diff --git a/ChangeLog b/ChangeLog index b93d1fcfc8..4cf9dd9e1b 100644 --- a/ChangeLog +++ b/ChangeLog @@ -1,3 +1,12 @@ +2006-10-21 Tim-Philipp Müller + + * tests/check/Makefile.am: + * tests/check/elements/.cvsignore: + * tests/check/elements/audiorate.c: (probe_cb), (got_buf), + (do_perfect_stream_test), (GST_START_TEST), (audiorate_suite): + Add some basic unit tests for audiorate. Disabled at the moment + since it doesn't pass yet (see bug #363119). + 2006-10-20 Tim-Philipp Müller * gst/subparse/gstsubparse.c: (subrip_fix_up_markup), diff --git a/common b/common index efcacf2625..ee0bb43e2b 160000 --- a/common +++ b/common @@ -1 +1 @@ -Subproject commit efcacf2625da231fbee99b68e0f5db6816cf6fad +Subproject commit ee0bb43e2b66781d04078e2210404da48f6c68f0 diff --git a/tests/check/Makefile.am b/tests/check/Makefile.am index f57e688fed..ad39289037 100644 --- a/tests/check/Makefile.am +++ b/tests/check/Makefile.am @@ -81,6 +81,7 @@ VALGRIND_TO_FIX = \ # these tests don't even pass noinst_PROGRAMS = \ + elements/audiorate \ elements/ffmpegcolorspace AM_CFLAGS = $(GST_CFLAGS) $(GST_CHECK_CFLAGS) @@ -111,6 +112,9 @@ elements_audioconvert_LDADD = \ $(top_builddir)/gst-libs/gst/audio/libgstaudio-@GST_MAJORMINOR@.la \ $(LDADD) +elements_audiorate_LDADD = $(LDADD) +elements_audiorate_CFLAGS = $(CFLAGS) $(AM_CFLAGS) + elements_gdpdepay_LDADD = $(GST_GDP_LIBS) $(LDADD) elements_gdppay_LDADD = $(GST_GDP_LIBS) $(LDADD) diff --git a/tests/check/elements/.gitignore b/tests/check/elements/.gitignore index 0e31e6beed..70fba4e8cb 100644 --- a/tests/check/elements/.gitignore +++ b/tests/check/elements/.gitignore @@ -2,6 +2,7 @@ adder alsa audioconvert +audiorate audioresample audiotestsrc gdpdepay diff --git a/tests/check/elements/audiorate.c b/tests/check/elements/audiorate.c new file mode 100644 index 0000000000..b15e1128c0 --- /dev/null +++ b/tests/check/elements/audiorate.c @@ -0,0 +1,219 @@ +/* GStreamer unit tests for audiorate + * + * Copyright (C) 2006 Tim-Philipp Müller + * + * 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. + */ + +#ifdef HAVE_CONFIG_H +# include +#endif + +#include + +static gboolean +probe_cb (GstPad * pad, GstBuffer * buf, gdouble * drop_probability) +{ + if (g_random_double () < *drop_probability) { + GST_LOG ("dropping buffer"); + return FALSE; /* drop buffer */ + } + + return TRUE; /* don't drop buffer */ +} + +static void +got_buf (GstElement * fakesink, GstBuffer * buf, GstPad * pad, GList ** p_bufs) +{ + *p_bufs = g_list_append (*p_bufs, gst_buffer_ref (buf)); +} + +static void +do_perfect_stream_test (guint rate, guint width, gdouble drop_probability) +{ + GstElement *pipe, *src, *conv, *filter, *audiorate, *sink; + GstMessage *msg; + GstCaps *caps; + GstPad *srcpad; + GList *l, *bufs = NULL; + GstClockTime next_time = GST_CLOCK_TIME_NONE; + gint64 next_offset = GST_BUFFER_OFFSET_NONE; + + caps = gst_caps_new_simple ("audio/x-raw-int", "rate", G_TYPE_INT, + rate, "width", G_TYPE_INT, width, NULL); + + GST_INFO ("-------- drop=%.0f%% caps = %" GST_PTR_FORMAT " ---------- ", + drop_probability * 100.0, caps); + + g_assert (drop_probability >= 0.0 && drop_probability <= 1.0); + g_assert (width > 0 && (width % 8) == 0); + + pipe = gst_pipeline_new ("pipeline"); + fail_unless (pipe != NULL); + + src = gst_element_factory_make ("audiotestsrc", "audiotestsrc"); + fail_unless (src != NULL); + + g_object_set (src, "num-buffers", 500, NULL); + + conv = gst_element_factory_make ("audioconvert", "audioconvert"); + fail_unless (conv != NULL); + + filter = gst_element_factory_make ("capsfilter", "capsfilter"); + fail_unless (filter != NULL); + + g_object_set (filter, "caps", caps, NULL); + + srcpad = gst_element_get_pad (filter, "src"); + fail_unless (srcpad != NULL); + gst_pad_add_buffer_probe (srcpad, G_CALLBACK (probe_cb), &drop_probability); + gst_object_unref (srcpad); + + audiorate = gst_element_factory_make ("audiorate", "audiorate"); + fail_unless (audiorate != NULL); + + sink = gst_element_factory_make ("fakesink", "fakesink"); + fail_unless (sink != NULL); + + g_object_set (sink, "signal-handoffs", TRUE, NULL); + + g_signal_connect (sink, "handoff", G_CALLBACK (got_buf), &bufs); + + gst_bin_add_many (GST_BIN (pipe), src, conv, filter, audiorate, sink, NULL); + gst_element_link_many (src, conv, filter, audiorate, sink, NULL); + + fail_unless_equals_int (gst_element_set_state (pipe, GST_STATE_PLAYING), + GST_STATE_CHANGE_ASYNC); + + fail_unless_equals_int (gst_element_get_state (pipe, NULL, NULL, -1), + GST_STATE_CHANGE_SUCCESS); + + msg = gst_bus_poll (GST_ELEMENT_BUS (pipe), + GST_MESSAGE_EOS | GST_MESSAGE_ERROR, -1); + fail_unless_equals_string (GST_MESSAGE_TYPE_NAME (msg), "eos"); + + for (l = bufs; l != NULL; l = l->next) { + GstBuffer *buf = GST_BUFFER (l->data); + guint num_samples; + + fail_unless (GST_BUFFER_TIMESTAMP_IS_VALID (buf)); + fail_unless (GST_BUFFER_DURATION_IS_VALID (buf)); + fail_unless (GST_BUFFER_OFFSET_IS_VALID (buf)); + fail_unless (GST_BUFFER_OFFSET_END_IS_VALID (buf)); + + GST_LOG ("buffer: ts=%" GST_TIME_FORMAT ", end_ts=%" GST_TIME_FORMAT + " off=%" G_GINT64_FORMAT ", end_off=%" G_GINT64_FORMAT, + GST_TIME_ARGS (GST_BUFFER_TIMESTAMP (buf)), + GST_TIME_ARGS (GST_BUFFER_TIMESTAMP (buf) + GST_BUFFER_DURATION (buf)), + GST_BUFFER_OFFSET (buf), GST_BUFFER_OFFSET_END (buf)); + + if (GST_CLOCK_TIME_IS_VALID (next_time)) { + fail_unless_equals_uint64 (next_time, GST_BUFFER_TIMESTAMP (buf)); + } + if (next_offset != GST_BUFFER_OFFSET_NONE) { + fail_unless_equals_uint64 (next_offset, GST_BUFFER_OFFSET (buf)); + } + + /* check buffer size for sanity */ + fail_unless_equals_int (GST_BUFFER_SIZE (buf) % (width / 8), 0); + + /* check there is actually as much data as there should be */ + num_samples = GST_BUFFER_OFFSET_END (buf) - GST_BUFFER_OFFSET (buf); + fail_unless_equals_int (GST_BUFFER_SIZE (buf), num_samples * (width / 8)); + + next_time = GST_BUFFER_TIMESTAMP (buf) + GST_BUFFER_DURATION (buf); + next_offset = GST_BUFFER_OFFSET_END (buf); + } + + gst_message_unref (msg); + gst_element_set_state (pipe, GST_STATE_NULL); + gst_object_unref (pipe); + + g_list_foreach (bufs, (GFunc) gst_mini_object_unref, NULL); + g_list_free (bufs); + + gst_caps_unref (caps); +} + +static const guint rates[] = { 8000, 11025, 16000, 22050, 32000, 44100, + 48000, 3333, 33333, 66666, 9999 +}; + +GST_START_TEST (test_perfect_stream_drop0) +{ + guint i; + + for (i = 0; i < G_N_ELEMENTS (rates); ++i) { + do_perfect_stream_test (rates[i], 8, 0.0); + do_perfect_stream_test (rates[i], 16, 0.0); + } +} + +GST_END_TEST; + +GST_START_TEST (test_perfect_stream_drop10) +{ + guint i; + + for (i = 0; i < G_N_ELEMENTS (rates); ++i) { + do_perfect_stream_test (rates[i], 8, 0.10); + do_perfect_stream_test (rates[i], 16, 0.10); + } +} + +GST_END_TEST; + +GST_START_TEST (test_perfect_stream_drop50) +{ + guint i; + + for (i = 0; i < G_N_ELEMENTS (rates); ++i) { + do_perfect_stream_test (rates[i], 8, 0.50); + do_perfect_stream_test (rates[i], 16, 0.50); + } +} + +GST_END_TEST; + +GST_START_TEST (test_perfect_stream_drop90) +{ + guint i; + + for (i = 0; i < G_N_ELEMENTS (rates); ++i) { + do_perfect_stream_test (rates[i], 8, 0.90); + do_perfect_stream_test (rates[i], 16, 0.90); + } +} + +GST_END_TEST; + +static Suite * +audiorate_suite (void) +{ + Suite *s = suite_create ("audiorate"); + TCase *tc_chain = tcase_create ("general"); + + suite_add_tcase (s, tc_chain); + + tcase_add_test (tc_chain, test_perfect_stream_drop0); + tcase_add_test (tc_chain, test_perfect_stream_drop10); + tcase_add_test (tc_chain, test_perfect_stream_drop50); + tcase_add_test (tc_chain, test_perfect_stream_drop90); + + return s; +} + +GST_CHECK_MAIN (audiorate);