add a check for audioresample

Original commit message from CVS:
add a check for audioresample
parent dcf4df2a
......@@ -353,6 +353,7 @@ static GstFlowReturn
guchar *data;
gulong size;
int outsize;
int outsamples;
/* FIXME: move to _inplace */
#if 0
......@@ -390,10 +391,17 @@ static GstFlowReturn
}
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)) {
......
......@@ -111,6 +111,9 @@ resample_scale_ref (ResampleState * r)
-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;
}
......
/* 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;
}
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