| /* GStreamer unit tests for audiorate | |
| * | |
| * Copyright (C) 2006 Tim-Philipp Müller <tim centricular net> | |
| * | |
| * 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., 51 Franklin St, Fifth Floor, | |
| * Boston, MA 02110-1301, USA. | |
| */ | |
| #ifdef HAVE_CONFIG_H | |
| # include <config.h> | |
| #endif | |
| #include <gst/check/gstcheck.h> | |
| #include <gst/audio/audio.h> | |
| #include <gst/app/gstappsrc.h> | |
| /* helper element to insert additional buffers overlapping with previous ones */ | |
| static gdouble injector_inject_probability = 0.0; | |
| typedef GstElement TestInjector; | |
| typedef GstElementClass TestInjectorClass; | |
| GType test_injector_get_type (void); | |
| G_DEFINE_TYPE (TestInjector, test_injector, GST_TYPE_ELEMENT); | |
| #define FORMATS "{ "GST_AUDIO_NE(F32)", S8, S16LE, S16BE, " \ | |
| "U16LE, U16NE, S32LE, S32BE, U32LE, U32BE }" | |
| #define INJECTOR_CAPS \ | |
| "audio/x-raw, " \ | |
| "format = (string) "FORMATS", " \ | |
| "rate = (int) [ 1, MAX ], " \ | |
| "channels = (int) [ 1, 8 ]" | |
| static GstStaticPadTemplate src_template = GST_STATIC_PAD_TEMPLATE ("src", | |
| GST_PAD_SRC, | |
| GST_PAD_ALWAYS, | |
| GST_STATIC_CAPS (INJECTOR_CAPS)); | |
| static GstStaticPadTemplate sink_template = GST_STATIC_PAD_TEMPLATE ("sink", | |
| GST_PAD_SINK, | |
| GST_PAD_ALWAYS, | |
| GST_STATIC_CAPS (INJECTOR_CAPS)); | |
| static void | |
| test_injector_class_init (TestInjectorClass * klass) | |
| { | |
| GstElementClass *element_class = GST_ELEMENT_CLASS (klass); | |
| gst_element_class_add_static_pad_template (element_class, &src_template); | |
| gst_element_class_add_static_pad_template (element_class, &sink_template); | |
| } | |
| static GstFlowReturn | |
| test_injector_chain (GstPad * pad, GstObject * parent, GstBuffer * buf) | |
| { | |
| GstFlowReturn ret; | |
| GstPad *srcpad; | |
| srcpad = gst_element_get_static_pad (GST_ELEMENT (parent), "src"); | |
| /* since we're increasing timestamp/offsets, push this one first */ | |
| GST_LOG (" passing buffer [t=%" GST_TIME_FORMAT "-%" GST_TIME_FORMAT | |
| "], offset=%" G_GINT64_FORMAT ", offset_end=%" 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)); | |
| gst_buffer_ref (buf); | |
| ret = gst_pad_push (srcpad, buf); | |
| if (g_random_double () < injector_inject_probability) { | |
| GstBuffer *ibuf; | |
| ibuf = gst_buffer_copy (buf); | |
| if (GST_BUFFER_OFFSET_IS_VALID (buf) && | |
| GST_BUFFER_OFFSET_END_IS_VALID (buf)) { | |
| guint64 delta; | |
| delta = GST_BUFFER_OFFSET_END (buf) - GST_BUFFER_OFFSET (buf); | |
| GST_BUFFER_OFFSET (ibuf) += delta / 4; | |
| GST_BUFFER_OFFSET_END (ibuf) += delta / 4; | |
| } else { | |
| GST_BUFFER_OFFSET (ibuf) = GST_BUFFER_OFFSET_NONE; | |
| GST_BUFFER_OFFSET_END (ibuf) = GST_BUFFER_OFFSET_NONE; | |
| } | |
| if (GST_BUFFER_TIMESTAMP_IS_VALID (buf) && | |
