mirrors/gstreamer
mirrors/gstreamer
1
1
Fork 0
mirror of https://gitlab.freedesktop.org/gstreamer/gstreamer.git synced 2024-11-23 18:21:04 +00:00
Code Issues 1 Projects Releases Wiki Activity
gstreamer/subprojects/gst-plugins-base/tests/check/libs/rtpbasepayload.c
Nirbheek Chauhan 146111d7c2 rtpbasepayload: Remove dead twcc code 
This feature was removed in 7a53fbad68,
but this code was left behind.

Part-of: <https://gitlab.freedesktop.org/gstreamer/gstreamer/-/merge_requests/1902>
2022-03-10 11:27:49 +00:00

2365 lines
76 KiB
C
Raw Blame History

/* GStreamer RTP base payloader unit tests
* Copyright (C) 2014 Sebastian Rasmussen <sebras@hotmail.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., 51 Franklin St, Fifth Floor,
* Boston, MA 02110-1301, USA.
*/
#ifdef HAVE_CONFIG_H
#include "config.h"
#endif
#include <gst/gst.h>
#include <gst/check/gstcheck.h>
#include <gst/check/gstharness.h>
#include <gst/rtp/rtp.h>
#include "rtpdummyhdrextimpl.c"
#define DEFAULT_CLOCK_RATE (42)
#define BUFFER_BEFORE_LIST (10)
/* GstRtpDummyPay */
#define GST_TYPE_RTP_DUMMY_PAY \
(gst_rtp_dummy_pay_get_type())
#define GST_RTP_DUMMY_PAY(obj) \
(G_TYPE_CHECK_INSTANCE_CAST((obj),GST_TYPE_RTP_DUMMY_PAY,GstRtpDummyPay))
#define GST_RTP_DUMMY_PAY_CLASS(klass) \
(G_TYPE_CHECK_CLASS_CAST((klass),GST_TYPE_RTP_DUMMY_PAY,GstRtpDummyPayClass))
#define GST_IS_RTP_DUMMY_PAY(obj) \
(G_TYPE_CHECK_INSTANCE_TYPE((obj),GST_TYPE_RTP_DUMMY_PAY))
#define GST_IS_RTP_DUMMY_PAY_CLASS(klass) \
(G_TYPE_CHECK_CLASS_TYPE((klass),GST_TYPE_RTP_DUMMY_PAY))
typedef struct _GstRtpDummyPay GstRtpDummyPay;
typedef struct _GstRtpDummyPayClass GstRtpDummyPayClass;
struct _GstRtpDummyPay
{
GstRTPBasePayload payload;
};
struct _GstRtpDummyPayClass
{
GstRTPBasePayloadClass parent_class;
};
GType gst_rtp_dummy_pay_get_type (void);
G_DEFINE_TYPE (GstRtpDummyPay, gst_rtp_dummy_pay, GST_TYPE_RTP_BASE_PAYLOAD);
static GstFlowReturn gst_rtp_dummy_pay_handle_buffer (GstRTPBasePayload * pay,
GstBuffer * buffer);
static GstStaticPadTemplate gst_rtp_dummy_pay_sink_template =
GST_STATIC_PAD_TEMPLATE ("sink",
GST_PAD_SINK,
GST_PAD_ALWAYS,
GST_STATIC_CAPS_ANY);
static GstStaticPadTemplate gst_rtp_dummy_pay_src_template =
GST_STATIC_PAD_TEMPLATE ("src",
GST_PAD_SRC,
GST_PAD_ALWAYS,
GST_STATIC_CAPS ("application/x-rtp"));
static void
gst_rtp_dummy_pay_class_init (GstRtpDummyPayClass * klass)
{
GstElementClass *gstelement_class;
GstRTPBasePayloadClass *gstrtpbasepayload_class;
gstelement_class = GST_ELEMENT_CLASS (klass);
gstrtpbasepayload_class = GST_RTP_BASE_PAYLOAD_CLASS (klass);
gst_element_class_add_static_pad_template (gstelement_class,
&gst_rtp_dummy_pay_sink_template);
gst_element_class_add_static_pad_template (gstelement_class,
&gst_rtp_dummy_pay_src_template);
gstrtpbasepayload_class->handle_buffer = gst_rtp_dummy_pay_handle_buffer;
}
static void
gst_rtp_dummy_pay_init (GstRtpDummyPay * pay)
{
gst_rtp_base_payload_set_options (GST_RTP_BASE_PAYLOAD (pay), "application",
TRUE, "dummy", DEFAULT_CLOCK_RATE);
}
static GstRtpDummyPay *
rtp_dummy_pay_new (void)
{
return g_object_new (GST_TYPE_RTP_DUMMY_PAY, NULL);
}
static GstFlowReturn
gst_rtp_dummy_pay_handle_buffer (GstRTPBasePayload * pay, GstBuffer * buffer)
{
GstBuffer *paybuffer;
GST_LOG ("payloading %" GST_PTR_FORMAT, buffer);
if (!gst_pad_has_current_caps (GST_RTP_BASE_PAYLOAD_SRCPAD (pay))) {
if (!gst_rtp_base_payload_set_outcaps (GST_RTP_BASE_PAYLOAD (pay),
"custom-caps", G_TYPE_UINT, DEFAULT_CLOCK_RATE, NULL)) {
gst_buffer_unref (buffer);
return GST_FLOW_NOT_NEGOTIATED;
}
}
paybuffer =
gst_rtp_base_payload_allocate_output_buffer (GST_RTP_BASE_PAYLOAD (pay),
0, 0, 0);
GST_BUFFER_PTS (paybuffer) = GST_BUFFER_PTS (buffer);
GST_BUFFER_OFFSET (paybuffer) = GST_BUFFER_OFFSET (buffer);
gst_buffer_append (paybuffer, buffer);
GST_LOG ("payloaded %" GST_PTR_FORMAT, paybuffer);
if (GST_BUFFER_PTS (paybuffer) < BUFFER_BEFORE_LIST) {
return gst_rtp_base_payload_push (pay, paybuffer);
} else {
GstBufferList *list = gst_buffer_list_new ();
gst_buffer_list_add (list, paybuffer);
return gst_rtp_base_payload_push_list (pay, list);
}
}
/* Helper functions and global state */
static GstStaticPadTemplate srctmpl = GST_STATIC_PAD_TEMPLATE ("src",
GST_PAD_SRC,
GST_PAD_ALWAYS,
GST_STATIC_CAPS_ANY);
static GstStaticPadTemplate sinktmpl = GST_STATIC_PAD_TEMPLATE ("sink",
GST_PAD_SINK,
GST_PAD_ALWAYS,
GST_STATIC_CAPS_ANY);
static GstStaticPadTemplate special_sinktmpl = GST_STATIC_PAD_TEMPLATE ("sink",
GST_PAD_SINK,
GST_PAD_ALWAYS,
GST_STATIC_CAPS ("application/x-rtp, payload=(int)98, ssrc=(uint)24, "
"timestamp-offset=(uint)212, seqnum-offset=(uint)2424"));
typedef struct State State;
struct State
{
GstElement *element;
GstPad *sinkpad;
GstPad *srcpad;
};
static GList *events;
static gboolean
event_func (GstPad * pad, GstObject * noparent, GstEvent * event)
{
events = g_list_append (events, gst_event_ref (event));
return gst_pad_event_default (pad, noparent, event);
}
static void
drop_events (void)
{
while (events != NULL) {
gst_event_unref (GST_EVENT (events->data));
events = g_list_delete_link (events, events);
}
}
static void
validate_events_received (guint received)
{
fail_unless_equals_int (g_list_length (events), received);
}
static void
validate_event (guint index, const gchar * name, const gchar * field, ...)
{
GstEvent *event;
va_list var_args;
fail_if (index >= g_list_length (events));
event = GST_EVENT (g_list_nth_data (events, index));
fail_if (event == NULL);
GST_TRACE ("%" GST_PTR_FORMAT, event);
fail_unless_equals_string (GST_EVENT_TYPE_NAME (event), name);
va_start (var_args, field);
while (field) {
if (!g_strcmp0 (field, "timestamp")) {
GstClockTime expected = va_arg (var_args, GstClockTime);
GstClockTime timestamp, duration;
gst_event_parse_gap (event, &timestamp, &duration);
fail_unless_equals_uint64 (timestamp, expected);
} else if (!g_strcmp0 (field, "duration")) {
GstClockTime expected = va_arg (var_args, GstClockTime);
GstClockTime timestamp, duration;
gst_event_parse_gap (event, &timestamp, &duration);
fail_unless_equals_uint64 (duration, expected);
} else if (!g_strcmp0 (field, "time")) {
GstClockTime expected = va_arg (var_args, GstClockTime);
const GstSegment *segment;
gst_event_parse_segment (event, &segment);
fail_unless_equals_uint64 (segment->time, expected);
} else if (!g_strcmp0 (field, "start")) {
GstClockTime expected = va_arg (var_args, GstClockTime);
const GstSegment *segment;
gst_event_parse_segment (event, &segment);
fail_unless_equals_uint64 (segment->start, expected);
} else if (!g_strcmp0 (field, "stop")) {
GstClockTime expected = va_arg (var_args, GstClockTime);
const GstSegment *segment;
gst_event_parse_segment (event, &segment);
fail_unless_equals_uint64 (segment->stop, expected);
} else if (!g_strcmp0 (field, "applied-rate")) {
gdouble expected = va_arg (var_args, gdouble);
const GstSegment *segment;
gst_event_parse_segment (event, &segment);
fail_unless_equals_float (segment->applied_rate, expected);
} else if (!g_strcmp0 (field, "rate")) {
gdouble expected = va_arg (var_args, gdouble);
const GstSegment *segment;
gst_event_parse_segment (event, &segment);
fail_unless_equals_float (segment->rate, expected);
} else if (!g_strcmp0 (field, "media-type")) {
const gchar *expected = va_arg (var_args, const gchar *);
GstCaps *caps;
const gchar *media_type;
gst_event_parse_caps (event, &caps);
media_type = gst_structure_get_name (gst_caps_get_structure (caps, 0));
fail_unless_equals_string (media_type, expected);
} else if (!g_strcmp0 (field, "npt-start")) {
GstClockTime expected = va_arg (var_args, GstClockTime);
GstCaps *caps;
GstClockTime start;
gst_event_parse_caps (event, &caps);
fail_unless (gst_structure_get_clock_time (gst_caps_get_structure (caps,
0), "npt-start", &start));
fail_unless_equals_uint64 (start, expected);
} else if (!g_strcmp0 (field, "npt-stop")) {
GstClockTime expected = va_arg (var_args, GstClockTime);
GstCaps *caps;
GstClockTime stop;
gst_event_parse_caps (event, &caps);
fail_unless (gst_structure_get_clock_time (gst_caps_get_structure (caps,
0), "npt-stop", &stop));
fail_unless_equals_uint64 (stop, expected);
} else if (!g_strcmp0 (field, "play-speed")) {
gdouble expected = va_arg (var_args, gdouble);
GstCaps *caps;
gdouble speed;
gst_event_parse_caps (event, &caps);
fail_unless (gst_structure_get_double (gst_caps_get_structure (caps, 0),
"play-speed", &speed));
fail_unless (speed == expected);
} else if (!g_strcmp0 (field, "play-scale")) {
gdouble expected = va_arg (var_args, gdouble);
GstCaps *caps;
gdouble scale;
gst_event_parse_caps (event, &caps);
fail_unless (gst_structure_get_double (gst_caps_get_structure (caps, 0),
"play-scale", &scale));
fail_unless (scale == expected);
} else if (!g_strcmp0 (field, "ssrc")) {
guint expected = va_arg (var_args, guint);
GstCaps *caps;
guint ssrc;
gst_event_parse_caps (event, &caps);
fail_unless (gst_structure_get_uint (gst_caps_get_structure (caps, 0),
"ssrc", &ssrc));
fail_unless_equals_int (ssrc, expected);
} else if (!g_strcmp0 (field, "a-framerate")) {
const gchar *expected = va_arg (var_args, const gchar *);
GstCaps *caps;
const gchar *framerate;
gst_event_parse_caps (event, &caps);
framerate = gst_structure_get_string (gst_caps_get_structure (caps, 0),
"a-framerate");
fail_unless_equals_string (framerate, expected);
} else if (!g_strcmp0 (field, "extmap-str")) {
guint ext_id = va_arg (var_args, guint);
const gchar *expected_ext_str = va_arg (var_args, const gchar *);
GstCaps *caps;
const gchar *ext_str;
gchar *ext_field = g_strdup_printf ("extmap-%u", ext_id);
gst_event_parse_caps (event, &caps);
ext_str = gst_structure_get_string (gst_caps_get_structure (caps, 0),
ext_field);
fail_unless_equals_string (ext_str, expected_ext_str);
g_free (ext_field);
} else {
fail ("test cannot validate unknown event field '%s'", field);
}
field = va_arg (var_args, const gchar *);
}
va_end (var_args);
}
static void
validate_normal_start_events (guint index)
{
validate_event (index, "stream-start", NULL);
validate_event (index + 1, "caps", "media-type", "application/x-rtp", NULL);
validate_event (index + 2, "segment",
"time", G_GUINT64_CONSTANT (0),
"start", G_GUINT64_CONSTANT (0), "stop", G_MAXUINT64, NULL);
}
#define push_buffer(state, field, ...) \
push_buffer_full ((state), GST_FLOW_OK, (field), __VA_ARGS__)
#define push_buffer_fails(state, error, field, ...) \
push_buffer_full ((state), (error), (field), __VA_ARGS__)
static void
push_buffer_full (State * state, GstFlowReturn expected,
const gchar * field, ...)
