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),
+ add a check for audioresample
+ (audioresample_suite), (main):
+ * check/elements/volume.c: (GST_START_TEST):
+ remove unused method
+ * gst/audioresample/gstaudioresample.c:
+ set correct buffer parameters since we're changing them
+ * gst/audioresample/resample_ref.c: (resample_scale_ref):
+ add some debug
+
+2005-08-25 Thomas Vander Stichele <thomas at apestaart dot org>
+
* gst/audioresample/debug.c:
* gst/audioresample/gstaudioresample.c:
add room for extra overlap samples when asked to transform size
# 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 audioresample
+ *
+ * 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 RESAMPLE_CAPS_TEMPLATE_STRING \
+ "audio/x-raw-int, " \
+ "channels = (int) [ 1, MAX ], " \
+ "rate = (int) [ 1, MAX ], " \
+ "endianness = (int) BYTE_ORDER, " \
+ "width = (int) 16, " \
+ "depth = (int) 16, " \
+ "signed = (bool) TRUE"
+
+static GstStaticPadTemplate sinktemplate = GST_STATIC_PAD_TEMPLATE ("sink",
+ GST_PAD_SINK,
+ GST_PAD_ALWAYS,
+ GST_STATIC_CAPS (RESAMPLE_CAPS_TEMPLATE_STRING)
+ );
+static GstStaticPadTemplate srctemplate = GST_STATIC_PAD_TEMPLATE ("src",
+ GST_PAD_SRC,
+ GST_PAD_ALWAYS,
+ GST_STATIC_CAPS (RESAMPLE_CAPS_TEMPLATE_STRING)
+ );
+
+GstElement *
+setup_audioresample (int inchannels, int inrate, int outchannels, int outrate)
+{
+ GstElement *audioresample;
+ GstCaps *caps;
+ GstStructure *structure;
+ GstPad *pad;
+
+ GST_DEBUG ("setup_audioresample");
+ audioresample = gst_check_setup_element ("audioresample");
+
+ caps = gst_caps_from_string (RESAMPLE_CAPS_TEMPLATE_STRING);
+ structure = gst_caps_get_structure (caps, 0);
+ gst_structure_set (structure, "channels", G_TYPE_INT, inchannels,
+ "rate", G_TYPE_INT, inrate, NULL);
+ fail_unless (gst_caps_is_fixed (caps));
+
+ mysrcpad = gst_check_setup_src_pad (audioresample, &srctemplate, caps);
+ pad = gst_pad_get_peer (mysrcpad);
+ gst_pad_set_caps (pad, caps);
+ gst_object_unref (GST_OBJECT (pad));
+ gst_caps_unref (caps);
+
+ caps = gst_caps_from_string (RESAMPLE_CAPS_TEMPLATE_STRING);
+ structure = gst_caps_get_structure (caps, 0);
+ gst_structure_set (structure, "channels", G_TYPE_INT, outchannels,
+ "rate", G_TYPE_INT, outrate, NULL);
+ fail_unless (gst_caps_is_fixed (caps));
+
+ mysinkpad = gst_check_setup_sink_pad (audioresample, &sinktemplate, caps);
+ /* this installs a getcaps func that will always return the caps we set
+ * later */
+ gst_pad_use_fixed_caps (mysinkpad);
+ pad = gst_pad_get_peer (mysinkpad);
+ gst_pad_set_caps (pad, caps);
+ gst_object_unref (GST_OBJECT (pad));
+ gst_caps_unref (caps);
+
+ return audioresample;
+}
+
+void
+cleanup_audioresample (GstElement * audioresample)
+{
+ GST_DEBUG ("cleanup_audioresample");
+
+ gst_check_teardown_src_pad (audioresample);
+ gst_check_teardown_sink_pad (audioresample);
+ gst_check_teardown_element (audioresample);
+}
+
+static void
+fail_unless_perfect_stream ()
+{
+ 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, GST_BUFFER_TIMESTAMP (buffer),
+ GST_BUFFER_DURATION (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;
+ }
+}
+
+static void
+test_perfect_stream_instance (int inrate, int outrate, int samples,
+ int numbuffers)
+{
+ GstElement *audioresample;
+ GstBuffer *inbuffer, *outbuffer;
+ GstCaps *caps;
+
+ int i, j;
