");
result = gst_pad_push (rtpsession->send_rtp_src, GST_BUFFER_CAST (data));
} else {
GST_LOG_OBJECT (rtpsession, "sending RTP list");
result = gst_pad_push_list (rtpsession->send_rtp_src,
GST_BUFFER_LIST_CAST (data));
}
} else {
gst_buffer_unref (buffer);
gst_mini_object_unref (GST_MINI_OBJECT_CAST (data));
result = GST_FLOW_OK;
}
return result;
@ -1642,11 +1647,12 @@ gst_rtp_session_setcaps_send_rtp (GstPad * pad, GstCaps * caps)
return TRUE;
}
/* Recieve an RTP packet to be send to the receivers, send to RTP session
* manager and forward to send_rtp_src.
/* Recieve an RTP packet or a list of packets to be send to the receivers,
* send to RTP session manager and forward to send_rtp_src.
*/
static GstFlowReturn
gst_rtp_session_chain_send_rtp (GstPad * pad, GstBuffer * buffer)
gst_rtp_session_chain_send_rtp_common (GstPad * pad, gpointer data,
gboolean is_list)
{
GstRtpSession *rtpsession;
GstRtpSessionPrivate *priv;
@ -1658,10 +1664,22 @@ gst_rtp_session_chain_send_rtp (GstPad * pad, GstBuffer * buffer)
rtpsession = GST_RTP_SESSION (gst_pad_get_parent (pad));
priv = rtpsession->priv;
GST_LOG_OBJECT (rtpsession, "received RTP packet");
GST_LOG_OBJECT (rtpsession, "received RTP %s", is_list ? "list" : "packet");
/* get NTP time when this packet was captured, this depends on the timestamp. */
timestamp = GST_BUFFER_TIMESTAMP (buffer);
if (is_list) {
GstBuffer *buffer = NULL;
/* All groups in an list have the same timestamp.
* So, just take it from the first group. */
buffer = gst_buffer_list_get (GST_BUFFER_LIST_CAST (data), 0, 0);
if (buffer)
timestamp = GST_BUFFER_TIMESTAMP (buffer);
else
timestamp = -1;
} else {
timestamp = GST_BUFFER_TIMESTAMP (GST_BUFFER_CAST (data));
}
if (GST_CLOCK_TIME_IS_VALID (timestamp)) {
/* convert to running time using the segment start value. */
ntpnstime =
@ -1676,7 +1694,9 @@ gst_rtp_session_chain_send_rtp (GstPad * pad, GstBuffer * buffer)
}
current_time = gst_clock_get_time (priv->sysclock);
ret = rtp_session_send_rtp (priv->session, buffer, current_time, ntpnstime);
ret =
rtp_session_send_rtp (priv->session, data, is_list, current_time,
ntpnstime);
if (ret != GST_FLOW_OK)
goto push_error;
@ -1694,6 +1714,18 @@ push_error:
}
}
static GstFlowReturn
gst_rtp_session_chain_send_rtp (GstPad * pad, GstBuffer * buffer)
{
return gst_rtp_session_chain_send_rtp_common (pad, buffer, FALSE);
}
static GstFlowReturn
gst_rtp_session_chain_send_rtp_list (GstPad * pad, GstBufferList * list)
{
return gst_rtp_session_chain_send_rtp_common (pad, list, TRUE);
}
/* Create sinkpad to receive RTP packets from senders. This will also create a
* srcpad for the RTP packets.