{
GstBuffer *buf = gst_buffer_new_allocate (0, 0, 0);
GstRTPBuffer rtp = GST_RTP_BUFFER_INIT;
gboolean mapped = FALSE;
va_list var_args;
va_start (var_args, field);
while (field) {
if (!g_strcmp0 (field, "pts")) {
GstClockTime pts = va_arg (var_args, GstClockTime);
GST_BUFFER_PTS (buf) = pts;
} else if (!g_strcmp0 (field, "offset")) {
guint64 offset = va_arg (var_args, guint64);
GST_BUFFER_OFFSET (buf) = offset;
} else if (!g_strcmp0 (field, "discont")) {
gboolean discont = va_arg (var_args, gboolean);
if (discont) {
GST_BUFFER_FLAG_SET (buf, GST_BUFFER_FLAG_DISCONT);
} else {
GST_BUFFER_FLAG_UNSET (buf, GST_BUFFER_FLAG_DISCONT);
}
} else {
if (!mapped) {
gst_rtp_buffer_map (buf, GST_MAP_WRITE, &rtp);
mapped = TRUE;
}
if (!g_strcmp0 (field, "rtptime")) {
guint32 rtptime = va_arg (var_args, guint);
gst_rtp_buffer_set_timestamp (&rtp, rtptime);
} else if (!g_strcmp0 (field, "payload-type")) {
guint payload_type = va_arg (var_args, guint);
gst_rtp_buffer_set_payload_type (&rtp, payload_type);
} else if (!g_strcmp0 (field, "seq")) {
guint seq = va_arg (var_args, guint);
gst_rtp_buffer_set_seq (&rtp, seq);
} else if (!g_strcmp0 (field, "ssrc")) {
guint32 ssrc = va_arg (var_args, guint);
gst_rtp_buffer_set_ssrc (&rtp, ssrc);
} else {
fail ("test cannot set unknown buffer field '%s'", field);
}
}
field = va_arg (var_args, const gchar *);
}
va_end (var_args);
if (mapped) {
gst_rtp_buffer_unmap (&rtp);
}
fail_unless_equals_int (gst_pad_push (state->srcpad, buf), expected);
}
static void
push_buffer_list (State * state, const gchar * field, ...)
{
GstBuffer *buf = gst_rtp_buffer_new_allocate (0, 0, 0);
GstRTPBuffer rtp = GST_RTP_BUFFER_INIT;
gboolean mapped = FALSE;
GstBufferList *list;
va_list var_args;
va_start (var_args, field);
while (field) {
if (!g_strcmp0 (field, "pts")) {
GstClockTime pts = va_arg (var_args, GstClockTime);
GST_BUFFER_PTS (buf) = pts;
} else if (!g_strcmp0 (field, "offset")) {
guint64 offset = va_arg (var_args, guint64);
GST_BUFFER_OFFSET (buf) = offset;
} else if (!g_strcmp0 (field, "discont")) {
gboolean discont = va_arg (var_args, gboolean);
if (discont) {
GST_BUFFER_FLAG_SET (buf, GST_BUFFER_FLAG_DISCONT);
} else {
GST_BUFFER_FLAG_UNSET (buf, GST_BUFFER_FLAG_DISCONT);
}
} else {
if (!mapped) {
gst_rtp_buffer_map (buf, GST_MAP_WRITE, &rtp);
mapped = TRUE;
}
if (!g_strcmp0 (field, "rtptime")) {
guint32 rtptime = va_arg (var_args, guint);
gst_rtp_buffer_set_timestamp (&rtp, rtptime);
} else if (!g_strcmp0 (field, "payload-type")) {
guint payload_type = va_arg (var_args, guint);
gst_rtp_buffer_set_payload_type (&rtp, payload_type);
} else if (!g_strcmp0 (field, "seq")) {
guint seq = va_arg (var_args, guint);
gst_rtp_buffer_set_seq (&rtp, seq);
} else if (!g_strcmp0 (field, "ssrc")) {
guint32 ssrc = va_arg (var_args, guint);
gst_rtp_buffer_set_ssrc (&rtp, ssrc);
} else {
fail ("test cannot set unknown buffer field '%s'", field);
}
}
field = va_arg (var_args, const gchar *);
}
va_end (var_args);
if (mapped) {
gst_rtp_buffer_unmap (&rtp);
}
list = gst_buffer_list_new ();
gst_buffer_list_add (list, buf);
fail_unless_equals_int (gst_pad_push_list (state->srcpad, list), GST_FLOW_OK);
}
static void
validate_buffers_received (guint received_buffers)
{
fail_unless_equals_int (g_list_length (buffers), received_buffers);
}
static void
validate_buffer_valist (GstBuffer * buf, const gchar * field, va_list var_args)
{
GstRTPBuffer rtp = GST_RTP_BUFFER_INIT;
gboolean mapped = FALSE;
while (field) {
if (!g_strcmp0 (field, "pts")) {
GstClockTime pts = va_arg (var_args, GstClockTime);
fail_unless_equals_uint64 (GST_BUFFER_PTS (buf), pts);
} else if (!g_strcmp0 (field, "offset")) {
guint64 offset = va_arg (var_args, guint64);
fail_unless_equals_uint64 (GST_BUFFER_OFFSET (buf), offset);
} else if (!g_strcmp0 (field, "discont")) {
gboolean discont = va_arg (var_args, gboolean);
if (discont) {
fail_unless (GST_BUFFER_FLAG_IS_SET (buf, GST_BUFFER_FLAG_DISCONT));
} else {
fail_if (GST_BUFFER_FLAG_IS_SET (buf, GST_BUFFER_FLAG_DISCONT));
}
} else if (!g_strcmp0 (field, "size")) {
gsize expected_size = va_arg (var_args, gsize);
fail_unless_equals_int64 ((guint64) expected_size,
gst_buffer_get_size (buf));
} else {
if (!mapped) {
gst_rtp_buffer_map (buf, GST_MAP_READ, &rtp);
mapped = TRUE;
}
if (!g_strcmp0 (field, "rtptime")) {
guint32 rtptime = va_arg (var_args, guint);
fail_unless_equals_int (gst_rtp_buffer_get_timestamp (&rtp), rtptime);
} else if (!g_strcmp0 (field, "payload-type")) {
guint pt = va_arg (var_args, guint);
fail_unless_equals_int (gst_rtp_buffer_get_payload_type (&rtp), pt);
} else if (!g_strcmp0 (field, "seq")) {
guint seq = va_arg (var_args, guint);
fail_unless_equals_int (gst_rtp_buffer_get_seq (&rtp), seq);
} else if (!g_strcmp0 (field, "ssrc")) {
guint32 ssrc = va_arg (var_args, guint);
fail_unless_equals_int (gst_rtp_buffer_get_ssrc (&rtp), ssrc);
} else if (!g_strcmp0 (field, "csrc")) {
guint idx = va_arg (var_args, guint);
guint csrc = va_arg (var_args, guint);
fail_unless_equals_int (gst_rtp_buffer_get_csrc (&rtp, idx), csrc);
} else if (!g_strcmp0 (field, "csrc-count")) {
guint csrc_count = va_arg (var_args, guint);
fail_unless_equals_int (gst_rtp_buffer_get_csrc_count (&rtp),
csrc_count);
} else if (!g_strcmp0 (field, "ext-data")) {
guint expected_bits = va_arg (var_args, guint) & 0xFFFF;
gsize expected_size = va_arg (var_args, gsize);
gpointer data;
guint word_len;
guint16 ext_bits;
gst_rtp_buffer_get_extension_data (&rtp, &ext_bits, &data, &word_len);
GST_MEMDUMP ("ext data", data, word_len * 4);
fail_unless_equals_int (expected_bits, ext_bits);
fail_unless_equals_int64 ((guint64) expected_size,
(guint64) word_len * 4);
} else {
fail ("test cannot validate unknown buffer field '%s'", field);
}
}
field = va_arg (var_args, const gchar *);
}
if (mapped) {
gst_rtp_buffer_unmap (&rtp);
}
}
static void
validate_buffer1 (GstBuffer * buf, const gchar * field, ...)
{
va_list var_args;
va_start (var_args, field);
validate_buffer_valist (buf, field, var_args);
va_end (var_args);
}
static void
validate_buffer (guint index, const gchar * field, ...)
