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check/: add a test for audioconvert
Original commit message from CVS: * 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
This commit is contained in:
Thomas Vander Stichele
parent
8f3a11d6f2
commit
6dff9c2cbd
13
ChangeLog
13
ChangeLog
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@ -1,3 +1,16 @@
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2005-08-25 Thomas Vander Stichele <thomas at apestaart dot org>
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* check/Makefile.am:
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* check/elements/audioconvert.c: (setup_audioconvert),
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(cleanup_audioconvert), (get_int_caps), (verify_convert),
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(GST_START_TEST), (audioconvert_suite), (main):
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add a test for audioconvert
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* gst/audioresample/gstaudioresample.c:
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* gst/audioresample/gstaudioresample.h:
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set DURATION so that TIMESTAMP(a) + DURATION(a) == TIMESTAMP(b);
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note that for buffers of 1/3 sec this means DURATION(c) is
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one nanosecond more than for a and b
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2005-08-25 Thomas Vander Stichele <thomas at apestaart dot org>
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* check/Makefile.am:
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@ -29,13 +29,14 @@ check_vorbis =
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endif
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check_PROGRAMS = \
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elements/audioconvert \
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elements/audioresample \
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elements/volume \
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$(check_vorbis)
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# these tests don't even pass
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# generic/states: elements need state fixin' before this can be added
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noinst_PROGRAMS = \
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elements/audioresample \
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generic/states
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AM_CFLAGS = $(GST_OBJ_CFLAGS) $(GST_CHECK_CFLAGS) $(CHECK_CFLAGS)
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209
check/elements/audioconvert.c
Normal file
209
check/elements/audioconvert.c
Normal file
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@ -0,0 +1,209 @@
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/* GStreamer
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*
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* unit test for audioconvert
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*
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* Copyright (C) <2005> Thomas Vander Stichele <thomas at apestaart dot org>
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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 Library General Public
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* License as published by the Free Software Foundation; either
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* version 2 of 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,
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* but 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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* Library General Public License for more details.
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*
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* You should have received a copy of the GNU Library General Public
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* License along with this library; if not, write to the
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* Free Software Foundation, Inc., 59 Temple Place - Suite 330,
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* Boston, MA 02111-1307, USA.
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*/
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#include <unistd.h>
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#include <gst/check/gstcheck.h>
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GList *buffers = NULL;
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gboolean have_eos = FALSE;
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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 CONVERT_CAPS_TEMPLATE_STRING \
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"audio/x-raw-float, " \
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"rate = (int) [ 1, MAX ], " \
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"channels = (int) [ 1, 8 ], " \
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"endianness = (int) BYTE_ORDER, " \
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"width = (int) 32, " \
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"buffer-frames = (int) [ 0, MAX ];" \
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"audio/x-raw-int, " \
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"rate = (int) [ 1, MAX ], " \
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"channels = (int) [ 1, 8 ], " \
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"endianness = (int) { LITTLE_ENDIAN, BIG_ENDIAN }, " \
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"width = (int) 32, " \
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"depth = (int) [ 1, 32 ], " \
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"signed = (boolean) { true, false }; " \
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"audio/x-raw-int, " \
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"rate = (int) [ 1, MAX ], " \
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"channels = (int) [ 1, 8 ], " \
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"endianness = (int) { LITTLE_ENDIAN, BIG_ENDIAN }, " \
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"width = (int) 24, " \
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"depth = (int) [ 1, 24 ], " \
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"signed = (boolean) { true, false }; " \
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"audio/x-raw-int, " \
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"rate = (int) [ 1, MAX ], " \
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"channels = (int) [ 1, 8 ], " \
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"endianness = (int) { LITTLE_ENDIAN, BIG_ENDIAN }, " \
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"width = (int) 16, " \
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"depth = (int) [ 1, 16 ], " \
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"signed = (boolean) { true, false }; " \
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"audio/x-raw-int, " \
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"rate = (int) [ 1, MAX ], " \
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"channels = (int) [ 1, 8 ], " \
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"endianness = (int) { LITTLE_ENDIAN, BIG_ENDIAN }, " \
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"width = (int) 8, " \
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"depth = (int) [ 1, 8 ], " \
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"signed = (boolean) { true, false } "
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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 (CONVERT_CAPS_TEMPLATE_STRING)
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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 (CONVERT_CAPS_TEMPLATE_STRING)
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);
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/* takes over reference for outcaps */
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GstElement *
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setup_audioconvert (GstCaps * outcaps)
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{
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GstElement *audioconvert;
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GST_DEBUG ("setup_audioconvert");
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audioconvert = gst_check_setup_element ("audioconvert");
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mysrcpad = gst_check_setup_src_pad (audioconvert, &srctemplate, NULL);
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mysinkpad = gst_check_setup_sink_pad (audioconvert, &sinktemplate, NULL);
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/* this installs a getcaps func that will always return the caps we set
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* later */
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gst_pad_use_fixed_caps (mysinkpad);
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gst_pad_set_caps (mysinkpad, outcaps);
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gst_caps_unref (outcaps);
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outcaps = gst_pad_get_negotiated_caps (mysinkpad);
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fail_unless (gst_caps_is_fixed (outcaps));
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gst_caps_unref (outcaps);
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return audioconvert;
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}
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void
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cleanup_audioconvert (GstElement * audioconvert)
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{
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GST_DEBUG ("cleanup_audioconvert");
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gst_check_teardown_src_pad (audioconvert);
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gst_check_teardown_sink_pad (audioconvert);
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gst_check_teardown_element (audioconvert);
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}
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GstCaps *
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get_int_caps (guint rate, guint channels, gchar * endianness, guint width,