*/
@ -1817,6 +1849,8 @@ create_send_rtp_sink (GstRtpSession * rtpsession)
"send_rtp_sink");
gst_pad_set_chain_function (rtpsession->send_rtp_sink,
gst_rtp_session_chain_send_rtp);
gst_pad_set_chain_list_function (rtpsession->send_rtp_sink,
gst_rtp_session_chain_send_rtp_list);
gst_pad_set_getcaps_function (rtpsession->send_rtp_sink,
gst_rtp_session_getcaps_send_rtp);
gst_pad_set_setcaps_function (rtpsession->send_rtp_sink,

37
gst/rtpmanager/rtpsession.c
View file

@ -958,7 +958,7 @@ rtp_session_get_sdes_string (RTPSession * sess, GstRTCPSDESType type)
}
static GstFlowReturn
source_push_rtp (RTPSource * source, GstBuffer * buffer, RTPSession * session)
source_push_rtp (RTPSource * source, gpointer data, RTPSession * session)
{
GstFlowReturn result = GST_FLOW_OK;
@ -969,21 +969,21 @@ source_push_rtp (RTPSource * source, GstBuffer * buffer, RTPSession * session)
if (session->callbacks.send_rtp)
result =
session->callbacks.send_rtp (session, source, buffer,
session->callbacks.send_rtp (session, source, data,
session->send_rtp_user_data);
else
gst_buffer_unref (buffer);
else {
gst_mini_object_unref (GST_MINI_OBJECT_CAST (data));
}
} else {
GST_LOG ("source %08x pushed receiver RTP packet", source->ssrc);
RTP_SESSION_UNLOCK (session);
if (session->callbacks.process_rtp)
result =
session->callbacks.process_rtp (session, source, buffer,
session->process_rtp_user_data);
session->callbacks.process_rtp (session, source,
GST_BUFFER_CAST (data), session->process_rtp_user_data);
else
gst_buffer_unref (buffer);
gst_buffer_unref (GST_BUFFER_CAST (data));
}
RTP_SESSION_LOCK (session);
@ -1962,7 +1962,7 @@ ignore:
/**
* rtp_session_send_rtp:
* @sess: an #RTPSession
* @buffer: an RTP buffer
* @data: pointer to either an RTP buffer or a list of RTP buffers
* @current_time: the current system time
* @ntpnstime: the NTP time in nanoseconds of when this buffer was captured.
* This is the buffer timestamp converted to NTP time.
@ -1973,20 +1973,27 @@ ignore:
* Returns: a #GstFlowReturn.
*/
GstFlowReturn
rtp_session_send_rtp (RTPSession * sess, GstBuffer * buffer,
rtp_session_send_rtp (RTPSession * sess, gpointer data, gboolean is_list,
GstClockTime current_time, guint64 ntpnstime)
{
GstFlowReturn result;
RTPSource *source;
gboolean prevsender;
gboolean valid_packet;
g_return_val_if_fail (RTP_IS_SESSION (sess), GST_FLOW_ERROR);
g_return_val_if_fail (GST_IS_BUFFER (buffer), GST_FLOW_ERROR);
g_return_val_if_fail (is_list || GST_IS_BUFFER (data), GST_FLOW_ERROR);
if (!gst_rtp_buffer_validate (buffer))
if (is_list) {
valid_packet = gst_rtp_buffer_list_validate (GST_BUFFER_LIST_CAST (data));
} else {
valid_packet = gst_rtp_buffer_validate (GST_BUFFER_CAST (data));
}
if (!valid_packet)
goto invalid_packet;
GST_LOG ("received RTP packet for sending");
GST_LOG ("received RTP %s for sending", is_list ? "list" : "packet");
RTP_SESSION_LOCK (sess);
source = sess->source;
@ -1997,7 +2004,7 @@ rtp_session_send_rtp (RTPSession * sess, GstBuffer * buffer,
prevsender = RTP_SOURCE_IS_SENDER (source);
/* we use our own source to send */
result = rtp_source_send_rtp (source, buffer, ntpnstime);
result = rtp_source_send_rtp (source, data, is_list, ntpnstime);
if (RTP_SOURCE_IS_SENDER (source) && !prevsender)
sess->stats.sender_sources++;
@ -2008,7 +2015,7 @@ rtp_session_send_rtp (RTPSession * sess, GstBuffer * buffer,
/* ERRORS */
invalid_packet:
{
gst_buffer_unref (buffer);
gst_mini_object_unref (GST_MINI_OBJECT_CAST (data));
GST_DEBUG ("invalid RTP packet received");
return GST_FLOW_OK;
}

4
gst/rtpmanager/rtpsession.h
View file

@ -65,7 +65,7 @@ typedef GstFlowReturn (*RTPSessionProcessRTP) (RTPSession *sess, RTPSource *src,
*
* Returns: a #GstFlowReturn.
