/* 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., 51 Franklin St, Fifth Floor, * Boston, MA 02110-1301, USA. */ #ifdef HAVE_CONFIG_H # include #endif #include #include #include /* 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);