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* Add caps change test to unit tests * Cleanup leftover buffers after each unit test * Add missing rawvideoparse entry in .gitignore https://bugzilla.gnome.org/show_bug.cgi?id=769637
395 lines
14 KiB
C
395 lines
14 KiB
C
/* GStreamer
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*
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* unit test for rawaudioparse
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*
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* Copyright (C) <2016> Carlos Rafael Giani <dv at pseudoterminal 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., 51 Franklin St, Fifth Floor,
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* Boston, MA 02110-1301, USA.
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*/
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/* FIXME: GValueArray is deprecated, but there is currently no viabla alternative
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* See https://bugzilla.gnome.org/show_bug.cgi?id=667228 */
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#define GLIB_DISABLE_DEPRECATION_WARNINGS
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#include <gst/check/gstcheck.h>
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#include <gst/audio/audio.h>
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/* Checks are hardcoded to expect stereo 16-bit data. The sample rate
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* however varies from the default of 40 kHz in some tests to see the
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* differences in calculated buffer durations. */
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#define NUM_TEST_SAMPLES 512
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#define NUM_TEST_CHANNELS 2
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#define TEST_SAMPLE_RATE 40000
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#define TEST_SAMPLE_FORMAT GST_AUDIO_FORMAT_S16
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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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static GstPad *mysrcpad, *mysinkpad;
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typedef struct
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{
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GstElement *rawaudioparse;
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GstAdapter *test_data_adapter;
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}
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RawAudParseTestCtx;
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/* Sets up a rawaudioparse element and a GstAdapter that contains 512 test
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* audio samples. The samples a monotonically increasing set from the values
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* 0 to 511 for the left and 512 to 1023 for the right channel. The result
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* is a GstAdapter that contains the interleaved 16-bit integer values:
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* 0,512,1,513,2,514, ... 511,1023 . This set is used in the checks to see
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* if rawaudioparse's output buffers contain valid data. */
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static void
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setup_rawaudioparse (RawAudParseTestCtx * testctx, gboolean use_sink_caps,
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gboolean set_properties, GstCaps * incaps, GstFormat format)
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{
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GstElement *rawaudioparse;
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GstAdapter *test_data_adapter;
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GstBuffer *buffer;
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guint i;
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guint16 samples[NUM_TEST_SAMPLES * NUM_TEST_CHANNELS];
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/* Setup the rawaudioparse element and the pads */
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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 (GST_AUDIO_CAPS_MAKE (GST_AUDIO_FORMATS_ALL))
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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_ANY);
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rawaudioparse = gst_check_setup_element ("rawaudioparse");
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g_object_set (G_OBJECT (rawaudioparse), "use-sink-caps", use_sink_caps, NULL);
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if (set_properties)
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g_object_set (G_OBJECT (rawaudioparse), "sample-rate", TEST_SAMPLE_RATE,
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"num-channels", NUM_TEST_CHANNELS, "pcm-format", TEST_SAMPLE_FORMAT,
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NULL);
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fail_unless (gst_element_set_state (rawaudioparse,
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GST_STATE_PAUSED) == GST_STATE_CHANGE_SUCCESS,
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"could not set to paused");
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mysrcpad = gst_check_setup_src_pad (rawaudioparse, &srctemplate);
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mysinkpad = gst_check_setup_sink_pad (rawaudioparse, &sinktemplate);
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gst_pad_set_active (mysrcpad, TRUE);
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gst_pad_set_active (mysinkpad, TRUE);
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gst_check_setup_events (mysrcpad, rawaudioparse, incaps, format);
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if (incaps)
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gst_caps_unref (incaps);
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/* Fill the adapter with the interleaved 0..511 and
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* 512..1023 samples */
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for (i = 0; i < NUM_TEST_SAMPLES; ++i) {
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guint c;
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for (c = 0; c < NUM_TEST_CHANNELS; ++c)
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samples[i * NUM_TEST_CHANNELS + c] = c * NUM_TEST_SAMPLES + i;
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}
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test_data_adapter = gst_adapter_new ();
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buffer = gst_buffer_new_allocate (NULL, sizeof (samples), NULL);
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gst_buffer_fill (buffer, 0, samples, sizeof (samples));
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gst_adapter_push (test_data_adapter, buffer);
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testctx->rawaudioparse = rawaudioparse;