*/
typedef GstFlowReturn (*RTPSessionSendRTP) (RTPSession *sess, RTPSource *src, GstBuffer *buffer, gpointer user_data);
typedef GstFlowReturn (*RTPSessionSendRTP) (RTPSession *sess, RTPSource *src, gpointer data, gpointer user_data);
/**
* RTPSessionSendRTCP:
@ -288,7 +288,7 @@ GstFlowReturn rtp_session_process_rtcp (RTPSession *sess, GstBuffer
GstClockTime current_time);
/* processing packets for sending */
GstFlowReturn rtp_session_send_rtp (RTPSession *sess, GstBuffer *buffer,
GstFlowReturn rtp_session_send_rtp (RTPSession *sess, gpointer data, gboolean is_list,
GstClockTime current_time, guint64 ntpnstime);
/* stopping the session */

120
gst/rtpmanager/rtpsource.c
View file

@ -26,7 +26,7 @@
GST_DEBUG_CATEGORY_STATIC (rtp_source_debug);
#define GST_CAT_DEFAULT rtp_source_debug
#define RTP_MAX_PROBATION_LEN 32
#define RTP_MAX_PROBATION_LEN 32
/* signals and args */
enum
@ -1091,41 +1091,73 @@ rtp_source_process_bye (RTPSource * src, const gchar * reason)
src->received_bye = TRUE;
}
static GstBufferListItem
set_ssrc (GstBuffer ** buffer, guint group, guint idx, RTPSource * src)
{
*buffer = gst_buffer_make_writable (*buffer);
gst_rtp_buffer_set_ssrc (*buffer, src->ssrc);
return GST_BUFFER_LIST_SKIP_GROUP;
}
/**
* rtp_source_send_rtp:
* @src: an #RTPSource
* @buffer: an RTP buffer
* @data: an RTP buffer or a list of RTP buffers
* @is_list: if @data is a buffer or list
* @ntpnstime: the NTP time when this buffer was captured in nanoseconds. This
* is the buffer timestamp converted to NTP time.
*
* Send an RTP @buffer originating from @src. This will make @src a sender.
* This function takes ownership of @buffer and modifies the SSRC in the RTP
* packet to that of @src when needed.
* Send @data (an RTP buffer or list of buffers) originating from @src.
* This will make @src a sender. This function takes ownership of @data and
* modifies the SSRC in the RTP packet to that of @src when needed.
*
* Returns: a #GstFlowReturn.
*/
GstFlowReturn
rtp_source_send_rtp (RTPSource * src, GstBuffer * buffer, guint64 ntpnstime)
rtp_source_send_rtp (RTPSource * src, gpointer data, gboolean is_list,
guint64 ntpnstime)
{
GstFlowReturn result = GST_FLOW_OK;
GstFlowReturn result;
guint len;
guint32 rtptime;
guint64 ext_rtptime;
guint64 ntp_diff, rtp_diff;
guint64 elapsed;
GstBufferList *list = NULL;
GstBuffer *buffer = NULL;
guint packets;
guint32 ssrc;
g_return_val_if_fail (RTP_IS_SOURCE (src), GST_FLOW_ERROR);
g_return_val_if_fail (GST_IS_BUFFER (buffer), GST_FLOW_ERROR);
g_return_val_if_fail (is_list || GST_IS_BUFFER (data), GST_FLOW_ERROR);
len = gst_rtp_buffer_get_payload_len (buffer);
if (is_list) {
list = GST_BUFFER_LIST_CAST (data);
/* We can grab the caps from the first group, since all
* groups of a buffer list have same caps. */
buffer = gst_buffer_list_get (list, 0, 0);
if (!buffer)
goto no_buffer;
} else {
buffer = GST_BUFFER_CAST (data);
}
rtp_source_update_caps (src, GST_BUFFER_CAPS (buffer));
