| /* GStreamer |
| * |
| * Copyright (C) 2007 Sebastian Dröge <slomo@circular-chaos.org> |
| * |
| * audiowsinclimit.c: Unit test for the audiowsinclimit element |
| * |
| * This library is free software; you can redistribute it and/or |
| * modify it under the terms of the GNU Lesser General Public License |
| * as published by the Free Software Foundation; either version 2.1 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 |
| * Lesser General Public License for more details. |
| * |
| * You should have received a copy of the GNU Lesser General Public |
| * License along with this library; if not, write to the Free Software |
| * Foundation, Inc., 51 Franklin Street, Fifth Floor, Boston, MA |
| * 02110-1301 USA |
| */ |
| |
| #include <gst/gst.h> |
| #include <gst/audio/audio.h> |
| #include <gst/base/gstbasetransform.h> |
| #include <gst/check/gstcheck.h> |
| |
| #include <math.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 */ |
| GstPad *mysrcpad, *mysinkpad; |
| |
| #define AUDIO_WSINC_LIMIT_CAPS_STRING_32 \ |
| "audio/x-raw, " \ |
| "format = (string) " GST_AUDIO_NE (F32) ", " \ |
| "layout = (string) interleaved, " \ |
| "channels = (int) 1, " \ |
| "rate = (int) 44100" |
| |
| #define AUDIO_WSINC_LIMIT_CAPS_STRING_64 \ |
| "audio/x-raw, " \ |
| "format = (string) " GST_AUDIO_NE (F64) ", " \ |
| "layout = (string) interleaved, " \ |
| "channels = (int) 1, " \ |
| "rate = (int) 44100" |
| |
| #define FORMATS "{ "GST_AUDIO_NE (F32)","GST_AUDIO_NE (F64)" }" |
| |
| static GstStaticPadTemplate sinktemplate = GST_STATIC_PAD_TEMPLATE ("sink", |
| GST_PAD_SINK, |
| GST_PAD_ALWAYS, |
| GST_STATIC_CAPS ("audio/x-raw, " |
| "format = (string) " FORMATS ", " |
| "layout = (string) interleaved, " |
| "channels = (int) 1, " "rate = (int) 44100") |
| ); |
| static GstStaticPadTemplate srctemplate = GST_STATIC_PAD_TEMPLATE ("src", |
| GST_PAD_SRC, |
| GST_PAD_ALWAYS, |
| GST_STATIC_CAPS ("audio/x-raw, " |
| "format = (string) " FORMATS ", " |
| "layout = (string) interleaved, " |
| "channels = (int) 1, " "rate = (int) 44100") |
| ); |
| |
| static GstElement * |
| setup_audiowsinclimit (void) |
| { |
| GstElement *audiowsinclimit; |
| |
| GST_DEBUG ("setup_audiowsinclimit"); |
| audiowsinclimit = gst_check_setup_element ("audiowsinclimit"); |
| mysrcpad = gst_check_setup_src_pad (audiowsinclimit, &srctemplate); |
| mysinkpad = gst_check_setup_sink_pad (audiowsinclimit, &sinktemplate); |
| gst_pad_set_active (mysrcpad, TRUE); |
| gst_pad_set_active (mysinkpad, TRUE); |
| |
| return audiowsinclimit; |
| } |
| |
| static void |
| cleanup_audiowsinclimit (GstElement * audiowsinclimit) |
| { |
| GST_DEBUG ("cleanup_audiowsinclimit"); |
| |
| g_list_foreach (buffers, (GFunc) gst_mini_object_unref, NULL); |
| g_list_free (buffers); |
| buffers = NULL; |
| |
| gst_pad_set_active (mysrcpad, FALSE); |
| gst_pad_set_active (mysinkpad, FALSE); |
| gst_check_teardown_src_pad (audiowsinclimit); |
| gst_check_teardown_sink_pad (audiowsinclimit); |
| gst_check_teardown_element (audiowsinclimit); |
| } |
| |
| /* Test if data containing only one frequency component |
| * at 0 is preserved with lowpass mode and a cutoff |
| * at rate/4 */ |
| GST_START_TEST (test_32_lp_0hz) |
| { |
| GstElement *audiowsinclimit; |
| GstBuffer *inbuffer, *outbuffer; |
| GstCaps *caps; |
| gfloat *in, *res, rms; |
| gint i; |
| GstMapInfo map; |
