2005-08-25 Thomas Vander Stichele <thomas at apestaart dot org>
* check/Makefile.am:
+ * check/elements/audioconvert.c: (setup_audioconvert),
+ (cleanup_audioconvert), (get_int_caps), (verify_convert),
+ (GST_START_TEST), (audioconvert_suite), (main):
+ add a test for audioconvert
+ * gst/audioresample/gstaudioresample.c:
+ * gst/audioresample/gstaudioresample.h:
+ set DURATION so that TIMESTAMP(a) + DURATION(a) == TIMESTAMP(b);
+ note that for buffers of 1/3 sec this means DURATION(c) is
+ one nanosecond more than for a and b
+
+2005-08-25 Thomas Vander Stichele <thomas at apestaart dot org>
+
+ * check/Makefile.am:
* check/elements/audioresample.c: (setup_audioresample),
(cleanup_audioresample), (fail_unless_perfect_stream),
(test_perfect_stream_instance), (GST_START_TEST),
endif
check_PROGRAMS = \
+ elements/audioconvert \
+ elements/audioresample \
elements/volume \
$(check_vorbis)
# these tests don't even pass
# generic/states: elements need state fixin' before this can be added
noinst_PROGRAMS = \
- elements/audioresample \
generic/states
AM_CFLAGS = $(GST_OBJ_CFLAGS) $(GST_CHECK_CFLAGS) $(CHECK_CFLAGS)
--- /dev/null
+/* GStreamer
+ *
+ * unit test for audioconvert
+ *
+ * Copyright (C) <2005> Thomas Vander Stichele <thomas at apestaart dot org>
+ *
+ * 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., 59 Temple Place - Suite 330,
+ * Boston, MA 02111-1307, USA.
+ */
+
+#include <unistd.h>
+
+#include <gst/check/gstcheck.h>
+
+GList *buffers = NULL;
+gboolean have_eos = FALSE;
+
+/* 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 CONVERT_CAPS_TEMPLATE_STRING \
+ "audio/x-raw-float, " \
+ "rate = (int) [ 1, MAX ], " \
+ "channels = (int) [ 1, 8 ], " \
+ "endianness = (int) BYTE_ORDER, " \
+ "width = (int) 32, " \
+ "buffer-frames = (int) [ 0, MAX ];" \
+ "audio/x-raw-int, " \
+ "rate = (int) [ 1, MAX ], " \
+ "channels = (int) [ 1, 8 ], " \
+ "endianness = (int) { LITTLE_ENDIAN, BIG_ENDIAN }, " \
+ "width = (int) 32, " \
+ "depth = (int) [ 1, 32 ], " \
+ "signed = (boolean) { true, false }; " \
+ "audio/x-raw-int, " \
+ "rate = (int) [ 1, MAX ], " \
+ "channels = (int) [ 1, 8 ], " \
+ "endianness = (int) { LITTLE_ENDIAN, BIG_ENDIAN }, " \
+ "width = (int) 24, " \
+ "depth = (int) [ 1, 24 ], " \
+ "signed = (boolean) { true, false }; " \
+ "audio/x-raw-int, " \
+ "rate = (int) [ 1, MAX ], " \
+ "channels = (int) [ 1, 8 ], " \
+ "endianness = (int) { LITTLE_ENDIAN, BIG_ENDIAN }, " \
+ "width = (int) 16, " \
+ "depth = (int) [ 1, 16 ], " \
+ "signed = (boolean) { true, false }; " \
+ "audio/x-raw-int, " \
+ "rate = (int) [ 1, MAX ], " \
+ "channels = (int) [ 1, 8 ], " \
+ "endianness = (int) { LITTLE_ENDIAN, BIG_ENDIAN }, " \
+ "width = (int) 8, " \
+ "depth = (int) [ 1, 8 ], " \
+ "signed = (boolean) { true, false } "
+
+static GstStaticPadTemplate sinktemplate = GST_STATIC_PAD_TEMPLATE ("sink",
+ GST_PAD_SINK,
+ GST_PAD_ALWAYS,
+ GST_STATIC_CAPS (CONVERT_CAPS_TEMPLATE_STRING)
+ );
+static GstStaticPadTemplate srctemplate = GST_STATIC_PAD_TEMPLATE ("src",
+ GST_PAD_SRC,
+ GST_PAD_ALWAYS,
+ GST_STATIC_CAPS (CONVERT_CAPS_TEMPLATE_STRING)
+ );
+
+/* takes over reference for outcaps */
+GstElement *
+setup_audioconvert (GstCaps * outcaps)
+{
+ GstElement *audioconvert;
+
+ GST_DEBUG ("setup_audioconvert");
+ audioconvert = gst_check_setup_element ("audioconvert");
+ mysrcpad = gst_check_setup_src_pad (audioconvert, &srctemplate, NULL);
