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https://gitlab.freedesktop.org/gstreamer/gstreamer.git
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d524439b35
Makes source code smaller, and ensures we go through common initialization path (like the one that sets up XML unit test output ...)
788 lines
25 KiB
C
788 lines
25 KiB
C
/* GStreamer
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*
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* Copyright (C) 2007 Sebastian Dröge <slomo@circular-chaos.org>
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*
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* audiowsinclimit.c: Unit test for the audiowsinclimit element
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*
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* This library is free software; you can redistribute it and/or
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* modify it under the terms of the GNU Lesser General Public License
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* as published by the Free Software Foundation; either version 2.1 of
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* the License, or (at your option) any later version.
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*
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* This library is distributed in the hope that it will be useful, but
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* WITHOUT ANY WARRANTY; without even the implied warranty of
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* MERCHANTABILITY or FITNESS FOR A PARTICULAR PURPOSE. See the GNU
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* Lesser General Public License for more details.
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*
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* You should have received a copy of the GNU Lesser General Public
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* License along with this library; if not, write to the Free Software
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* Foundation, Inc., 51 Franklin Street, Fifth Floor, Boston, MA
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* 02110-1301 USA
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*/
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#include <gst/gst.h>
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#include <gst/audio/audio.h>
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#include <gst/base/gstbasetransform.h>
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#include <gst/check/gstcheck.h>
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#include <math.h>
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/* For ease of programming we use globals to keep refs for our floating
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* src and sink pads we create; otherwise we always have to do get_pad,
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* get_peer, and then remove references in every test function */
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GstPad *mysrcpad, *mysinkpad;
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#define AUDIO_WSINC_LIMIT_CAPS_STRING_32 \
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"audio/x-raw, " \
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"format = (string) " GST_AUDIO_NE (F32) ", " \
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"layout = (string) interleaved, " \
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"channels = (int) 1, " \
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"rate = (int) 44100"
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#define AUDIO_WSINC_LIMIT_CAPS_STRING_64 \
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"audio/x-raw, " \
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"format = (string) " GST_AUDIO_NE (F64) ", " \
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"layout = (string) interleaved, " \
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"channels = (int) 1, " \
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"rate = (int) 44100"
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#define FORMATS "{ "GST_AUDIO_NE (F32)","GST_AUDIO_NE (F64)" }"
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static GstStaticPadTemplate sinktemplate = GST_STATIC_PAD_TEMPLATE ("sink",
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GST_PAD_SINK,
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GST_PAD_ALWAYS,
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GST_STATIC_CAPS ("audio/x-raw, "
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"format = (string) " FORMATS ", "
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"layout = (string) interleaved, "
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"channels = (int) 1, " "rate = (int) 44100")
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);
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static GstStaticPadTemplate srctemplate = GST_STATIC_PAD_TEMPLATE ("src",
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GST_PAD_SRC,
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GST_PAD_ALWAYS,
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GST_STATIC_CAPS ("audio/x-raw, "
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"format = (string) " FORMATS ", "
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"layout = (string) interleaved, "
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"channels = (int) 1, " "rate = (int) 44100")
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);
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static GstElement *
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setup_audiowsinclimit (void)
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{
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GstElement *audiowsinclimit;
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GST_DEBUG ("setup_audiowsinclimit");
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audiowsinclimit = gst_check_setup_element ("audiowsinclimit");
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mysrcpad = gst_check_setup_src_pad (audiowsinclimit, &srctemplate);
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mysinkpad = gst_check_setup_sink_pad (audiowsinclimit, &sinktemplate);
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gst_pad_set_active (mysrcpad, TRUE);
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gst_pad_set_active (mysinkpad, TRUE);
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return audiowsinclimit;
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}
