/* GStreamer * * unit test for audioresample, based on the audioresample unit test * * Copyright (C) <2005> Thomas Vander Stichele <thomas at apestaart dot org> * Copyright (C) <2006> Tim-Philipp Müller <tim at 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/fft/gstfft.h> #include <gst/fft/gstffts16.h> #include <gst/fft/gstffts32.h> #include <gst/fft/gstfftf32.h> #include <gst/fft/gstfftf64.h> /* For ease of programming we use globals to keep refs for our floating * src and sink pads we create; otherwise we always have to do get_pad, * get_peer, and then remove references in every test function */ static GstPad *mysrcpad, *mysinkpad; #if G_BYTE_ORDER == G_LITTLE_ENDIAN #define FORMATS "{ F32LE, F64LE, S16LE, S32LE }" #else #define FORMATS "{ F32BE, F64BE, S16BE, S32BE }" #endif #define RESAMPLE_CAPS \ "audio/x-raw, " \ "format = (string) "FORMATS", " \ "channels = (int) [ 1, MAX ], " \ "rate = (int) [ 1, MAX ], " \ "layout = (string) interleaved" static GstElement * setup_audioresample (int channels, guint64 mask, int inrate, int outrate, const gchar * format) { GstPadTemplate *sinktemplate; static GstStaticPadTemplate srctemplate = GST_STATIC_PAD_TEMPLATE ("src", GST_PAD_SRC, GST_PAD_ALWAYS, GST_STATIC_CAPS (RESAMPLE_CAPS) ); GstElement *audioresample; GstCaps *caps; GstStructure *structure; GST_DEBUG ("setup_audioresample"); audioresample = gst_check_setup_element ("audioresample"); caps = gst_caps_from_string (RESAMPLE_CAPS); structure = gst_caps_get_structure (caps, 0); gst_structure_set (structure, "channels", G_TYPE_INT, channels, "rate", G_TYPE_INT, inrate, "format", G_TYPE_STRING, format, "channel-mask", GST_TYPE_BITMASK, mask, NULL); fail_unless (gst_caps_is_fixed (caps)); fail_unless (gst_element_set_state (audioresample, GST_STATE_PAUSED) == GST_STATE_CHANGE_SUCCESS, "could not set to paused"); mysrcpad = gst_check_setup_src_pad (audioresample, &srctemplate); gst_pad_set_active (mysrcpad, TRUE); gst_check_setup_events (mysrcpad, audioresample, caps, GST_FORMAT_TIME); gst_caps_unref (caps); caps = gst_caps_from_string (RESAMPLE_CAPS); structure = gst_caps_get_structure (caps, 0); gst_structure_set (structure, "channels", G_TYPE_INT, channels, "rate", G_TYPE_INT, outrate, "format", G_TYPE_STRING, format, NULL); fail_unless (gst_caps_is_fixed (caps)); sinktemplate = gst_pad_template_new ("sink", GST_PAD_SINK, GST_PAD_ALWAYS, caps); mysinkpad = gst_check_setup_sink_pad_from_template (audioresample, sinktemplate); gst_pad_set_active (mysinkpad, TRUE); /* this installs a getcaps func that will always return the caps we set * later */ gst_pad_use_fixed_caps (mysinkpad); gst_caps_unref (caps); gst_object_unref (sinktemplate); return audioresample; } static void cleanup_audioresample (GstElement * audioresample) { GST_DEBUG ("cleanup_audioresample"); fail_unless (gst_element_set_state (audioresample, GST_STATE_NULL) == GST_STATE_CHANGE_SUCCESS, "could not set to NULL"); gst_pad_set_active (mysrcpad, FALSE); gst_pad_set_active (mysinkpad, FALSE); gst_check_teardown_src_pad (audioresample); gst_check_teardown_sink_pad (audioresample); gst_check_teardown_element (audioresample); gst_check_drop_buffers (); } static void fail_unless_perfect_stream (void) { guint64 timestamp = 0L, duration = 0L; guint64 offset = 0L, offset_end = 0L; GList *l; GstBuffer *buffer; for (l = buffers; l; l = l->next) { buffer = GST_BUFFER (l->data); ASSERT_BUFFER_REFCOUNT (buffer, "buffer", 1); GST_DEBUG ("buffer timestamp %" G_GUINT64_FORMAT ", duration %" G_GUINT64_FORMAT " offset %" G_GUINT64_FORMAT " offset_end %" G_GUINT64_FORMAT, GST_BUFFER_TIMESTAMP (buffer), GST_BUFFER_DURATION (buffer), GST_BUFFER_OFFSET (buffer), GST_BUFFER_OFFSET_END (buffer)); fail_unless_equals_uint64 (timestamp, GST_BUFFER_TIMESTAMP (buffer)); fail_unless_equals_uint64 (offset, GST_BUFFER_OFFSET (buffer)); duration = GST_BUFFER_DURATION (buffer); offset_end = GST_BUFFER_OFFSET_END (buffer); timestamp += duration; offset = offset_end; gst_buffer_unref (buffer); } g_list_free (buffers); buffers = NULL; } /* this tests that the output is a perfect stream if the input is */ static void test_perfect_stream_instance (int inrate, int outrate, int samples, int numbuffers) { GstElement *audioresample; GstBuffer *inbuffer, *outbuffer; GstCaps *caps; guint64 offset = 0; int i, j; GstMapInfo map; gint16 *p; audioresample = setup_audioresample (2, 0x3, inrate, outrate, GST_AUDIO_NE (S16)); caps = gst_pad_get_current_caps (mysrcpad); fail_unless (gst_caps_is_fixed (caps)); fail_unless (gst_element_set_state (audioresample, GST_STATE_PLAYING) == GST_STATE_CHANGE_SUCCESS, "could not set to playing"); for (j = 1; j <= numbuffers; ++j) { inbuffer = gst_buffer_new_and_alloc (samples * 4); GST_BUFFER_DURATION (inbuffer) = GST_FRAMES_TO_CLOCK_TIME (samples, inrate); GST_BUFFER_TIMESTAMP (inbuffer) = GST_BUFFER_DURATION (inbuffer) * (j - 1); GST_BUFFER_OFFSET (inbuffer) = offset; offset += samples; GST_BUFFER_OFFSET_END (inbuffer) = offset; gst_buffer_map (inbuffer, &map, GST_MAP_WRITE); p = (gint16 *) map.data; /* create a 16 bit signed ramp */ for (i = 0; i < samples; ++i) { *p = -32767 + i * (65535 / samples); ++p; *p = -32767 + i * (65535 / samples); ++p; } gst_buffer_unmap (inbuffer, &map); /* pushing gives away my reference ... */ fail_unless (gst_pad_push (mysrcpad, inbuffer) == GST_FLOW_OK); /* ... but it ends up being collected on the global buffer list */ fail_unless_equals_int (g_list_length (buffers), j); } /* FIXME: we should make audioresample handle eos by flushing out the last * samples, which will give us one more, small, buffer */ fail_if ((outbuffer = (GstBuffer *) buffers->data) == NULL); ASSERT_BUFFER_REFCOUNT (outbuffer, "outbuffer", 1); fail_unless_perfect_stream (); /* cleanup */ gst_caps_unref (caps); cleanup_audioresample (audioresample); } /* make sure that outgoing buffers are contiguous in timestamp/duration and * offset/offsetend */ GST_START_TEST (test_perfect_stream) { /* integral scalings */ test_perfect_stream_instance (48000, 24000, 500, 20); test_perfect_stream_instance (48000, 12000, 500, 20); test_perfect_stream_instance (12000, 24000, 500, 20); test_perfect_stream_instance (12000, 48000, 500, 20); /* non-integral scalings */ test_perfect_stream_instance (44100, 8000, 500, 20); test_perfect_stream_instance (8000, 44100, 500, 20); /* wacky scalings */ test_perfect_stream_instance (12345, 54321, 500, 20); test_perfect_stream_instance (101, 99, 500, 20); } GST_END_TEST; /* this tests that the output is a correct discontinuous stream * if the input is; ie input drops in time come out the same way */ static void test_discont_stream_instance (int inrate, int outrate, int samples, int numbuffers) { GstElement *audioresample; GstBuffer *inbuffer, *outbuffer; GstCaps *caps; GstClockTime ints; int i, j; GstMapInfo map; gint16 *p; GST_DEBUG ("inrate:%d outrate:%d samples:%d numbuffers:%d", inrate, outrate, samples, numbuffers); audioresample = setup_audioresample (2, 3, inrate, outrate, GST_AUDIO_NE (S16)); caps = gst_pad_get_current_caps (mysrcpad); fail_unless (gst_caps_is_fixed (caps)); fail_unless (gst_element_set_state (audioresample, GST_STATE_PLAYING) == GST_STATE_CHANGE_SUCCESS, "could not set to playing"); for (j = 1; j <= numbuffers; ++j) { inbuffer = gst_buffer_new_and_alloc (samples * 4); GST_BUFFER_DURATION (inbuffer) = samples * GST_SECOND / inrate; /* "drop" half the buffers */ ints = GST_BUFFER_DURATION (inbuffer) * 2 * (j - 1); GST_BUFFER_TIMESTAMP (inbuffer) = ints; GST_BUFFER_OFFSET (inbuffer) = (j - 1) * 2 * samples; GST_BUFFER_OFFSET_END (inbuffer) = j * 2 * samples + samples; gst_buffer_map (inbuffer, &map, GST_MAP_WRITE); p = (gint16 *) map.data; /* create a 16 bit signed ramp */ for (i = 0; i < samples; ++i) { *p = -32767 + i * (65535 / samples); ++p; *p = -32767 + i * (65535 / samples); ++p; } gst_buffer_unmap (inbuffer, &map); GST_DEBUG ("Sending Buffer time:%" G_GUINT64_FORMAT " duration:%" G_GINT64_FORMAT " discont:%d offset:%" G_GUINT64_FORMAT " offset_end:%" G_GUINT64_FORMAT, GST_BUFFER_TIMESTAMP (inbuffer), GST_BUFFER_DURATION (inbuffer), GST_BUFFER_IS_DISCONT (inbuffer), GST_BUFFER_OFFSET (inbuffer), GST_BUFFER_OFFSET_END (inbuffer)); /* pushing gives away my reference ... */ fail_unless (gst_pad_push (mysrcpad, inbuffer) == GST_FLOW_OK); /* check if the timestamp of the pushed buffer matches the incoming one */ outbuffer = g_list_nth_data (buffers, g_list_length (buffers) - 1); fail_if (outbuffer == NULL); fail_unless_equals_uint64 (ints, GST_BUFFER_TIMESTAMP (outbuffer)); GST_DEBUG ("Got Buffer time:%" G_GUINT64_FORMAT " duration:%" G_GINT64_FORMAT " discont:%d offset:%" G_GUINT64_FORMAT " offset_end:%" G_GUINT64_FORMAT, GST_BUFFER_TIMESTAMP (outbuffer), GST_BUFFER_DURATION (outbuffer), GST_BUFFER_IS_DISCONT (outbuffer), GST_BUFFER_OFFSET (outbuffer), GST_BUFFER_OFFSET_END (outbuffer)); if (j > 1) { fail_unless (GST_BUFFER_IS_DISCONT (outbuffer), "expected discont for buffer #%d", j); } } /* cleanup */ gst_caps_unref (caps); cleanup_audioresample (audioresample); } GST_START_TEST (test_discont_stream) { /* integral scalings */ test_discont_stream_instance (48000, 24000, 5000, 20); test_discont_stream_instance (48000, 12000, 5000, 20); test_discont_stream_instance (12000, 24000, 5000, 20); test_discont_stream_instance (12000, 48000, 5000, 20); /* non-integral scalings */ test_discont_stream_instance (44100, 8000, 5000, 20); test_discont_stream_instance (8000, 44100, 5000, 20); /* wacky scalings */ test_discont_stream_instance (12345, 54321, 5000, 20); test_discont_stream_instance (101, 99, 5000, 20); } GST_END_TEST; GST_START_TEST (test_reuse) { GstElement *audioresample; GstEvent *newseg; GstBuffer *inbuffer; GstCaps *caps; GstSegment segment; audioresample = setup_audioresample (1, 0, 9343, 48000, GST_AUDIO_NE (S16)); caps = gst_pad_get_current_caps (mysrcpad); fail_unless (gst_caps_is_fixed (caps)); fail_unless (gst_element_set_state (audioresample, GST_STATE_PLAYING) == GST_STATE_CHANGE_SUCCESS, "could not set to playing"); gst_segment_init (&segment, GST_FORMAT_TIME); newseg = gst_event_new_segment (&segment); fail_unless (gst_pad_push_event (mysrcpad, newseg) != FALSE); inbuffer = gst_buffer_new_and_alloc (9343 * 4); gst_buffer_memset (inbuffer, 0, 0, 9343 * 4); GST_BUFFER_DURATION (inbuffer) = GST_SECOND; GST_BUFFER_TIMESTAMP (inbuffer) = 0; GST_BUFFER_OFFSET (inbuffer) = 0; /* pushing gives away my reference ... */ fail_unless (gst_pad_push (mysrcpad, inbuffer) == GST_FLOW_OK); /* ... but it ends up being collected on the global buffer list */ fail_unless_equals_int (g_list_length (buffers), 1); /* now reset and try again ... */ fail_unless (gst_element_set_state (audioresample, GST_STATE_NULL) == GST_STATE_CHANGE_SUCCESS, "could not set to NULL"); fail_unless (gst_element_set_state (audioresample, GST_STATE_PLAYING) == GST_STATE_CHANGE_SUCCESS, "could not set to playing"); newseg = gst_event_new_segment (&segment); fail_unless (gst_pad_push_event (mysrcpad, newseg) != FALSE); inbuffer = gst_buffer_new_and_alloc (9343 * 4); gst_buffer_memset (inbuffer, 0, 0, 9343 * 4); GST_BUFFER_DURATION (inbuffer) = GST_SECOND; GST_BUFFER_TIMESTAMP (inbuffer) = 0; GST_BUFFER_OFFSET (inbuffer) = 0; fail_unless (gst_pad_push (mysrcpad, inbuffer) == GST_FLOW_OK); /* ... it also ends up being collected on the global buffer list. If we * now have more than 2 buffers, then audioresample probably didn't clean * up its internal buffer properly and tried to push the remaining samples * when it got the second NEWSEGMENT event */ fail_unless_equals_int (g_list_length (buffers), 2); cleanup_audioresample (audioresample); gst_caps_unref (caps); } GST_END_TEST; GST_START_TEST (test_shutdown) { GstElement *pipeline, *src, *cf1, *ar, *cf2, *sink; GstCaps *caps; guint i; /* create pipeline, force audioresample to actually resample */ pipeline = gst_pipeline_new (NULL); src = gst_check_setup_element ("audiotestsrc"); cf1 = gst_check_setup_element ("capsfilter"); ar = gst_check_setup_element ("audioresample"); cf2 = gst_check_setup_element ("capsfilter"); g_object_set (cf2, "name", "capsfilter2", NULL); sink = gst_check_setup_element ("fakesink"); caps = gst_caps_new_simple ("audio/x-raw", "rate", G_TYPE_INT, 11025, NULL); g_object_set (cf1, "caps", caps, NULL); gst_caps_unref (caps); caps = gst_caps_new_simple ("audio/x-raw", "rate", G_TYPE_INT, 48000, NULL); g_object_set (cf2, "caps", caps, NULL); gst_caps_unref (caps); /* don't want to sync against the clock, the more throughput the better */ g_object_set (src, "is-live", FALSE, NULL); g_object_set (sink, "sync", FALSE, NULL); gst_bin_add_many (GST_BIN (pipeline), src, cf1, ar, cf2, sink, NULL); fail_if (!gst_element_link_many (src, cf1, ar, cf2, sink, NULL)); /* now, wait until pipeline is running and then shut it down again; repeat */ for (i = 0; i < 20; ++i) { gst_element_set_state (pipeline, GST_STATE_PAUSED); gst_element_get_state (pipeline, NULL, NULL, -1); gst_element_set_state (pipeline, GST_STATE_PLAYING); g_usleep (100); gst_element_set_state (pipeline, GST_STATE_NULL); } gst_object_unref (pipeline); } GST_END_TEST; static void live_switch_push (gint pts, gint rate, GstCaps * caps) { GstBuffer *inbuffer; GstCaps *desired; desired = gst_caps_copy (caps); gst_caps_set_simple (desired, "rate", G_TYPE_INT, rate, NULL); gst_pad_set_caps (mysrcpad, desired); inbuffer = gst_buffer_new_and_alloc (rate * 4 * 2); gst_buffer_memset (inbuffer, 0, 0, rate * 4 * 2); GST_BUFFER_DURATION (inbuffer) = GST_SECOND; GST_BUFFER_TIMESTAMP (inbuffer) = pts * GST_SECOND; GST_BUFFER_OFFSET (inbuffer) = 0; GST_BUFFER_OFFSET_END (inbuffer) = rate - 1; /* pushing gives away my reference ... */ fail_unless_equals_int (gst_pad_push (mysrcpad, inbuffer), GST_FLOW_OK); /* ... but it ends up being collected on the global buffer list */ gst_caps_unref (desired); } #if !GLIB_CHECK_VERSION(2,58,0) #define G_APPROX_VALUE(a, b, epsilon) \ (((a) > (b) ? (a) - (b) : (b) - (a)) < (epsilon)) #endif GST_START_TEST (test_live_switch) { GstElement *audioresample; GstEvent *newseg; GstCaps *caps; GstSegment segment; GList *l; guint i; audioresample = setup_audioresample (4, 0xf, 48000, 48000, GST_AUDIO_NE (S16)); caps = gst_pad_get_current_caps (mysrcpad); fail_unless (gst_caps_is_fixed (caps)); fail_unless (gst_element_set_state (audioresample, GST_STATE_PLAYING) == GST_STATE_CHANGE_SUCCESS, "could not set to playing"); gst_segment_init (&segment, GST_FORMAT_TIME); newseg = gst_event_new_segment (&segment); fail_unless (gst_pad_push_event (mysrcpad, newseg) != FALSE); /* downstream can accept the requested rate */ live_switch_push (0, 48000, caps); /* buffer is directly passed through */ fail_unless_equals_int (g_list_length (buffers), 1); /* Downstream can never accept this rate */ live_switch_push (1, 40000, caps); /* one additional buffer is provided with the new sample rate */ fail_unless_equals_int (g_list_length (buffers), 2); /* Downstream can never accept this rate */ live_switch_push (2, 50000, caps); /* two additional buffers are provided. One is the drained remainder of * the previous sample rate, the second is the buffer with the new sample * rate */ fail_unless_equals_int (g_list_length (buffers), 4); /* Send EOS to drain the remaining samples */ fail_unless (gst_pad_push_event (mysrcpad, gst_event_new_eos ())); fail_unless_equals_int (g_list_length (buffers), 5); /* Now test that each buffer has the expected samples. We simply check this * by checking whether the timestamps, durations and sizes are matching */ for (l = buffers, i = 0; l; l = l->next, i++) { GstBuffer *buffer = GST_BUFFER (l->data); switch (i) { case 0: fail_unless_equals_uint64 (GST_BUFFER_PTS (buffer), 0 * GST_SECOND); fail_unless_equals_uint64 (GST_BUFFER_DURATION (buffer), 1 * GST_SECOND); fail_unless_equals_int (gst_buffer_get_size (buffer), 48000 * 4 * 2); break; case 1: fail_unless_equals_uint64 (GST_BUFFER_PTS (buffer), 1 * GST_SECOND); fail_unless_equals_int (gst_buffer_get_size (buffer), 47961 * 4 * 2); break; case 2: fail_unless (G_APPROX_VALUE (GST_BUFFER_PTS (buffer) + GST_BUFFER_DURATION (buffer), 2 * GST_SECOND, GST_SECOND / 48000 + 1)); fail_unless_equals_int (gst_buffer_get_size (buffer), 38 * 4 * 2); break; case 3: fail_unless_equals_uint64 (GST_BUFFER_PTS (buffer), 2 * GST_SECOND); fail_unless_equals_int (gst_buffer_get_size (buffer), 47969 * 4 * 2); break; case 4: fail_unless (G_APPROX_VALUE (GST_BUFFER_PTS (buffer) + GST_BUFFER_DURATION (buffer), 3 * GST_SECOND, GST_SECOND / 48000 + 1)); fail_unless_equals_int (gst_buffer_get_size (buffer), 30 * 4 * 2); break; default: g_assert_not_reached (); break; } gst_buffer_unref (buffer); } g_list_free (buffers); buffers = NULL; cleanup_audioresample (audioresample); gst_caps_unref (caps); } GST_END_TEST; static gint current_rate = 0; static gboolean live_switch_sink_query (GstPad * pad, GstObject * parent, GstQuery * query) { switch (GST_QUERY_TYPE (query)) { case GST_QUERY_ACCEPT_CAPS:{ GstCaps *acceptable_caps; GstCaps *caps; acceptable_caps = gst_pad_get_current_caps (mysrcpad); acceptable_caps = gst_caps_make_writable (acceptable_caps); gst_caps_set_simple (acceptable_caps, "rate", G_TYPE_INT, current_rate, NULL); gst_query_parse_accept_caps (query, &caps); gst_query_set_accept_caps_result (query, gst_caps_can_intersect (caps, acceptable_caps)); gst_caps_unref (acceptable_caps); return TRUE; } case GST_QUERY_CAPS:{ GstCaps *acceptable_caps; GstCaps *filter; GstCaps *caps; acceptable_caps = gst_pad_get_current_caps (mysrcpad); acceptable_caps = gst_caps_make_writable (acceptable_caps); gst_caps_set_simple (acceptable_caps, "rate", G_TYPE_INT, current_rate, NULL); gst_query_parse_caps (query, &filter); if (filter) caps = gst_caps_intersect_full (filter, acceptable_caps, GST_CAPS_INTERSECT_FIRST); else caps = gst_caps_ref (acceptable_caps); gst_query_set_caps_result (query, caps); gst_caps_unref (caps); gst_caps_unref (acceptable_caps); return TRUE; } default: return gst_pad_query_default (pad, parent, query); } } static void live_switch_push_downstream (gint pts, gint rate) { GstBuffer *inbuffer; current_rate = rate; gst_pad_push_event (mysinkpad, gst_event_new_reconfigure ()); inbuffer = gst_buffer_new_and_alloc (48000 * 4 * 2); gst_buffer_memset (inbuffer, 0, 0, 48000 * 4 * 2); GST_BUFFER_DURATION (inbuffer) = GST_SECOND; GST_BUFFER_TIMESTAMP (inbuffer) = pts * GST_SECOND; GST_BUFFER_OFFSET (inbuffer) = 0; GST_BUFFER_OFFSET_END (inbuffer) = 47999; /* pushing gives away my reference ... */ fail_unless_equals_int (gst_pad_push (mysrcpad, inbuffer), GST_FLOW_OK); /* ... but it ends up being collected on the global buffer list */ } GST_START_TEST (test_live_switch_downstream) { GstElement *audioresample; GstEvent *newseg; GstCaps *caps; GstSegment segment; GList *l; guint i; audioresample = setup_audioresample (4, 0xf, 48000, 48000, GST_AUDIO_NE (S16)); gst_pad_set_query_function (mysinkpad, live_switch_sink_query); caps = gst_pad_get_current_caps (mysrcpad); fail_unless (gst_caps_is_fixed (caps)); fail_unless (gst_element_set_state (audioresample, GST_STATE_PLAYING) == GST_STATE_CHANGE_SUCCESS, "could not set to playing"); gst_segment_init (&segment, GST_FORMAT_TIME); newseg = gst_event_new_segment (&segment); fail_unless (gst_pad_push_event (mysrcpad, newseg) != FALSE); /* buffer