+ mysinkpad = gst_check_setup_sink_pad (audioconvert, &sinktemplate, NULL);
+ /* this installs a getcaps func that will always return the caps we set
+ * later */
+ gst_pad_use_fixed_caps (mysinkpad);
+ gst_pad_set_caps (mysinkpad, outcaps);
+ gst_caps_unref (outcaps);
+ outcaps = gst_pad_get_negotiated_caps (mysinkpad);
+ fail_unless (gst_caps_is_fixed (outcaps));
+ gst_caps_unref (outcaps);
+
+ return audioconvert;
+}
+
+void
+cleanup_audioconvert (GstElement * audioconvert)
+{
+ GST_DEBUG ("cleanup_audioconvert");
+
+ gst_check_teardown_src_pad (audioconvert);
+ gst_check_teardown_sink_pad (audioconvert);
+ gst_check_teardown_element (audioconvert);
+}
+
+GstCaps *
+get_int_caps (guint rate, guint channels, gchar * endianness, guint width,
+ guint depth, gboolean signedness)
+{
+ GstCaps *caps;
+ gchar *string;
+
+ string = g_strdup_printf ("audio/x-raw-int, "
+ "rate = (int) %d, "
+ "channels = (int) %d, "
+ "endianness = (int) %s, "
+ "width = (int) %d, "
+ "depth = (int) %d, "
+ "signed = (boolean) %s ",
+ rate, channels, endianness, width, depth, signedness ? "true" : "false");
+ GST_DEBUG ("creating caps from %s", string);
+ caps = gst_caps_from_string (string);
+ fail_unless (caps != NULL);
+ g_free (string);
+ return caps;
+}
+
+static void
+verify_convert (GstElement * audioconvert, void *in, int inlength, void *out,
+ int outlength, GstCaps * incaps)
+{
+ GstBuffer *inbuffer, *outbuffer;
+
+ fail_unless (gst_element_set_state (audioconvert,
+ GST_STATE_PLAYING) == GST_STATE_SUCCESS, "could not set to playing");
+
+ GST_DEBUG ("Creating buffer of %d bytes", inlength);
+ inbuffer = gst_buffer_new_and_alloc (inlength);
+ memcpy (GST_BUFFER_DATA (inbuffer), in, inlength);
+ gst_buffer_set_caps (inbuffer, incaps);
+ ASSERT_BUFFER_REFCOUNT (inbuffer, "inbuffer", 1);
+
+ /* pushing gives away my reference ... */
+ fail_unless (gst_pad_push (mysrcpad, inbuffer) == GST_FLOW_OK);
+ /* ... and puts a new buffer on the global list */
+ fail_unless (g_list_length (buffers) == 1);
+ fail_if ((outbuffer = (GstBuffer *) buffers->data) == NULL);
+
+ ASSERT_BUFFER_REFCOUNT (outbuffer, "outbuffer", 1);
+ fail_unless_equals_int (GST_BUFFER_SIZE (outbuffer), outlength);
+ fail_unless (memcmp (GST_BUFFER_DATA (outbuffer), out, outlength) == 0);
+}
+
+GST_START_TEST (test_unity)
+{
+ GstElement *audioconvert;
+ GstCaps *incaps, *outcaps;
+
+ gint16 in[] = { 16384, -256 };
+ gint16 out[] = { 8064 };
+
+ outcaps = get_int_caps (44100, 1, "LITTLE_ENDIAN", 16, 16, TRUE);
+ audioconvert = setup_audioconvert (outcaps);
+
+ incaps = get_int_caps (44100, 2, "LITTLE_ENDIAN", 16, 16, TRUE);
+ verify_convert (audioconvert, in, sizeof (in), out, sizeof (out), incaps);
+
+ /* cleanup */
+ cleanup_audioconvert (audioconvert);
+}
+
+GST_END_TEST;
+
+Suite *
+audioconvert_suite (void)
+{
+ Suite *s = suite_create ("audioconvert");
+ TCase *tc_chain = tcase_create ("general");
+
+ suite_add_tcase (s, tc_chain);
+ tcase_add_test (tc_chain, test_unity);
+
+ return s;
+}
+
+int
+main (int argc, char **argv)
+{
+ int nf;
+
+ Suite *s = audioconvert_suite ();
+ SRunner *sr = srunner_create (s);
+
+ gst_check_init (&argc, &argv);
+
+ srunner_run_all (sr, CK_NORMAL);
+ nf = srunner_ntests_failed (sr);
+ srunner_free (sr);
+
+ return nf;
+}
GST_LOG_OBJECT (audioresample, "resample gave me %d bytes or %d samples",
outsize, outsamples);
+ GST_BUFFER_OFFSET (outbuf) = audioresample->offset;
