gstreamer/tests/check/elements/rtp-payloading.c
Tim-Philipp Müller cf1135694d tests/check/Makefile.am: Add rtp-payloading test to VALGRIND_TO_FIX.
Original commit message from CVS:
* tests/check/Makefile.am:
Add rtp-payloading test to VALGRIND_TO_FIX.
* tests/check/elements/rtp-payloading.c:
Add semicolons after GST_TEST_END so gst-indent gets the
formatting right; make test less verbose in general, but
more verbose in the error case (which should probably
make the test fail anyway).
2008-02-02 18:06:19 +00:00

552 lines
15 KiB
C

/* GStreamer RTP payloader unit tests
* Copyright (C) 2008 Nokia Corporation and its subsidary(-ies)
* contact: <stefan.kost@nokia.com>
*
* This library is free software; you can redistribute it and/or
* modify it under the terms of the GNU Library General Public
* License 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 <stdlib.h>
#include <unistd.h>
#define RELEASE_ELEMENT(x) if(x) {gst_object_unref(x); x = NULL;}
#define LOOP_COUNT 1
/*
* RTP pipeline structure to store the required elements.
*/
typedef struct
{
GstElement *pipeline;
GstElement *fdsrc;
GstElement *capsfilter;
GstElement *rtppay;
GstElement *rtpdepay;
GstElement *fakesink;
int fd[2];
const char *frame_data;
int frame_data_size;
int frame_count;
} rtp_pipeline;
/*
* RTP bus callback.
*/
static gboolean
rtp_bus_callback (GstBus * bus, GstMessage * message, gpointer data)
{
GMainLoop *mainloop = (GMainLoop *) data;
switch (GST_MESSAGE_TYPE (message)) {
case GST_MESSAGE_ERROR:
{
GError *err;
gchar *debug;
gchar *element_name;
element_name = (message->src) ? gst_object_get_name (message->src) : NULL;
gst_message_parse_error (message, &err, &debug);
/* FIXME: should we fail the test here? */
g_print ("\nError from element %s: %s\n%s\n\n",
GST_STR_NULL (element_name), err->message, (debug) ? debug : "");
g_error_free (err);
g_free (debug);
g_free (element_name);
g_main_loop_quit (mainloop);
}
break;
case GST_MESSAGE_EOS:
{
g_main_loop_quit (mainloop);
}
break;
break;
default:
{
}
break;
}
return TRUE;
}
/*
* Creates a RTP pipeline for one test.
* @param frame_data Pointer to the frame data which is used to pass thru pay/depayloaders.
* @param frame_data_size Frame data size in bytes.
* @param frame_count Frame count.
* @param filtercaps Caps filters.
* @param pay Payloader name.
* @param depay Depayloader name.
* @return
* Returns pointer to the RTP pipeline.
* The user must free the RTP pipeline when it's not used anymore.
*/
static rtp_pipeline *
rtp_pipeline_create (const char *frame_data, int frame_data_size,
int frame_count, const char *filtercaps, const char *pay, const char *depay)
{
gchar *pipeline_name;
/* Check parameters. */
if (!frame_data || !pay || !depay) {
return NULL;
}
/* Allocate memory for the RTP pipeline. */
rtp_pipeline *p = (rtp_pipeline *) malloc (sizeof (rtp_pipeline));
p->frame_data = frame_data;
p->frame_data_size = frame_data_size;
p->frame_count = frame_count;
/* Create elements. */
pipeline_name = g_strdup_printf ("%s-%s-pipeline", pay, depay);
p->pipeline = gst_pipeline_new (pipeline_name);
g_free (pipeline_name);
p->fdsrc = gst_element_factory_make ("fdsrc", NULL);
p->capsfilter = gst_element_factory_make ("capsfilter", NULL);
p->rtppay = gst_element_factory_make (pay, NULL);
p->rtpdepay = gst_element_factory_make (depay, NULL);
