From: George Kiagiadakis Date: Wed, 31 Jan 2018 15:41:32 +0000 (+0200) Subject: tests: audiotestsrc: add unit test for non-interleaved audio output X-Git-Tag: 1.19.3~511^2~1598 X-Git-Url: http://review.tizen.org/git/?a=commitdiff_plain;h=48fa908e15ba474b97a3d2d08d092dee6a4bcd00;p=platform%2Fupstream%2Fgstreamer.git tests: audiotestsrc: add unit test for non-interleaved audio output https://bugzilla.gnome.org/show_bug.cgi?id=796739 --- diff --git a/tests/check/elements/audiotestsrc.c b/tests/check/elements/audiotestsrc.c index bb70c67..e99583a 100644 --- a/tests/check/elements/audiotestsrc.c +++ b/tests/check/elements/audiotestsrc.c @@ -25,6 +25,7 @@ #endif #include +#include #include /* For ease of programming we use globals to keep refs for our floating @@ -114,6 +115,188 @@ GST_START_TEST (test_all_waves) GST_END_TEST; + +#define TEST_LAYOUT_CHANNELS 6 + +static GstStaticPadTemplate sinktemplate_interleaved = +GST_STATIC_PAD_TEMPLATE ("sink", + GST_PAD_SINK, + GST_PAD_ALWAYS, + GST_STATIC_CAPS ("audio/x-raw, " + "format = (string) " GST_AUDIO_NE (S16) ", " + "channels = (int) " G_STRINGIFY (TEST_LAYOUT_CHANNELS) ", " + "rate = (int) [ 1, MAX ], layout = (string) interleaved") + ); + +static GstStaticPadTemplate sinktemplate_planar = +GST_STATIC_PAD_TEMPLATE ("sink", + GST_PAD_SINK, + GST_PAD_ALWAYS, + GST_STATIC_CAPS ("audio/x-raw, " + "format = (string) " GST_AUDIO_NE (S16) ", " + "channels = (int) " G_STRINGIFY (TEST_LAYOUT_CHANNELS) ", " + "rate = (int) [ 1, MAX ], layout = (string) non-interleaved") + ); + +typedef enum +{ + GST_AUDIO_TEST_SRC_WAVE_SINE, + GST_AUDIO_TEST_SRC_WAVE_SQUARE, + GST_AUDIO_TEST_SRC_WAVE_SAW, + GST_AUDIO_TEST_SRC_WAVE_TRIANGLE, + GST_AUDIO_TEST_SRC_WAVE_SILENCE, + GST_AUDIO_TEST_SRC_WAVE_WHITE_NOISE, + GST_AUDIO_TEST_SRC_WAVE_PINK_NOISE, + GST_AUDIO_TEST_SRC_WAVE_SINE_TAB, + GST_AUDIO_TEST_SRC_WAVE_TICKS, + GST_AUDIO_TEST_SRC_WAVE_GAUSSIAN_WHITE_NOISE, + GST_AUDIO_TEST_SRC_WAVE_RED_NOISE, + GST_AUDIO_TEST_SRC_WAVE_BLUE_NOISE, + GST_AUDIO_TEST_SRC_WAVE_VIOLET_NOISE, + + _GST_AUDIO_TEST_SRC_WAVE_LAST +} GstAudioTestSrcWave; + +GST_START_TEST (test_layout) +{ + GstHarness *interleavedsrc, *plannarsrc; + GObjectClass *oclass; + GParamSpec *property; + GEnumValue *values; + guint i, j; + + interleavedsrc = gst_harness_new_with_templates ("audiotestsrc", NULL, + &sinktemplate_interleaved); + plannarsrc = gst_harness_new_with_templates ("audiotestsrc", NULL, + &sinktemplate_planar); + + gst_harness_use_testclock (interleavedsrc); + gst_harness_use_testclock (plannarsrc); + g_object_set (interleavedsrc->element, "is-live", TRUE, NULL); + g_object_set (plannarsrc->element, "is-live", TRUE, NULL); + + oclass = G_OBJECT_GET_CLASS (interleavedsrc->element); + property = g_object_class_find_property (oclass, "wave"); + fail_unless (G_IS_PARAM_SPEC_ENUM (property)); + values = G_ENUM_CLASS (g_type_class_ref (property->value_type))->values; + + for (j = 0; values[j].value_name; j++) { + /* these produce random values by definition, + * so we can't compare channels */ + switch (j) { + case GST_AUDIO_TEST_SRC_WAVE_WHITE_NOISE: + case GST_AUDIO_TEST_SRC_WAVE_PINK_NOISE: + case GST_AUDIO_TEST_SRC_WAVE_GAUSSIAN_WHITE_NOISE: + case GST_AUDIO_TEST_SRC_WAVE_RED_NOISE: + case GST_AUDIO_TEST_SRC_WAVE_BLUE_NOISE: + case GST_AUDIO_TEST_SRC_WAVE_VIOLET_NOISE: + continue; + default: + break; + } + + GST_DEBUG ("layout test with wave %s", values[j].value_name); + g_object_set (interleavedsrc->element, "wave", values[j].value, NULL); + g_object_set (plannarsrc->element, "wave", values[j].value, NULL); + + if (j == 0) { + GST_DEBUG ("gst_harness_play"); + gst_harness_play (interleavedsrc); + gst_harness_play (plannarsrc); + } else { + GST_DEBUG ("discarding buffers with old wave"); + fail_unless (gst_harness_crank_single_clock_wait (interleavedsrc)); + fail_unless (gst_harness_crank_single_clock_wait (plannarsrc)); + gst_buffer_unref (gst_harness_pull (interleavedsrc)); + gst_buffer_unref (gst_harness_pull (plannarsrc)); + } + + for (i = 0; i < 10; i++) { + GstBuffer *ibuf, *pbuf; + GstMapInfo imap, pmap; + GstAudioMeta *meta; + GstAudioBuffer pabuf; + gint16 *iptr, *pptr; + guint isamples, psamples, s, c; + + GST_DEBUG ("waiting on clock"); + fail_unless (gst_harness_crank_single_clock_wait (interleavedsrc)); + fail_unless (gst_harness_crank_single_clock_wait (plannarsrc)); + + ibuf = gst_harness_pull (interleavedsrc); + pbuf = gst_harness_pull (plannarsrc); + + gst_buffer_map (ibuf, &imap, GST_MAP_READ); + gst_buffer_map (pbuf, &pmap, GST_MAP_READ); + + /* buffers should have the same size in bytes and in samples */ + fail_unless_equals_int (imap.size, pmap.size); + isamples = imap.size / TEST_LAYOUT_CHANNELS; + isamples /= 2; /* S16 -> 2 bytes per sample */ + fail_unless_equals_int (imap.size % TEST_LAYOUT_CHANNELS, 0); + psamples = pmap.size / TEST_LAYOUT_CHANNELS; + psamples /= 2; /* S16 -> 2 bytes per sample */ + fail_unless_equals_int (pmap.size % TEST_LAYOUT_CHANNELS, 0); + fail_unless_equals_int (isamples, psamples); + + iptr = (gint16 *) imap.data; + pptr = (gint16 *) pmap.data; + + GST_DEBUG ("verifying contents of buffers; samples=%d, channels=%d", + isamples, TEST_LAYOUT_CHANNELS); + + for (s = 0; s < isamples; s++) { + for (c = 0; c < TEST_LAYOUT_CHANNELS; c++) { + guint iidx = s * TEST_LAYOUT_CHANNELS + c; + guint pidx = c * isamples + s; + + GST_TRACE ("s = %u | c = %u | iidx (s * channels + c) = %u | " + "pidx (c * samples + s) = %u", s, c, iidx, pidx); + + fail_unless (iidx < imap.size / 2); + fail_unless (pidx < pmap.size / 2); + fail_unless_equals_int (iptr[iidx], pptr[pidx]); + } + } + + gst_buffer_unmap (pbuf, &pmap); + + GST_DEBUG ("verify that mapping through GstAudioBuffer works the same"); + + meta = gst_buffer_get_audio_meta (pbuf); + fail_unless (meta); + + gst_audio_buffer_map (&pabuf, &meta->info, pbuf, GST_MAP_READ); + + for (s = 0; s < isamples; s++) { + for (c = 0; c < TEST_LAYOUT_CHANNELS; c++) { + guint iidx = s * TEST_LAYOUT_CHANNELS + c; + + fail_unless_equals_int (iptr[iidx], ((gint16 *) pabuf.planes[c])[s]); + } + } + + gst_audio_buffer_unmap (&pabuf); + gst_buffer_unmap (ibuf, &imap); + + gst_buffer_unref (ibuf); + gst_buffer_unref (pbuf); + } + + /* ensure the audiotestsrcs are not in fill() while we change the wave */ + fail_unless (gst_harness_wait_for_clock_id_waits (interleavedsrc, 1, 1)); + fail_unless (gst_harness_wait_for_clock_id_waits (plannarsrc, 1, 1)); + } + + /* make sure we ran the test */ + fail_unless_equals_int (j, _GST_AUDIO_TEST_SRC_WAVE_LAST); + + gst_harness_teardown (interleavedsrc); + gst_harness_teardown (plannarsrc); +} + +GST_END_TEST; + static Suite * audiotestsrc_suite (void) { @@ -122,6 +305,7 @@ audiotestsrc_suite (void) suite_add_tcase (s, tc_chain); tcase_add_test (tc_chain, test_all_waves); + tcase_add_test (tc_chain, test_layout); return s; }