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static void
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cleanup_audiowsinclimit (GstElement * audiowsinclimit)
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{
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GST_DEBUG ("cleanup_audiowsinclimit");
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g_list_foreach (buffers, (GFunc) gst_mini_object_unref, NULL);
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g_list_free (buffers);
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buffers = NULL;
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gst_pad_set_active (mysrcpad, FALSE);
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gst_pad_set_active (mysinkpad, FALSE);
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gst_check_teardown_src_pad (audiowsinclimit);
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gst_check_teardown_sink_pad (audiowsinclimit);
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gst_check_teardown_element (audiowsinclimit);
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}
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/* Test if data containing only one frequency component
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* at 0 is preserved with lowpass mode and a cutoff
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* at rate/4 */
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GST_START_TEST (test_32_lp_0hz)
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{
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GstElement *audiowsinclimit;
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GstBuffer *inbuffer, *outbuffer;
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GstCaps *caps;
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gfloat *in, *res, rms;
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gint i;
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GstMapInfo map;
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GList *node;
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GstSegment segment;
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audiowsinclimit = setup_audiowsinclimit ();
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/* Set to lowpass */
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g_object_set (G_OBJECT (audiowsinclimit), "mode", 0, NULL);
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g_object_set (G_OBJECT (audiowsinclimit), "length", 21, NULL);
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fail_unless (gst_element_set_state (audiowsinclimit,
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GST_STATE_PLAYING) == GST_STATE_CHANGE_SUCCESS,
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"could not set to playing");
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/* cutoff = sampling rate / 4, data = 0 */
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g_object_set (G_OBJECT (audiowsinclimit), "cutoff", 44100 / 4.0, NULL);
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inbuffer = gst_buffer_new_and_alloc (128 * sizeof (gfloat));
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GST_BUFFER_TIMESTAMP (inbuffer) = 0;
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gst_buffer_map (inbuffer, &map, GST_MAP_WRITE);
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in = (gfloat *) map.data;
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for (i = 0; i < 128; i++)
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in[i] = 1.0;
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gst_buffer_unmap (inbuffer, &map);
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caps = gst_caps_from_string (AUDIO_WSINC_LIMIT_CAPS_STRING_32);
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gst_check_setup_events (mysrcpad, audiowsinclimit, caps, GST_FORMAT_TIME);
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gst_caps_unref (caps);
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ASSERT_BUFFER_REFCOUNT (inbuffer, "inbuffer", 1);
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/* ensure segment (format) properly setup */
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gst_segment_init (&segment, GST_FORMAT_TIME);
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fail_unless (gst_pad_push_event (mysrcpad, gst_event_new_segment (&segment)));
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/* pushing gives away my reference ... */
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fail_unless (gst_pad_push (mysrcpad, inbuffer) == GST_FLOW_OK);
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fail_unless (gst_pad_push_event (mysrcpad, gst_event_new_eos ()));
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/* ... and puts a new buffer on the global list */
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fail_unless (g_list_length (buffers) >= 1);
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for (node = buffers; node; node = node->next) {
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gint buffer_length;
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fail_if ((outbuffer = (GstBuffer *) node->data) == NULL);
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gst_buffer_map (outbuffer, &map, GST_MAP_READ);
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res = (gfloat *) map.data;
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buffer_length = map.size / sizeof (gfloat);
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rms = 0.0;
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for (i = 0; i < buffer_length; i++)
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rms += res[i] * res[i];
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rms = sqrt (rms / buffer_length);
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gst_buffer_unmap (outbuffer, &map);
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fail_unless (rms >= 0.9);
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}
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/* cleanup */
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cleanup_audiowsinclimit (audiowsinclimit);
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}
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GST_END_TEST;
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/* Test if data containing only one frequency component
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* at rate/2 is erased with lowpass mode and a cutoff
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* at rate/4 */