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testctx->test_data_adapter = test_data_adapter;
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}
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static void
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cleanup_rawaudioparse (RawAudParseTestCtx * testctx)
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{
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int num_buffers, i;
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gst_pad_set_active (mysrcpad, FALSE);
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gst_pad_set_active (mysinkpad, FALSE);
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gst_check_teardown_src_pad (testctx->rawaudioparse);
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gst_check_teardown_sink_pad (testctx->rawaudioparse);
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gst_check_teardown_element (testctx->rawaudioparse);
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g_object_unref (G_OBJECT (testctx->test_data_adapter));
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if (buffers != NULL) {
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num_buffers = g_list_length (buffers);
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for (i = 0; i < num_buffers; ++i) {
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GstBuffer *buf = GST_BUFFER (buffers->data);
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buffers = g_list_remove (buffers, buf);
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gst_buffer_unref (buf);
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}
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g_list_free (buffers);
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buffers = NULL;
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}
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}
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static void
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push_data_and_check_output (RawAudParseTestCtx * testctx, gsize num_in_bytes,
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gsize expected_num_out_bytes, gint64 expected_pts, gint64 expected_dur,
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guint expected_num_buffers_in_list, guint bpf, guint16 channel0_start,
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guint16 channel1_start)
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{
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GstBuffer *inbuf, *outbuf;
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guint num_buffers;
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/* Simulate upstream input by taking num_in_bytes bytes from the adapter */
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inbuf = gst_adapter_take_buffer (testctx->test_data_adapter, num_in_bytes);
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fail_unless (inbuf != NULL);
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/* Push the input data and check that the output buffers list grew as
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* expected */
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fail_unless (gst_pad_push (mysrcpad, inbuf) == GST_FLOW_OK);
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num_buffers = g_list_length (buffers);
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fail_unless_equals_int (num_buffers, expected_num_buffers_in_list);
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/* Take the latest output buffer */
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outbuf = g_list_nth_data (buffers, num_buffers - 1);
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fail_unless (outbuf != NULL);
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/* Verify size, PTS, duration of the output buffer */
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fail_unless_equals_uint64 (expected_num_out_bytes,
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gst_buffer_get_size (outbuf));
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fail_unless_equals_uint64 (expected_pts, GST_BUFFER_PTS (outbuf));
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fail_unless_equals_uint64 (expected_dur, GST_BUFFER_DURATION (outbuf));
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/* Go through all of the samples in the output buffer and check that they are
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* valid. The samples are interleaved. The offsets specified by channel0_start
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* and channel1_start are the expected values of the first sample for each
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* channel in the buffer. So, if channel0_start is 512, then sample #0 in the
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* buffer must have value 512, and if channel1_start is 700, then sample #1
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* in the buffer must have value 700 etc. */
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{
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guint i, num_frames;
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guint16 *s;
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GstMapInfo map_info;
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guint channel_starts[2] = { channel0_start, channel1_start };
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gst_buffer_map (outbuf, &map_info, GST_MAP_READ);
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num_frames = map_info.size / bpf;
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s = (guint16 *) (map_info.data);
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for (i = 0; i < num_frames; ++i) {
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guint c;
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for (c = 0; i < NUM_TEST_CHANNELS; ++i) {
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guint16 expected = channel_starts[c] + i;
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guint16 actual = s[i * NUM_TEST_CHANNELS + c];
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fail_unless_equals_int (expected, actual);
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}
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}
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gst_buffer_unmap (outbuf, &map_info);
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}
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}
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GST_START_TEST (test_push_unaligned_data_properties_config)
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{
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RawAudParseTestCtx testctx;
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setup_rawaudioparse (&testctx, FALSE, TRUE, NULL, GST_FORMAT_BYTES);
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/* Send in data buffers that are not aligned to multiples of the
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* frame size (= sample size * num_channels). This tests if rawaudioparse
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* aligns output data properly.