| GList *node; |
| GstSegment segment; |
| |
| audiowsinclimit = setup_audiowsinclimit (); |
| /* Set to lowpass */ |
| g_object_set (G_OBJECT (audiowsinclimit), "mode", 0, NULL); |
| g_object_set (G_OBJECT (audiowsinclimit), "length", 21, NULL); |
| |
| fail_unless (gst_element_set_state (audiowsinclimit, |
| GST_STATE_PLAYING) == GST_STATE_CHANGE_SUCCESS, |
| "could not set to playing"); |
| |
| /* cutoff = sampling rate / 4, data = 0 */ |
| g_object_set (G_OBJECT (audiowsinclimit), "cutoff", 44100 / 4.0, NULL); |
| inbuffer = gst_buffer_new_and_alloc (128 * sizeof (gfloat)); |
| GST_BUFFER_TIMESTAMP (inbuffer) = 0; |
| gst_buffer_map (inbuffer, &map, GST_MAP_WRITE); |
| in = (gfloat *) map.data; |
| for (i = 0; i < 128; i++) |
| in[i] = 1.0; |
| gst_buffer_unmap (inbuffer, &map); |
| |
| caps = gst_caps_from_string (AUDIO_WSINC_LIMIT_CAPS_STRING_32); |
| gst_check_setup_events (mysrcpad, audiowsinclimit, caps, GST_FORMAT_TIME); |
| gst_caps_unref (caps); |
| ASSERT_BUFFER_REFCOUNT (inbuffer, "inbuffer", 1); |
| |
| /* ensure segment (format) properly setup */ |
| gst_segment_init (&segment, GST_FORMAT_TIME); |
| fail_unless (gst_pad_push_event (mysrcpad, gst_event_new_segment (&segment))); |
| |
| /* pushing gives away my reference ... */ |
| fail_unless (gst_pad_push (mysrcpad, inbuffer) == GST_FLOW_OK); |
| fail_unless (gst_pad_push_event (mysrcpad, gst_event_new_eos ())); |
| /* ... and puts a new buffer on the global list */ |
| fail_unless (g_list_length (buffers) >= 1); |
| |
| for (node = buffers; node; node = node->next) { |
| gint buffer_length; |
| |
| fail_if ((outbuffer = (GstBuffer *) node->data) == NULL); |
| |
| gst_buffer_map (outbuffer, &map, GST_MAP_READ); |
| res = (gfloat *) map.data; |
| buffer_length = map.size / sizeof (gfloat); |
| rms = 0.0; |
| for (i = 0; i < buffer_length; i++) |
| rms += res[i] * res[i]; |
| rms = sqrt (rms / buffer_length); |
| gst_buffer_unmap (outbuffer, &map); |
| fail_unless (rms >= 0.9); |
| } |
| |
| /* cleanup */ |
| cleanup_audiowsinclimit (audiowsinclimit); |
| } |
| |
| GST_END_TEST; |
| |
| /* Test if data containing only one frequency component |
| * at rate/2 is erased with lowpass mode and a cutoff |
| * at rate/4 */ |
| GST_START_TEST (test_32_lp_22050hz) |
| { |
| GstElement *audiowsinclimit; |
| GstBuffer *inbuffer, *outbuffer; |
| GstCaps *caps; |
| gfloat *in, *res, rms; |
| gint i; |
| GstMapInfo map; |
| GList *node; |
| GstSegment segment; |
| |
| audiowsinclimit = setup_audiowsinclimit (); |
| /* Set to lowpass */ |
| g_object_set (G_OBJECT (audiowsinclimit), "mode", 0, NULL); |
| g_object_set (G_OBJECT (audiowsinclimit), "length", 21, NULL); |
| |
| fail_unless (gst_element_set_state (audiowsinclimit, |
| GST_STATE_PLAYING) == GST_STATE_CHANGE_SUCCESS, |
| "could not set to playing"); |
| |
| g_object_set (G_OBJECT (audiowsinclimit), "cutoff", 44100 / 4.0, NULL); |
| inbuffer = gst_buffer_new_and_alloc (128 * sizeof (gfloat)); |
| GST_BUFFER_TIMESTAMP (inbuffer) = 0; |
| gst_buffer_map (inbuffer, &map, GST_MAP_WRITE); |
| in = (gfloat *) map.data; |
| for (i = 0; i < 128; i += 2) { |
| in[i] = 1.0; |
| in[i + 1] = -1.0; |
| } |
| gst_buffer_unmap (inbuffer, &map); |
| |
| caps = gst_caps_from_string (AUDIO_WSINC_LIMIT_CAPS_STRING_32); |
| gst_check_setup_events (mysrcpad, audiowsinclimit, caps, GST_FORMAT_TIME); |