is directly passed through */ live_switch_push_downstream (0, 48000); fail_unless_equals_int (g_list_length (buffers), 1); /* Reconfigure downstream to 40000 Hz */ live_switch_push_downstream (1, 40000); /* one additional buffer is provided with the new sample rate */ fail_unless_equals_int (g_list_length (buffers), 2); /* Reconfigure downstream to 50000 Hz */ live_switch_push_downstream (2, 50000); /* two additional buffers are provided. One is the drained remainder of * the previous sample rate, the second is the buffer with the new sample * rate */ fail_unless_equals_int (g_list_length (buffers), 4); /* Send EOS to drain the remaining samples */ fail_unless (gst_pad_push_event (mysrcpad, gst_event_new_eos ())); fail_unless_equals_int (g_list_length (buffers), 5); /* Now test that each buffer has the expected samples. We simply check this * by checking whether the timestamps, durations and sizes are matching */ for (l = buffers, i = 0; l; l = l->next, i++) { GstBuffer *buffer = GST_BUFFER (l->data); switch (i) { case 0: fail_unless_equals_uint64 (GST_BUFFER_PTS (buffer), 0 * GST_SECOND); fail_unless_equals_uint64 (GST_BUFFER_DURATION (buffer), 1 * GST_SECOND); fail_unless_equals_int (gst_buffer_get_size (buffer), 48000 * 4 * 2); break; case 1: fail_unless_equals_uint64 (GST_BUFFER_PTS (buffer), 1 * GST_SECOND); fail_unless_equals_int (gst_buffer_get_size (buffer), 39966 * 4 * 2); break; case 2: fail_unless (G_APPROX_VALUE (GST_BUFFER_PTS (buffer) + GST_BUFFER_DURATION (buffer), 2 * GST_SECOND, GST_SECOND / 40000 + 1)); fail_unless_equals_int (gst_buffer_get_size (buffer), 34 * 4 * 2); break; case 3: fail_unless_equals_uint64 (GST_BUFFER_PTS (buffer), 2 * GST_SECOND); fail_unless_equals_int (gst_buffer_get_size (buffer), 49966 * 4 * 2); break; case 4: fail_unless (G_APPROX_VALUE (GST_BUFFER_PTS (buffer) + GST_BUFFER_DURATION (buffer), 3 * GST_SECOND, GST_SECOND / 50000 + 1)); fail_unless_equals_int (gst_buffer_get_size (buffer), 33 * 4 * 2); break; default: g_assert_not_reached (); break; } gst_buffer_unref (buffer); } g_list_free (buffers); buffers = NULL; cleanup_audioresample (audioresample); gst_caps_unref (caps); } GST_END_TEST; #ifndef GST_DISABLE_PARSE static GMainLoop *loop; static gint messages = 0; static void element_message_cb (GstBus * bus, GstMessage * message, gpointer user_data) { gchar *s; s = gst_structure_to_string (gst_message_get_structure (message)); GST_DEBUG ("Received message: %s", s); g_free (s); messages++; } static void eos_message_cb (GstBus * bus, GstMessage * message, gpointer user_data) { GST_DEBUG ("Received eos"); g_main_loop_quit (loop); } static void test_pipeline (const gchar * format, gint inrate, gint outrate, gint quality) { GstElement *pipeline; GstBus *bus; GError *error = NULL; gchar *pipe_str; pipe_str = g_strdup_printf ("audiotestsrc num-buffers=10 ! audioconvert ! audio/x-raw,format=%s,rate=%d,channels=2 ! audioresample quality=%d ! audio/x-raw,format=%s,rate=%d ! identity check-imperfect-timestamp=TRUE ! fakesink", format, inrate, quality, format, outrate); pipeline = gst_parse_launch (pipe_str, &error); fail_unless (pipeline != NULL, "Error parsing pipeline: %s", error ? error->message : "(invalid error)"); g_free (pipe_str); bus = gst_element_get_bus (pipeline); fail_if (bus == NULL); gst_bus_add_signal_watch (bus); g_signal_connect (bus, "message::element", (GCallback) element_message_cb, NULL); g_signal_connect (bus, "message::eos", (GCallback) eos_message_cb, NULL); gst_element_set_state (pipeline, GST_STATE_PLAYING); /* run until we receive EOS */ loop = g_main_loop_new (NULL, FALSE); g_main_loop_run (loop); g_main_loop_unref (loop); loop = NULL; gst_element_set_state (pipeline, GST_STATE_NULL); gst_bus_remove_signal_watch (bus); gst_object_unref (bus); fail_if (messages > 0, "Received imperfect timestamp messages"); gst_object_unref (pipeline); } GST_START_TEST (test_pipelines) { gint quality; /* Test qualities 0, 5 and 10 */ for (quality = 0; quality < 11; quality += 5) { GST_DEBUG ("Checking with quality %d", quality); test_pipeline ("S8", 44100, 48000, quality); test_pipeline ("S8", 48000, 44100, quality); test_pipeline (GST_AUDIO_NE (S16), 44100, 48000, quality); test_pipeline (GST_AUDIO_NE (S16), 48000, 44100, quality); test_pipeline (GST_AUDIO_NE (S24), 44100, 48000, quality); test_pipeline (GST_AUDIO_NE (S24), 48000, 44100, quality); test_pipeline (GST_AUDIO_NE (S32), 44100, 48000, quality); test_pipeline (GST_AUDIO_NE (S32), 48000, 44100, quality); test_pipeline (GST_AUDIO_NE (F32), 44100, 48000, quality); test_pipeline (GST_AUDIO_NE (F32), 48000, 44100, quality); test_pipeline (GST_AUDIO_NE (F64), 44100, 48000, quality); test_pipeline (GST_AUDIO_NE (F64), 48000, 44100, quality); } } GST_END_TEST; GST_START_TEST (test_preference_passthrough) { GstStateChangeReturn