| GST_BUFFER_DURATION_IS_VALID (buf)) { | |
| GstClockTime delta; | |
| delta = GST_BUFFER_DURATION (buf); | |
| GST_BUFFER_TIMESTAMP (ibuf) += delta / 4; | |
| } else { | |
| GST_BUFFER_TIMESTAMP (ibuf) = GST_CLOCK_TIME_NONE; | |
| GST_BUFFER_DURATION (ibuf) = GST_CLOCK_TIME_NONE; | |
| } | |
| if (GST_BUFFER_TIMESTAMP_IS_VALID (ibuf) || | |
| GST_BUFFER_OFFSET_IS_VALID (ibuf)) { | |
| GST_LOG ("injecting buffer [t=%" GST_TIME_FORMAT "-%" GST_TIME_FORMAT | |
| "], offset=%" G_GINT64_FORMAT ", offset_end=%" G_GINT64_FORMAT, | |
| GST_TIME_ARGS (GST_BUFFER_TIMESTAMP (ibuf)), | |
| GST_TIME_ARGS (GST_BUFFER_TIMESTAMP (ibuf) + | |
| GST_BUFFER_DURATION (ibuf)), GST_BUFFER_OFFSET (ibuf), | |
| GST_BUFFER_OFFSET_END (ibuf)); | |
| if (gst_pad_push (srcpad, ibuf) != GST_FLOW_OK) { | |
| /* ignore return value */ | |
| } | |
| } else { | |
| GST_WARNING ("couldn't inject buffer, no incoming timestamps or offsets"); | |
| gst_buffer_unref (ibuf); | |
| } | |
| } | |
| gst_buffer_unref (buf); | |
| gst_object_unref (srcpad); | |
| return ret; | |
| } | |
| static void | |
| test_injector_init (TestInjector * injector) | |
| { | |
| GstPad *pad; | |
| pad = gst_pad_new_from_static_template (&sink_template, "sink"); | |
| gst_pad_set_chain_function (pad, test_injector_chain); | |
| GST_PAD_SET_PROXY_CAPS (pad); | |
| gst_element_add_pad (GST_ELEMENT (injector), pad); | |
| pad = gst_pad_new_from_static_template (&src_template, "src"); | |
| GST_PAD_SET_PROXY_CAPS (pad); | |
| gst_element_add_pad (GST_ELEMENT (injector), pad); | |
| } | |
| static GstPadProbeReturn | |
| probe_cb (GstPad * pad, GstPadProbeInfo * info, gpointer user_data) | |
| { | |
| GstBuffer *buf = GST_PAD_PROBE_INFO_BUFFER (info); | |
| gdouble *drop_probability = user_data; | |
| if (g_random_double () < *drop_probability) { | |
| GST_LOG ("dropping buffer [t=%" GST_TIME_FORMAT "-%" GST_TIME_FORMAT "], " | |
| "offset=%" G_GINT64_FORMAT ", offset_end=%" 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)); | |
| return GST_PAD_PROBE_DROP; /* drop buffer */ | |
| } | |
| return GST_PAD_PROBE_OK; /* 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, const gchar * format, | |
| gdouble drop_probability, gdouble inject_probability) | |
| { | |
| GstElement *pipe, *src, *conv, *filter, *injector, *audiorate, *sink; | |
| GstMessage *msg; | |
| GstCaps *caps; | |
| GstPad *srcpad; | |
| GList *l, *bufs = NULL; | |
| GstClockTime next_time = GST_CLOCK_TIME_NONE; | |
| guint64 next_offset = GST_BUFFER_OFFSET_NONE; | |
| GstAudioFormat fmt; | |
| const GstAudioFormatInfo *finfo; | |
| gint width; | |
| fmt = gst_audio_format_from_string (format); | |
| fail_unless (fmt != GST_AUDIO_FORMAT_UNKNOWN); | |
| finfo = gst_audio_format_get_info (fmt); | |
| width = GST_AUDIO_FORMAT_INFO_WIDTH (finfo); | |
| caps = gst_caps_new_simple ("audio/x-raw", "rate", G_TYPE_INT, | |