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GST_START_TEST (test_32_lp_22050hz)
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{
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GstElement *audiowsinclimit;
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GstBuffer *inbuffer, *outbuffer;
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GstCaps *caps;
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gfloat *in, *res, rms;
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gint i;
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GstMapInfo map;
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GList *node;
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GstSegment segment;
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audiowsinclimit = setup_audiowsinclimit ();
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/* Set to lowpass */
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g_object_set (G_OBJECT (audiowsinclimit), "mode", 0, NULL);
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g_object_set (G_OBJECT (audiowsinclimit), "length", 21, NULL);
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fail_unless (gst_element_set_state (audiowsinclimit,
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GST_STATE_PLAYING) == GST_STATE_CHANGE_SUCCESS,
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"could not set to playing");
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g_object_set (G_OBJECT (audiowsinclimit), "cutoff", 44100 / 4.0, NULL);
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inbuffer = gst_buffer_new_and_alloc (128 * sizeof (gfloat));
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GST_BUFFER_TIMESTAMP (inbuffer) = 0;
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gst_buffer_map (inbuffer, &map, GST_MAP_WRITE);
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in = (gfloat *) map.data;
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for (i = 0; i < 128; i += 2) {
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in[i] = 1.0;
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in[i + 1] = -1.0;
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}
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gst_buffer_unmap (inbuffer, &map);
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caps = gst_caps_from_string (AUDIO_WSINC_LIMIT_CAPS_STRING_32);
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gst_check_setup_events (mysrcpad, audiowsinclimit, caps, GST_FORMAT_TIME);
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gst_caps_unref (caps);
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ASSERT_BUFFER_REFCOUNT (inbuffer, "inbuffer", 1);
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/* ensure segment (format) properly setup */
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gst_segment_init (&segment, GST_FORMAT_TIME);
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fail_unless (gst_pad_push_event (mysrcpad, gst_event_new_segment (&segment)));
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/* pushing gives away my reference ... */
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fail_unless (gst_pad_push (mysrcpad, inbuffer) == GST_FLOW_OK);
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fail_unless (gst_pad_push_event (mysrcpad, gst_event_new_eos ()));
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/* ... and puts a new buffer on the global list */
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fail_unless (g_list_length (buffers) >= 1);
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for (node = buffers; node; node = node->next) {
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gint buffer_length;
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fail_if ((outbuffer = (GstBuffer *) node->data) == NULL);
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gst_buffer_map (outbuffer, &map, GST_MAP_READ);
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res = (gfloat *) map.data;
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buffer_length = map.size / sizeof (gfloat);
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rms = 0.0;
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for (i = 0; i < buffer_length; i++)
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rms += res[i] * res[i];
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rms = sqrt (rms / buffer_length);
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gst_buffer_unmap (outbuffer, &map);
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fail_unless (rms <= 0.1);
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}
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/* cleanup */
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cleanup_audiowsinclimit (audiowsinclimit);
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}
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GST_END_TEST;
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/* Test if data containing only one frequency component
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* at 0 is erased with highpass mode and a cutoff
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* at rate/4 */
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GST_START_TEST (test_32_hp_0hz)
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{
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GstElement *audiowsinclimit;
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GstBuffer *inbuffer, *outbuffer;
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GstCaps *caps;
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gfloat *in, *res, rms;
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gint i;
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GstMapInfo map;
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GList *node;
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GstSegment segment;
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audiowsinclimit = setup_audiowsinclimit ();
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/* Set to highpass */
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g_object_set (G_OBJECT (audiowsinclimit), "mode", 1, NULL);
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g_object_set (G_OBJECT (audiowsinclimit), "length", 21, NULL);
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fail_unless (gst_element_set_state (audiowsinclimit,
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GST_STATE_PLAYING) == GST_STATE_CHANGE_SUCCESS,
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"could not set to playing");