| gst_caps_unref (caps); |
| ASSERT_BUFFER_REFCOUNT (inbuffer, "inbuffer", 1); |
| |
| /* ensure segment (format) properly setup */ |
| gst_segment_init (&segment, GST_FORMAT_TIME); |
| fail_unless (gst_pad_push_event (mysrcpad, gst_event_new_segment (&segment))); |
| |
| /* pushing gives away my reference ... */ |
| fail_unless (gst_pad_push (mysrcpad, inbuffer) == GST_FLOW_OK); |
| fail_unless (gst_pad_push_event (mysrcpad, gst_event_new_eos ())); |
| /* ... and puts a new buffer on the global list */ |
| fail_unless (g_list_length (buffers) >= 1); |
| |
| for (node = buffers; node; node = node->next) { |
| gint buffer_length; |
| |
| fail_if ((outbuffer = (GstBuffer *) node->data) == NULL); |
| |
| gst_buffer_map (outbuffer, &map, GST_MAP_READ); |
| res = (gfloat *) map.data; |
| buffer_length = map.size / sizeof (gfloat); |
| rms = 0.0; |
| for (i = 0; i < buffer_length; i++) |
| rms += res[i] * res[i]; |
| rms = sqrt (rms / buffer_length); |
| gst_buffer_unmap (outbuffer, &map); |
| fail_unless (rms <= 0.1); |
| } |
| |
| /* cleanup */ |
| cleanup_audiowsinclimit (audiowsinclimit); |
| } |
| |
| GST_END_TEST; |
| |
| /* Test if data containing only one frequency component |
| * at 0 is erased with highpass mode and a cutoff |
| * at rate/4 */ |
| GST_START_TEST (test_32_hp_0hz) |
| { |
| GstElement *audiowsinclimit; |
| GstBuffer *inbuffer, *outbuffer; |
| GstCaps *caps; |
| gfloat *in, *res, rms; |
| gint i; |
| GstMapInfo map; |
| GList *node; |
| GstSegment segment; |
| |
| audiowsinclimit = setup_audiowsinclimit (); |
| /* Set to highpass */ |
| g_object_set (G_OBJECT (audiowsinclimit), "mode", 1, NULL); |
| g_object_set (G_OBJECT (audiowsinclimit), "length", 21, NULL); |
| |
| fail_unless (gst_element_set_state (audiowsinclimit, |
| GST_STATE_PLAYING) == GST_STATE_CHANGE_SUCCESS, |
| "could not set to playing"); |
| |
| g_object_set (G_OBJECT (audiowsinclimit), "cutoff", 44100 / 4.0, NULL); |
| inbuffer = gst_buffer_new_and_alloc (128 * sizeof (gfloat)); |
| GST_BUFFER_TIMESTAMP (inbuffer) = 0; |
| gst_buffer_map (inbuffer, &map, GST_MAP_WRITE); |
| in = (gfloat *) map.data; |
| for (i = 0; i < 128; i++) |
| in[i] = 1.0; |
| gst_buffer_unmap (inbuffer, &map); |
| |
| caps = gst_caps_from_string (AUDIO_WSINC_LIMIT_CAPS_STRING_32); |
| gst_check_setup_events (mysrcpad, audiowsinclimit, caps, GST_FORMAT_TIME); |
| gst_caps_unref (caps); |
| ASSERT_BUFFER_REFCOUNT (inbuffer, "inbuffer", 1); |
| |
| /* ensure segment (format) properly setup */ |
| gst_segment_init (&segment, GST_FORMAT_TIME); |
| fail_unless (gst_pad_push_event (mysrcpad, gst_event_new_segment (&segment))); |
| |
| /* pushing gives away my reference ... */ |
| fail_unless (gst_pad_push (mysrcpad, inbuffer) == GST_FLOW_OK); |
| fail_unless (gst_pad_push_event (mysrcpad, gst_event_new_eos ())); |
| /* ... and puts a new buffer on the global list */ |
| fail_unless (g_list_length (buffers) >= 1); |
| |
| for (node = buffers; node; node = node->next) { |
| gint buffer_length; |
| |
| fail_if ((outbuffer = (GstBuffer *) node->data) == NULL); |
| |
| gst_buffer_map (outbuffer, &map, GST_MAP_READ); |
| res = (gfloat *) map.data; |
| buffer_length = map.size / sizeof (gfloat); |
| rms = 0.0; |
| for (i = 0; i < buffer_length; i++) |
| rms += res[i] * res[i]; |
| rms = sqrt (rms / buffer_length); |
| gst_buffer_unmap (outbuffer, &map); |
| fail_unless (rms <= 0.1); |
