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https://gitlab.freedesktop.org/gstreamer/gstreamer.git
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tests: add decodebin2 test for parser autoplugging
Make sure decodebin2 doesn't try to plug the same parser twice in a row.
This commit is contained in:
Tim-Philipp Müller
parent
b3a39daed6
commit
69dcacf148
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@ -1,6 +1,7 @@
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/* GStreamer unit tests for decodebin2
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*
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* Copyright (C) 2006 Tim-Philipp Müller <tim centricular net>
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* Copyright (C) 2011 Hewlett-Packard Development Company, L.P.
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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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@ -23,6 +24,7 @@
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#endif
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#include <gst/check/gstcheck.h>
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#include <gst/base/gstbaseparse.h>
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#include <unistd.h>
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static const gchar dummytext[] =
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@ -187,6 +189,177 @@ GST_START_TEST (test_reuse_without_decoders)
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GST_END_TEST;
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/* Fake mp3 parser for test */
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typedef GstBaseParse TestMpegAudioParse;
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typedef GstBaseParseClass TestMpegAudioParseClass;
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static GstStaticPadTemplate src_template = 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 ("audio/mpeg, mpegversion=1, layer=[1,3], parsed=(b)true")
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);
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static GstStaticPadTemplate sink_template = 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 ("audio/mpeg, mpegversion=1, parsed=(bool) { false, true }")
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);
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static GType test_mpeg_audio_parse_get_type (void);
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static gboolean test_mpeg_audio_parse_start (GstBaseParse * parse);
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static gboolean test_mpeg_audio_parse_stop (GstBaseParse * parse);
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static gboolean test_mpeg_audio_parse_check_valid_frame (GstBaseParse * parse,
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GstBaseParseFrame * frame, guint * size, gint * skipsize);
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static GstFlowReturn test_mpeg_audio_parse_parse_frame (GstBaseParse * parse,
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GstBaseParseFrame * frame);
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GST_BOILERPLATE (TestMpegAudioParse, test_mpeg_audio_parse, GstBaseParse,
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GST_TYPE_BASE_PARSE);
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static void
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test_mpeg_audio_parse_base_init (gpointer klass)
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{
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GstElementClass *element_class = GST_ELEMENT_CLASS (klass);
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gst_element_class_add_pad_template (element_class,
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gst_static_pad_template_get (&sink_template));
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gst_element_class_add_pad_template (element_class,
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gst_static_pad_template_get (&src_template));
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gst_element_class_set_details_simple (element_class, "MPEG1 Audio Parser",
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"Codec/Parser/Audio", "Pretends to parse mpeg1 audio stream",
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"Foo Bar <foo@bar.com>");
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}
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static void
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test_mpeg_audio_parse_class_init (TestMpegAudioParseClass * klass)
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{
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GstBaseParseClass *parse_class = GST_BASE_PARSE_CLASS (klass);
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parse_class->start = test_mpeg_audio_parse_start;
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parse_class->stop = test_mpeg_audio_parse_stop;
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parse_class->check_valid_frame = test_mpeg_audio_parse_check_valid_frame;
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parse_class->parse_frame = test_mpeg_audio_parse_parse_frame;
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}
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static gint num_parse_instances = 0;
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static void
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test_mpeg_audio_parse_init (TestMpegAudioParse * mp3parse,
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TestMpegAudioParseClass * klass)
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{
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/* catch decodebin plugging parsers in a loop early */
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fail_unless (++num_parse_instances < 10);
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}
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static gboolean
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test_mpeg_audio_parse_start (GstBaseParse * parse)
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{
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gst_base_parse_set_min_frame_size (parse, 6);
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return TRUE;
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}
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static gboolean
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test_mpeg_audio_parse_stop (GstBaseParse * parse)
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{
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return TRUE;
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}
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static gboolean
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test_mpeg_audio_parse_check_valid_frame (GstBaseParse * parse,
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GstBaseParseFrame * frame, guint * framesize, gint * skipsize)
