add a check for audioresample

Original commit message from CVS:
add a check for audioresample

ChangeLog

@@ -1,3 +1,18 @@
+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:

check/Makefile.am

@@ -35,6 +35,7 @@ check_PROGRAMS = \
 # 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)

check/elements/audioresample.c

@@ -0,0 +1,227 @@
+/* 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;
+}

check/elements/volume.c

@@ -72,15 +72,6 @@ static GstStaticPadTemplate srctemplate = GST_STATIC_PAD_TEMPLATE ("src",
     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 ()
 {

gst/audioresample/gstaudioresample.c

@@ -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)) {

gst/audioresample/resample_ref.c

@@ -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;
       }

tests/check/Makefile.am

@@ -35,6 +35,7 @@ check_PROGRAMS = \
 # 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)

tests/check/elements/audioresample.c

@@ -0,0 +1,227 @@
+/* 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;
+}

tests/check/elements/volume.c

@@ -72,15 +72,6 @@ static GstStaticPadTemplate srctemplate = GST_STATIC_PAD_TEMPLATE ("src",
     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 ()
 {