| } |
| |
| /* cleanup */ |
| cleanup_audiowsinclimit (audiowsinclimit); |
| } |
| |
| GST_END_TEST; |
| |
| /* Test if data containing only one frequency component |
| * at rate/2 is preserved with highpass mode and a cutoff |
| * at rate/4 */ |
| GST_START_TEST (test_32_hp_22050hz) |
| { |
| GstElement *audiowsinclimit; |
| GstBuffer *inbuffer, *outbuffer; |
| GstCaps *caps; |
| gfloat *in, *res, rms; |
| gint i; |
| GstMapInfo map; |
| GList *node; |
| GstSegment segment; |
| |
| audiowsinclimit = setup_audiowsinclimit (); |
| /* Set to highpass */ |
| g_object_set (G_OBJECT (audiowsinclimit), "mode", 1, NULL); |
| g_object_set (G_OBJECT (audiowsinclimit), "length", 21, NULL); |
| |
| fail_unless (gst_element_set_state (audiowsinclimit, |
| GST_STATE_PLAYING) == GST_STATE_CHANGE_SUCCESS, |
| "could not set to playing"); |
| |
| g_object_set (G_OBJECT (audiowsinclimit), "cutoff", 44100 / 4.0, NULL); |
| inbuffer = gst_buffer_new_and_alloc (128 * sizeof (gfloat)); |
| GST_BUFFER_TIMESTAMP (inbuffer) = 0; |
| gst_buffer_map (inbuffer, &map, GST_MAP_WRITE); |
| in = (gfloat *) map.data; |
| for (i = 0; i < 128; i += 2) { |
| in[i] = 1.0; |
| in[i + 1] = -1.0; |
| } |
| gst_buffer_unmap (inbuffer, &map); |
| |
| caps = gst_caps_from_string (AUDIO_WSINC_LIMIT_CAPS_STRING_32); |
| gst_check_setup_events (mysrcpad, audiowsinclimit, caps, GST_FORMAT_TIME); |
| gst_caps_unref (caps); |
| ASSERT_BUFFER_REFCOUNT (inbuffer, "inbuffer", 1); |
| |
| /* ensure segment (format) properly setup */ |
| gst_segment_init (&segment, GST_FORMAT_TIME); |
| fail_unless (gst_pad_push_event (mysrcpad, gst_event_new_segment (&segment))); |
| |
| /* pushing gives away my reference ... */ |
| fail_unless (gst_pad_push (mysrcpad, inbuffer) == GST_FLOW_OK); |
| fail_unless (gst_pad_push_event (mysrcpad, gst_event_new_eos ())); |
| /* ... and puts a new buffer on the global list */ |
| fail_unless (g_list_length (buffers) >= 1); |
| fail_if ((outbuffer = (GstBuffer *) buffers->data) == NULL); |
| |
| for (node = buffers; node; node = node->next) { |
| gint buffer_length; |
| |
| fail_if ((outbuffer = (GstBuffer *) node->data) == NULL); |
| |
| gst_buffer_map (outbuffer, &map, GST_MAP_READ); |
| res = (gfloat *) map.data; |
| buffer_length = map.size / sizeof (gfloat); |
| rms = 0.0; |
| for (i = 0; i < buffer_length; i++) |
| rms += res[i] * res[i]; |
| rms = sqrt (rms / buffer_length); |
| gst_buffer_unmap (outbuffer, &map); |
| fail_unless (rms >= 0.9); |
| } |
| |
| /* cleanup */ |
| cleanup_audiowsinclimit (audiowsinclimit); |
| } |
| |
| GST_END_TEST; |
| |
| /* Test if buffers smaller than the kernel size are handled |
| * correctly without accessing wrong memory areas */ |
| GST_START_TEST (test_32_small_buffer) |
| { |
| GstElement *audiowsinclimit; |
| GstBuffer *inbuffer, *outbuffer; |
| GstCaps *caps; |
| gfloat *in; |
| gint i; |
| GstMapInfo map; |
| GstSegment segment; |
| |
| audiowsinclimit = setup_audiowsinclimit (); |
| /* Set to lowpass */ |
| g_object_set (G_OBJECT (audiowsinclimit), "mode", 0, NULL); |
| g_object_set (G_OBJECT (audiowsinclimit), "length", 101, NULL); |
| |
| fail_unless (gst_element_set_state (audiowsinclimit, |
| GST_STATE_PLAYING) == GST_STATE_CHANGE_SUCCESS, |
| "could not set to playing"); |
| |
| g_object_set (G_OBJECT (audiowsinclimit), "cutoff", 44100 / 4.0, NULL); |