p->fakesink = gst_element_factory_make ("fakesink", NULL);
/* One or more elements are not created successfully or failed to create p? */
if (!p->pipeline || !p->fdsrc || !p->capsfilter || !p->rtppay || !p->rtpdepay
|| !p->fakesink || pipe (p->fd) == -1) {
/* Release created elements. */
RELEASE_ELEMENT (p->pipeline);
RELEASE_ELEMENT (p->fdsrc);
RELEASE_ELEMENT (p->capsfilter);
RELEASE_ELEMENT (p->rtppay);
RELEASE_ELEMENT (p->rtpdepay);
RELEASE_ELEMENT (p->fakesink);
/* Close pipe. */
if (p->fd[0]) {
close (p->fd[0]);
}
if (p->fd[1]) {
close (p->fd[1]);
}
/* Release allocated memory. */
free (p);
return NULL;
}
/* Set fdsrc properties. */
g_object_set (p->fdsrc, "fd", p->fd[0], NULL);
g_object_set (p->fdsrc, "do-timestamp", TRUE, NULL);
g_object_set (p->fdsrc, "blocksize", p->frame_data_size, NULL);
g_object_set (p->fdsrc, "num-buffers", p->frame_count * LOOP_COUNT, NULL);
/* Set caps filters. */
GstCaps *caps = gst_caps_from_string (filtercaps);
g_object_set (p->capsfilter, "caps", caps, NULL);
gst_caps_unref (caps);
/* Add elements to the pipeline. */
gst_bin_add (GST_BIN (p->pipeline), p->fdsrc);
gst_bin_add (GST_BIN (p->pipeline), p->capsfilter);
gst_bin_add (GST_BIN (p->pipeline), p->rtppay);
gst_bin_add (GST_BIN (p->pipeline), p->rtpdepay);
gst_bin_add (GST_BIN (p->pipeline), p->fakesink);
/* Link elements. */
gst_element_link (p->fdsrc, p->capsfilter);
gst_element_link (p->capsfilter, p->rtppay);
gst_element_link (p->rtppay, p->rtpdepay);
gst_element_link (p->rtpdepay, p->fakesink);
return p;
}
/*
* Destroys the RTP pipeline.
* @param p Pointer to the RTP pipeline.
*/
static void
rtp_pipeline_destroy (rtp_pipeline * p)
{
/* Check parameters. */
if (p == NULL) {
return;
}
/* Release pipeline. */
RELEASE_ELEMENT (p->pipeline);
/* Close pipe. */
if (p->fd[0]) {
close (p->fd[0]);
}
if (p->fd[1]) {
close (p->fd[1]);
}
/* Release allocated memory. */
free (p);
}
/*
* Runs the RTP pipeline.
* @param p Pointer to the RTP pipeline.
*/
static void
rtp_pipeline_run (rtp_pipeline * p)
{
GMainLoop *mainloop = NULL;
/* Check parameters. */
if (p == NULL) {
return;
}
/* Create mainloop. */
mainloop = g_main_loop_new (NULL, FALSE);
if (!mainloop) {
return;
}
/* Add bus callback. */
GstBus *bus = gst_pipeline_get_bus (GST_PIPELINE (p->pipeline));
gst_bus_add_watch (bus, rtp_bus_callback, (gpointer) mainloop);
gst_object_unref (bus);
/* Set pipeline to PLAYING. */
gst_element_set_state (p->pipeline, GST_STATE_PLAYING);
/* TODO: Writing may need some changes... */
int i = 0;
for (; i < LOOP_COUNT; i++) {
const char *frame_data_pointer = p->frame_data;
int frame_count = p->frame_count;
/* Write in to the pipe. */
while (frame_count > 0) {
write (p->fd[1], frame_data_pointer, p->frame_data_size);
frame_data_pointer += p->frame_data_size;
frame_count--;
}
}
/* Run mainloop. */
g_main_loop_run (mainloop);
/* Set pipeline to NULL. */
gst_element_set_state (p->pipeline, GST_STATE_NULL);
/* Release mainloop. */
g_main_loop_unref (mainloop);
}
/*
* Creates the RTP pipeline and runs the test using the pipeline.
* @param frame_data Pointer to the frame data which is used to pass thru pay/depayloaders.
* @param frame_data_size Frame data size in bytes.
* @param frame_count Frame count.
* @param filtercaps Caps filters.
* @param pay Payloader name.
* @param depay Depayloader name.