ret; GstElement *pipeline, *src; GstStructure *s; GstMessage *msg; GstCaps *caps; GstPad *pad; GstBus *bus; GError *error = NULL; gint rate = 0; pipeline = gst_parse_launch ("audiotestsrc num-buffers=1 name=src ! " "audioresample ! audio/x-raw,format=" GST_AUDIO_NE (S16) ",channels=1," "rate=8000 ! fakesink can-activate-pull=false", &error); fail_unless (pipeline != NULL, "Error parsing pipeline: %s", error ? error->message : "(invalid error)"); ret = gst_element_set_state (pipeline, GST_STATE_PLAYING); fail_unless_equals_int (ret, GST_STATE_CHANGE_ASYNC); /* run until we receive EOS */ bus = gst_element_get_bus (pipeline); fail_if (bus == NULL); msg = gst_bus_timed_pop_filtered (bus, -1, GST_MESSAGE_EOS); gst_message_unref (msg); gst_object_unref (bus); src = gst_bin_get_by_name (GST_BIN (pipeline), "src"); fail_unless (src != NULL); pad = gst_element_get_static_pad (src, "src"); fail_unless (pad != NULL); caps = gst_pad_get_current_caps (pad); GST_LOG ("current audiotestsrc caps: %" GST_PTR_FORMAT, caps); fail_unless (caps != NULL); s = gst_caps_get_structure (caps, 0); fail_unless (gst_structure_get_int (s, "rate", &rate)); /* there's no need to resample, audiotestsrc supports any rate, so make * sure audioresample provided upstream with the right caps to negotiate * this correctly */ fail_unless_equals_int (rate, 8000); gst_caps_unref (caps); gst_object_unref (pad); gst_object_unref (src); gst_element_set_state (pipeline, GST_STATE_NULL); gst_object_unref (pipeline); } GST_END_TEST; #endif static void _message_cb (GstBus * bus, GstMessage * message, gpointer user_data) { GMainLoop *loop = user_data; switch (GST_MESSAGE_TYPE (message)) { case GST_MESSAGE_ERROR: case GST_MESSAGE_WARNING: g_assert_not_reached (); break; case GST_MESSAGE_EOS: g_main_loop_quit (loop); break; default: break; } } typedef struct { guint64 latency; GstClockTime in_ts; GstClockTime next_out_ts; guint64 next_out_off; guint64 in_buffer_count, out_buffer_count; } TimestampDriftCtx; static void fakesink_handoff_cb (GstElement * object, GstBuffer * buffer, GstPad * pad, gpointer user_data) { TimestampDriftCtx *ctx = user_data; ctx->out_buffer_count++; if (ctx->latency == GST_CLOCK_TIME_NONE) { ctx->latency = 1000 - gst_buffer_get_size (buffer) / 8; } /* Check if we have a perfectly timestamped stream */ if (ctx->next_out_ts != GST_CLOCK_TIME_NONE) fail_unless (ctx->next_out_ts == GST_BUFFER_TIMESTAMP (buffer), "expected timestamp %" GST_TIME_FORMAT " got timestamp %" GST_TIME_FORMAT, GST_TIME_ARGS (ctx->next_out_ts), GST_TIME_ARGS (GST_BUFFER_TIMESTAMP (buffer))); /* Check if we have a perfectly offsetted stream */ fail_unless (GST_BUFFER_OFFSET_END (buffer) == GST_BUFFER_OFFSET (buffer) + gst_buffer_get_size (buffer) / 8, "expected offset end %" G_GUINT64_FORMAT " got offset end %" G_GUINT64_FORMAT, GST_BUFFER_OFFSET (buffer) + gst_buffer_get_size (buffer) / 8, GST_BUFFER_OFFSET_END (buffer)); if (ctx->next_out_off != GST_BUFFER_OFFSET_NONE) { fail_unless (GST_BUFFER_OFFSET (buffer) == ctx->next_out_off, "expected offset %" G_GUINT64_FORMAT " got offset %" G_GUINT64_FORMAT, ctx->next_out_off, GST_BUFFER_OFFSET (buffer)); } if (ctx->in_buffer_count != ctx->out_buffer_count) { GST_INFO ("timestamp %" GST_TIME_FORMAT, GST_TIME_ARGS (GST_BUFFER_TIMESTAMP (buffer))); } if (ctx->in_ts != GST_CLOCK_TIME_NONE && ctx->in_buffer_count > 1 && ctx->in_buffer_count == ctx->out_buffer_count) { fail_unless (GST_BUFFER_TIMESTAMP (buffer) == ctx->in_ts - gst_util_uint64_scale_round (ctx->latency, GST_SECOND, 4096), "expected output timestamp %" GST_TIME_FORMAT " (%" G_GUINT64_FORMAT ") got output timestamp %" GST_TIME_FORMAT " (%" G_GUINT64_FORMAT ")", GST_TIME_ARGS (ctx->in_ts - gst_util_uint64_scale_round (ctx->latency, GST_SECOND, 4096)), ctx->in_ts - gst_util_uint64_scale_round (ctx->latency, GST_SECOND, 4096), GST_TIME_ARGS (GST_BUFFER_TIMESTAMP (buffer)), GST_BUFFER_TIMESTAMP (buffer)); } ctx->next_out_ts = GST_BUFFER_TIMESTAMP (buffer) + GST_BUFFER_DURATION (buffer); ctx->next_out_off = GST_BUFFER_OFFSET_END (buffer); } static void identity_handoff_cb (GstElement * object, GstBuffer * buffer, gpointer user_data) { TimestampDriftCtx *ctx = user_data; ctx->in_ts = GST_BUFFER_TIMESTAMP (buffer); ctx->in_buffer_count++; } GST_START_TEST (test_timestamp_drift) { TimestampDriftCtx ctx = { GST_CLOCK_TIME_NONE, GST_CLOCK_TIME_NONE, GST_CLOCK_TIME_NONE, GST_BUFFER_OFFSET_NONE, 0, 0 }; GstElement *pipeline; GstElement *audiotestsrc, *capsfilter1, *identity, *audioresample, *capsfilter2, *fakesink; GstBus *bus; GMainLoop *loop; GstCaps *caps; pipeline = gst_pipeline_new ("pipeline"); fail_unless (pipeline != NULL); audiotestsrc = gst_element_factory_make ("audiotestsrc", "src"); fail_unless (audiotestsrc != NULL); g_object_set (G_OBJECT (audiotestsrc), "num-buffers", 