| inbuffer = gst_buffer_new_and_alloc (20 * sizeof (gfloat)); |
| GST_BUFFER_TIMESTAMP (inbuffer) = 0; |
| gst_buffer_map (inbuffer, &map, GST_MAP_WRITE); |
| in = (gfloat *) map.data; |
| for (i = 0; i < 20; i++) |
| in[i] = 1.0; |
| gst_buffer_unmap (inbuffer, &map); |
| |
| caps = gst_caps_from_string (AUDIO_WSINC_LIMIT_CAPS_STRING_32); |
| gst_check_setup_events (mysrcpad, audiowsinclimit, caps, GST_FORMAT_TIME); |
| gst_caps_unref (caps); |
| ASSERT_BUFFER_REFCOUNT (inbuffer, "inbuffer", 1); |
| |
| /* ensure segment (format) properly setup */ |
| gst_segment_init (&segment, GST_FORMAT_TIME); |
| fail_unless (gst_pad_push_event (mysrcpad, gst_event_new_segment (&segment))); |
| |
| /* pushing gives away my reference ... */ |
| fail_unless (gst_pad_push (mysrcpad, inbuffer) == GST_FLOW_OK); |
| fail_unless (gst_pad_push_event (mysrcpad, gst_event_new_eos ())); |
| /* ... and puts a new buffer on the global list */ |
| fail_unless (g_list_length (buffers) >= 1); |
| fail_if ((outbuffer = (GstBuffer *) buffers->data) == NULL); |
| |
| /* cleanup */ |
| cleanup_audiowsinclimit (audiowsinclimit); |
| } |
| |
| GST_END_TEST; |
| |
| /* Test if data containing only one frequency component |
| * at 0 is preserved with lowpass mode and a cutoff |
| * at rate/4 */ |
| GST_START_TEST (test_64_lp_0hz) |
| { |
| GstElement *audiowsinclimit; |
| GstBuffer *inbuffer, *outbuffer; |
| GstCaps *caps; |
| gdouble *in, *res, rms; |
| gint i; |
| GstMapInfo map; |
| GList *node; |
| GstSegment segment; |
| |
| audiowsinclimit = setup_audiowsinclimit (); |
| /* Set to lowpass */ |
| g_object_set (G_OBJECT (audiowsinclimit), "mode", 0, NULL); |
| g_object_set (G_OBJECT (audiowsinclimit), "length", 21, NULL); |
| |
| fail_unless (gst_element_set_state (audiowsinclimit, |
| GST_STATE_PLAYING) == GST_STATE_CHANGE_SUCCESS, |
| "could not set to playing"); |
| |
| /* cutoff = sampling rate / 4, data = 0 */ |
| g_object_set (G_OBJECT (audiowsinclimit), "cutoff", 44100 / 4.0, NULL); |
| inbuffer = gst_buffer_new_and_alloc (128 * sizeof (gdouble)); |
| GST_BUFFER_TIMESTAMP (inbuffer) = 0; |
| gst_buffer_map (inbuffer, &map, GST_MAP_WRITE); |
| in = (gdouble *) map.data; |
| for (i = 0; i < 128; i++) |
| in[i] = 1.0; |
| gst_buffer_unmap (inbuffer, &map); |
| |
| caps = gst_caps_from_string (AUDIO_WSINC_LIMIT_CAPS_STRING_64); |
| gst_check_setup_events (mysrcpad, audiowsinclimit, caps, GST_FORMAT_TIME); |
| gst_caps_unref (caps); |
| ASSERT_BUFFER_REFCOUNT (inbuffer, "inbuffer", 1); |
| |
| /* ensure segment (format) properly setup */ |
| gst_segment_init (&segment, GST_FORMAT_TIME); |
| fail_unless (gst_pad_push_event (mysrcpad, gst_event_new_segment (&segment))); |
| |
| /* pushing gives away my reference ... */ |
| fail_unless (gst_pad_push (mysrcpad, inbuffer) == GST_FLOW_OK); |
| fail_unless (gst_pad_push_event (mysrcpad, gst_event_new_eos ())); |
| /* ... and puts a new buffer on the global list */ |
| fail_unless (g_list_length (buffers) >= 1); |
| |
| for (node = buffers; node; node = node->next) { |
| gint buffer_length; |
| |
| fail_if ((outbuffer = (GstBuffer *) node->data) == NULL); |
| |
| gst_buffer_map (outbuffer, &map, GST_MAP_READ); |
| res = (gdouble *) map.data; |
| buffer_length = map.size / sizeof (gdouble); |
| rms = 0.0; |
| for (i = 0; i < buffer_length; i++) |
| rms += res[i] * res[i]; |
| rms = sqrt (rms / buffer_length); |