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*
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* The second line sends in 99 bytes, and expects 100 bytes in the
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* output buffer. This is because the first buffer contains 45 bytes,
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* and rawaudioparse is expected to output 44 bytes (which is an integer
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* multiple of the frame size). The leftover 1 byte then gets prepended
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* to the input buffer with 99 bytes, resulting in 100 bytes, which is
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* an integer multiple of the frame size.
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*/
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push_data_and_check_output (&testctx, 45, 44, GST_USECOND * 0,
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GST_USECOND * 275, 1, 4, 0, 512);
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push_data_and_check_output (&testctx, 99, 100, GST_USECOND * 275,
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GST_USECOND * 625, 2, 4, 11, 523);
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push_data_and_check_output (&testctx, 18, 16, GST_USECOND * 900,
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GST_USECOND * 100, 3, 4, 36, 548);
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cleanup_rawaudioparse (&testctx);
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}
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GST_END_TEST;
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GST_START_TEST (test_push_unaligned_data_sink_caps_config)
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{
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RawAudParseTestCtx testctx;
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GstAudioInfo ainfo;
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GstCaps *caps;
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/* This test is essentially the same as test_push_unaligned_data_properties_config,
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* except that rawaudioparse uses the sink caps config instead of the property config. */
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gst_audio_info_set_format (&ainfo, TEST_SAMPLE_FORMAT, TEST_SAMPLE_RATE,
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NUM_TEST_CHANNELS, NULL);
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caps = gst_audio_info_to_caps (&ainfo);
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setup_rawaudioparse (&testctx, TRUE, FALSE, caps, GST_FORMAT_BYTES);
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push_data_and_check_output (&testctx, 45, 44, GST_USECOND * 0,
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GST_USECOND * 275, 1, 4, 0, 512);
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push_data_and_check_output (&testctx, 99, 100, GST_USECOND * 275,
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GST_USECOND * 625, 2, 4, 11, 523);
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push_data_and_check_output (&testctx, 18, 16, GST_USECOND * 900,
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GST_USECOND * 100, 3, 4, 36, 548);
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cleanup_rawaudioparse (&testctx);
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}
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GST_END_TEST;
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GST_START_TEST (test_push_swapped_channels)
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{
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RawAudParseTestCtx testctx;
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GValueArray *valarray;
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GValue val = G_VALUE_INIT;
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/* Send in 40 bytes and use a nonstandard channel order (left and right channels
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* swapped). Expected behavior is for rawaudioparse to reorder the samples inside
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* output buffers to conform to the GStreamer channel order. For this reason,
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* channel0 offset is 512 and channel1 offset is 0 in the check below. */
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setup_rawaudioparse (&testctx, FALSE, TRUE, NULL, GST_FORMAT_BYTES);
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valarray = g_value_array_new (2);
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g_value_init (&val, GST_TYPE_AUDIO_CHANNEL_POSITION);
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g_value_set_enum (&val, GST_AUDIO_CHANNEL_POSITION_FRONT_RIGHT);
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g_value_array_insert (valarray, 0, &val);
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g_value_set_enum (&val, GST_AUDIO_CHANNEL_POSITION_FRONT_LEFT);
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g_value_array_insert (valarray, 1, &val);
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g_object_set (G_OBJECT (testctx.rawaudioparse), "channel-positions",
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valarray, NULL);
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g_value_array_free (valarray);
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g_value_unset (&val);
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push_data_and_check_output (&testctx, 40, 40, GST_USECOND * 0,
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GST_USECOND * 250, 1, 4, 512, 0);
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cleanup_rawaudioparse (&testctx);
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}
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GST_END_TEST;
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GST_START_TEST (test_config_switch)
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{
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RawAudParseTestCtx testctx;
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GstAudioInfo ainfo;
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GstCaps *caps;
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/* Start processing with the properties config active, then mid-stream switch to
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* the sink caps config. The properties config is altered to have a different