/* we are a sender now */
src->is_sender = TRUE;
if (is_list) {
/* Each group makes up a network packet. */
packets = gst_buffer_list_n_groups (list);
len = gst_rtp_buffer_list_get_payload_len (list);
} else {
packets = 1;
len = gst_rtp_buffer_get_payload_len (buffer);
}
/* update stats for the SR */
src->stats.packets_sent++;
src->stats.packets_sent += packets;
src->stats.octets_sent += len;
src->bytes_sent += len;
@ -1156,7 +1188,11 @@ rtp_source_send_rtp (RTPSource * src, GstBuffer * buffer, guint64 ntpnstime)
src->bitrate = 0;
}
rtptime = gst_rtp_buffer_get_timestamp (buffer);
if (is_list) {
rtptime = gst_rtp_buffer_list_get_timestamp (list);
} else {
rtptime = gst_rtp_buffer_get_timestamp (buffer);
}
ext_rtptime = src->last_rtptime;
ext_rtptime = gst_rtp_buffer_ext_timestamp (&ext_rtptime, rtptime);
@ -1180,31 +1216,53 @@ rtp_source_send_rtp (RTPSource * src, GstBuffer * buffer, guint64 ntpnstime)
src->last_ntpnstime = ntpnstime;
/* push packet */
if (src->callbacks.push_rtp) {
guint32 ssrc;
if (!src->callbacks.push_rtp)
goto no_callback;
ssrc = gst_rtp_buffer_get_ssrc (buffer);
if (ssrc != src->ssrc) {
/* the SSRC of the packet is not correct, make a writable buffer and
* update the SSRC. This could involve a complete copy of the packet when
* it is not writable. Usually the payloader will use caps negotiation to
* get the correct SSRC from the session manager before pushing anything. */
buffer = gst_buffer_make_writable (buffer);
/* FIXME, we don't want to warn yet because we can't inform any payloader
* of the changes SSRC yet because we don't implement pad-alloc. */
GST_LOG ("updating SSRC from %08x to %08x, fix the payloader", ssrc,
src->ssrc);
gst_rtp_buffer_set_ssrc (buffer, src->ssrc);
}
GST_LOG ("pushing RTP packet %" G_GUINT64_FORMAT, src->stats.packets_sent);
result = src->callbacks.push_rtp (src, buffer, src->user_data);
if (is_list) {
ssrc = gst_rtp_buffer_list_get_ssrc (list);
} else {
GST_WARNING ("no callback installed, dropping packet");
gst_buffer_unref (buffer);
ssrc = gst_rtp_buffer_get_ssrc (buffer);
}
if (ssrc != src->ssrc) {
/* the SSRC of the packet is not correct, make a writable buffer and
* update the SSRC. This could involve a complete copy of the packet when
* it is not writable. Usually the payloader will use caps negotiation to
* get the correct SSRC from the session manager before pushing anything. */
/* FIXME, we don't want to warn yet because we can't inform any payloader
* of the changes SSRC yet because we don't implement pad-alloc. */
GST_LOG ("updating SSRC from %08x to %08x, fix the payloader", ssrc,
src->ssrc);
if (is_list) {
list = gst_buffer_list_make_writable (list);
gst_buffer_list_foreach (list, (GstBufferListFunc) set_ssrc, src);
} else {
set_ssrc (&buffer, 0, 0, src);
}
}
GST_LOG ("pushing RTP %s %" G_GUINT64_FORMAT, is_list ? "list" : "packet",
src->stats.packets_sent);
result = src->callbacks.push_rtp (src, data, src->user_data);