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{
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const guint8 *data = GST_BUFFER_DATA (frame->buffer);
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if ((GST_READ_UINT16_BE (data) & 0xffe0) == 0xffe0) {
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/* this framesize is hard-coded for ../test.mp3 */
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*framesize = 1045;
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return TRUE;
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} else {
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*skipsize = 1;
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return FALSE;
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}
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}
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static GstFlowReturn
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test_mpeg_audio_parse_parse_frame (GstBaseParse * parse,
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GstBaseParseFrame * frame)
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{
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if (GST_BUFFER_OFFSET (frame->buffer) == 0) {
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GstCaps *caps;
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caps = gst_caps_new_simple ("audio/mpeg", "mpegversion", G_TYPE_INT, 1,
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"mpegaudioversion", G_TYPE_INT, 1, "layer", G_TYPE_INT, 3,
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"rate", G_TYPE_INT, 44100, "channels", G_TYPE_INT, 2, NULL);
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gst_buffer_set_caps (frame->buffer, caps);
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gst_pad_set_caps (GST_BASE_PARSE_SRC_PAD (parse), caps);
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gst_caps_unref (caps);
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}
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return GST_FLOW_OK;
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}
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static gboolean
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plugin_init (GstPlugin * plugin)
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{
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return gst_element_register (plugin, "testmpegaudioparse", GST_RANK_NONE,
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test_mpeg_audio_parse_get_type ());
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}
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GST_START_TEST (test_mp3_parser_loop)
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{
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GstStateChangeReturn sret;
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GstPluginFeature *feature;
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GstMessage *msg;
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GstElement *pipe, *src, *dec;
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gchar *path;
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num_parse_instances = 0;
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gst_plugin_register_static (GST_VERSION_MAJOR, GST_VERSION_MINOR,
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"fakemp3parse", "fakemp3parse", plugin_init, VERSION, "LGPL",
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"gst-plugins-base", GST_PACKAGE_NAME, GST_PACKAGE_ORIGIN);
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feature = gst_default_registry_find_feature ("testmpegaudioparse",
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GST_TYPE_ELEMENT_FACTORY);
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gst_plugin_feature_set_rank (feature, GST_RANK_PRIMARY + 100);
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pipe = gst_pipeline_new (NULL);
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src = gst_element_factory_make ("filesrc", NULL);
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fail_unless (src != NULL);
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path = g_build_filename (GST_TEST_FILES_PATH, "test.mp3", NULL);
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g_object_set (src, "location", path, NULL);
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g_free (path);
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dec = gst_element_factory_make ("decodebin2", NULL);
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fail_unless (dec != NULL);
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gst_bin_add_many (GST_BIN (pipe), src, dec, NULL);
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gst_element_link_many (src, dec, NULL);
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sret = gst_element_set_state (pipe, GST_STATE_PLAYING);
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fail_unless_equals_int (sret, GST_STATE_CHANGE_ASYNC);
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/* wait for unlinked error */
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msg = gst_bus_timed_pop_filtered (GST_ELEMENT_BUS (pipe),
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GST_CLOCK_TIME_NONE, GST_MESSAGE_ERROR);
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gst_message_unref (msg);
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gst_element_set_state (pipe, GST_STATE_NULL);
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gst_object_unref (pipe);
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/* make sure out parser got plugged at all though */
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fail_unless_equals_int (num_parse_instances, 1);
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/* don't want to interfere with any other of the other tests */
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gst_plugin_feature_set_rank (feature, GST_RANK_NONE);
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gst_object_unref (feature);
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}
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GST_END_TEST;
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static Suite *
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decodebin2_suite (void)
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{
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@ -196,6 +369,7 @@ decodebin2_suite (void)
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suite_add_tcase (s, tc_chain);
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tcase_add_test (tc_chain, test_text_plain_streams);
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tcase_add_test (tc_chain, test_reuse_without_decoders);
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tcase_add_test (tc_chain, test_mp3_parser_loop);
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return s;
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}
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