GST_BUFFER_TIMESTAMP (outbuf) =
audioresample->offset * GST_SECOND / audioresample->o_rate;
- GST_BUFFER_DURATION (outbuf) =
- outsamples * GST_SECOND / audioresample->o_rate;
- GST_BUFFER_OFFSET (outbuf) = audioresample->offset;
+
audioresample->offset += outsamples;
GST_BUFFER_OFFSET_END (outbuf) = audioresample->offset;
+ /* we calculate DURATION as the difference between "next" timestamp
+ * and current timestamp so we ensure a contiguous stream, instead of
+ * having rounding errors. */
+ GST_BUFFER_DURATION (outbuf) =
+ audioresample->offset * GST_SECOND / audioresample->o_rate -
+ GST_BUFFER_TIMESTAMP (outbuf);
+
/* check for possible mem corruption */
if (outsize > GST_BUFFER_SIZE (outbuf)) {
/* this is an error that when it happens, would need fixing in the
gboolean passthru;
- gint64 offset;
+ guint64 offset;
int channels;
int i_rate;
endif
check_PROGRAMS = \
+ elements/audioconvert \
+ elements/audioresample \
elements/volume \
$(check_vorbis)
# these tests don't even pass
# generic/states: elements need state fixin' before this can be added
noinst_PROGRAMS = \
- elements/audioresample \
generic/states
AM_CFLAGS = $(GST_OBJ_CFLAGS) $(GST_CHECK_CFLAGS) $(CHECK_CFLAGS)
--- /dev/null
+/* GStreamer
+ *
+ * unit test for audioconvert
+ *
+ * Copyright (C) <2005> Thomas Vander Stichele <thomas at apestaart dot org>
+ *
+ * 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., 59 Temple Place - Suite 330,
+ * Boston, MA 02111-1307, USA.
+ */
+
+#include <unistd.h>
+
+#include <gst/check/gstcheck.h>
+
+GList *buffers = NULL;
+gboolean have_eos = FALSE;
+
+/* 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 CONVERT_CAPS_TEMPLATE_STRING \
+ "audio/x-raw-float, " \
+ "rate = (int) [ 1, MAX ], " \
+ "channels = (int) [ 1, 8 ], " \
+ "endianness = (int) BYTE_ORDER, " \
+ "width = (int) 32, " \
+ "buffer-frames = (int) [ 0, MAX ];" \
+ "audio/x-raw-int, " \
+ "rate = (int) [ 1, MAX ], " \
+ "channels = (int) [ 1, 8 ], " \
+ "endianness = (int) { LITTLE_ENDIAN, BIG_ENDIAN }, " \
+ "width = (int) 32, " \
+ "depth = (int) [ 1, 32 ], " \
+ "signed = (boolean) { true, false }; " \
+ "audio/x-raw-int, " \
+ "rate = (int) [ 1, MAX ], " \
+ "channels = (int) [ 1, 8 ], " \
+ "endianness = (int) { LITTLE_ENDIAN, BIG_ENDIAN }, " \
+ "width = (int) 24, " \
+ "depth = (int) [ 1, 24 ], " \
+ "signed = (boolean) { true, false }; " \
+ "audio/x-raw-int, " \
+ "rate = (int) [ 1, MAX ], " \
+ "channels = (int) [ 1, 8 ], " \
+ "endianness = (int) { LITTLE_ENDIAN, BIG_ENDIAN }, " \
+ "width = (int) 16, " \
+ "depth = (int) [ 1, 16 ], " \
+ "signed = (boolean) { true, false }; " \
+ "audio/x-raw-int, " \
+ "rate = (int) [ 1, MAX ], " \
+ "channels = (int) [ 1, 8 ], " \
+ "endianness = (int) { LITTLE_ENDIAN, BIG_ENDIAN }, " \
+ "width = (int) 8, " \
+ "depth = (int) [ 1, 8 ], " \
+ "signed = (boolean) { true, false } "
+
+static GstStaticPadTemplate sinktemplate = GST_STATIC_PAD_TEMPLATE ("sink",
+ GST_PAD_SINK,
+ GST_PAD_ALWAYS,
+ GST_STATIC_CAPS (CONVERT_CAPS_TEMPLATE_STRING)
+ );
+static GstStaticPadTemplate srctemplate = GST_STATIC_PAD_TEMPLATE ("src",
+ GST_PAD_SRC,
+ GST_PAD_ALWAYS,
+ GST_STATIC_CAPS (CONVERT_CAPS_TEMPLATE_STRING)
+ );
+
+/* takes over reference for outcaps */
+GstElement *
+setup_audioconvert (GstCaps * outcaps)
+{
+ GstElement *audioconvert;
+
+ GST_DEBUG ("setup_audioconvert");
+ audioconvert = gst_check_setup_element ("audioconvert");