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* sample rate than the sink caps to be able to detect the switch. The net effect
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* is that output buffer durations are altered. For example, 40 bytes equal
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* 10 samples, and this equals 500 us with 20 kHz or 250 us with 40 kHz. */
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gst_audio_info_set_format (&ainfo, TEST_SAMPLE_FORMAT, TEST_SAMPLE_RATE,
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NUM_TEST_CHANNELS, NULL);
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caps = gst_audio_info_to_caps (&ainfo);
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setup_rawaudioparse (&testctx, FALSE, TRUE, caps, GST_FORMAT_BYTES);
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g_object_set (G_OBJECT (testctx.rawaudioparse), "sample-rate", 20000, NULL);
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/* Push in data with properties config active, expecting duration calculations
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* to be based on the 20 kHz sample rate */
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push_data_and_check_output (&testctx, 40, 40, GST_USECOND * 0,
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GST_USECOND * 500, 1, 4, 0, 512);
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push_data_and_check_output (&testctx, 20, 20, GST_USECOND * 500,
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GST_USECOND * 250, 2, 4, 10, 522);
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/* Perform the switch */
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g_object_set (G_OBJECT (testctx.rawaudioparse), "use-sink-caps", TRUE, NULL);
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/* Push in data with sink caps config active, expecting duration calculations
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* to be based on the 40 kHz sample rate */
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push_data_and_check_output (&testctx, 40, 40, GST_USECOND * 750,
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GST_USECOND * 250, 3, 4, 15, 527);
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cleanup_rawaudioparse (&testctx);
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}
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GST_END_TEST;
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GST_START_TEST (test_change_caps)
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{
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RawAudParseTestCtx testctx;
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GstAudioInfo ainfo;
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GstCaps *caps;
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/* Start processing with the sink caps config active, using the
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* default channel count and sample format and 20 kHz sample rate
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* for the caps. Push some data, then change caps (20 kHz -> 40 kHz).
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* Check that the changed caps are handled properly. */
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gst_audio_info_set_format (&ainfo, TEST_SAMPLE_FORMAT, 20000,
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NUM_TEST_CHANNELS, NULL);
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caps = gst_audio_info_to_caps (&ainfo);
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setup_rawaudioparse (&testctx, TRUE, FALSE, caps, GST_FORMAT_BYTES);
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/* Push in data with caps sink config active, expecting duration calculations
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* to be based on the 20 kHz sample rate */
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push_data_and_check_output (&testctx, 40, 40, GST_USECOND * 0,
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GST_USECOND * 500, 1, 4, 0, 512);
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push_data_and_check_output (&testctx, 20, 20, GST_USECOND * 500,
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GST_USECOND * 250, 2, 4, 10, 522);
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/* Change caps */
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gst_audio_info_set_format (&ainfo, TEST_SAMPLE_FORMAT, 40000,
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NUM_TEST_CHANNELS, NULL);
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caps = gst_audio_info_to_caps (&ainfo);
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fail_unless (gst_pad_push_event (mysrcpad, gst_event_new_caps (caps)));
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gst_caps_unref (caps);
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/* Push in data with the new caps, expecting duration calculations
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* to be based on the 40 kHz sample rate */
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push_data_and_check_output (&testctx, 40, 40, GST_USECOND * 750,
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GST_USECOND * 250, 3, 4, 15, 527);
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cleanup_rawaudioparse (&testctx);
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}
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GST_END_TEST;
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static Suite *
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rawaudioparse_suite (void)
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{
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Suite *s = suite_create ("rawaudioparse");
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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_push_unaligned_data_properties_config);
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tcase_add_test (tc_chain, test_push_unaligned_data_sink_caps_config);
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tcase_add_test (tc_chain, test_push_swapped_channels);
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tcase_add_test (tc_chain, test_config_switch);
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tcase_add_test (tc_chain, test_change_caps);
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return s;
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}
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GST_CHECK_MAIN (rawaudioparse);
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