{
GstBuffer *buf;
va_list var_args;
fail_if (index >= g_list_length (buffers));
buf = GST_BUFFER (g_list_nth_data (buffers, index));
fail_if (buf == NULL);
GST_TRACE ("%" GST_PTR_FORMAT, buf);
va_start (var_args, field);
validate_buffer_valist (buf, field, var_args);
va_end (var_args);
}
static void
get_buffer_field (guint index, const gchar * field, ...)
{
GstBuffer *buf;
GstRTPBuffer rtp = GST_RTP_BUFFER_INIT;
gboolean mapped = FALSE;
va_list var_args;
fail_if (index >= g_list_length (buffers));
buf = GST_BUFFER (g_list_nth_data (buffers, (index)));
fail_if (buf == NULL);
va_start (var_args, field);
while (field) {
if (!g_strcmp0 (field, "pts")) {
GstClockTime *pts = va_arg (var_args, GstClockTime *);
*pts = GST_BUFFER_PTS (buf);
} else if (!g_strcmp0 (field, "offset")) {
guint64 *offset = va_arg (var_args, guint64 *);
*offset = GST_BUFFER_OFFSET (buf);
} else if (!g_strcmp0 (field, "discont")) {
gboolean *discont = va_arg (var_args, gboolean *);
*discont = GST_BUFFER_FLAG_IS_SET (buf, GST_BUFFER_FLAG_DISCONT);
} else {
if (!mapped) {
gst_rtp_buffer_map (buf, GST_MAP_READ, &rtp);
mapped = TRUE;
}
if (!g_strcmp0 (field, "rtptime")) {
guint32 *rtptime = va_arg (var_args, guint32 *);
*rtptime = gst_rtp_buffer_get_timestamp (&rtp);
} else if (!g_strcmp0 (field, "payload-type")) {
guint *pt = va_arg (var_args, guint *);
*pt = gst_rtp_buffer_get_payload_type (&rtp);
} else if (!g_strcmp0 (field, "seq")) {
guint16 *seq = va_arg (var_args, guint16 *);
*seq = gst_rtp_buffer_get_seq (&rtp);
} else if (!g_strcmp0 (field, "ssrc")) {
guint32 *ssrc = va_arg (var_args, guint32 *);
*ssrc = gst_rtp_buffer_get_ssrc (&rtp);
} else {
fail ("test retrieve validate unknown buffer field '%s'", field);
}
}
field = va_arg (var_args, const gchar *);
}
va_end (var_args);
if (mapped)
gst_rtp_buffer_unmap (&rtp);
}
static State *
create_payloader (const gchar * caps_str,
GstStaticPadTemplate * sinktmpl, const gchar * property, ...)
{
va_list var_args;
GstCaps *caps;
State *state;
state = g_new0 (State, 1);
state->element = GST_ELEMENT (rtp_dummy_pay_new ());
fail_unless (GST_IS_RTP_DUMMY_PAY (state->element));
va_start (var_args, property);
g_object_set_valist (G_OBJECT (state->element), property, var_args);
va_end (var_args);
state->srcpad = gst_check_setup_src_pad (state->element, &srctmpl);
state->sinkpad = gst_check_setup_sink_pad (state->element, sinktmpl);
fail_unless (gst_pad_set_active (state->srcpad, TRUE));
fail_unless (gst_pad_set_active (state->sinkpad, TRUE));
caps = gst_caps_from_string (caps_str);
gst_check_setup_events (state->srcpad, state->element, caps, GST_FORMAT_TIME);
gst_caps_unref (caps);
gst_pad_set_chain_function (state->sinkpad, gst_check_chain_func);
gst_pad_set_event_function (state->sinkpad, event_func);
return state;
}
static void
set_state (State * state, GstState new_state)
{
fail_unless_equals_int (gst_element_set_state (state->element, new_state),
GST_STATE_CHANGE_SUCCESS);
}
static void
validate_would_not_be_filled (State * state, guint size, GstClockTime duration)
{
GstRTPBasePayload *basepay;
basepay = GST_RTP_BASE_PAYLOAD (state->element);
fail_if (gst_rtp_base_payload_is_filled (basepay, size, duration));
}
static void
validate_would_be_filled (State * state, guint size, GstClockTime duration)
{
GstRTPBasePayload *basepay;
basepay = GST_RTP_BASE_PAYLOAD (state->element);
fail_unless (gst_rtp_base_payload_is_filled (basepay, size, duration));
}
static void
ssrc_collision (State * state, guint ssrc,
gboolean have_new_ssrc, guint new_ssrc)
{
GstStructure *s;
GstEvent *event;
if (have_new_ssrc) {
s = gst_structure_new ("GstRTPCollision",
"ssrc", G_TYPE_UINT, ssrc,
"suggested-ssrc", G_TYPE_UINT, new_ssrc, NULL);
} else {
s = gst_structure_new ("GstRTPCollision", "ssrc", G_TYPE_UINT, ssrc, NULL);
}
event = gst_event_new_custom (GST_EVENT_CUSTOM_UPSTREAM, s);
fail_unless (gst_pad_push_event (state->sinkpad, event));
}
static void
reconfigure (State * state)
{
GstEvent *event;
event = gst_event_new_reconfigure ();
fail_unless (gst_pad_push_event (state->sinkpad, event));
}
static void
validate_stats (State * state, guint clock_rate,
GstClockTime running_time, guint16 seq, guint32 rtptime)
{
GstStructure *stats;
g_object_get (state->element, "stats", &stats, NULL);
fail_unless_equals_int (g_value_get_uint (gst_structure_get_value (stats,
"clock-rate")), clock_rate);
fail_unless_equals_uint64 (g_value_get_uint64 (gst_structure_get_value (stats,
"running-time")), running_time);
fail_unless_equals_int (g_value_get_uint (gst_structure_get_value (stats,
"seqnum")), seq);
fail_unless_equals_int (g_value_get_uint (gst_structure_get_value (stats,
"timestamp")), rtptime);
gst_structure_free (stats);
}
static void
destroy_payloader (State * state)
{
gst_check_teardown_sink_pad (state->element);
gst_check_teardown_src_pad (state->element);
gst_check_drop_buffers ();
drop_events ();
g_object_unref (state->element);
g_free (state);
}
/* Tests */
/* push two buffers to the payloader which should successfully payload them
* into RTP packets. the first packet will have a random rtptime and sequence
* number, but the last packet should have an rtptime incremented by
* DEFAULT_CLOCK_RATE and a sequence number incremented by one because the
* packets are sequential. besides the two payloaded RTP packets there should
* be the three events initial events: stream-start, caps and segment.
*/
GST_START_TEST (rtp_base_payload_buffer_test)
{
State *state;
guint32 rtptime;
guint16 seq;
state = create_payloader ("application/x-rtp", &sinktmpl,
"perfect-rtptime", FALSE, NULL);
set_state (state, GST_STATE_PLAYING);
push_buffer (state, "pts", 0 * GST_SECOND, NULL);
push_buffer (state, "pts", 1 * GST_SECOND, NULL);
set_state (state, GST_STATE_NULL);
validate_buffers_received (2);
validate_buffer (0, "pts", 0 * GST_SECOND, NULL);
get_buffer_field (0, "rtptime", &rtptime, "seq", &seq, NULL);
validate_buffer (1,
"pts", 1 * GST_SECOND,
"rtptime", rtptime + 1 * DEFAULT_CLOCK_RATE, "seq", seq + 1, NULL);
validate_events_received (3);
validate_normal_start_events (0);
destroy_payloader (state);
}
GST_END_TEST;
/* push single buffers in buffer lists to the payloader to be payloaded into
* RTP packets. the dummy payloader will start pushing buffer lists itself
* after BUFFER_BEFORE_LIST payloaded RTP packets. any RTP packets included in
* buffer lists should have rtptime and sequence numbers incrementting in the
* same way as for separate RTP packets.
*/
GST_START_TEST (rtp_base_payload_buffer_list_test)
{
State *state;
guint32 rtptime;
guint16 seq;
guint i;
state = create_payloader ("application/x-rtp", &sinktmpl, NULL);
set_state (state, GST_STATE_PLAYING);
for (i = 0; i < BUFFER_BEFORE_LIST + 1; i++) {
push_buffer_list (state, "pts", i * GST_SECOND, NULL);
}
set_state (state, GST_STATE_NULL);
validate_buffers_received (11);
validate_buffer (0, "pts", 0 * GST_SECOND, NULL);
get_buffer_field (0, "rtptime", &rtptime, "seq", &seq, NULL);
for (i = 1; i < BUFFER_BEFORE_LIST + 1; i++) {
validate_buffer (i,
"pts", i * GST_SECOND,
"rtptime", rtptime + i * DEFAULT_CLOCK_RATE, "seq", seq + i, NULL);
}
validate_events_received (3);
validate_normal_start_events (0);
destroy_payloader (state);
}
GST_END_TEST;
/* push two buffers. because the payloader is using non-perfect rtptime the
* second buffer will be timestamped with the default clock and ignore any
* offset set on the buffers being payloaded.
*/
GST_START_TEST (rtp_base_payload_normal_rtptime_test)
{
guint32 rtptime;
State *state;
state = create_payloader ("application/x-rtp", &sinktmpl,
"perfect-rtptime", FALSE, NULL);
set_state (state, GST_STATE_PLAYING);
push_buffer (state,
"pts", 0 * GST_SECOND, "offset", GST_BUFFER_OFFSET_NONE, NULL);
push_buffer (state,
"pts", 1 * GST_SECOND, "offset", GST_BUFFER_OFFSET_NONE, NULL);
set_state (state, GST_STATE_NULL);
validate_buffers_received (2);
validate_buffer (0,
"pts", 0 * GST_SECOND, "offset", GST_BUFFER_OFFSET_NONE, NULL);
get_buffer_field (0, "rtptime", &rtptime, NULL);
validate_buffer (1,
"pts", 1 * GST_SECOND,
"offset", GST_BUFFER_OFFSET_NONE,
"rtptime", rtptime + DEFAULT_CLOCK_RATE, NULL);
validate_events_received (3);
validate_normal_start_events (0);
destroy_payloader (state);
}
GST_END_TEST;
/* push two buffers. because the payloader is using perfect rtptime the
* second buffer will be timestamped with a timestamp incremented with the
* difference in offset between the first and second buffer. the pts will be
* ignored for any buffer after the first buffer.
*/
GST_START_TEST (rtp_base_payload_perfect_rtptime_test)
{
guint32 rtptime;
State *state;
state = create_payloader ("application/x-rtp", &sinktmpl,
"perfect-rtptime", TRUE, NULL);
set_state (state, GST_STATE_PLAYING);
push_buffer (state, "pts", 0 * GST_SECOND, "offset", G_GINT64_CONSTANT (0),
NULL);
push_buffer (state, "pts", GST_CLOCK_TIME_NONE, "offset",
G_GINT64_CONSTANT (21), NULL);
set_state (state, GST_STATE_NULL);
validate_buffers_received (2);
validate_buffer (0, "pts", 0 * GST_SECOND, "offset", G_GINT64_CONSTANT (0),
NULL);
get_buffer_field (0, "rtptime", &rtptime, NULL);
validate_buffer (1,
"pts", GST_CLOCK_TIME_NONE, "offset", G_GINT64_CONSTANT (21), "rtptime",
rtptime + 21, NULL);
validate_events_received (3);
validate_normal_start_events (0);
destroy_payloader (state);
}
GST_END_TEST;
/* validate that a payloader will re-use the last used timestamp when a buffer
* is using perfect rtptime and both the pushed buffers timestamp and the offset
* is NONE. the payloader is configuered to start with a specific timestamp.
