mirror of
https://gitlab.freedesktop.org/gstreamer/gstreamer.git
synced 2024-10-17 07:53:55 +00:00
995a135df6
gstcheck is declaring its own buffers glist which ends up overwritten, loks like the code meant to use that already gcc 10 is also complaining about this, but only on static builds for some reason ``` FAILED: subprojects/gst-plugins-ugly/tests/check/elements_amrnbenc /usr/bin/ld: subprojects/gstreamer/libs/gst/check/libgstcheck-1.0.a(gstcheck.c.o):(.bss+0x38): multiple definition of `buffers'; subprojects/gst-plugins-ugly/tests/check/708af1f@@elements_amrnbenc@exe/elements_amrnbenc.c.o:(.bss+0x18): first defined here collect2: error: ld returned 1 exit status ``` also remove unused var `current_buf` Part-of: <https://gitlab.freedesktop.org/gstreamer/gst-plugins-ugly/-/merge_requests/62>
144 lines
3.7 KiB
C
144 lines
3.7 KiB
C
/*
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* GStreamer
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*
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* unit test for amrnbenc
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*
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* Copyright (C) 2006 Thomas Vander Stichele <thomas at apestaart dot org>
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*
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* This library is free software; you can redistribute it and/or
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* modify it under the terms of the GNU Library General Public
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* License as published by the Free Software Foundation; either
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* version 2 of the License, or (at your option) any later version.
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*
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* This library is distributed in the hope that it will be useful,
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* but WITHOUT ANY WARRANTY; without even the implied warranty of
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* MERCHANTABILITY or FITNESS FOR A PARTICULAR PURPOSE. See the GNU
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* Library General Public License for more details.
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*
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* You should have received a copy of the GNU Library General Public
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* License along with this library; if not, write to the
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* Free Software Foundation, Inc., 51 Franklin St, Fifth Floor,
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* Boston, MA 02110-1301, USA.
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*/
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#include <gst/check/gstcheck.h>
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#include <gst/audio/audio.h>
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#define SRC_CAPS "audio/x-raw, format = (string)" GST_AUDIO_NE (S16) ", " \
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"layout = (string) interleaved, channels = (int) 1, rate = (int) 8000"
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#define SINK_CAPS "audio/AMR"
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static GstPad *srcpad, *sinkpad;
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static GstStaticPadTemplate sinktemplate = GST_STATIC_PAD_TEMPLATE ("sink",
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GST_PAD_SINK,
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GST_PAD_ALWAYS,
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GST_STATIC_CAPS (SINK_CAPS)
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);
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static GstStaticPadTemplate srctemplate = GST_STATIC_PAD_TEMPLATE ("src",
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GST_PAD_SRC,
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GST_PAD_ALWAYS,
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GST_STATIC_CAPS (SRC_CAPS)
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);
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static void
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buffer_unref (void *buffer, void *user_data)
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{
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gst_buffer_unref (GST_BUFFER (buffer));
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}
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static GstElement *
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setup_amrnbenc (void)
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{
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GstElement *amrnbenc;
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GstCaps *caps;
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GstBus *bus;
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GST_DEBUG ("setup_amrnbenc");
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amrnbenc = gst_check_setup_element ("amrnbenc");
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srcpad = gst_check_setup_src_pad (amrnbenc, &srctemplate);
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sinkpad = gst_check_setup_sink_pad (amrnbenc, &sinktemplate);
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gst_pad_set_active (srcpad, TRUE);
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gst_pad_set_active (sinkpad, TRUE);
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bus = gst_bus_new ();
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gst_element_set_bus (amrnbenc, bus);
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fail_unless (gst_element_set_state (amrnbenc,
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GST_STATE_PLAYING) != GST_STATE_CHANGE_FAILURE,
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"could not set to playing");
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caps = gst_caps_from_string (SRC_CAPS);
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gst_check_setup_events (srcpad, amrnbenc, caps, GST_FORMAT_TIME);
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gst_caps_unref (caps);
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buffers = NULL;
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return amrnbenc;
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}
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static void
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cleanup_amrnbenc (GstElement * amrnbenc)
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{
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GstBus *bus;
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/* free encoded buffers */
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g_list_foreach (buffers, buffer_unref, NULL);
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g_list_free (buffers);
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buffers = NULL;
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bus = GST_ELEMENT_BUS (amrnbenc);
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gst_bus_set_flushing (bus, TRUE);
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gst_object_unref (bus);
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GST_DEBUG ("cleanup_amrnbenc");
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gst_pad_set_active (srcpad, FALSE);
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gst_pad_set_active (sinkpad, FALSE);
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gst_check_teardown_src_pad (amrnbenc);
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gst_check_teardown_sink_pad (amrnbenc);
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gst_check_teardown_element (amrnbenc);
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}
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/* push a random block of audio of the given size */
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static void
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push_data (gint size, GstFlowReturn expected_return)
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{
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GstBuffer *buffer;
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GstFlowReturn res;
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buffer = gst_buffer_new_and_alloc (size);
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/* make valgrind happier */
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gst_buffer_memset (buffer, 0, 0, size);
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res = gst_pad_push (srcpad, buffer);
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fail_unless (res == expected_return,
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"pushing audio returned %d (%s) not %d (%s)", res,
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gst_flow_get_name (res), expected_return,
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gst_flow_get_name (expected_return));
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}
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GST_START_TEST (test_enc)
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{
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GstElement *amrnbenc;
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amrnbenc = setup_amrnbenc ();
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push_data (1000, GST_FLOW_OK);
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cleanup_amrnbenc (amrnbenc);
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}
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GST_END_TEST;
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static Suite *
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amrnbenc_suite ()
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{
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Suite *s = suite_create ("amrnbenc");
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TCase *tc_chain = tcase_create ("general");
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suite_add_tcase (s, tc_chain);
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tcase_add_test (tc_chain, test_enc);
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return s;
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}
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GST_CHECK_MAIN (amrnbenc);
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