| gst_buffer_unmap (outbuffer, &map); |
| fail_unless (rms >= 0.9); |
| } |
| |
| /* cleanup */ |
| cleanup_audiowsinclimit (audiowsinclimit); |
| } |
| |
| GST_END_TEST; |
| |
| /* Test if data containing only one frequency component |
| * at rate/2 is erased with lowpass mode and a cutoff |
| * at rate/4 */ |
| GST_START_TEST (test_64_lp_22050hz) |
| { |
| GstElement *audiowsinclimit; |
| GstBuffer *inbuffer, *outbuffer; |
| GstCaps *caps; |
| gdouble *in, *res, rms; |
| gint i; |
| GstMapInfo map; |
| GList *node; |
| GstSegment segment; |
| |
| audiowsinclimit = setup_audiowsinclimit (); |
| /* Set to lowpass */ |
| g_object_set (G_OBJECT (audiowsinclimit), "mode", 0, NULL); |
| g_object_set (G_OBJECT (audiowsinclimit), "length", 21, NULL); |
| |
| fail_unless (gst_element_set_state (audiowsinclimit, |
| GST_STATE_PLAYING) == GST_STATE_CHANGE_SUCCESS, |
| "could not set to playing"); |
| |
| g_object_set (G_OBJECT (audiowsinclimit), "cutoff", 44100 / 4.0, NULL); |
| inbuffer = gst_buffer_new_and_alloc (128 * sizeof (gdouble)); |
| GST_BUFFER_TIMESTAMP (inbuffer) = 0; |
| gst_buffer_map (inbuffer, &map, GST_MAP_WRITE); |
| in = (gdouble *) map.data; |
| for (i = 0; i < 128; i += 2) { |
| in[i] = 1.0; |
| in[i + 1] = -1.0; |
| } |
| gst_buffer_unmap (inbuffer, &map); |
| |
| caps = gst_caps_from_string (AUDIO_WSINC_LIMIT_CAPS_STRING_64); |
| gst_check_setup_events (mysrcpad, audiowsinclimit, caps, GST_FORMAT_TIME); |
| gst_caps_unref (caps); |
| ASSERT_BUFFER_REFCOUNT (inbuffer, "inbuffer", 1); |
| |
| /* ensure segment (format) properly setup */ |
| gst_segment_init (&segment, GST_FORMAT_TIME); |
| fail_unless (gst_pad_push_event (mysrcpad, gst_event_new_segment (&segment))); |
| |
| /* pushing gives away my reference ... */ |
| fail_unless (gst_pad_push (mysrcpad, inbuffer) == GST_FLOW_OK); |
| fail_unless (gst_pad_push_event (mysrcpad, gst_event_new_eos ())); |
| /* ... and puts a new buffer on the global list */ |
| fail_unless (g_list_length (buffers) >= 1); |
| |
| for (node = buffers; node; node = node->next) { |
| gint buffer_length; |
| |
| fail_if ((outbuffer = (GstBuffer *) node->data) == NULL); |
| |
| gst_buffer_map (outbuffer, &map, GST_MAP_READ); |
| res = (gdouble *) map.data; |
| buffer_length = map.size / sizeof (gdouble); |
| rms = 0.0; |
| for (i = 0; i < buffer_length; i++) |
| rms += res[i] * res[i]; |
| rms = sqrt (rms / buffer_length); |
| gst_buffer_unmap (outbuffer, &map); |
| fail_unless (rms <= 0.1); |
| } |
| |
| /* cleanup */ |
| cleanup_audiowsinclimit (audiowsinclimit); |
| } |
| |
| GST_END_TEST; |
| |
| /* Test if data containing only one frequency component |
| * at 0 is erased with highpass mode and a cutoff |
| * at rate/4 */ |
| GST_START_TEST (test_64_hp_0hz) |
| { |
| GstElement *audiowsinclimit; |
| GstBuffer *inbuffer, *outbuffer; |
| GstCaps *caps; |
| gdouble *in, *res, rms; |
| gint i; |
| GstMapInfo map; |
| GList *node; |
| GstSegment segment; |
| |
| audiowsinclimit = setup_audiowsinclimit (); |
| /* Set to highpass */ |
| g_object_set (G_OBJECT (audiowsinclimit), "mode", 1, NULL); |
| g_object_set (G_OBJECT (audiowsinclimit), "length", 21, NULL); |
| |
| fail_unless (gst_element_set_state (audiowsinclimit, |
| GST_STATE_PLAYING) == GST_STATE_CHANGE_SUCCESS, |
| "could not set to playing"); |
| |
| g_object_set (G_OBJECT (audiowsinclimit), "cutoff", 44100 / 4.0, NULL); |