* then a buffer is sent with a valid timestamp but without any offset. the
* payloded RTP packet is expected to use the specific timestamp. next another
* buffer is pushed with a normal timestamp set to illustrate that the payloaded
* RTP packet will have an increased timestamp. finally a buffer without any
* timestamp or offset is pushed. in this case the payloaded RTP packet is
* expected to have the same timestamp as the previously payloaded RTP packet.
*/
GST_START_TEST (rtp_base_payload_no_pts_no_offset_test)
{
State *state;
state = create_payloader ("application/x-rtp", &sinktmpl,
"timestamp-offset", 0x42, NULL);
set_state (state, GST_STATE_PLAYING);
push_buffer (state,
"pts", 0 * GST_SECOND, "offset", GST_BUFFER_OFFSET_NONE, NULL);
push_buffer (state,
"pts", 1 * GST_SECOND, "offset", GST_BUFFER_OFFSET_NONE, NULL);
push_buffer (state,
"pts", GST_CLOCK_TIME_NONE, "offset", GST_BUFFER_OFFSET_NONE, NULL);
set_state (state, GST_STATE_NULL);
validate_buffers_received (3);
validate_buffer (0,
"pts", 0 * GST_SECOND,
"offset", GST_BUFFER_OFFSET_NONE, "rtptime", 0x42, NULL);
validate_buffer (1,
"pts", 1 * GST_SECOND,
"offset", GST_BUFFER_OFFSET_NONE,
"rtptime", 0x42 + 1 * DEFAULT_CLOCK_RATE, NULL);
validate_buffer (2,
"pts", GST_CLOCK_TIME_NONE,
"offset", GST_BUFFER_OFFSET_NONE,
"rtptime", 0x42 + 1 * DEFAULT_CLOCK_RATE, NULL);
validate_events_received (3);
validate_normal_start_events (0);
destroy_payloader (state);
}
GST_END_TEST;
/* validate that a downstream element with caps on its sink pad can effectively
* configure the payloader's payload-type, ssrc, timestamp-offset and
* seqnum-offset properties and therefore also affect the payloaded RTP packets.
* this is done by connecting to a sink pad with template caps setting the
* relevant fields and then pushing a buffer and making sure that the payloaded
* RTP packet has the expected properties.
*/
GST_START_TEST (rtp_base_payload_downstream_caps_test)
{
State *state;
state = create_payloader ("application/x-rtp", &special_sinktmpl, NULL);
set_state (state, GST_STATE_PLAYING);
push_buffer (state, "pts", 0 * GST_SECOND, NULL);
set_state (state, GST_STATE_NULL);
validate_buffers_received (1);
validate_buffer (0,
"pts", 0 * GST_SECOND,
"seq", 2424, "payload-type", 98, "ssrc", 24, "rtptime", 212, NULL);
validate_events_received (3);
validate_normal_start_events (0);
destroy_payloader (state);
}
GST_END_TEST;
/* when a payloader receives a GstRTPCollision upstream event it should try to
* switch to a new ssrc for the next payloaded RTP packets. GstRTPCollision can
* supply a suggested new ssrc. if a suggested new ssrc is supplied then the
* payloaded is supposed to use this new ssrc, otherwise it should generate a
* new random ssrc which is not identical to the one that collided.
*
* this is tested by first setting the ssrc to a specific value and pushing a
* buffer. the payloaded RTP packet is validate to have the set ssrc. then a
* GstRTPCollision event is generated to instruct the payloader that the
* previously set ssrc collided. this event suggests a new ssrc and it is
* verified that a pushed buffer results in a payloaded RTP packet that actually
* uses this new ssrc. finally a new GstRTPCollision event is generated to
* indicate another ssrc collision. this time the event does not suggest a new
* ssrc. the payloaded RTP packet is then expected to have a new random ssrc
* different from the collided one.
*/
GST_START_TEST (rtp_base_payload_ssrc_collision_test)
{
State *state;
guint32 ssrc;
state = create_payloader ("application/x-rtp", &sinktmpl, NULL);
g_object_set (state->element, "ssrc", 0x4242, NULL);
g_object_get (state->element, "ssrc", &ssrc, NULL);
fail_unless_equals_int (ssrc, 0x4242);
set_state (state, GST_STATE_PLAYING);
push_buffer (state, "pts", 0 * GST_SECOND, NULL);
ssrc_collision (state, 0x4242, TRUE, 0x4343);
push_buffer (state, "pts", 1 * GST_SECOND, NULL);
ssrc_collision (state, 0x4343, FALSE, 0);
push_buffer (state, "pts", 2 * GST_SECOND, NULL);
set_state (state, GST_STATE_NULL);
validate_buffers_received (3);
validate_buffer (0, "pts", 0 * GST_SECOND, "ssrc", 0x4242, NULL);
validate_buffer (1, "pts", 1 * GST_SECOND, "ssrc", 0x4343, NULL);
validate_buffer (2, "pts", 2 * GST_SECOND, NULL);
get_buffer_field (2, "ssrc", &ssrc, NULL);
fail_if (ssrc == 0x4343);
validate_events_received (5);
validate_normal_start_events (0);
validate_event (3, "caps",
"media-type", "application/x-rtp", "ssrc", 0x4343, NULL);
validate_event (4, "caps",
"media-type", "application/x-rtp", "ssrc", ssrc, NULL);
destroy_payloader (state);
}
GST_END_TEST;
/* validate that an upstream event different from GstRTPCollision is successfully
* forwarded to upstream elements. in this test a caps reconfiguration event is
* pushed upstream to validate the behaviour.
*/
GST_START_TEST (rtp_base_payload_reconfigure_test)
{
State *state;
state = create_payloader ("application/x-rtp", &sinktmpl, NULL);
set_state (state, GST_STATE_PLAYING);
push_buffer (state, "pts", 0 * GST_SECOND, NULL);
reconfigure (state);
push_buffer (state, "pts", 1 * GST_SECOND, NULL);
set_state (state, GST_STATE_NULL);
validate_buffers_received (2);
validate_buffer (0, "pts", 0 * GST_SECOND, NULL);
validate_buffer (1, "pts", 1 * GST_SECOND, NULL);
validate_events_received (4);
validate_normal_start_events (0);
destroy_payloader (state);
}
GST_END_TEST;
/* validate that changing the mtu actually affects whether buffers are
* considered to be filled. first detect the default mtu and check that having
* buffers slightly less or equal to the size will not be considered to be
* filled, and that going over this size will be filling the buffers. then
* change the mtu slightly and validate that the boundary actually changed.
* lastly try the boundary values and make sure that they work as expected.
*/
GST_START_TEST (rtp_base_payload_property_mtu_test)
{
State *state;
guint mtu, check;
state = create_payloader ("application/x-rtp", &sinktmpl, NULL);
g_object_get (state->element, "mtu", &mtu, NULL);
validate_would_not_be_filled (state, mtu - 1, GST_CLOCK_TIME_NONE);
validate_would_not_be_filled (state, mtu, GST_CLOCK_TIME_NONE);
validate_would_be_filled (state, mtu + 1, GST_CLOCK_TIME_NONE);
g_object_set (state->element, "mtu", mtu - 1, NULL);
g_object_get (state->element, "mtu", &check, NULL);
fail_unless_equals_int (check, mtu - 1);
validate_would_not_be_filled (state, mtu - 1, GST_CLOCK_TIME_NONE);
validate_would_be_filled (state, mtu, GST_CLOCK_TIME_NONE);
validate_would_be_filled (state, mtu + 1, GST_CLOCK_TIME_NONE);
g_object_set (state->element, "mtu", 28, NULL);
g_object_get (state->element, "mtu", &check, NULL);
fail_unless_equals_int (check, 28);
validate_would_not_be_filled (state, 28, GST_CLOCK_TIME_NONE);
validate_would_be_filled (state, 29, GST_CLOCK_TIME_NONE);
g_object_set (state->element, "mtu", G_MAXUINT, NULL);
g_object_get (state->element, "mtu", &check, NULL);
fail_unless_equals_int (check, G_MAXUINT);
validate_would_not_be_filled (state, G_MAXUINT - 1, GST_CLOCK_TIME_NONE);
validate_would_not_be_filled (state, G_MAXUINT, GST_CLOCK_TIME_NONE);
destroy_payloader (state);
}
GST_END_TEST;
/* validate that changing the payload-type will actually affect the
* payload-type of the payloaded RTP packets. first get the default, then send
* a buffer with this payload-type. increment the payload-type and send another
* buffer. then test the boundary values for the payload-type and make sure
* that these are all carried over to the payloaded RTP packets.
*/
GST_START_TEST (rtp_base_payload_property_pt_test)
{
State *state;
guint payload_type, check;
state = create_payloader ("application/x-rtp", &sinktmpl, NULL);
set_state (state, GST_STATE_PLAYING);
g_object_get (state->element, "pt", &payload_type, NULL);
push_buffer (state, "pts", 0 * GST_SECOND, NULL);
g_object_set (state->element, "pt", payload_type + 1, NULL);
g_object_get (state->element, "pt", &check, NULL);
fail_unless_equals_int (check, payload_type + 1);
push_buffer (state, "pts", 1 * GST_SECOND, NULL);
g_object_set (state->element, "pt", 0, NULL);
g_object_get (state->element, "pt", &check, NULL);
fail_unless_equals_int (check, 0);
push_buffer (state, "pts", 2 * GST_SECOND, NULL);
g_object_set (state->element, "pt", 0x7f, NULL);
g_object_get (state->element, "pt", &check, NULL);
fail_unless_equals_int (check, 0x7f);
push_buffer (state, "pts", 3 * GST_SECOND, NULL);
set_state (state, GST_STATE_NULL);
validate_buffers_received (4);
validate_buffer (0,
"pts", 0 * GST_SECOND, "payload-type", payload_type, NULL);
validate_buffer (1,
"pts", 1 * GST_SECOND, "payload-type", payload_type + 1, NULL);
validate_buffer (2, "pts", 2 * GST_SECOND, "payload-type", 0, NULL);
validate_buffer (3, "pts", 3 * GST_SECOND, "payload-type", 0x7f, NULL);
validate_events_received (3);
validate_normal_start_events (0);
destroy_payloader (state);
}
GST_END_TEST;
/* validate that changing the ssrc will actually affect the ssrc of the
* payloaded RTP packets. first get the current ssrc which should indicate
* random ssrcs. send two buffers and expect their ssrcs to be random but
* identical. since setting the ssrc will only take effect when the pipeline
* goes READY->PAUSED, bring the pipeline to NULL state, set the ssrc to a given
* value and make sure that this is carried over to the payloaded RTP packets.