*/
static void
rtp_pipeline_test (const char *frame_data, int frame_data_size, int frame_count,
const char *filtercaps, const char *pay, const char *depay)
{
/* Create RTP pipeline. */
rtp_pipeline *p =
rtp_pipeline_create (frame_data, frame_data_size, frame_count, filtercaps,
pay, depay);
if (p == NULL) {
return;
}
/* Run RTP pipeline. */
rtp_pipeline_run (p);
/* Destroy RTP pipeline. */
rtp_pipeline_destroy (p);
}
static char rtp_ilbc_frame_data[] =
{ 0x00, 0x00, 0x00, 0x00, 0x00, 0x00, 0x00, 0x00, 0x00, 0x00, 0x00, 0x00,
0x00, 0x00, 0x00, 0x00, 0x00, 0x00, 0x00, 0x00, 0x00
};
static int rtp_ilbc_frame_data_size = 20;
static int rtp_ilbc_frame_count = 1;
GST_START_TEST (rtp_ilbc)
{
rtp_pipeline_test (rtp_ilbc_frame_data, rtp_ilbc_frame_data_size,
rtp_ilbc_frame_count, "audio/x-iLBC,mode=20", "rtpilbcpay",
"rtpilbcdepay");
}
GST_END_TEST;
static char rtp_gsm_frame_data[] =
{ 0x00, 0x00, 0x00, 0x00, 0x00, 0x00, 0x00, 0x00, 0x00, 0x00, 0x00, 0x00,
0x00, 0x00, 0x00, 0x00, 0x00, 0x00, 0x00, 0x00, 0x00
};
static int rtp_gsm_frame_data_size = 20;
static int rtp_gsm_frame_count = 1;
GST_START_TEST (rtp_gsm)
{
rtp_pipeline_test (rtp_gsm_frame_data, rtp_gsm_frame_data_size,
rtp_gsm_frame_count, "audio/x-gsm,rate=8000,channels=1", "rtpgsmpay",
"rtpgsmdepay");
}
GST_END_TEST;
static char rtp_amr_frame_data[] =
{ 0x00, 0x00, 0x00, 0x00, 0x00, 0x00, 0x00, 0x00, 0x00, 0x00, 0x00, 0x00,
0x00, 0x00, 0x00, 0x00, 0x00, 0x00, 0x00, 0x00, 0x00
};
static int rtp_amr_frame_data_size = 20;
static int rtp_amr_frame_count = 1;
GST_START_TEST (rtp_amr)
{
rtp_pipeline_test (rtp_amr_frame_data, rtp_amr_frame_data_size,
rtp_amr_frame_count, "audio/AMR,channels=1,rate=8000", "rtpamrpay",
"rtpamrdepay");
}
GST_END_TEST;
static char rtp_pcma_frame_data[] =
{ 0x00, 0x00, 0x00, 0x00, 0x00, 0x00, 0x00, 0x00, 0x00, 0x00, 0x00, 0x00,
0x00, 0x00, 0x00, 0x00, 0x00, 0x00, 0x00, 0x00, 0x00
};
static int rtp_pcma_frame_data_size = 20;
static int rtp_pcma_frame_count = 1;
GST_START_TEST (rtp_pcma)
{
rtp_pipeline_test (rtp_pcma_frame_data, rtp_pcma_frame_data_size,
rtp_pcma_frame_count, "audio/x-alaw,channels=1,rate=8000", "rtppcmapay",
"rtppcmadepay");
}
GST_END_TEST;
static char rtp_pcmu_frame_data[] =
{ 0x00, 0x00, 0x00, 0x00, 0x00, 0x00, 0x00, 0x00, 0x00, 0x00, 0x00, 0x00,
0x00, 0x00, 0x00, 0x00, 0x00, 0x00, 0x00, 0x00, 0x00
};
static int rtp_pcmu_frame_data_size = 20;
static int rtp_pcmu_frame_count = 1;
GST_START_TEST (rtp_pcmu)
{
rtp_pipeline_test (rtp_pcmu_frame_data, rtp_pcmu_frame_data_size,
rtp_pcmu_frame_count, "audio/x-mulaw,channels=1,rate=8000", "rtppcmupay",
"rtppcmudepay");
}
GST_END_TEST;
static char rtp_mpa_frame_data[] =