+ gint16 *p;
+
+ audioresample = setup_audioresample (2, inrate, 2, outrate);
+ caps = gst_pad_get_negotiated_caps (mysrcpad);
+ fail_unless (gst_caps_is_fixed (caps));
+
+ fail_unless (gst_element_set_state (audioresample,
+ GST_STATE_PLAYING) == GST_STATE_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;
+ GST_BUFFER_TIMESTAMP (inbuffer) = GST_BUFFER_DURATION (inbuffer) * (j - 1);
+ GST_BUFFER_OFFSET (inbuffer) = 0;
+ GST_BUFFER_OFFSET_END (inbuffer) = samples;
+
+ gst_buffer_set_caps (inbuffer, caps);
+
+ p = (gint16 *) GST_BUFFER_DATA (inbuffer);
+
+ /* create a 16 bit signed ramp */
+ for (i = 0; i < samples; ++i) {
+ *p = -32767 + i * (65535 / samples);
+ ++p;
+ *p = -32767 + i * (65535 / samples);
+ ++p;
+ }
+
+ /* 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 (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)
+{
+ test_perfect_stream_instance (4000, 2000, 1000, 20);
+}
+
+GST_END_TEST;
+
+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);
+
+ return s;
+}
+
+int
+main (int argc, char **argv)
+{
+ int nf;
+
+ Suite *s = audioresample_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_STATIC_CAPS (VOLUME_CAPS_TEMPLATE_STRING)
);
-GstFlowReturn
-chain_func (GstPad * pad, GstBuffer * buffer)
-{
- GST_DEBUG ("chain_func: received buffer %p", buffer);
- buffers = g_list_append (buffers, buffer);
-
- return GST_FLOW_OK;
-}
-
GstElement *
setup_volume ()
{
guchar *data;
gulong size;
int outsize;
+ int outsamples;
/* FIXME: move to _inplace */
#if 0
}
outsize = resample_get_output_data (r, GST_BUFFER_DATA (outbuf), outsize);
+ outsamples = outsize / r->sample_size;
+ GST_LOG_OBJECT (audioresample, "resample gave me %d bytes or %d samples",
+ outsize, outsamples);
+
GST_BUFFER_TIMESTAMP (outbuf) =
audioresample->offset * GST_SECOND / audioresample->o_rate;
- audioresample->offset += outsize / sizeof (gint16) / audioresample->channels;
- GST_BUFFER_DURATION (outbuf) = outsize * 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;
/* check for possible mem corruption */
if (outsize > GST_BUFFER_SIZE (outbuf)) {
-0.5 * r->i_inc, r->i_inc);
buffer = audioresample_buffer_queue_pull (r->queue, r->sample_size);
if (buffer == NULL) {
+ /* FIXME: for the first buffer, this isn't necessarily an error,
+ * since because of the filter length we'll output less buffers.
+ * deal with that so we don't print to console */
RESAMPLE_ERROR ("buffer_queue_pull returned NULL");
return;
}
# 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 audioresample
+ *
+ * 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 RESAMPLE_CAPS_TEMPLATE_STRING \
+ "audio/x-raw-int, " \
+ "channels = (int) [ 1, MAX ], " \
+ "rate = (int) [ 1, MAX ], " \
+ "endianness = (int) BYTE_ORDER, " \
+ "width = (int) 16, " \
+ "depth = (int) 16, " \
+ "signed = (bool) TRUE"
+
+static GstStaticPadTemplate sinktemplate = GST_STATIC_PAD_TEMPLATE ("sink",
+ GST_PAD_SINK,
+ GST_PAD_ALWAYS,
+ GST_STATIC_CAPS (RESAMPLE_CAPS_TEMPLATE_STRING)
+ );
+static GstStaticPadTemplate srctemplate = GST_STATIC_PAD_TEMPLATE ("src",
+ GST_PAD_SRC,
+ GST_PAD_ALWAYS,
+ GST_STATIC_CAPS (RESAMPLE_CAPS_TEMPLATE_STRING)
+ );
+
+GstElement *
+setup_audioresample (int inchannels, int inrate, int outchannels, int outrate)
+{
+ GstElement *audioresample;
+ GstCaps *caps;
+ GstStructure *structure;
+ GstPad *pad;
+