* the last step is to test the boundary values.
*/
GST_START_TEST (rtp_base_payload_property_ssrc_test)
{
State *state;
guint32 ssrc;
state = create_payloader ("application/x-rtp", &sinktmpl, NULL);
set_state (state, GST_STATE_PLAYING);
g_object_get (state->element, "ssrc", &ssrc, NULL);
fail_unless_equals_int (ssrc, -1);
push_buffer (state, "pts", 0 * GST_SECOND, NULL);
push_buffer (state, "pts", 1 * GST_SECOND, NULL);
set_state (state, GST_STATE_NULL);
g_object_set (state->element, "ssrc", 0x4242, NULL);
g_object_get (state->element, "ssrc", &ssrc, NULL);
fail_unless_equals_int (ssrc, 0x4242);
set_state (state, GST_STATE_PLAYING);
push_buffer (state, "pts", 2 * GST_SECOND, NULL);
set_state (state, GST_STATE_NULL);
g_object_set (state->element, "ssrc", 0, NULL);
g_object_get (state->element, "ssrc", &ssrc, NULL);
fail_unless_equals_int (ssrc, 0);
set_state (state, GST_STATE_PLAYING);
push_buffer (state, "pts", 3 * GST_SECOND, NULL);
set_state (state, GST_STATE_NULL);
g_object_set (state->element, "ssrc", G_MAXUINT32, NULL);
g_object_get (state->element, "ssrc", &ssrc, NULL);
fail_unless_equals_int (ssrc, G_MAXUINT32);
set_state (state, GST_STATE_PLAYING);
push_buffer (state, "pts", 4 * GST_SECOND, NULL);
set_state (state, GST_STATE_NULL);
validate_buffers_received (5);
validate_buffer (0, "pts", 0 * GST_SECOND, NULL);
get_buffer_field (0, "ssrc", &ssrc, NULL);
validate_buffer (1, "pts", 1 * GST_SECOND, "ssrc", ssrc, NULL);
validate_buffer (2, "pts", 2 * GST_SECOND, "ssrc", 0x4242, NULL);
validate_buffer (3, "pts", 3 * GST_SECOND, "ssrc", 0, NULL);
validate_buffer (4, "pts", 4 * GST_SECOND, "ssrc", G_MAXUINT32, NULL);
validate_events_received (12);
validate_normal_start_events (0);
validate_normal_start_events (3);
validate_normal_start_events (6);
validate_normal_start_events (9);
destroy_payloader (state);
}
GST_END_TEST;
/* validate that changing the timestamp-offset will actually effect the rtptime
* of the payloaded RTP packets. unfortunately setting the timestamp-offset
* property will only take effect when the payloader goes from READY to PAUSED.
* so the test starts by making sure that the default timestamp-offset indicates
* random timestamps. then a buffer is pushed which is expected to be payloaded
* as an RTP packet with a random timestamp. then the timestamp-offset is
* modified without changing the state of the pipeline. therefore the next
* buffer pushed is expected to result in an RTP packet with a timestamp equal
* to the previous RTP packet incremented by DEFAULT_CLOCK_RATE. next the
* pipeline is brought to NULL state and the timestamp-offset is set to a
* specific value, the pipeline is then brought back to PLAYING state and the
* two buffers pushed are expected to result in payloaded RTP packets that have
* timestamps based on the set timestamp-offset incremented by multiples of
* DEFAULT_CLOCK_RATE. next the boundary values of the timestamp-offset are
* tested. again the pipeline state needs to be modified and buffers are pushed
* and the resulting payloaded RTP packets' timestamps are validated. note that
* the maximum timestamp-offset value will wrap around for the very last
* payloaded RTP packet.
*/
GST_START_TEST (rtp_base_payload_property_timestamp_offset_test)
{
guint32 rtptime;
guint32 offset;
State *state;
state = create_payloader ("application/x-rtp", &sinktmpl, NULL);
set_state (state, GST_STATE_PLAYING);
g_object_get (state->element, "timestamp-offset", &offset, NULL);
fail_unless_equals_int (offset, -1);
push_buffer (state, "pts", 0 * GST_SECOND, NULL);
g_object_set (state->element, "timestamp-offset", 0x42, NULL);
g_object_get (state->element, "timestamp-offset", &offset, NULL);
fail_unless_equals_int (offset, 0x42);
push_buffer (state, "pts", 1 * GST_SECOND, NULL);
set_state (state, GST_STATE_NULL);
g_object_set (state->element, "timestamp-offset", 0x4242, NULL);
g_object_get (state->element, "timestamp-offset", &offset, NULL);
fail_unless_equals_int (offset, 0x4242);
set_state (state, GST_STATE_PLAYING);
push_buffer (state, "pts", 2 * GST_SECOND, NULL);
push_buffer (state, "pts", 3 * GST_SECOND, NULL);
set_state (state, GST_STATE_NULL);
g_object_set (state->element, "timestamp-offset", 0, NULL);
set_state (state, GST_STATE_PLAYING);
push_buffer (state, "pts", 4 * GST_SECOND, NULL);
push_buffer (state, "pts", 5 * GST_SECOND, NULL);
set_state (state, GST_STATE_NULL);
g_object_set (state->element, "timestamp-offset", G_MAXUINT32, NULL);
set_state (state, GST_STATE_PLAYING);
push_buffer (state, "pts", 6 * GST_SECOND, NULL);
push_buffer (state, "pts", 7 * GST_SECOND, NULL);
set_state (state, GST_STATE_NULL);
validate_buffers_received (8);
validate_buffer (0, "pts", 0 * GST_SECOND, NULL);
get_buffer_field (0, "rtptime", &rtptime, NULL);
validate_buffer (1,
"pts", 1 * GST_SECOND, "rtptime", rtptime + 1 * DEFAULT_CLOCK_RATE, NULL);
validate_buffer (2,
"pts", 2 * GST_SECOND, "rtptime", 0x4242 + 2 * DEFAULT_CLOCK_RATE, NULL);
validate_buffer (3,
"pts", 3 * GST_SECOND, "rtptime", 0x4242 + 3 * DEFAULT_CLOCK_RATE, NULL);
validate_buffer (4,
"pts", 4 * GST_SECOND, "rtptime", 0 + 4 * DEFAULT_CLOCK_RATE, NULL);
validate_buffer (5,
"pts", 5 * GST_SECOND, "rtptime", 0 + 5 * DEFAULT_CLOCK_RATE, NULL);
validate_buffer (6,
"pts", 6 * GST_SECOND, "rtptime", 6 * DEFAULT_CLOCK_RATE - 1, NULL);
validate_buffer (7,
"pts", 7 * GST_SECOND, "rtptime", 7 * DEFAULT_CLOCK_RATE - 1, NULL);
validate_events_received (12);
validate_normal_start_events (0);
validate_normal_start_events (3);
validate_normal_start_events (6);
validate_normal_start_events (9);
destroy_payloader (state);
}
GST_END_TEST;
/* as for timestamp-offset above setting the seqnum-offset property of a
* payloader will only take effect when the payloader goes from READY to PAUSED
* state. this test starts by validating that seqnum-offset indicates random
* sequence numbers and that the random sequence numbers increment by one for
* each payloaded RTP packet. also it is verified that setting seqnum-offset
* without bringing the pipeline to READY will not affect the payloaded RTP
* packets' sequence numbers. next the pipeline is brought to NULL state,
* seqnum-offset is set to a specific value before bringing the pipeline back to
* PLAYING state. the next two buffers pushed are expected to resulting in
* payloaded RTP packets that start with sequence numbers relating to the set
* seqnum-offset value, and that again increment by one for each packet. finally
* the boundary values of seqnum-offset are tested. this means bringing the
* pipeline to NULL state, setting the seqnum-offset and bringing the pipeline
* back to PLAYING state. note that for the very last payloded RTP packet the
* sequence number will have wrapped around because the previous packet is
* expected to have the maximum sequence number value.
*/
GST_START_TEST (rtp_base_payload_property_seqnum_offset_test)
{
State *state;
guint16 seq;
gint offset;
state = create_payloader ("application/x-rtp", &sinktmpl, NULL);
set_state (state, GST_STATE_PLAYING);
g_object_get (state->element, "seqnum-offset", &offset, NULL);
fail_unless_equals_int (offset, -1);
push_buffer (state, "pts", 0 * GST_SECOND, NULL);
g_object_set (state->element, "seqnum-offset", 0x42, NULL);
g_object_get (state->element, "seqnum-offset", &offset, NULL);
fail_unless_equals_int (offset, 0x42);
push_buffer (state, "pts", 1 * GST_SECOND, NULL);
set_state (state, GST_STATE_NULL);
g_object_set (state->element, "seqnum-offset", 0x4242, NULL);
g_object_get (state->element, "seqnum-offset", &offset, NULL);
fail_unless_equals_int (offset, 0x4242);
set_state (state, GST_STATE_PLAYING);
push_buffer (state, "pts", 2 * GST_SECOND, NULL);
push_buffer (state, "pts", 3 * GST_SECOND, NULL);
set_state (state, GST_STATE_NULL);
g_object_set (state->element, "seqnum-offset", -1, NULL);
set_state (state, GST_STATE_PLAYING);
push_buffer (state, "pts", 4 * GST_SECOND, NULL);
push_buffer (state, "pts", 5 * GST_SECOND, NULL);
set_state (state, GST_STATE_NULL);
g_object_set (state->element, "seqnum-offset", G_MAXUINT16, NULL);
set_state (state, GST_STATE_PLAYING);
push_buffer (state, "pts", 6 * GST_SECOND, NULL);
push_buffer (state, "pts", 7 * GST_SECOND, NULL);
set_state (state, GST_STATE_NULL);
validate_buffers_received (8);
validate_buffer (0, "pts", 0 * GST_SECOND, NULL);
get_buffer_field (0, "seq", &seq, NULL);
validate_buffer (1, "pts", 1 * GST_SECOND, "seq", seq + 1, NULL);
validate_buffer (2, "pts", 2 * GST_SECOND, "seq", 0x4242, NULL);
validate_buffer (3, "pts", 3 * GST_SECOND, "seq", 0x4242 + 1, NULL);
validate_buffer (4, "pts", 4 * GST_SECOND, NULL);
get_buffer_field (4, "seq", &seq, NULL);
validate_buffer (5, "pts", 5 * GST_SECOND, "seq", seq + 1, NULL);
validate_buffer (6, "pts", 6 * GST_SECOND, "seq", G_MAXUINT16, NULL);
validate_buffer (7, "pts", 7 * GST_SECOND, "seq", 0, NULL);
validate_events_received (12);
validate_normal_start_events (0);
validate_normal_start_events (3);
validate_normal_start_events (6);
validate_normal_start_events (9);
destroy_payloader (state);
}
GST_END_TEST;
/* a payloader's max-ptime property is linked to its MTU property. whenever a
* packet is larger than MTU or has a duration longer than max-ptime it will be
* considered to be full. so this test first validates that the default value of
* max-ptime is unspecified. then it retrieves the MTU and validates that a
* packet of size MTU will not be considered full even if the duration is at its
* maximum value. however incrementing the size to exceed the MTU will result in
* the packet being full. next max-ptime is set to a value and it is verified
* that only if both the size and duration are below the allowed values then the
* packet will be considered not to be full, otherwise it will be reported as
* being full. finally the boundary values of the property are tested in a
* similar fashion.