| rate, "format", G_TYPE_STRING, format, 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 (inject_probability >= 0.0 && inject_probability <= 1.0); | |
| 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", 10, 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); | |
| injector_inject_probability = inject_probability; | |
| injector = GST_ELEMENT (g_object_new (test_injector_get_type (), NULL)); | |
| srcpad = gst_element_get_static_pad (injector, "src"); | |
| fail_unless (srcpad != NULL); | |
| gst_pad_add_probe (srcpad, GST_PAD_PROBE_TYPE_BUFFER, probe_cb, | |
| &drop_probability, NULL); | |
| 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, injector, audiorate, | |
| sink, NULL); | |
| gst_element_link_many (src, conv, filter, injector, 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_get_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_get_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], "S8", 0.0, 0.0); | |
| do_perfect_stream_test (rates[i], GST_AUDIO_NE (S16), 0.0, 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], "S8", 0.10, 0.0); | |
| do_perfect_stream_test (rates[i], GST_AUDIO_NE (S16), 0.10, 0.0); | |
| } | |
| } | |
| 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], "S8", 0.50, 0.0); | |
| do_perfect_stream_test (rates[i], GST_AUDIO_NE (S16), 0.50, 0.0); | |
| } | |
| } | |
| 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], "S8", 0.90, 0.0); | |
| do_perfect_stream_test (rates[i], GST_AUDIO_NE (S16), 0.90, 0.0); | |
| } | |
| } | |
| GST_END_TEST; | |
| GST_START_TEST (test_perfect_stream_inject10) | |
| { | |
| guint i; | |
| for (i = 0; i < G_N_ELEMENTS (rates); ++i) { | |
| do_perfect_stream_test (rates[i], "S8", 0.0, 0.10); | |
| do_perfect_stream_test (rates[i], GST_AUDIO_NE (S16), 0.0, 0.10); | |
| } | |
| } | |
| GST_END_TEST; | |
| GST_START_TEST (test_perfect_stream_inject90) | |
| { | |
| guint i; | |
| for (i = 0; i < G_N_ELEMENTS (rates); ++i) { | |
| do_perfect_stream_test (rates[i], "S8", 0.0, 0.90); | |
| do_perfect_stream_test (rates[i], GST_AUDIO_NE (S16), 0.0, 0.90); | |
| } | |
| } | |
| GST_END_TEST; | |
| GST_START_TEST (test_perfect_stream_drop45_inject25) | |
| { | |
| guint i; | |
| for (i = 0; i < G_N_ELEMENTS (rates); ++i) { | |
| do_perfect_stream_test (rates[i], "S8", 0.45, 0.25); | |
| do_perfect_stream_test (rates[i], GST_AUDIO_NE (S16), 0.45, 0.25); | |
| } | |
| } | |
| GST_END_TEST; | |
| /* TODO: also do all tests with channels=1 and channels=2 */ | |
| static GstStaticPadTemplate srctemplate = GST_STATIC_PAD_TEMPLATE ("src", | |
| GST_PAD_SRC, | |
| GST_PAD_ALWAYS, | |
| GST_STATIC_CAPS ("audio/x-raw,format=" GST_AUDIO_NE (F32) | |
| ",channels=1,rate=44100") | |
| ); | |
| static GstStaticPadTemplate sinktemplate = GST_STATIC_PAD_TEMPLATE ("sink", | |
| GST_PAD_SINK, | |