+ mysrcpad = gst_check_setup_src_pad (audioconvert, &srctemplate, NULL);
+ mysinkpad = gst_check_setup_sink_pad (audioconvert, &sinktemplate, NULL);
+ /* this installs a getcaps func that will always return the caps we set
+ * later */
+ gst_pad_use_fixed_caps (mysinkpad);
+ gst_pad_set_caps (mysinkpad, outcaps);
+ gst_caps_unref (outcaps);
+ outcaps = gst_pad_get_negotiated_caps (mysinkpad);
+ fail_unless (gst_caps_is_fixed (outcaps));
+ gst_caps_unref (outcaps);
+
+ return audioconvert;
+}
+
+void
+cleanup_audioconvert (GstElement * audioconvert)
+{
+ GST_DEBUG ("cleanup_audioconvert");
+
+ gst_check_teardown_src_pad (audioconvert);
+ gst_check_teardown_sink_pad (audioconvert);
+ gst_check_teardown_element (audioconvert);
+}
+
+GstCaps *
+get_int_caps (guint rate, guint channels, gchar * endianness, guint width,
+ guint depth, gboolean signedness)
+{
+ GstCaps *caps;
+ gchar *string;
+
+ string = g_strdup_printf ("audio/x-raw-int, "
+ "rate = (int) %d, "
+ "channels = (int) %d, "
+ "endianness = (int) %s, "
+ "width = (int) %d, "
+ "depth = (int) %d, "
+ "signed = (boolean) %s ",
+ rate, channels, endianness, width, depth, signedness ? "true" : "false");
+ GST_DEBUG ("creating caps from %s", string);
+ caps = gst_caps_from_string (string);
+ fail_unless (caps != NULL);
+ g_free (string);
+ return caps;
+}
+
+static void
+verify_convert (GstElement * audioconvert, void *in, int inlength, void *out,
+ int outlength, GstCaps * incaps)
+{
+ GstBuffer *inbuffer, *outbuffer;
+
+ fail_unless (gst_element_set_state (audioconvert,
+ GST_STATE_PLAYING) == GST_STATE_SUCCESS, "could not set to playing");
+
+ GST_DEBUG ("Creating buffer of %d bytes", inlength);
+ inbuffer = gst_buffer_new_and_alloc (inlength);
+ memcpy (GST_BUFFER_DATA (inbuffer), in, inlength);
+ gst_buffer_set_caps (inbuffer, incaps);
+ ASSERT_BUFFER_REFCOUNT (inbuffer, "inbuffer", 1);
+
+ /* pushing gives away my reference ... */
+ fail_unless (gst_pad_push (mysrcpad, inbuffer) == GST_FLOW_OK);
+ /* ... and puts a new buffer on the global list */
+ fail_unless (g_list_length (buffers) == 1);
+ fail_if ((outbuffer = (GstBuffer *) buffers->data) == NULL);
+
+ ASSERT_BUFFER_REFCOUNT (outbuffer, "outbuffer", 1);
+ fail_unless_equals_int (GST_BUFFER_SIZE (outbuffer), outlength);
+ fail_unless (memcmp (GST_BUFFER_DATA (outbuffer), out, outlength) == 0);
+}
+
+GST_START_TEST (test_unity)
+{
+ GstElement *audioconvert;
+ GstCaps *incaps, *outcaps;
+
+ gint16 in[] = { 16384, -256 };
+ gint16 out[] = { 8064 };
+
+ outcaps = get_int_caps (44100, 1, "LITTLE_ENDIAN", 16, 16, TRUE);
+ audioconvert = setup_audioconvert (outcaps);
+
+ incaps = get_int_caps (44100, 2, "LITTLE_ENDIAN", 16, 16, TRUE);
+ verify_convert (audioconvert, in, sizeof (in), out, sizeof (out), incaps);
+
+ /* cleanup */
+ cleanup_audioconvert (audioconvert);
+}
+
+GST_END_TEST;
+
+Suite *
+audioconvert_suite (void)
+{
+ Suite *s = suite_create ("audioconvert");
+ TCase *tc_chain = tcase_create ("general");
+
+ suite_add_tcase (s, tc_chain);
+ tcase_add_test (tc_chain, test_unity);
+
+ return s;
+}
+
+int
+main (int argc, char **argv)
+{
+ int nf;
+
+ Suite *s = audioconvert_suite ();
+ SRunner *sr = srunner_create (s);
+
+ gst_check_init (&argc, &argv);
+
+ srunner_run_all (sr, CK_NORMAL);
+ nf = srunner_ntests_failed (sr);
+ srunner_free (sr);
+
+ return nf;
+}
projects / platform / upstream / gstreamer.git / commitdiff