*/
GST_START_TEST (rtp_base_payload_property_max_ptime_test)
{
gint64 max_ptime;
State *state;
guint mtu;
state = create_payloader ("application/x-rtp", &sinktmpl, NULL);
g_object_get (state->element, "max-ptime", &max_ptime, NULL);
fail_unless_equals_int64 (max_ptime, -1);
g_object_get (state->element, "mtu", &mtu, NULL);
validate_would_not_be_filled (state, mtu, G_MAXINT64 - 1);
validate_would_be_filled (state, mtu + 1, G_MAXINT64 - 1);
g_object_set (state->element, "max-ptime", GST_SECOND, NULL);
g_object_get (state->element, "max-ptime", &max_ptime, NULL);
fail_unless_equals_int64 (max_ptime, GST_SECOND);
validate_would_not_be_filled (state, mtu, GST_SECOND - 1);
validate_would_be_filled (state, mtu, GST_SECOND);
validate_would_be_filled (state, mtu + 1, GST_SECOND - 1);
validate_would_be_filled (state, mtu + 1, GST_SECOND);
g_object_set (state->element, "max-ptime", G_MAXUINT64, NULL);
g_object_get (state->element, "max-ptime", &max_ptime, NULL);
fail_unless_equals_int64 (max_ptime, G_MAXUINT64);
validate_would_not_be_filled (state, mtu, G_MAXINT64 - 1);
validate_would_be_filled (state, mtu + 1, G_MAXINT64 - 1);
g_object_set (state->element, "max-ptime", G_MAXINT64, NULL);
g_object_get (state->element, "max-ptime", &max_ptime, NULL);
fail_unless_equals_int64 (max_ptime, G_MAXINT64);
validate_would_be_filled (state, mtu, G_MAXINT64);
destroy_payloader (state);
}
GST_END_TEST;
/* a basepayloader has a min-ptime property with an allowed range, the property
* itself is never checked by the payloader but is meant to be used by
* inheriting classes. therefore this test only validates that setting the
* property will mean that retrieveing the property results in the value
* previously being set. first the default value is validated, then a new
* specific value, before finally testing the boundary values.
*/
GST_START_TEST (rtp_base_payload_property_min_ptime_test)
{
State *state;
guint64 reference, min_ptime;
state = create_payloader ("application/x-rtp", &sinktmpl, NULL);
g_object_get (state->element, "min-ptime", &reference, NULL);
fail_unless_equals_int (reference, 0);
g_object_set (state->element, "min-ptime", reference + 1, NULL);
g_object_get (state->element, "min-ptime", &min_ptime, NULL);
fail_unless_equals_int (min_ptime, reference + 1);
g_object_set (state->element, "min-ptime", G_GUINT64_CONSTANT (0), NULL);
g_object_get (state->element, "min-ptime", &min_ptime, NULL);
fail_unless_equals_int (min_ptime, 0);
g_object_set (state->element, "min-ptime", G_MAXINT64, NULL);
g_object_get (state->element, "min-ptime", &min_ptime, NULL);
fail_unless_equals_int64 (min_ptime, G_MAXINT64);
destroy_payloader (state);
}
GST_END_TEST;
/* paylaoders have a timestamp property that reflects the timestamp of the last
* payloaded RTP packet. in this test the timestamp-offset is set to a specific
* value so that when the first buffer is pushed its timestamp can be predicted
* and thus that the timestamp property also has this value. (if
* timestamp-offset was not set the timestamp would be random). another buffer
* is then pushed and its timestamp is expected to increment by
* DEFAULT_CLOCK_RATE.
*/
GST_START_TEST (rtp_base_payload_property_timestamp_test)
{
State *state;
guint32 timestamp;
state = create_payloader ("application/x-rtp", &sinktmpl,
"timestamp-offset", 0, NULL);
set_state (state, GST_STATE_PLAYING);
push_buffer (state, "pts", 0 * GST_SECOND, NULL);
g_object_get (state->element, "timestamp", &timestamp, NULL);
fail_unless_equals_int (timestamp, 0);
push_buffer (state, "pts", 1 * GST_SECOND, NULL);
g_object_get (state->element, "timestamp", &timestamp, NULL);
fail_unless_equals_int (timestamp, DEFAULT_CLOCK_RATE);
set_state (state, GST_STATE_NULL);
validate_buffers_received (2);
validate_buffer (0, "pts", 0 * GST_SECOND, "rtptime", 0, NULL);
validate_buffer (1,
"pts", 1 * GST_SECOND, "rtptime", DEFAULT_CLOCK_RATE, NULL);
validate_events_received (3);
validate_normal_start_events (0);
destroy_payloader (state);
}
GST_END_TEST;
/* basepayloaders have a seqnum property that is supposed to contain the
* sequence number of the last payloaded RTP packet. so therefore this test
* initializes the seqnum-offset property to a know value and pushes a buffer.
* the payloaded RTP packet is expected to have a sequence number equal to the
* set seqnum-offset, as is the seqnum property. next another buffer is pushed
* and then both the payloaded RTP packet and the seqnum property value are
* expected to increment by one compared to the previous packet.
*/
GST_START_TEST (rtp_base_payload_property_seqnum_test)
{
State *state;
guint seq;
state = create_payloader ("application/x-rtp", &sinktmpl,
"seqnum-offset", 0, NULL);
set_state (state, GST_STATE_PLAYING);
push_buffer (state, "pts", 0 * GST_SECOND, NULL);
g_object_get (state->element, "seqnum", &seq, NULL);
fail_unless_equals_int (seq, 0);
push_buffer (state, "pts", 1 * GST_SECOND, NULL);
g_object_get (state->element, "seqnum", &seq, NULL);
fail_unless_equals_int (seq, 1);
set_state (state, GST_STATE_NULL);
validate_buffers_received (2);
validate_buffer (0, "pts", 0 * GST_SECOND, "seq", 0, NULL);
validate_buffer (1, "pts", 1 * GST_SECOND, "seq", 1, NULL);
validate_events_received (3);
validate_normal_start_events (0);
destroy_payloader (state);
}
GST_END_TEST;
/* basepayloader has a perfect-rtptime property when it is set to FALSE
* the timestamps of payloaded RTP packets will determined by initial
* timestamp-offset (usually random) as well as the clock-rate. when
* perfect-rtptime is set to TRUE the timestamps of payloaded RTP packets are
* instead determined by the timestamp of the first packet and then the
* difference in offset of the input buffers.
*
* to verify that this test starts by setting the timestamp-offset to a specific
* value to prevent random timestamps of the RTP packets. next perfect-rtptime
* is set to FALSE. the two buffers pushed will result in two payloaded RTP
* packets whose timestamps differ based on the current clock-rate
* DEFAULT_CLOCK_RATE. the next step is to set perfect-rtptime to TRUE. the two
* buffers that are pushed will result in two payloaded RTP packets. the first
* of these RTP packets has a timestamp that relates to the previous packet and
* the difference in offset between the middle two input buffers. the latter of
* the two RTP packets has a timestamp that instead relates to the offset of the
* last two input buffers.
*/
GST_START_TEST (rtp_base_payload_property_perfect_rtptime_test)
{
State *state;
guint32 timestamp_base = 0;
gboolean perfect;
state = create_payloader ("application/x-rtp", &sinktmpl,
"timestamp-offset", timestamp_base, NULL);
set_state (state, GST_STATE_PLAYING);
g_object_set (state->element, "perfect-rtptime", FALSE, NULL);
g_object_get (state->element, "perfect-rtptime", &perfect, NULL);
fail_unless (!perfect);
push_buffer (state, "pts", 0 * GST_SECOND, "offset", G_GINT64_CONSTANT (0),
NULL);
push_buffer (state, "pts", 1 * GST_SECOND, "offset", G_GINT64_CONSTANT (17),
NULL);
g_object_set (state->element, "perfect-rtptime", TRUE, NULL);
g_object_get (state->element, "perfect-rtptime", &perfect, NULL);
fail_unless (perfect);
push_buffer (state, "pts", 2 * GST_SECOND, "offset", G_GINT64_CONSTANT (31),
NULL);
push_buffer (state, "pts", 3 * GST_SECOND, "offset", G_GINT64_CONSTANT (67),
NULL);
set_state (state, GST_STATE_NULL);
validate_buffers_received (4);
validate_buffer (0,
"pts", 0 * GST_SECOND, "offset", G_GINT64_CONSTANT (0), "rtptime",
timestamp_base, NULL);
validate_buffer (1,
"pts", 1 * GST_SECOND,
"offset", G_GINT64_CONSTANT (17), "rtptime",
timestamp_base + 1 * DEFAULT_CLOCK_RATE, NULL);
validate_buffer (2,
"pts", 2 * GST_SECOND,
"offset", G_GINT64_CONSTANT (31),
"rtptime", timestamp_base + 1 * DEFAULT_CLOCK_RATE + (31 - 17), NULL);
validate_buffer (3,
"pts", 3 * GST_SECOND,
"offset", G_GINT64_CONSTANT (67),
"rtptime", timestamp_base + 1 * DEFAULT_CLOCK_RATE + (67 - 17), NULL);
validate_events_received (3);
validate_normal_start_events (0);
destroy_payloader (state);
}
GST_END_TEST;
/* basepayloaders have a ptime-multiple property but its value does not affect
* any payloaded RTP packets as this is supposed to be done by inherited
* classes. therefore this test only validates the default value of the
* property, makes sure that a set value actually sticks and that the boundary
* values are indeed allowed to be set.
*/
GST_START_TEST (rtp_base_payload_property_ptime_multiple_test)
{
State *state;
gint64 multiple;
state = create_payloader ("application/x-rtp", &sinktmpl, NULL);
g_object_get (state->element, "ptime-multiple", &multiple, NULL);
fail_unless_equals_int64 (multiple, 0);
g_object_set (state->element, "ptime-multiple", G_GINT64_CONSTANT (42), NULL);
g_object_get (state->element, "ptime-multiple", &multiple, NULL);
fail_unless_equals_int64 (multiple, 42);
g_object_set (state->element, "ptime-multiple", G_GINT64_CONSTANT (0), NULL);
g_object_get (state->element, "ptime-multiple", &multiple, NULL);
fail_unless_equals_int64 (multiple, 0);
g_object_set (state->element, "ptime-multiple", G_MAXINT64, NULL);
g_object_get (state->element, "ptime-multiple", &multiple, NULL);
fail_unless_equals_int64 (multiple, G_MAXINT64);
destroy_payloader (state);
}
GST_END_TEST;
/* basepayloaders have a property called stats that is used to atomically
* retrieve several values (clock-rate, running-time, seqnum and timestamp) that
* relate to the stream and its current progress. this test is meant to test
* retrieval of these values.