| GST_PAD_ALWAYS, | |
| GST_STATIC_CAPS ("audio/x-raw,format=" GST_AUDIO_NE (F32) | |
| ",channels=1,rate=44100") | |
| ); | |
| GST_START_TEST (test_large_discont) | |
| { | |
| GstElement *audiorate; | |
| GstCaps *caps; | |
| GstPad *srcpad, *sinkpad; | |
| GstBuffer *buf; | |
| audiorate = gst_check_setup_element ("audiorate"); | |
| caps = gst_caps_new_simple ("audio/x-raw", | |
| "format", G_TYPE_STRING, GST_AUDIO_NE (F32), | |
| "layout", G_TYPE_STRING, "interleaved", | |
| "channels", G_TYPE_INT, 1, "rate", G_TYPE_INT, 44100, NULL); | |
| srcpad = gst_check_setup_src_pad (audiorate, &srctemplate); | |
| sinkpad = gst_check_setup_sink_pad (audiorate, &sinktemplate); | |
| gst_pad_set_active (srcpad, TRUE); | |
| gst_check_setup_events (srcpad, audiorate, caps, GST_FORMAT_TIME); | |
| gst_pad_set_active (sinkpad, TRUE); | |
| fail_unless (gst_element_set_state (audiorate, | |
| GST_STATE_PLAYING) == GST_STATE_CHANGE_SUCCESS, | |
| "failed to set audiorate playing"); | |
| buf = gst_buffer_new_and_alloc (4); | |
| GST_BUFFER_TIMESTAMP (buf) = 0; | |
| gst_pad_push (srcpad, buf); | |
| fail_unless_equals_int (g_list_length (buffers), 1); | |
| buf = gst_buffer_new_and_alloc (4); | |
| GST_BUFFER_TIMESTAMP (buf) = 2 * GST_SECOND; | |
| gst_pad_push (srcpad, buf); | |
| /* Now we should have 3 more buffers: the one we injected, plus _two_ filler | |
| * buffers, because the gap is > 1 second (but less than 2 seconds) */ | |
| fail_unless_equals_int (g_list_length (buffers), 4); | |
| gst_element_set_state (audiorate, GST_STATE_NULL); | |
| gst_caps_unref (caps); | |
| gst_check_drop_buffers (); | |
| gst_check_teardown_sink_pad (audiorate); | |
| gst_check_teardown_src_pad (audiorate); | |
| gst_object_unref (audiorate); | |
| } | |
| GST_END_TEST; | |
| #define FIRST_CAPS \ | |
| "audio/x-raw,format=S16LE,layout=interleaved,rate=48000,channels=1" | |
| #define SECOND_CAPS \ | |
| "audio/x-raw,format=S16LE,layout=interleaved,rate=8000,channels=1" | |
| #define BUFFERS_BEFORE_CHANGE 10 | |
| #define TOTAL_BUFFERS (BUFFERS_BEFORE_CHANGE * 2) | |
| static GList * | |
| generate_buffers (gint from_rate, gint to_rate) | |
| { | |
| GQueue q = G_QUEUE_INIT; | |
| GstBuffer *buf; | |
| guint i; | |
| GstClockTime pts = 0; | |
| for (i = 0; i < BUFFERS_BEFORE_CHANGE; i++) { | |
| buf = gst_buffer_new_allocate (NULL, 2 * from_rate / 100, NULL); | |
| gst_buffer_memset (buf, 0, 1, gst_buffer_get_size (buf)); | |
| GST_BUFFER_PTS (buf) = pts; | |
| GST_BUFFER_DURATION (buf) = GST_SECOND / 100; | |
| pts += GST_BUFFER_DURATION (buf); | |
| g_queue_push_tail (&q, buf); | |
| } | |
| for (; i < TOTAL_BUFFERS; i++) { | |
| buf = gst_buffer_new_allocate (NULL, 2 * to_rate / 100, NULL); | |
| gst_buffer_memset (buf, 0, 1, gst_buffer_get_size (buf)); | |
| GST_BUFFER_PTS (buf) = pts; | |
| GST_BUFFER_DURATION (buf) = GST_SECOND / 100; | |