{ 0x00, 0x00, 0x00, 0x00, 0x00, 0x00, 0x00, 0x00, 0x00, 0x00, 0x00, 0x00,
0x00, 0x00, 0x00, 0x00, 0x00, 0x00, 0x00, 0x00, 0x00
};
static int rtp_mpa_frame_data_size = 20;
static int rtp_mpa_frame_count = 1;
GST_START_TEST (rtp_mpa)
{
rtp_pipeline_test (rtp_mpa_frame_data, rtp_mpa_frame_data_size,
rtp_mpa_frame_count, "audio/mpeg", "rtpmpapay", "rtpmpadepay");
}
GST_END_TEST;
static char rtp_h263_frame_data[] =
{ 0x00, 0x00, 0x00, 0x00, 0x00, 0x00, 0x00, 0x00, 0x00, 0x00, 0x00, 0x00,
0x00, 0x00, 0x00, 0x00, 0x00, 0x00, 0x00, 0x00, 0x00
};
static int rtp_h263_frame_data_size = 20;
static int rtp_h263_frame_count = 1;
GST_START_TEST (rtp_h263)
{
rtp_pipeline_test (rtp_h263_frame_data, rtp_h263_frame_data_size,
rtp_h263_frame_count, "video/x-h263,variant=itu,h263version=h263",
"rtph263pay", "rtph263depay");
}
GST_END_TEST;
static char rtp_h263p_frame_data[] =
{ 0x00, 0x00, 0x00, 0x00, 0x00, 0x00, 0x00, 0x00, 0x00, 0x00, 0x00, 0x00,
0x00, 0x00, 0x00, 0x00, 0x00, 0x00, 0x00, 0x00, 0x00
};
static int rtp_h263p_frame_data_size = 20;
static int rtp_h263p_frame_count = 1;
GST_START_TEST (rtp_h263p)
{
rtp_pipeline_test (rtp_h263p_frame_data, rtp_h263p_frame_data_size,
rtp_h263p_frame_count, "video/x-h263,variant=itu", "rtph263ppay",
"rtph263pdepay");
}
GST_END_TEST;
static char rtp_h264_frame_data[] =
{ 0x00, 0x00, 0x00, 0x00, 0x00, 0x00, 0x00, 0x00, 0x00, 0x00, 0x00, 0x00,
0x00, 0x00, 0x00, 0x00, 0x00, 0x00, 0x00, 0x00, 0x00
};
static int rtp_h264_frame_data_size = 20;
static int rtp_h264_frame_count = 1;
GST_START_TEST (rtp_h264)
{
rtp_pipeline_test (rtp_h264_frame_data, rtp_h264_frame_data_size,
rtp_h264_frame_count, "video/x-h264", "rtph264pay", "rtph264depay");
}
GST_END_TEST;
static char rtp_L16_frame_data[] =
{ 0x00, 0x00, 0x00, 0x00, 0x00, 0x00, 0x00, 0x00, 0x00, 0x00, 0x00, 0x00,
0x00, 0x00, 0x00, 0x00, 0x00, 0x00, 0x00, 0x00, 0x00
};
static int rtp_L16_frame_data_size = 20;
static int rtp_L16_frame_count = 1;
GST_START_TEST (rtp_L16)
{
rtp_pipeline_test (rtp_L16_frame_data, rtp_L16_frame_data_size,
rtp_L16_frame_count,
"audio/x-raw-int,endianess=4321,signed=true,width=16,depth=16,rate=1,channels=1",
"rtpL16pay", "rtpL16depay");
}
GST_END_TEST;
static char rtp_mp2t_frame_data[] =
{ 0x00, 0x00, 0x00, 0x00, 0x00, 0x00, 0x00, 0x00, 0x00, 0x00, 0x00, 0x00,
0x00, 0x00, 0x00, 0x00, 0x00, 0x00, 0x00, 0x00, 0x00
};
static int rtp_mp2t_frame_data_size = 20;
static int rtp_mp2t_frame_count = 1;
GST_START_TEST (rtp_mp2t)
{
rtp_pipeline_test (rtp_mp2t_frame_data, rtp_mp2t_frame_data_size,
rtp_mp2t_frame_count, "video/mpegts,packetsize=188,systemstream=true",
"rtpmp2tpay", "rtpmp2tdepay");
}
GST_END_TEST;
static char rtp_mp4v_frame_data[] =