return result;
/* ERRORS */
no_buffer:
{
GST_WARNING ("no buffers in buffer list");
gst_mini_object_unref (GST_MINI_OBJECT_CAST (data));
return GST_FLOW_OK;
}
no_callback:
{
GST_WARNING ("no callback installed, dropping packet");
gst_mini_object_unref (GST_MINI_OBJECT_CAST (data));
return GST_FLOW_OK;
}
}
/**

2
gst/rtpmanager/rtpsource.h
View file

@ -194,7 +194,7 @@ void rtp_source_set_rtcp_from (RTPSource *src, GstNetAddress *a
/* handling RTP */
GstFlowReturn rtp_source_process_rtp (RTPSource *src, GstBuffer *buffer, RTPArrivalStats *arrival);
GstFlowReturn rtp_source_send_rtp (RTPSource *src, GstBuffer *buffer, guint64 ntpnstime);
GstFlowReturn rtp_source_send_rtp (RTPSource *src, gpointer data, gboolean is_list, guint64 ntpnstime);
/* RTCP messages */
void rtp_source_process_bye (RTPSource *src, const gchar *reason);

8
tests/check/Makefile.am
View file

@ -84,6 +84,7 @@ check_PROGRAMS = \
elements/legacyresample \
elements/qtmux \
elements/rtpbin \
elements/rtpbin_buffer_list \
elements/selector \
elements/shapewipe \
elements/mxfdemux \
@ -108,6 +109,13 @@ elements_camerabin_LDADD = \
-lgstinterfaces-@GST_MAJORMINOR@
elements_camerabin_SOURCES = elements/camerabin.c
elements_rtpbin_buffer_list_CFLAGS = $(GST_PLUGINS_BASE_CFLAGS) $(GST_CFLAGS) \
$(ERROR_CFLAGS) $(GST_CHECK_CFLAGS)
elements_rtpbin_buffer_list_LDADD = $(GST_PLUGINS_BASE_LIBS) \
-lgstnetbuffer-@GST_MAJORMINOR@ -lgstrtp-@GST_MAJORMINOR@ \
$(GST_BASE_LIBS) $(GST_LIBS_LIBS) $(GST_CHECK_LIBS)
elements_rtpbin_buffer_list_SOURCES = elements/rtpbin_buffer_list.c
elements_timidity_CFLAGS = $(GST_BASE_CFLAGS) $(AM_CFLAGS)
elements_timidity_LDADD = $(GST_BASE_LIBS) $(LDADD)

331
tests/check/elements/rtpbin_buffer_list.c Normal file
View file

@ -0,0 +1,331 @@
/* GStreamer
*
* Unit test for gstrtpbin sending rtp packets using GstBufferList.
* Copyright (C) 2009 Branko Subasic <branko dot subasic at axis dot 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.
*/
#include <gst/check/gstcheck.h>
#include <gst/rtp/gstrtpbuffer.h>
/* This test makes sure that RTP packets sent as buffer lists are sent through
* the rtpbin as they are supposed to, and not corrupted in any way.
*/
#define TEST_CAPS \
"application/x-rtp, " \
"media=(string)video, " \
"clock-rate=(int)90000, " \
"encoding-name=(string)H264, " \
"profile-level-id=(string)4d4015, " \
"payload=(int)96, " \
"ssrc=(guint)2633237432, " \
"clock-base=(guint)1868267015, " \
"seqnum-base=(guint)54229"
/* RTP headers and the first 2 bytes of the payload (FU indicator and FU header)
*/
static const guint8 rtp_header[2][14] = {
{0x80, 0x60, 0xbb, 0xb7, 0x5c, 0xe9, 0x09,
0x0d, 0xf5, 0x9c, 0x43, 0x55, 0x1c, 0x86},
{0x80, 0x60, 0xbb, 0xb8, 0x5c, 0xe9, 0x09,
0x0d, 0xf5, 0x9c, 0x43, 0x55, 0x1c, 0x46}
};
static const guint rtp_header_len[] = {
sizeof rtp_header[0],
sizeof rtp_header[1]
};
static GstBuffer *header_buffer[2] = { NULL, NULL };
/* Some payload.