+ GST_DEBUG ("setup_audioresample");
+ audioresample = gst_check_setup_element ("audioresample");
+
+ caps = gst_caps_from_string (RESAMPLE_CAPS_TEMPLATE_STRING);
+ structure = gst_caps_get_structure (caps, 0);
+ gst_structure_set (structure, "channels", G_TYPE_INT, inchannels,
+ "rate", G_TYPE_INT, inrate, NULL);
+ fail_unless (gst_caps_is_fixed (caps));
+
+ mysrcpad = gst_check_setup_src_pad (audioresample, &srctemplate, caps);
+ pad = gst_pad_get_peer (mysrcpad);
+ gst_pad_set_caps (pad, caps);
+ gst_object_unref (GST_OBJECT (pad));
+ gst_caps_unref (caps);
+
+ caps = gst_caps_from_string (RESAMPLE_CAPS_TEMPLATE_STRING);
+ structure = gst_caps_get_structure (caps, 0);
+ gst_structure_set (structure, "channels", G_TYPE_INT, outchannels,
+ "rate", G_TYPE_INT, outrate, NULL);
+ fail_unless (gst_caps_is_fixed (caps));
+
+ mysinkpad = gst_check_setup_sink_pad (audioresample, &sinktemplate, caps);
+ /* this installs a getcaps func that will always return the caps we set
+ * later */
+ gst_pad_use_fixed_caps (mysinkpad);
+ pad = gst_pad_get_peer (mysinkpad);
+ gst_pad_set_caps (pad, caps);
+ gst_object_unref (GST_OBJECT (pad));
+ gst_caps_unref (caps);
+
+ return audioresample;
+}
+
+void
+cleanup_audioresample (GstElement * audioresample)
+{
+ GST_DEBUG ("cleanup_audioresample");
+
+ gst_check_teardown_src_pad (audioresample);
+ gst_check_teardown_sink_pad (audioresample);
+ gst_check_teardown_element (audioresample);
+}
+
+static void
+fail_unless_perfect_stream ()
+{
+ 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, GST_BUFFER_TIMESTAMP (buffer),
+ GST_BUFFER_DURATION (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;
+ }
+}
+
+static void
+test_perfect_stream_instance (int inrate, int outrate, int samples,
+ int numbuffers)
+{
+ GstElement *audioresample;
+ GstBuffer *inbuffer, *outbuffer;
+ GstCaps *caps;
+
+ int i, j;
+ gint16 *p;
+
+ audioresample = setup_audioresample (2, inrate, 2, outrate);
+ caps = gst_pad_get_negotiated_caps (mysrcpad);
+ fail_unless (gst_caps_is_fixed (caps));
+
+ fail_unless (gst_element_set_state (audioresample,
+ GST_STATE_PLAYING) == GST_STATE_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;
+ GST_BUFFER_TIMESTAMP (inbuffer) = GST_BUFFER_DURATION (inbuffer) * (j - 1);
+ GST_BUFFER_OFFSET (inbuffer) = 0;
+ GST_BUFFER_OFFSET_END (inbuffer) = samples;
+
+ gst_buffer_set_caps (inbuffer, caps);
+
+ p = (gint16 *) GST_BUFFER_DATA (inbuffer);
+
+ /* create a 16 bit signed ramp */
+ for (i = 0; i < samples; ++i) {
+ *p = -32767 + i * (65535 / samples);
+ ++p;
+ *p = -32767 + i * (65535 / samples);
+ ++p;
+ }
+
+ /* 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 (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)
+{
+ test_perfect_stream_instance (4000, 2000, 1000, 20);
+}
+
+GST_END_TEST;
+
+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);
+
+ return s;
+}
+
+int
+main (int argc, char **argv)
+{
+ int nf;
+
+ Suite *s = audioresample_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_STATIC_CAPS (VOLUME_CAPS_TEMPLATE_STRING)
);
-GstFlowReturn
-chain_func (GstPad * pad, GstBuffer * buffer)
-{
- GST_DEBUG ("chain_func: received buffer %p", buffer);
- buffers = g_list_append (buffers, buffer);
-
- return GST_FLOW_OK;
-}
-
GstElement *
setup_volume ()
{
projects / platform / upstream / gstreamer.git / commitdiff