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g_object_set (G_OBJECT (audiowsinclimit), "cutoff", 44100 / 4.0, NULL);
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inbuffer = gst_buffer_new_and_alloc (128 * sizeof (gfloat));
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GST_BUFFER_TIMESTAMP (inbuffer) = 0;
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gst_buffer_map (inbuffer, &map, GST_MAP_WRITE);
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in = (gfloat *) map.data;
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for (i = 0; i < 128; i++)
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in[i] = 1.0;
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gst_buffer_unmap (inbuffer, &map);
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caps = gst_caps_from_string (AUDIO_WSINC_LIMIT_CAPS_STRING_32);
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gst_check_setup_events (mysrcpad, audiowsinclimit, caps, GST_FORMAT_TIME);
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gst_caps_unref (caps);
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ASSERT_BUFFER_REFCOUNT (inbuffer, "inbuffer", 1);
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/* ensure segment (format) properly setup */
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gst_segment_init (&segment, GST_FORMAT_TIME);
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fail_unless (gst_pad_push_event (mysrcpad, gst_event_new_segment (&segment)));
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/* pushing gives away my reference ... */
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fail_unless (gst_pad_push (mysrcpad, inbuffer) == GST_FLOW_OK);
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fail_unless (gst_pad_push_event (mysrcpad, gst_event_new_eos ()));
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/* ... and puts a new buffer on the global list */
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fail_unless (g_list_length (buffers) >= 1);
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for (node = buffers; node; node = node->next) {
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gint buffer_length;
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fail_if ((outbuffer = (GstBuffer *) node->data) == NULL);
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gst_buffer_map (outbuffer, &map, GST_MAP_READ);
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res = (gfloat *) map.data;
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buffer_length = map.size / sizeof (gfloat);
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rms = 0.0;
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for (i = 0; i < buffer_length; i++)
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rms += res[i] * res[i];
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rms = sqrt (rms / buffer_length);
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gst_buffer_unmap (outbuffer, &map);
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fail_unless (rms <= 0.1);
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}
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/* cleanup */
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cleanup_audiowsinclimit (audiowsinclimit);
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}
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GST_END_TEST;
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/* Test if data containing only one frequency component
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* at rate/2 is preserved with highpass mode and a cutoff
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* at rate/4 */
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GST_START_TEST (test_32_hp_22050hz)
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{
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GstElement *audiowsinclimit;
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GstBuffer *inbuffer, *outbuffer;
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GstCaps *caps;
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gfloat *in, *res, rms;
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gint i;
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GstMapInfo map;
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GList *node;
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GstSegment segment;
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audiowsinclimit = setup_audiowsinclimit ();
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/* Set to highpass */
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g_object_set (G_OBJECT (audiowsinclimit), "mode", 1, NULL);
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g_object_set (G_OBJECT (audiowsinclimit), "length", 21, NULL);
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fail_unless (gst_element_set_state (audiowsinclimit,
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GST_STATE_PLAYING) == GST_STATE_CHANGE_SUCCESS,
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"could not set to playing");
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g_object_set (G_OBJECT (audiowsinclimit), "cutoff", 44100 / 4.0, NULL);
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inbuffer = gst_buffer_new_and_alloc (128 * sizeof (gfloat));
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GST_BUFFER_TIMESTAMP (inbuffer) = 0;
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gst_buffer_map (inbuffer, &map, GST_MAP_WRITE);
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in = (gfloat *) map.data;
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for (i = 0; i < 128; i += 2) {
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in[i] = 1.0;
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in[i + 1] = -1.0;
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}
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gst_buffer_unmap (inbuffer, &map);
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caps = gst_caps_from_string (AUDIO_WSINC_LIMIT_CAPS_STRING_32);
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gst_check_setup_events (mysrcpad, audiowsinclimit, caps, GST_FORMAT_TIME);
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gst_caps_unref (caps);
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ASSERT_BUFFER_REFCOUNT (inbuffer, "inbuffer", 1);
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/* ensure segment (format) properly setup */
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gst_segment_init (&segment, GST_FORMAT_TIME);
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fail_unless (gst_pad_push_event (mysrcpad, gst_event_new_segment (&segment)));