*/
static char *payload =
"0123456789ABSDEF0123456789ABSDEF0123456789ABSDEF0123456789ABSDEF0123456789ABSDEF"
"0123456789ABSDEF0123456789ABSDEF0123456789ABSDEF0123456789ABSDEF0123456789ABSDEF"
"0123456789ABSDEF0123456789ABSDEF0123456789ABSDEF0123456789ABSDEF0123456789ABSDEF"
"0123456789ABSDEF0123456789ABSDEF0123456789ABSDEF0123456789ABSDEF0123456789ABSDEF"
"0123456789ABSDEF0123456789ABSDEF0123456789ABSDEF0123456789ABSDEF0123456789ABSDEF"
"0123456789ABSDEF0123456789ABSDEF0123456789ABSDEF0123456789ABSDEF0123456789ABSDEF"
"0123456789ABSDEF0123456";
static const guint payload_offset[] = {
0, 498
};
static const guint payload_len[] = {
498, 5
};
static GstBuffer *original_buffer = NULL;
static GstStaticPadTemplate sinktemplate = GST_STATIC_PAD_TEMPLATE ("sink",
GST_PAD_SINK,
GST_PAD_ALWAYS,
GST_STATIC_CAPS ("application/x-rtp"));
static GstStaticPadTemplate srctemplate = GST_STATIC_PAD_TEMPLATE ("src",
GST_PAD_SRC,
GST_PAD_ALWAYS,
GST_STATIC_CAPS ("application/x-rtp"));
static GstBuffer *
_create_original_buffer (void)
{
GstCaps *caps;
if (original_buffer != NULL)
return original_buffer;
original_buffer = gst_buffer_new ();
fail_unless (original_buffer != NULL);
gst_buffer_set_data (original_buffer, (guint8 *) payload, strlen (payload));
GST_BUFFER_TIMESTAMP (original_buffer) =
gst_clock_get_internal_time (gst_system_clock_obtain ());
caps = gst_caps_from_string (TEST_CAPS);
fail_unless (caps != NULL);
gst_buffer_set_caps (original_buffer, caps);
gst_caps_unref (caps);
return original_buffer;
}
static GstBufferList *
_create_buffer_list (void)
{
GstBufferList *list;
GstBufferListIterator *it;
GstBuffer *orig_buffer;
GstBuffer *buffer;
orig_buffer = _create_original_buffer ();
fail_if (orig_buffer == NULL);
list = gst_buffer_list_new ();
fail_if (list == NULL);
it = gst_buffer_list_iterate (list);
fail_if (it == NULL);
/*** First group, i.e. first packet. **/
gst_buffer_list_iterator_add_group (it);
/* Create buffer with RTP header and add it to the 1st group */
buffer = gst_buffer_new ();
GST_BUFFER_MALLOCDATA (buffer) = g_memdup (&rtp_header[0], rtp_header_len[0]);
GST_BUFFER_DATA (buffer) = GST_BUFFER_MALLOCDATA (buffer);
GST_BUFFER_SIZE (buffer) = rtp_header_len[0];
gst_buffer_copy_metadata (buffer, orig_buffer, GST_BUFFER_COPY_ALL);
header_buffer[0] = buffer;
gst_buffer_list_iterator_add (it, buffer);
/* Create the payload buffer and add it to the 1st group
*/
buffer =
gst_buffer_create_sub (orig_buffer, payload_offset[0], payload_len[0]);
fail_if (buffer == NULL);
gst_buffer_list_iterator_add (it, buffer);
/*** Second group, i.e. second packet. ***/
/* Create a new group to hold the rtp header and the payload */
gst_buffer_list_iterator_add_group (it);
/* Create buffer with RTP header and add it to the 2nd group */
buffer = gst_buffer_new ();
GST_BUFFER_MALLOCDATA (buffer) = g_memdup (&rtp_header[1], rtp_header_len[1]);
GST_BUFFER_DATA (buffer) = GST_BUFFER_MALLOCDATA (buffer);
GST_BUFFER_SIZE (buffer) = rtp_header_len[1];
gst_buffer_copy_metadata (buffer, orig_buffer, GST_BUFFER_COPY_ALL);
header_buffer[1] = buffer;
/* Add the rtp header to the buffer list */
gst_buffer_list_iterator_add (it, buffer);
/* Create the payload buffer and add it to the 2d group
*/
buffer =
gst_buffer_create_sub (orig_buffer, payload_offset[1], payload_len[1]);
fail_if (buffer == NULL);
gst_buffer_list_iterator_add (it, buffer);
gst_buffer_list_iterator_free (it);
return list;
}
static void
_check_header (GstBuffer * buffer, guint index)
{
guint8 *data;
fail_if (buffer == NULL);
fail_unless (index < 2);
fail_unless (GST_BUFFER_SIZE (buffer) == rtp_header_len[index]);
/* Can't do a memcmp() on the whole header, cause the SSRC (bytes 8-11) will
* most likely be changed in gstrtpbin.