| inbuffer = gst_buffer_new_and_alloc (128 * sizeof (gdouble)); |
| GST_BUFFER_TIMESTAMP (inbuffer) = 0; |
| gst_buffer_map (inbuffer, &map, GST_MAP_WRITE); |
| in = (gdouble *) map.data; |
| for (i = 0; i < 128; i++) |
| in[i] = 1.0; |
| gst_buffer_unmap (inbuffer, &map); |
| |
| caps = gst_caps_from_string (AUDIO_WSINC_LIMIT_CAPS_STRING_64); |
| gst_check_setup_events (mysrcpad, audiowsinclimit, caps, GST_FORMAT_TIME); |
| gst_caps_unref (caps); |
| ASSERT_BUFFER_REFCOUNT (inbuffer, "inbuffer", 1); |
| |
| /* ensure segment (format) properly setup */ |
| gst_segment_init (&segment, GST_FORMAT_TIME); |
| fail_unless (gst_pad_push_event (mysrcpad, gst_event_new_segment (&segment))); |
| |
| /* pushing gives away my reference ... */ |
| fail_unless (gst_pad_push (mysrcpad, inbuffer) == GST_FLOW_OK); |
| fail_unless (gst_pad_push_event (mysrcpad, gst_event_new_eos ())); |
| /* ... and puts a new buffer on the global list */ |
| fail_unless (g_list_length (buffers) >= 1); |
| |
| for (node = buffers; node; node = node->next) { |
| gint buffer_length; |
| |
| fail_if ((outbuffer = (GstBuffer *) node->data) == NULL); |
| |
| gst_buffer_map (outbuffer, &map, GST_MAP_READ); |
| res = (gdouble *) map.data; |
| buffer_length = map.size / sizeof (gdouble); |
| rms = 0.0; |
| for (i = 0; i < buffer_length; i++) |
| rms += res[i] * res[i]; |
| rms = sqrt (rms / buffer_length); |
| gst_buffer_unmap (outbuffer, &map); |
| fail_unless (rms <= 0.1); |
| } |
| |
| /* cleanup */ |
| cleanup_audiowsinclimit (audiowsinclimit); |
| } |
| |
| GST_END_TEST; |
| |
| /* Test if data containing only one frequency component |
| * at rate/2 is preserved with highpass mode and a cutoff |
| * at rate/4 */ |
| GST_START_TEST (test_64_hp_22050hz) |
| { |
| GstElement *audiowsinclimit; |
| GstBuffer *inbuffer, *outbuffer; |
| GstCaps *caps; |
| gdouble *in, *res, rms; |
| gint i; |
| GstMapInfo map; |
| GList *node; |
| GstSegment segment; |
| |
| audiowsinclimit = setup_audiowsinclimit (); |
| /* Set to highpass */ |
| g_object_set (G_OBJECT (audiowsinclimit), "mode", 1, NULL); |
| g_object_set (G_OBJECT (audiowsinclimit), "length", 21, NULL); |
| |
| fail_unless (gst_element_set_state (audiowsinclimit, |
| GST_STATE_PLAYING) == GST_STATE_CHANGE_SUCCESS, |
| "could not set to playing"); |
| |
| g_object_set (G_OBJECT (audiowsinclimit), "cutoff", 44100 / 4.0, NULL); |
| inbuffer = gst_buffer_new_and_alloc (128 * sizeof (gdouble)); |
| GST_BUFFER_TIMESTAMP (inbuffer) = 0; |
| gst_buffer_map (inbuffer, &map, GST_MAP_WRITE); |
| in = (gdouble *) map.data; |
| for (i = 0; i < 128; i += 2) { |
| in[i] = 1.0; |
| in[i + 1] = -1.0; |
| } |
| gst_buffer_unmap (inbuffer, &map); |
| |
| caps = gst_caps_from_string (AUDIO_WSINC_LIMIT_CAPS_STRING_64); |
| gst_check_setup_events (mysrcpad, audiowsinclimit, caps, GST_FORMAT_TIME); |
| gst_caps_unref (caps); |
| ASSERT_BUFFER_REFCOUNT (inbuffer, "inbuffer", 1); |
| |
| /* ensure segment (format) properly setup */ |
| gst_segment_init (&segment, GST_FORMAT_TIME); |
| fail_unless (gst_pad_push_event (mysrcpad, gst_event_new_segment (&segment))); |
| |
| /* pushing gives away my reference ... */ |
| fail_unless (gst_pad_push (mysrcpad, inbuffer) == GST_FLOW_OK); |
| fail_unless (gst_pad_push_event (mysrcpad, gst_event_new_eos ())); |
| /* ... and puts a new buffer on the global list */ |