*
* first of all perfect-rtptime is set to TRUE, next the the test starts out by
* setting seqnum-offset and timestamp-offset to known values to prevent that
* sequence numbers and timestamps of payloaded RTP packets are random. next the
* stats property is retrieved. the clock-rate must be at the default
* DEFAULT_CLOCK_RATE, while running-time must be equal to the first buffers
* PTS. the sequence number should be equal to the initialized value of
* seqnum-offset and the timestamp should be equal to the initialized value of
* timestamp-offset. after pushing a second buffer the stats property is
* validate again. this time running-time, seqnum and timestamp should have
* advanced as expected. next the pipeline is brought to NULL state to be able
* to change the perfect-rtptime property to FALSE before going back to PLAYING
* state. this is done to validate that the stats values reflect normal
* timestamp updates that are not based on input buffer offsets as expected.
* lastly two buffers are pushed and the stats property retrieved after each
* time. here it is expected that the sequence numbers values are restarted at
* the initial value while the timestamps and running-time reflect the input
* buffers.
*/
GST_START_TEST (rtp_base_payload_property_stats_test)
{
State *state;
state = create_payloader ("application/x-rtp", &sinktmpl,
"perfect-rtptime", TRUE, "seqnum-offset", 0, "timestamp-offset", 0, NULL);
set_state (state, GST_STATE_PLAYING);
push_buffer (state, "pts", 0 * GST_SECOND, NULL);
validate_stats (state,
DEFAULT_CLOCK_RATE, 0 * GST_SECOND, 0, 0 * DEFAULT_CLOCK_RATE);
push_buffer (state, "pts", 1 * GST_SECOND, NULL);
validate_stats (state,
DEFAULT_CLOCK_RATE, 1 * DEFAULT_CLOCK_RATE, 1, 1 * DEFAULT_CLOCK_RATE);
set_state (state, GST_STATE_NULL);
g_object_set (state->element, "perfect-rtptime", FALSE, NULL);
set_state (state, GST_STATE_PLAYING);
push_buffer (state, "pts", 2 * GST_SECOND, NULL);
validate_stats (state,
DEFAULT_CLOCK_RATE, 2 * GST_SECOND, 0, 2 * DEFAULT_CLOCK_RATE);
push_buffer (state, "pts", 3 * GST_SECOND, NULL);
validate_stats (state,
DEFAULT_CLOCK_RATE, 3 * GST_SECOND, 1, 3 * DEFAULT_CLOCK_RATE);
set_state (state, GST_STATE_NULL);
validate_buffers_received (4);
validate_buffer (0, "pts", 0 * GST_SECOND, NULL);
validate_buffer (1, "pts", 1 * GST_SECOND, NULL);
validate_buffer (2, "pts", 2 * GST_SECOND, NULL);
validate_buffer (3, "pts", 3 * GST_SECOND, NULL);
validate_events_received (6);
validate_normal_start_events (0);
validate_normal_start_events (3);
destroy_payloader (state);
}
GST_END_TEST;
/* basepayloader has a property source-info that makes it aware of RTP
* source information passed as GstRTPSourceMeta on the input buffers. All
* sources found in the meta will be added to the list of CSRCs in the RTP
* header. A useful scenario for this is, for instance, to signal which
* sources contributed to a mixed audio stream. */
GST_START_TEST (rtp_base_payload_property_source_info_test)
{
GstHarness *h;
GstRtpDummyPay *pay;
GstBuffer *buffer;
guint csrc_count = 2;
const guint32 csrc[] = { 0x11, 0x22 };
const guint32 ssrc = 0x33;
pay = rtp_dummy_pay_new ();
h = gst_harness_new_with_element (GST_ELEMENT_CAST (pay), "sink", "src");
gst_harness_set_src_caps_str (h, "application/x-rtp");
/* Input buffer has no meta, payloader should not add CSRC */
g_object_set (pay, "source-info", TRUE, NULL);
buffer = gst_rtp_buffer_new_allocate (0, 0, 0);
buffer = gst_harness_push_and_pull (h, buffer);
validate_buffer1 (buffer, "csrc-count", 0, NULL);
fail_if (gst_buffer_get_rtp_source_meta (buffer));
gst_buffer_unref (buffer);
/* Input buffer has meta, payloader should add CSRC */
buffer = gst_rtp_buffer_new_allocate (0, 0, 0);
fail_unless (gst_buffer_add_rtp_source_meta (buffer, &ssrc, csrc,
csrc_count));
buffer = gst_harness_push_and_pull (h, buffer);
/* The meta SSRC should be added as the last contributing source */
validate_buffer1 (buffer, "csrc-count", 3, "csrc", 0, csrc[0],
"csrc", 1, csrc[1], "csrc", 2, ssrc, NULL);
fail_if (gst_buffer_get_rtp_source_meta (buffer));
gst_buffer_unref (buffer);
/* When property is disabled, the meta should be ignored and no CSRC
* added. */
g_object_set (pay, "source-info", FALSE, NULL);
buffer = gst_rtp_buffer_new_allocate (0, 0, 0);
fail_unless (gst_buffer_add_rtp_source_meta (buffer, NULL, csrc, csrc_count));
buffer = gst_harness_push_and_pull (h, buffer);
validate_buffer1 (buffer, "csrc-count", 0, NULL);
fail_if (gst_buffer_get_rtp_source_meta (buffer));
gst_buffer_unref (buffer);
g_object_unref (pay);
gst_harness_teardown (h);
}
GST_END_TEST;
/* push a single buffer to the payloader which should successfully payload it
* into an RTP packet. besides the payloaded RTP packet there should be the
* three events initial events: stream-start, caps and segment. because of that
* the input caps has framerate this will be propagated to an a-framerate field
* on the output caps.
*/
GST_START_TEST (rtp_base_payload_framerate_attribute)
{
State *state;
state = create_payloader ("video/x-raw,framerate=(fraction)1/4", &sinktmpl,
"perfect-rtptime", FALSE, NULL);
set_state (state, GST_STATE_PLAYING);
push_buffer (state, "pts", 0 * GST_SECOND, NULL);
set_state (state, GST_STATE_NULL);
validate_buffers_received (1);
validate_buffer (0, "pts", 0 * GST_SECOND, NULL);
validate_events_received (3);
validate_normal_start_events (0);
validate_event (1, "caps", "a-framerate", "0.25", NULL);
destroy_payloader (state);
}
GST_END_TEST;
/* push a single buffer to the payloader which should successfully payload it
* into an RTP packet. besides the payloaded RTP packet there should be the
* three events initial events: stream-start, caps and segment. because of that
* the input caps has both framerate and max-framerate set the a-framerate field
* on the output caps will correspond to the value of the max-framerate field.
*/
GST_START_TEST (rtp_base_payload_max_framerate_attribute)
{
State *state;
state =
create_payloader
("video/x-raw,framerate=(fraction)0/1,max-framerate=(fraction)1/8",
&sinktmpl, "perfect-rtptime", FALSE, NULL);
set_state (state, GST_STATE_PLAYING);
push_buffer (state, "pts", 0 * GST_SECOND, NULL);
set_state (state, GST_STATE_NULL);
validate_buffers_received (1);
validate_buffer (0, "pts", 0 * GST_SECOND, NULL);
validate_events_received (3);
validate_normal_start_events (0);
validate_event (1, "caps", "a-framerate", "0.125", NULL);
destroy_payloader (state);
}
GST_END_TEST;
GST_START_TEST (rtp_base_payload_segment_time)
{
State *state;
guint32 timestamp_base = 0;
guint segment_time = 10;
GstEvent *event;
GstSegment *segment = gst_segment_new ();
state =
create_payloader
("application/x-rtp",
&sinktmpl, "onvif-no-rate-control", TRUE, "timestamp-offset",
timestamp_base, NULL);
set_state (state, GST_STATE_PLAYING);
gst_segment_init (segment, GST_FORMAT_TIME);
segment->time = segment_time * GST_SECOND;
event = gst_event_new_segment (segment);
fail_unless (gst_pad_push_event (state->srcpad, event));
push_buffer (state, "pts", 0 * GST_SECOND, NULL);
push_buffer (state, "pts", 1 * GST_SECOND, NULL);
push_buffer (state, "pts", 2 * GST_SECOND, NULL);
push_buffer (state, "pts", 3 * GST_SECOND, NULL);
set_state (state, GST_STATE_NULL);
validate_buffers_received (4);
validate_buffer (0, "rtptime",
timestamp_base + (segment_time) * DEFAULT_CLOCK_RATE, NULL);
validate_buffer (1, "rtptime",
timestamp_base + (1 + segment_time) * DEFAULT_CLOCK_RATE, NULL);
validate_buffer (2, "rtptime",
timestamp_base + (2 + segment_time) * DEFAULT_CLOCK_RATE, NULL);
validate_buffer (3, "rtptime",
timestamp_base + (3 + segment_time) * DEFAULT_CLOCK_RATE, NULL);
destroy_payloader (state);
gst_segment_free (segment);
}
GST_END_TEST;
GST_START_TEST (rtp_base_payload_one_byte_hdr_ext)
{
GstRTPHeaderExtension *ext;
State *state;
state = create_payloader ("application/x-rtp", &sinktmpl, NULL);
ext = rtp_dummy_hdr_ext_new ();
GST_RTP_DUMMY_HDR_EXT (ext)->supported_flags =
GST_RTP_HEADER_EXTENSION_ONE_BYTE;
gst_rtp_header_extension_set_id (ext, 1);
g_signal_emit_by_name (state->element, "add-extension", ext);
set_state (state, GST_STATE_PLAYING);
push_buffer (state, "pts", 0 * GST_SECOND, NULL);
set_state (state, GST_STATE_NULL);
validate_buffers_received (1);
validate_buffer (0, "pts", 0 * GST_SECOND, NULL);
validate_events_received (3);
validate_normal_start_events (0);
fail_unless_equals_int (GST_RTP_DUMMY_HDR_EXT (ext)->write_count, 1);
gst_object_unref (ext);
destroy_payloader (state);
}
GST_END_TEST;
GST_START_TEST (rtp_base_payload_two_byte_hdr_ext)
{
GstRTPHeaderExtension *ext;
State *state;
state = create_payloader ("application/x-rtp", &sinktmpl, NULL);
ext = rtp_dummy_hdr_ext_new ();
GST_RTP_DUMMY_HDR_EXT (ext)->supported_flags =
GST_RTP_HEADER_EXTENSION_TWO_BYTE;
gst_rtp_header_extension_set_id (ext, 1);