*/
fail_unless ((data = GST_BUFFER_DATA (buffer)) != NULL);
fail_unless_equals_uint64 (*(guint64 *) data, *(guint64 *) rtp_header[index]);
fail_unless (*(guint16 *) (data + 12) ==
*(guint16 *) (rtp_header[index] + 12));
}
static void
_check_payload (GstBuffer * buffer, guint index)
{
fail_if (buffer == NULL);
fail_unless (index < 2);
fail_unless (GST_BUFFER_SIZE (buffer) == payload_len[index]);
fail_if (GST_BUFFER_DATA (buffer) !=
(gpointer) (payload + payload_offset[index]));
fail_if (memcmp (GST_BUFFER_DATA (buffer), payload + payload_offset[index],
payload_len[index]));
}
static void
_check_group (GstBufferListIterator * it, guint index, GstCaps * caps)
{
GstBuffer *buffer;
fail_unless (it != NULL);
fail_unless (gst_buffer_list_iterator_n_buffers (it) == 2);
fail_unless (caps != NULL);
fail_unless ((buffer = gst_buffer_list_iterator_next (it)) != NULL);
fail_unless (GST_BUFFER_TIMESTAMP (buffer) ==
GST_BUFFER_TIMESTAMP (original_buffer));
fail_unless (gst_caps_is_equal (GST_BUFFER_CAPS (original_buffer),
GST_BUFFER_CAPS (buffer)));
_check_header (buffer, index);
fail_unless ((buffer = gst_buffer_list_iterator_next (it)) != NULL);
_check_payload (buffer, index);
}
static GstFlowReturn
_sink_chain_list (GstPad * pad, GstBufferList * list)
{
GstCaps *caps;
GstBufferListIterator *it;
caps = gst_caps_from_string (TEST_CAPS);
fail_unless (caps != NULL);
fail_unless (GST_IS_BUFFER_LIST (list));
fail_unless (gst_buffer_list_n_groups (list) == 2);
it = gst_buffer_list_iterate (list);
fail_if (it == NULL);
fail_unless (gst_buffer_list_iterator_next_group (it));
_check_group (it, 0, caps);
fail_unless (gst_buffer_list_iterator_next_group (it));
_check_group (it, 1, caps);
gst_caps_unref (caps);
gst_buffer_list_iterator_free (it);
gst_buffer_list_unref (list);
return GST_FLOW_OK;
}
static void
_set_chain_functions (GstPad * pad)
{
gst_pad_set_chain_list_function (pad, _sink_chain_list);
}
GST_START_TEST (test_bufferlist)
{
GstElement *rtpbin;
GstPad *sinkpad;
GstPad *srcpad;
GstBufferList *list;
list = _create_buffer_list ();
fail_unless (list != NULL);
rtpbin = gst_check_setup_element ("gstrtpbin");
srcpad =
gst_check_setup_src_pad_by_name (rtpbin, &srctemplate, "send_rtp_sink_0");
fail_if (srcpad == NULL);
sinkpad =
gst_check_setup_sink_pad_by_name (rtpbin, &sinktemplate,
"send_rtp_src_0");
fail_if (sinkpad == NULL);
_set_chain_functions (sinkpad);
gst_pad_set_active (sinkpad, TRUE);
gst_element_set_state (rtpbin, GST_STATE_PLAYING);
fail_unless (gst_pad_push_list (srcpad, list) == GST_FLOW_OK);
gst_pad_set_active (sinkpad, FALSE);
gst_check_teardown_pad_by_name (rtpbin, "send_rtp_src_0");
gst_check_teardown_pad_by_name (rtpbin, "send_rtp_sink_0");
gst_check_teardown_element (rtpbin);
}
GST_END_TEST;
static Suite *
bufferlist_suite (void)
{
Suite *s = suite_create ("BufferList");
TCase *tc_chain = tcase_create ("general");
/* time out after 30s. */
tcase_set_timeout (tc_chain, 10);
suite_add_tcase (s, tc_chain);
tcase_add_test (tc_chain, test_bufferlist);
return s;
}
GST_CHECK_MAIN (bufferlist);