| fail_unless (g_list_length (buffers) >= 1); |
| fail_if ((outbuffer = (GstBuffer *) buffers->data) == NULL); |
| |
| for (node = buffers; node; node = node->next) { |
| gint buffer_length; |
| |
| fail_if ((outbuffer = (GstBuffer *) node->data) == NULL); |
| |
| gst_buffer_map (outbuffer, &map, GST_MAP_READ); |
| res = (gdouble *) map.data; |
| buffer_length = map.size / sizeof (gdouble); |
| rms = 0.0; |
| for (i = 0; i < buffer_length; i++) |
| rms += res[i] * res[i]; |
| rms = sqrt (rms / buffer_length); |
| gst_buffer_unmap (outbuffer, &map); |
| fail_unless (rms >= 0.9); |
| } |
| |
| /* cleanup */ |
| cleanup_audiowsinclimit (audiowsinclimit); |
| } |
| |
| GST_END_TEST; |
| |
| /* Test if buffers smaller than the kernel size are handled |
| * correctly without accessing wrong memory areas */ |
| GST_START_TEST (test_64_small_buffer) |
| { |
| GstElement *audiowsinclimit; |
| GstBuffer *inbuffer, *outbuffer; |
| GstCaps *caps; |
| gdouble *in; |
| gint i; |
| GstMapInfo map; |
| GstSegment segment; |
| |
| audiowsinclimit = setup_audiowsinclimit (); |
| /* Set to lowpass */ |
| g_object_set (G_OBJECT (audiowsinclimit), "mode", 0, NULL); |
| g_object_set (G_OBJECT (audiowsinclimit), "length", 101, NULL); |
| |
| fail_unless (gst_element_set_state (audiowsinclimit, |
| GST_STATE_PLAYING) == GST_STATE_CHANGE_SUCCESS, |
| "could not set to playing"); |
| |
| g_object_set (G_OBJECT (audiowsinclimit), "cutoff", 44100 / 4.0, NULL); |
| inbuffer = gst_buffer_new_and_alloc (20 * sizeof (gdouble)); |
| GST_BUFFER_TIMESTAMP (inbuffer) = 0; |
| gst_buffer_map (inbuffer, &map, GST_MAP_WRITE); |
| in = (gdouble *) map.data; |
| for (i = 0; i < 20; i++) |
| in[i] = 1.0; |
| gst_buffer_unmap (inbuffer, &map); |
| |
| caps = gst_caps_from_string (AUDIO_WSINC_LIMIT_CAPS_STRING_64); |
| gst_check_setup_events (mysrcpad, audiowsinclimit, caps, GST_FORMAT_TIME); |
| gst_caps_unref (caps); |
| ASSERT_BUFFER_REFCOUNT (inbuffer, "inbuffer", 1); |
| |
| /* ensure segment (format) properly setup */ |
| gst_segment_init (&segment, GST_FORMAT_TIME); |
| fail_unless (gst_pad_push_event (mysrcpad, gst_event_new_segment (&segment))); |
| |
| /* pushing gives away my reference ... */ |
| fail_unless (gst_pad_push (mysrcpad, inbuffer) == GST_FLOW_OK); |
| fail_unless (gst_pad_push_event (mysrcpad, gst_event_new_eos ())); |
| /* ... and puts a new buffer on the global list */ |
| fail_unless (g_list_length (buffers) >= 1); |
| fail_if ((outbuffer = (GstBuffer *) buffers->data) == NULL); |
| |
| /* cleanup */ |
| cleanup_audiowsinclimit (audiowsinclimit); |
| } |
| |
| GST_END_TEST; |
| |
| static Suite * |
| audiowsinclimit_suite (void) |
| { |
| Suite *s = suite_create ("audiowsinclimit"); |
| TCase *tc_chain = tcase_create ("general"); |
| |
| suite_add_tcase (s, tc_chain); |
| tcase_add_test (tc_chain, test_32_lp_0hz); |
| tcase_add_test (tc_chain, test_32_lp_22050hz); |
| tcase_add_test (tc_chain, test_32_hp_0hz); |
| tcase_add_test (tc_chain, test_32_hp_22050hz); |
| tcase_add_test (tc_chain, test_32_small_buffer); |
| tcase_add_test (tc_chain, test_64_lp_0hz); |
| tcase_add_test (tc_chain, test_64_lp_22050hz); |
| tcase_add_test (tc_chain, test_64_hp_0hz); |
| tcase_add_test (tc_chain, test_64_hp_22050hz); |
| tcase_add_test (tc_chain, test_64_small_buffer); |
| |
| return s; |
| } |
| |
| GST_CHECK_MAIN (audiowsinclimit); |