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guint depth, gboolean signedness)
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{
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GstCaps *caps;
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gchar *string;
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string = g_strdup_printf ("audio/x-raw-int, "
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"rate = (int) %d, "
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"channels = (int) %d, "
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"endianness = (int) %s, "
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"width = (int) %d, "
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"depth = (int) %d, "
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"signed = (boolean) %s ",
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rate, channels, endianness, width, depth, signedness ? "true" : "false");
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GST_DEBUG ("creating caps from %s", string);
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caps = gst_caps_from_string (string);
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fail_unless (caps != NULL);
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g_free (string);
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return caps;
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}
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static void
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verify_convert (GstElement * audioconvert, void *in, int inlength, void *out,
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int outlength, GstCaps * incaps)
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{
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GstBuffer *inbuffer, *outbuffer;
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fail_unless (gst_element_set_state (audioconvert,
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GST_STATE_PLAYING) == GST_STATE_SUCCESS, "could not set to playing");
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GST_DEBUG ("Creating buffer of %d bytes", inlength);
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inbuffer = gst_buffer_new_and_alloc (inlength);
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memcpy (GST_BUFFER_DATA (inbuffer), in, inlength);
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gst_buffer_set_caps (inbuffer, incaps);
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ASSERT_BUFFER_REFCOUNT (inbuffer, "inbuffer", 1);
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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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/* ... 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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ASSERT_BUFFER_REFCOUNT (outbuffer, "outbuffer", 1);
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fail_unless_equals_int (GST_BUFFER_SIZE (outbuffer), outlength);
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fail_unless (memcmp (GST_BUFFER_DATA (outbuffer), out, outlength) == 0);
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}
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GST_START_TEST (test_unity)
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{
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GstElement *audioconvert;
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GstCaps *incaps, *outcaps;
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gint16 in[] = { 16384, -256 };
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gint16 out[] = { 8064 };
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outcaps = get_int_caps (44100, 1, "LITTLE_ENDIAN", 16, 16, TRUE);
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audioconvert = setup_audioconvert (outcaps);
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incaps = get_int_caps (44100, 2, "LITTLE_ENDIAN", 16, 16, TRUE);
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verify_convert (audioconvert, in, sizeof (in), out, sizeof (out), incaps);
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/* cleanup */
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cleanup_audioconvert (audioconvert);
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}
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GST_END_TEST;
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Suite *
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audioconvert_suite (void)
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{
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Suite *s = suite_create ("audioconvert");
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TCase *tc_chain = tcase_create ("general");
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suite_add_tcase (s, tc_chain);
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tcase_add_test (tc_chain, test_unity);
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return s;
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}
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int
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main (int argc, char **argv)
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{
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int nf;
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Suite *s = audioconvert_suite ();
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SRunner *sr = srunner_create (s);
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gst_check_init (&argc, &argv);
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srunner_run_all (sr, CK_NORMAL);
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nf = srunner_ntests_failed (sr);
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srunner_free (sr);
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return nf;
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}
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@ -395,14 +395,20 @@ static GstFlowReturn
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GST_LOG_OBJECT (audioresample, "resample gave me %d bytes or %d samples",
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outsize, outsamples);
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GST_BUFFER_OFFSET (outbuf) = audioresample->offset;
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GST_BUFFER_TIMESTAMP (outbuf) =
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audioresample->offset * GST_SECOND / audioresample->o_rate;
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GST_BUFFER_DURATION (outbuf) =
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outsamples * GST_SECOND / audioresample->o_rate;
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GST_BUFFER_OFFSET (outbuf) = audioresample->offset;
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audioresample->offset += outsamples;
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GST_BUFFER_OFFSET_END (outbuf) = audioresample->offset;
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/* we calculate DURATION as the difference between "next" timestamp
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* and current timestamp so we ensure a contiguous stream, instead of
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* having rounding errors. */
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GST_BUFFER_DURATION (outbuf) =
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audioresample->offset * GST_SECOND / audioresample->o_rate -
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GST_BUFFER_TIMESTAMP (outbuf);
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/* check for possible mem corruption */
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if (outsize > GST_BUFFER_SIZE (outbuf)) {
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/* this is an error that when it happens, would need fixing in the
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@ -52,7 +52,7 @@ struct _GstAudioresample {
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gboolean passthru;
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gint64 offset;
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guint64 offset;
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int channels;
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int i_rate;
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@ -29,13 +29,14 @@ check_vorbis =
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endif
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check_PROGRAMS = \
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elements/audioconvert \
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elements/audioresample \
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elements/volume \
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$(check_vorbis)
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# these tests don't even pass
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# generic/states: elements need state fixin' before this can be added
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noinst_PROGRAMS = \
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elements/audioresample \
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generic/states
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AM_CFLAGS = $(GST_OBJ_CFLAGS) $(GST_CHECK_CFLAGS) $(CHECK_CFLAGS)
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209
tests/check/elements/audioconvert.c
Normal file
209
tests/check/elements/audioconvert.c
Normal file
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@ -0,0 +1,209 @@
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/* GStreamer
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*
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* unit test for audioconvert
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*
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* Copyright (C) <2005> Thomas Vander Stichele <thomas at apestaart dot org>
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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 Library General Public
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* License as published by the Free Software Foundation; either
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* version 2 of 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,
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* but 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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* Library General Public License for more details.