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/* pushing gives away my reference ... */
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fail_unless (gst_pad_push (mysrcpad, inbuffer) == GST_FLOW_OK);
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fail_unless (gst_pad_push_event (mysrcpad, gst_event_new_eos ()));
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/* ... and puts a new buffer on the global list */
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fail_unless (g_list_length (buffers) >= 1);
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fail_if ((outbuffer = (GstBuffer *) buffers->data) == NULL);
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for (node = buffers; node; node = node->next) {
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gint buffer_length;
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fail_if ((outbuffer = (GstBuffer *) node->data) == NULL);
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gst_buffer_map (outbuffer, &map, GST_MAP_READ);
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res = (gfloat *) map.data;
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buffer_length = map.size / sizeof (gfloat);
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rms = 0.0;
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for (i = 0; i < buffer_length; i++)
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rms += res[i] * res[i];
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rms = sqrt (rms / buffer_length);
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gst_buffer_unmap (outbuffer, &map);
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fail_unless (rms >= 0.9);
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}
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/* cleanup */
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cleanup_audiowsinclimit (audiowsinclimit);
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}
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GST_END_TEST;
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/* Test if buffers smaller than the kernel size are handled
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* correctly without accessing wrong memory areas */
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GST_START_TEST (test_32_small_buffer)
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{
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GstElement *audiowsinclimit;
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GstBuffer *inbuffer, *outbuffer;
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GstCaps *caps;
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gfloat *in;
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gint i;
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GstMapInfo map;
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GstSegment segment;
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audiowsinclimit = setup_audiowsinclimit ();
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/* Set to lowpass */
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g_object_set (G_OBJECT (audiowsinclimit), "mode", 0, NULL);
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g_object_set (G_OBJECT (audiowsinclimit), "length", 101, NULL);
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fail_unless (gst_element_set_state (audiowsinclimit,
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GST_STATE_PLAYING) == GST_STATE_CHANGE_SUCCESS,
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"could not set to playing");
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g_object_set (G_OBJECT (audiowsinclimit), "cutoff", 44100 / 4.0, NULL);
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inbuffer = gst_buffer_new_and_alloc (20 * sizeof (gfloat));
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GST_BUFFER_TIMESTAMP (inbuffer) = 0;
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gst_buffer_map (inbuffer, &map, GST_MAP_WRITE);
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in = (gfloat *) map.data;
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for (i = 0; i < 20; i++)
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in[i] = 1.0;
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gst_buffer_unmap (inbuffer, &map);
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caps = gst_caps_from_string (AUDIO_WSINC_LIMIT_CAPS_STRING_32);
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gst_check_setup_events (mysrcpad, audiowsinclimit, caps, GST_FORMAT_TIME);
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gst_caps_unref (caps);
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ASSERT_BUFFER_REFCOUNT (inbuffer, "inbuffer", 1);
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/* ensure segment (format) properly setup */
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gst_segment_init (&segment, GST_FORMAT_TIME);
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fail_unless (gst_pad_push_event (mysrcpad, gst_event_new_segment (&segment)));
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/* pushing gives away my reference ... */
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fail_unless (gst_pad_push (mysrcpad, inbuffer) == GST_FLOW_OK);
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fail_unless (gst_pad_push_event (mysrcpad, gst_event_new_eos ()));
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/* ... and puts a new buffer on the global list */
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fail_unless (g_list_length (buffers) >= 1);
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fail_if ((outbuffer = (GstBuffer *) buffers->data) == NULL);
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/* cleanup */
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cleanup_audiowsinclimit (audiowsinclimit);
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}
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GST_END_TEST;
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/* Test if data containing only one frequency component
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* at 0 is preserved with lowpass mode and a cutoff
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* at rate/4 */
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GST_START_TEST (test_64_lp_0hz)
|
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{
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GstElement *audiowsinclimit;
|
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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);
|