10000, "samplesperbuffer", 4000, NULL); capsfilter1 = gst_element_factory_make ("capsfilter", "capsfilter1"); fail_unless (capsfilter1 != NULL); caps = gst_caps_from_string ("audio/x-raw, format=" GST_AUDIO_NE (F64) ", channels=1, rate=16384"); g_object_set (G_OBJECT (capsfilter1), "caps", caps, NULL); gst_caps_unref (caps); identity = gst_element_factory_make ("identity", "identity"); fail_unless (identity != NULL); g_object_set (G_OBJECT (identity), "sync", FALSE, "signal-handoffs", TRUE, NULL); g_signal_connect (identity, "handoff", (GCallback) identity_handoff_cb, &ctx); audioresample = gst_element_factory_make ("audioresample", "resample"); fail_unless (audioresample != NULL); capsfilter2 = gst_element_factory_make ("capsfilter", "capsfilter2"); fail_unless (capsfilter2 != NULL); caps = gst_caps_from_string ("audio/x-raw, format=" GST_AUDIO_NE (F64) ", channels=1, rate=4096"); g_object_set (G_OBJECT (capsfilter2), "caps", caps, NULL); gst_caps_unref (caps); fakesink = gst_element_factory_make ("fakesink", "sink"); fail_unless (fakesink != NULL); g_object_set (G_OBJECT (fakesink), "sync", FALSE, "async", FALSE, "signal-handoffs", TRUE, NULL); g_signal_connect (fakesink, "handoff", (GCallback) fakesink_handoff_cb, &ctx); gst_bin_add_many (GST_BIN (pipeline), audiotestsrc, capsfilter1, identity, audioresample, capsfilter2, fakesink, NULL); fail_unless (gst_element_link_many (audiotestsrc, capsfilter1, identity, audioresample, capsfilter2, fakesink, NULL)); loop = g_main_loop_new (NULL, FALSE); bus = gst_element_get_bus (pipeline); gst_bus_add_signal_watch (bus); g_signal_connect (bus, "message", (GCallback) _message_cb, loop); fail_unless (gst_element_set_state (pipeline, GST_STATE_PLAYING) == GST_STATE_CHANGE_SUCCESS); g_main_loop_run (loop); fail_unless (gst_element_set_state (pipeline, GST_STATE_NULL) == GST_STATE_CHANGE_SUCCESS); g_main_loop_unref (loop); gst_bus_remove_signal_watch (bus); gst_object_unref (bus); gst_object_unref (pipeline); } GST_END_TEST; #define FFT_HELPERS(type,ffttag,ffttag2,scale); \ static gdouble magnitude##ffttag (const GstFFT##ffttag##Complex *c) \ { \ gdouble mag = (gdouble) c->r * (gdouble) c->r; \ mag += (gdouble) c->i * (gdouble) c->i; \ mag /= scale * scale; \ mag = 10.0 * log10 (mag); \ return mag; \ } \ static gdouble find_main_frequency_spot_##ffttag (const GstFFT##ffttag##Complex *v, \ int elements) \ { \ int i; \ gdouble maxmag = -9999; \ int maxidx = 0; \ for (i=0; i<elements; ++i) { \ gdouble mag = magnitude##ffttag (v+i); \ if (mag > maxmag) { \ maxmag = mag; \ maxidx = i; \ } \ } \ return maxidx / (gdouble) elements; \ } \ static gboolean is_zero_except_##ffttag (const GstFFT##ffttag##Complex *v, int elements, \ gdouble spot) \ { \ int i; \ for (i=0; i<elements; ++i) { \ gdouble pos = i / (gdouble) elements; \ gdouble mag = magnitude##ffttag (v+i); \ if (fabs (pos - spot) > 0.01) { \ if (mag > -55.0) { \ return FALSE; \ } \ } \ } \ return TRUE; \ } \ static void compare_ffts_##ffttag (GstBuffer *inbuffer, GstBuffer *outbuffer) \ { \ GstMapInfo inmap, outmap; \ int insamples, outsamples; \ gdouble inspot, outspot; \ GstFFT##ffttag *inctx, *outctx; \ GstFFT##ffttag##Complex *in, *out; \ \ gst_buffer_map (inbuffer, &inmap, GST_MAP_READ); \ gst_buffer_map (outbuffer, &outmap, GST_MAP_READWRITE); \ \ insamples = inmap.size / sizeof(type) & ~1; \ outsamples = outmap.size / sizeof(type) & ~1; \ inctx = gst_fft_##ffttag2##_new (insamples, FALSE); \ outctx = gst_fft_##ffttag2##_new (outsamples, FALSE); \ in = g_new (GstFFT##ffttag##Complex, insamples / 2 + 1); \ out = g_new (GstFFT##ffttag##Complex, outsamples / 2 + 1); \ \ gst_fft_##ffttag2##_window (inctx, (type*)inmap.data, \ GST_FFT_WINDOW_HAMMING); \ gst_fft_##ffttag2##_fft (inctx, (type*)inmap.data, in); \ gst_fft_##ffttag2##_window (outctx, (type*)outmap.data, \ GST_FFT_WINDOW_HAMMING); \ gst_fft_##ffttag2##_fft (outctx, (type*)outmap.data, out); \ \ inspot = find_main_frequency_spot_##ffttag (in, insamples / 2 + 1); \ outspot = find_main_frequency_spot_##ffttag (out, outsamples / 2 + 1); \ GST_LOG ("Spots are %.3f and %.3f", inspot, outspot); \ fail_unless (fabs (outspot - inspot) < 0.05); \ fail_unless (is_zero_except_##ffttag (in, insamples / 2 + 1, inspot)); \ fail_unless (is_zero_except_##ffttag (out, outsamples / 2 + 1, outspot)); \ \ gst_buffer_unmap (inbuffer, &inmap); \ gst_buffer_unmap (outbuffer, &outmap); \ \ gst_fft_##ffttag2##_free (inctx); \ gst_fft_##ffttag2##_free (outctx); \ g_free (in); \ g_free (out); \ } FFT_HELPERS (float, F32, f32, 2048.0f); FFT_HELPERS (double, F64, f64, 2048.0); FFT_HELPERS (gint16, S16, s16, 32767.0); FFT_HELPERS (gint32, S32, s32, 2147483647.0); #define FILL_BUFFER(type, desc, value); \ static void init_##type##_##desc (GstBuffer *buffer) \ { \ GstMapInfo map; \ type *ptr; \ int i, nsamples; \ gst_buffer_map (buffer, &map, GST_MAP_WRITE); \ ptr = (type *)map.data; \ nsamples = map.size / sizeof (type); \ for (i = 0; i < nsamples; ++i) { \ *ptr++ = value; \ } \ gst_buffer_unmap (buffer, &map); \ } FILL_BUFFER (float, silence, 0.0f); FILL_BUFFER (double, silence, 0.0); FILL_BUFFER (gint16, silence, 0); FILL_BUFFER (gint32, silence, 0); FILL_BUFFER (float, sine, sinf (i * 0.01f)); FILL_BUFFER (float, sine2, sinf (i * 1.8f)); FILL_BUFFER (double, sine, sin (i * 0.01)); FILL_BUFFER (double, sine2, sin (i * 1.8)); FILL_BUFFER (gint16, sine, (gint16) (32767 * sinf (i * 0.01f))); FILL_BUFFER (gint16, sine2, (gint16) (32767 * sinf (i * 1.8f))); FILL_BUFFER (gint32, sine, (gint32) (2147483647.0 * sin (i * 0.01))); FILL_BUFFER (gint32, sine2, (gint32) (2147483647.0 * sin (i * 1.8))); static void run_fft_pipeline (int inrate, int outrate, int quality, int width, const gchar * format, void (*init) (GstBuffer *), void (*compare_ffts) (GstBuffer *, GstBuffer *)) { GstElement *audioresample; GstBuffer *inbuffer, *outbuffer; const int nsamples = 2048; audioresample = setup_audioresample (1, 0, inrate, outrate, format); fail_unless (audioresample != NULL); g_object_set (audioresample, "quality", quality, NULL); fail_unless (gst_element_set_state (audioresample, GST_STATE_PLAYING) == GST_STATE_CHANGE_SUCCESS, "could not set to playing"); inbuffer = gst_buffer_new_and_alloc (nsamples * width / 8); GST_BUFFER_DURATION (inbuffer) = GST_FRAMES_TO_CLOCK_TIME (nsamples, inrate); GST_BUFFER_TIMESTAMP (inbuffer) = 0; (*init) (inbuffer); gst_buffer_ref (inbuffer); /* pushing gives away my reference ... */ fail_unless (gst_pad_push (mysrcpad, inbuffer) == GST_FLOW_OK); /* ... but it ends up being collected on the global buffer list */ fail_unless_equals_int (g_list_length (buffers), 1); /* retrieve out buffer */ fail_if ((outbuffer = (GstBuffer *) buffers->data) == NULL); fail_unless (gst_element_set_state (audioresample, GST_STATE_NULL) == GST_STATE_CHANGE_SUCCESS, "could not set to null"); if (inbuffer == outbuffer) gst_buffer_unref (inbuffer); (*compare_ffts) (inbuffer, outbuffer); /* cleanup */ cleanup_audioresample (audioresample); } GST_START_TEST (test_fft) { int quality; size_t f0, f1; static const int frequencies[] = { 8000, 16000, 44100, 48000, 128000, 12345, 54321 }; /* audioresample uses a mixed float/double code path for floats with quality>8, make sure we test it */ for (quality = 0; quality <= 10; quality += 5) { for (f0 = 0; f0 < G_N_ELEMENTS (frequencies); ++f0) { for (f1 = 0; f1 < G_N_ELEMENTS (frequencies); ++f1) { run_fft_pipeline (frequencies[f0], frequencies[f0], quality, 32, GST_AUDIO_NE (F32), &init_float_silence, &compare_ffts_F32); run_fft_pipeline (frequencies[f0], frequencies[f0], quality, 32, GST_AUDIO_NE (F32), &init_float_sine, &compare_ffts_F32); run_fft_pipeline (frequencies[f0], frequencies[f0], quality, 32, GST_AUDIO_NE (F32), &init_float_sine2, &compare_ffts_F32); run_fft_pipeline (frequencies[f0], frequencies[f0], quality, 64, GST_AUDIO_NE (F64), &init_double_silence, &compare_ffts_F64); run_fft_pipeline (frequencies[f0], frequencies[f0], quality, 64, GST_AUDIO_NE (F64), &init_double_sine, &compare_ffts_F64); run_fft_pipeline (frequencies[f0], frequencies[f0], quality, 64, GST_AUDIO_NE (F64), &init_double_sine2, &compare_ffts_F64); run_fft_pipeline (frequencies[f0], frequencies[f0], quality, 16, GST_AUDIO_NE (S16), &init_gint16_silence, &compare_ffts_S16); run_fft_pipeline (frequencies[f0], frequencies[f0], quality, 16, GST_AUDIO_NE (S16), &init_gint16_sine, &compare_ffts_S16); run_fft_pipeline (frequencies[f0], frequencies[f0], quality, 16, GST_AUDIO_NE (S16), &init_gint16_sine2, &compare_ffts_S16); run_fft_pipeline (frequencies[f0], frequencies[f0], quality, 32, GST_AUDIO_NE (S32), &init_gint32_silence, &compare_ffts_S32); run_fft_pipeline (frequencies[f0], frequencies[f0], quality, 32, GST_AUDIO_NE (S32), &init_gint32_sine, &compare_ffts_S32); run_fft_pipeline (frequencies[f0], frequencies[f0], quality, 32, GST_AUDIO_NE (S32), &init_gint32_sine2, &compare_ffts_S32); } } } } GST_END_TEST; static Suite * audioresample_suite (void) { Suite *s = suite_create ("audioresample"); TCase *tc_chain = tcase_create ("general"); suite_add_tcase (s, tc_chain); tcase_add_test (tc_chain, test_perfect_stream); tcase_add_test (tc_chain, test_discont_stream); tcase_add_test (tc_chain, test_reuse); tcase_add_test (tc_chain, test_shutdown); tcase_add_test (tc_chain, test_live_switch); tcase_add_test (tc_chain, test_live_switch_downstream); tcase_add_test (tc_chain, test_timestamp_drift); tcase_add_test (tc_chain, test_fft); #ifndef GST_DISABLE_PARSE tcase_set_timeout (tc_chain, 360); tcase_add_test (tc_chain, test_pipelines); tcase_add_test (tc_chain, test_preference_passthrough); #endif return s; } GST_CHECK_MAIN (audioresample);