g_signal_emit_by_name (state->element, "add-extension", ext);
set_state (state, GST_STATE_PLAYING);
push_buffer (state, "pts", 0 * GST_SECOND, NULL);
set_state (state, GST_STATE_NULL);
validate_buffers_received (1);
validate_buffer (0, "pts", 0 * GST_SECOND, NULL);
validate_events_received (3);
validate_normal_start_events (0);
fail_unless_equals_int (GST_RTP_DUMMY_HDR_EXT (ext)->write_count, 1);
gst_object_unref (ext);
destroy_payloader (state);
}
GST_END_TEST;
GST_START_TEST (rtp_base_payload_clear_extensions)
{
GstRTPHeaderExtension *ext;
State *state;
state = create_payloader ("application/x-rtp", &sinktmpl, NULL);
ext = rtp_dummy_hdr_ext_new ();
gst_rtp_header_extension_set_id (ext, 1);
g_signal_emit_by_name (state->element, "add-extension", ext);
set_state (state, GST_STATE_PLAYING);
push_buffer (state, "pts", 0 * GST_SECOND, NULL);
g_signal_emit_by_name (state->element, "clear-extensions", ext);
push_buffer (state, "pts", 1 * GST_SECOND, NULL);
set_state (state, GST_STATE_NULL);
validate_buffers_received (2);
validate_buffer (0, "pts", 0 * GST_SECOND, NULL);
validate_buffer (1, "pts", 1 * GST_SECOND, NULL);
validate_events_received (3);
validate_normal_start_events (0);
fail_unless_equals_int (GST_RTP_DUMMY_HDR_EXT (ext)->write_count, 1);
gst_object_unref (ext);
destroy_payloader (state);
}
GST_END_TEST;
GST_START_TEST (rtp_base_payload_multiple_exts)
{
GstRTPHeaderExtension *ext1, *ext2;
State *state;
state = create_payloader ("application/x-rtp", &sinktmpl, NULL);
ext1 = rtp_dummy_hdr_ext_new ();
gst_rtp_header_extension_set_id (ext1, 1);
ext2 = rtp_dummy_hdr_ext_new ();
gst_rtp_header_extension_set_id (ext2, 2);
g_signal_emit_by_name (state->element, "add-extension", ext1);
g_signal_emit_by_name (state->element, "add-extension", ext2);
set_state (state, GST_STATE_PLAYING);
push_buffer (state, "pts", 0 * GST_SECOND, NULL);
set_state (state, GST_STATE_NULL);
validate_buffers_received (1);
validate_buffer (0, "pts", 0 * GST_SECOND, NULL);
validate_events_received (3);
validate_normal_start_events (0);
fail_unless_equals_int (GST_RTP_DUMMY_HDR_EXT (ext1)->write_count, 1);
fail_unless_equals_int (GST_RTP_DUMMY_HDR_EXT (ext2)->write_count, 1);
gst_object_unref (ext1);
gst_object_unref (ext2);
destroy_payloader (state);
}
GST_END_TEST;
static GstStaticPadTemplate sinktmpl_with_extmap_str =
GST_STATIC_PAD_TEMPLATE ("sink",
GST_PAD_SINK,
GST_PAD_ALWAYS,
GST_STATIC_CAPS ("application/x-rtp, payload=(int)98, ssrc=(uint)24, "
"timestamp-offset=(uint)212, seqnum-offset=(uint)2424, extmap-4=(string)"
DUMMY_HDR_EXT_URI));
static GstRTPHeaderExtension *
request_extension (GstRTPBasePayload * depayload, guint ext_id,
const gchar * ext_uri, gpointer user_data)
{
GstRTPHeaderExtension *ext = user_data;
if (ext && gst_rtp_header_extension_get_id (ext) == ext_id
&& g_strcmp0 (ext_uri, gst_rtp_header_extension_get_uri (ext)) == 0)
return gst_object_ref (ext);
return NULL;
}
GST_START_TEST (rtp_base_payload_caps_request)
{
GstRTPHeaderExtension *ext;
GstRTPDummyHdrExt *dummy;
State *state;
state =
create_payloader ("application/x-rtp", &sinktmpl_with_extmap_str, NULL);
ext = rtp_dummy_hdr_ext_new ();
dummy = GST_RTP_DUMMY_HDR_EXT (ext);
gst_rtp_header_extension_set_id (ext, 4);
g_signal_connect (state->element, "request-extension",
G_CALLBACK (request_extension), ext);
fail_unless (dummy->set_attributes_count == 0);
set_state (state, GST_STATE_PLAYING);
push_buffer (state, "pts", 0 * GST_SECOND, NULL);
set_state (state, GST_STATE_NULL);
validate_buffers_received (1);
validate_buffer (0, "pts", 0 * GST_SECOND, NULL);
validate_events_received (3);
validate_normal_start_events (0);
fail_unless_equals_int (GST_RTP_DUMMY_HDR_EXT (ext)->write_count, 1);
fail_unless (dummy->set_attributes_count == 1);
gst_object_unref (ext);
destroy_payloader (state);
}
GST_END_TEST;
static GstRTPHeaderExtension *
request_extension_ignored (GstRTPBasePayload * depayload, guint ext_id,
const gchar * ext_uri, gpointer user_data)
{
guint *request_counter = user_data;
*request_counter += 1;
return NULL;
}
GST_START_TEST (rtp_base_payload_caps_request_ignored)
{
State *state;
guint request_counter = 0;
state =
create_payloader ("application/x-rtp", &sinktmpl_with_extmap_str, NULL);
g_signal_connect (state->element, "request-extension",
G_CALLBACK (request_extension_ignored), &request_counter);
set_state (state, GST_STATE_PLAYING);
push_buffer (state, "pts", 0 * GST_SECOND, NULL);
set_state (state, GST_STATE_NULL);
fail_unless_equals_int (request_counter, 1);
validate_buffers_received (1);
destroy_payloader (state);
}
GST_END_TEST;
GST_START_TEST (rtp_base_payload_extensions_in_output_caps)
{
GstRTPHeaderExtension *ext;
State *state;
state = create_payloader ("application/x-rtp", &sinktmpl, NULL);
ext = rtp_dummy_hdr_ext_new ();
GST_RTP_DUMMY_HDR_EXT (ext)->supported_flags =
GST_RTP_HEADER_EXTENSION_TWO_BYTE;
gst_rtp_header_extension_set_id (ext, 1);
g_signal_emit_by_name (state->element, "add-extension", ext);
set_state (state, GST_STATE_PLAYING);
push_buffer (state, "pts", 0 * GST_SECOND, NULL);
set_state (state, GST_STATE_NULL);
validate_buffers_received (1);
validate_buffer (0, "pts", 0 * GST_SECOND, NULL);
validate_events_received (3);
validate_normal_start_events (0);
validate_event (1, "caps", "extmap-str", 1, DUMMY_HDR_EXT_URI, NULL);
fail_unless_equals_int (GST_RTP_DUMMY_HDR_EXT (ext)->write_count, 1);
gst_object_unref (ext);
ext = NULL;
destroy_payloader (state);
}
GST_END_TEST;
GST_START_TEST (rtp_base_payload_extensions_shrink_ext_data)
{
GstRTPHeaderExtension *ext;
State *state;
state = create_payloader ("application/x-rtp", &sinktmpl, NULL);
ext = rtp_dummy_hdr_ext_new ();
GST_RTP_DUMMY_HDR_EXT (ext)->supported_flags =
GST_RTP_HEADER_EXTENSION_ONE_BYTE;
GST_RTP_DUMMY_HDR_EXT (ext)->max_size = 5;
gst_rtp_header_extension_set_id (ext, 1);
g_signal_emit_by_name (state->element, "add-extension", ext);
set_state (state, GST_STATE_PLAYING);
push_buffer (state, "pts", 0 * GST_SECOND, NULL);
set_state (state, GST_STATE_NULL);
validate_buffers_received (1);
validate_buffer (0, "pts", 0 * GST_SECOND, "size", (gsize) 20, "ext-data",
(guint) 0xBEDE, (gsize) 4, NULL);
validate_events_received (3);
validate_normal_start_events (0);
fail_unless_equals_int (GST_RTP_DUMMY_HDR_EXT (ext)->write_count, 1);
gst_object_unref (ext);
ext = NULL;
destroy_payloader (state);
}
GST_END_TEST;
static Suite *
rtp_basepayloading_suite (void)
{
Suite *s = suite_create ("rtp_base_payloading_test");
TCase *tc_chain = tcase_create ("payloading tests");
tcase_set_timeout (tc_chain, 60);
suite_add_tcase (s, tc_chain);
tcase_add_test (tc_chain, rtp_base_payload_buffer_test);
tcase_add_test (tc_chain, rtp_base_payload_buffer_list_test);
tcase_add_test (tc_chain, rtp_base_payload_normal_rtptime_test);
tcase_add_test (tc_chain, rtp_base_payload_perfect_rtptime_test);
tcase_add_test (tc_chain, rtp_base_payload_no_pts_no_offset_test);
tcase_add_test (tc_chain, rtp_base_payload_downstream_caps_test);
tcase_add_test (tc_chain, rtp_base_payload_ssrc_collision_test);
tcase_add_test (tc_chain, rtp_base_payload_reconfigure_test);
tcase_add_test (tc_chain, rtp_base_payload_property_mtu_test);
tcase_add_test (tc_chain, rtp_base_payload_property_pt_test);
tcase_add_test (tc_chain, rtp_base_payload_property_ssrc_test);
tcase_add_test (tc_chain, rtp_base_payload_property_timestamp_offset_test);
tcase_add_test (tc_chain, rtp_base_payload_property_seqnum_offset_test);
tcase_add_test (tc_chain, rtp_base_payload_property_max_ptime_test);
tcase_add_test (tc_chain, rtp_base_payload_property_min_ptime_test);
tcase_add_test (tc_chain, rtp_base_payload_property_timestamp_test);
tcase_add_test (tc_chain, rtp_base_payload_property_seqnum_test);
tcase_add_test (tc_chain, rtp_base_payload_property_perfect_rtptime_test);
tcase_add_test (tc_chain, rtp_base_payload_property_ptime_multiple_test);
tcase_add_test (tc_chain, rtp_base_payload_property_stats_test);
tcase_add_test (tc_chain, rtp_base_payload_property_source_info_test);
tcase_add_test (tc_chain, rtp_base_payload_framerate_attribute);
tcase_add_test (tc_chain, rtp_base_payload_max_framerate_attribute);
tcase_add_test (tc_chain, rtp_base_payload_segment_time);
tcase_add_test (tc_chain, rtp_base_payload_one_byte_hdr_ext);
tcase_add_test (tc_chain, rtp_base_payload_two_byte_hdr_ext);
tcase_add_test (tc_chain, rtp_base_payload_clear_extensions);
tcase_add_test (tc_chain, rtp_base_payload_multiple_exts);
tcase_add_test (tc_chain, rtp_base_payload_caps_request);
tcase_add_test (tc_chain, rtp_base_payload_caps_request_ignored);
tcase_add_test (tc_chain, rtp_base_payload_extensions_in_output_caps);
tcase_add_test (tc_chain, rtp_base_payload_extensions_shrink_ext_data);
return s;
}
GST_CHECK_MAIN (rtp_basepayloading)
Reference in a new issue View git blame Copy permalink