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*
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* You should have received a copy of the GNU Library General Public
|
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* License along with this library; if not, write to the
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* Free Software Foundation, Inc., 59 Temple Place - Suite 330,
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* Boston, MA 02111-1307, USA.
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*/
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#include <unistd.h>
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#include <gst/check/gstcheck.h>
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GList *buffers = NULL;
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gboolean have_eos = FALSE;
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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 CONVERT_CAPS_TEMPLATE_STRING \
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"audio/x-raw-float, " \
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"rate = (int) [ 1, MAX ], " \
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"channels = (int) [ 1, 8 ], " \
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"endianness = (int) BYTE_ORDER, " \
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"width = (int) 32, " \
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"buffer-frames = (int) [ 0, MAX ];" \
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"audio/x-raw-int, " \
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"rate = (int) [ 1, MAX ], " \
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"channels = (int) [ 1, 8 ], " \
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"endianness = (int) { LITTLE_ENDIAN, BIG_ENDIAN }, " \
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"width = (int) 32, " \
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"depth = (int) [ 1, 32 ], " \
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"signed = (boolean) { true, false }; " \
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"audio/x-raw-int, " \
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"rate = (int) [ 1, MAX ], " \
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"channels = (int) [ 1, 8 ], " \
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"endianness = (int) { LITTLE_ENDIAN, BIG_ENDIAN }, " \
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"width = (int) 24, " \
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"depth = (int) [ 1, 24 ], " \
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"signed = (boolean) { true, false }; " \
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"audio/x-raw-int, " \
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"rate = (int) [ 1, MAX ], " \
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"channels = (int) [ 1, 8 ], " \
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"endianness = (int) { LITTLE_ENDIAN, BIG_ENDIAN }, " \
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"width = (int) 16, " \
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"depth = (int) [ 1, 16 ], " \
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"signed = (boolean) { true, false }; " \
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"audio/x-raw-int, " \
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"rate = (int) [ 1, MAX ], " \
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"channels = (int) [ 1, 8 ], " \
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"endianness = (int) { LITTLE_ENDIAN, BIG_ENDIAN }, " \
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"width = (int) 8, " \
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"depth = (int) [ 1, 8 ], " \
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"signed = (boolean) { true, false } "
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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 (CONVERT_CAPS_TEMPLATE_STRING)
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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 (CONVERT_CAPS_TEMPLATE_STRING)
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);
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/* takes over reference for outcaps */
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GstElement *
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setup_audioconvert (GstCaps * outcaps)
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{
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GstElement *audioconvert;
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GST_DEBUG ("setup_audioconvert");
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audioconvert = gst_check_setup_element ("audioconvert");
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mysrcpad = gst_check_setup_src_pad (audioconvert, &srctemplate, NULL);
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mysinkpad = gst_check_setup_sink_pad (audioconvert, &sinktemplate, NULL);
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/* this installs a getcaps func that will always return the caps we set
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* later */
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gst_pad_use_fixed_caps (mysinkpad);
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gst_pad_set_caps (mysinkpad, outcaps);
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gst_caps_unref (outcaps);
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outcaps = gst_pad_get_negotiated_caps (mysinkpad);
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fail_unless (gst_caps_is_fixed (outcaps));
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gst_caps_unref (outcaps);
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return audioconvert;
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}
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void
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cleanup_audioconvert (GstElement * audioconvert)
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{
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GST_DEBUG ("cleanup_audioconvert");
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gst_check_teardown_src_pad (audioconvert);
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gst_check_teardown_sink_pad (audioconvert);
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gst_check_teardown_element (audioconvert);
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}
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GstCaps *
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get_int_caps (guint rate, guint channels, gchar * endianness, guint width,
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guint depth, gboolean signedness)
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{
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GstCaps *caps;
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gchar *string;
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string = g_strdup_printf ("audio/x-raw-int, "
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"rate = (int) %d, "
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"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;
|
||||
}
|
||||
Loading…
Reference in a new issue