| pts += GST_BUFFER_DURATION (buf); | |
| g_queue_push_tail (&q, buf); | |
| } | |
| return q.head; | |
| } | |
| GST_START_TEST (test_rate_change_down) | |
| { | |
| GList *l, *rbufs = NULL, *bufs = NULL; | |
| GstElement *pipeline; | |
| GstElement *sink; | |
| GstElement *src; | |
| GstElement *audiorate; | |
| GstCaps *caps1, *caps2; | |
| int i = 0; | |
| gint64 drop, in, out; | |
| GstBus *bus; | |
| caps1 = gst_caps_from_string (FIRST_CAPS); | |
| caps2 = gst_caps_from_string (SECOND_CAPS); | |
| bufs = generate_buffers (48000, 8000); | |
| pipeline = | |
| gst_parse_launch | |
| ("appsrc name=src is-live=true format=time !" | |
| " audiorate name=audiorate ! fakesink name=sink signal-handoffs=true", | |
| NULL); | |
| fail_if (pipeline == NULL); | |
| sink = gst_bin_get_by_name (GST_BIN (pipeline), "sink"); | |
| g_signal_connect (sink, "handoff", G_CALLBACK (got_buf), &rbufs); | |
| gst_object_unref (sink); | |
| src = gst_bin_get_by_name (GST_BIN (pipeline), "src"); | |
| gst_app_src_set_caps (GST_APP_SRC (src), caps1); | |
| gst_element_set_state (pipeline, GST_STATE_PLAYING); | |
| for (l = bufs; l != NULL; l = l->next) { | |
| if (i++ == BUFFERS_BEFORE_CHANGE) { | |
| gst_app_src_set_caps (GST_APP_SRC (src), caps2); | |
| } | |
| GST_LOG ("Position: %" GST_TIME_FORMAT " Duration: %" GST_TIME_FORMAT "\n", | |
| GST_TIME_ARGS (GST_BUFFER_PTS (l->data)), | |
| GST_TIME_ARGS (GST_BUFFER_DURATION (l->data))); | |
| fail_unless_equals_int (gst_app_src_push_buffer (GST_APP_SRC (src), | |
| GST_BUFFER (l->data)), GST_FLOW_OK); | |
| } | |
| g_list_free (bufs); | |
| gst_app_src_end_of_stream (GST_APP_SRC (src)); | |
| gst_object_unref (src); | |
| /* Give some time to the appsrc loop to push the buffers */ | |
| bus = gst_pipeline_get_bus (GST_PIPELINE (pipeline)); | |
| gst_message_unref (gst_bus_timed_pop_filtered (bus, GST_CLOCK_TIME_NONE, | |
| GST_MESSAGE_EOS)); | |
| gst_object_unref (bus); | |
| audiorate = gst_bin_get_by_name (GST_BIN (pipeline), "audiorate"); | |
| g_object_get (audiorate, "drop", &drop, "out", &out, "in", &in, NULL); | |
| gst_object_unref (audiorate); | |
| fail_unless_equals_int64 (drop, 0); | |
| g_list_foreach (rbufs, (GFunc) gst_mini_object_unref, NULL); | |
| g_list_free (rbufs); | |
| gst_element_set_state (pipeline, GST_STATE_NULL); | |
| gst_object_unref (pipeline); | |
| gst_caps_unref (caps1); | |
| gst_caps_unref (caps2); | |
| } | |
| 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); | |
| tcase_add_test (tc_chain, test_perfect_stream_inject10); | |
| tcase_add_test (tc_chain, test_perfect_stream_inject90); | |
| tcase_add_test (tc_chain, test_perfect_stream_drop45_inject25); | |
| tcase_add_test (tc_chain, test_large_discont); | |
| tcase_add_test (tc_chain, test_rate_change_down); | |
| return s; | |
| } | |
| GST_CHECK_MAIN (audiorate); |