{ 0x00, 0x00, 0x00, 0x00, 0x00, 0x00, 0x00, 0x00, 0x00, 0x00, 0x00, 0x00,
0x00, 0x00, 0x00, 0x00, 0x00, 0x00, 0x00, 0x00, 0x00
};
static int rtp_mp4v_frame_data_size = 20;
static int rtp_mp4v_frame_count = 1;
GST_START_TEST (rtp_mp4v)
{
rtp_pipeline_test (rtp_mp4v_frame_data, rtp_mp4v_frame_data_size,
rtp_mp4v_frame_count, "video/mpeg,mpegversion=4,systemstream=false",
"rtpmp4vpay", "rtpmp4vdepay");
}
GST_END_TEST;
static char rtp_mp4g_frame_data[] =
{ 0x00, 0x00, 0x00, 0x00, 0x00, 0x00, 0x00, 0x00, 0x00, 0x00, 0x00, 0x00,
0x00, 0x00, 0x00, 0x00, 0x00, 0x00, 0x00, 0x00, 0x00
};
static int rtp_mp4g_frame_data_size = 20;
static int rtp_mp4g_frame_count = 1;
GST_START_TEST (rtp_mp4g)
{
rtp_pipeline_test (rtp_mp4g_frame_data, rtp_mp4g_frame_data_size,
rtp_mp4g_frame_count, "video/mpeg,mpegversion=4", "rtpmp4gpay",
"rtpmp4gdepay");
}
GST_END_TEST;
static char rtp_theora_frame_data[] =
{ 0x00, 0x00, 0x00, 0x00, 0x00, 0x00, 0x00, 0x00, 0x00, 0x00, 0x00, 0x00,
0x00, 0x00, 0x00, 0x00, 0x00, 0x00, 0x00, 0x00, 0x00
};
static int rtp_theora_frame_data_size = 20;
static int rtp_theora_frame_count = 1;
GST_START_TEST (rtp_theora)
{
rtp_pipeline_test (rtp_theora_frame_data, rtp_theora_frame_data_size,
rtp_theora_frame_count, "video/x-theora", "rtptheorapay",
"rtptheoradepay");
}
GST_END_TEST;
static char rtp_vorbis_frame_data[] =
{ 0x00, 0x00, 0x00, 0x00, 0x00, 0x00, 0x00, 0x00, 0x00, 0x00, 0x00, 0x00,
0x00, 0x00, 0x00, 0x00, 0x00, 0x00, 0x00, 0x00, 0x00
};
static int rtp_vorbis_frame_data_size = 20;
static int rtp_vorbis_frame_count = 1;
GST_START_TEST (rtp_vorbis)
{
rtp_pipeline_test (rtp_vorbis_frame_data, rtp_vorbis_frame_data_size,
rtp_vorbis_frame_count, "audio/x-vorbis", "rtpvorbispay",
"rtpvorbisdepay");
}
GST_END_TEST;
/*
* Creates the test suite.
*
* Returns: pointer to the test suite.
*/
static Suite *
rtp_payloading_suite ()
{
Suite *s = suite_create ("rtp_data_test");
TCase *tc_chain = tcase_create ("linear");
/* Set timeout to 60 seconds. */
tcase_set_timeout (tc_chain, 60);
suite_add_tcase (s, tc_chain);
tcase_add_test (tc_chain, rtp_ilbc);
tcase_add_test (tc_chain, rtp_gsm);
tcase_add_test (tc_chain, rtp_amr);
tcase_add_test (tc_chain, rtp_pcma);
tcase_add_test (tc_chain, rtp_pcmu);
tcase_add_test (tc_chain, rtp_mpa);
tcase_add_test (tc_chain, rtp_h263);
tcase_add_test (tc_chain, rtp_h263p);
tcase_add_test (tc_chain, rtp_h264);
tcase_add_test (tc_chain, rtp_L16);
tcase_add_test (tc_chain, rtp_mp2t);
tcase_add_test (tc_chain, rtp_mp4v);
tcase_add_test (tc_chain, rtp_mp4g);
tcase_add_test (tc_chain, rtp_theora);
tcase_add_test (tc_chain, rtp_vorbis);
return s;
}
GST_CHECK_MAIN (rtp_payloading)