gstreamer/tests/check/libs/audiodecoder.c

708 lines
19 KiB
C

/* GStreamer
*
* Copyright (C) 2014 Samsung Electronics. All rights reserved.
* Author: Thiago Santos <ts.santos@sisa.samsung.com>
*
* 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., 51 Franklin St, Fifth Floor,
* Boston, MA 02110-1301, USA.
*/
#ifdef HAVE_CONFIG_H
#include "config.h"
#endif
#include <gst/gst.h>
#include <gst/check/gstcheck.h>
#include <gst/audio/audio.h>
#include <gst/app/app.h>
static GstPad *mysrcpad, *mysinkpad;
static GstElement *dec;
static GList *events = NULL;
#define TEST_MSECS_PER_SAMPLE 44100
#define GST_AUDIO_DECODER_TESTER_TYPE gst_audio_decoder_tester_get_type()
static GType gst_audio_decoder_tester_get_type (void);
typedef struct _GstAudioDecoderTester GstAudioDecoderTester;
typedef struct _GstAudioDecoderTesterClass GstAudioDecoderTesterClass;
struct _GstAudioDecoderTester
{
GstAudioDecoder parent;
gboolean setoutputformat_on_decoding;
gboolean output_too_many_frames;
};
struct _GstAudioDecoderTesterClass
{
GstAudioDecoderClass parent_class;
};
G_DEFINE_TYPE (GstAudioDecoderTester, gst_audio_decoder_tester,
GST_TYPE_AUDIO_DECODER);
static gboolean
gst_audio_decoder_tester_start (GstAudioDecoder * dec)
{
return TRUE;
}
static gboolean
gst_audio_decoder_tester_stop (GstAudioDecoder * dec)
{
return TRUE;
}
static void
gst_audio_decoder_tester_flush (GstAudioDecoder * dec, gboolean hard)
{
}
static gboolean
gst_audio_decoder_tester_set_format (GstAudioDecoder * dec, GstCaps * caps)
{
GstAudioDecoderTester *tester = (GstAudioDecoderTester *) dec;
GstAudioInfo info;
gst_caps_unref (caps);
if (!tester->setoutputformat_on_decoding) {
caps = gst_caps_new_simple ("audio/x-raw", "format", G_TYPE_STRING, "S32LE",
"channels", G_TYPE_INT, 2, "rate", G_TYPE_INT, 44100,
"layout", G_TYPE_STRING, "interleaved", NULL);
gst_audio_info_from_caps (&info, caps);
gst_caps_unref (caps);
gst_audio_decoder_set_output_format (dec, &info);
}
return TRUE;
}
static GstFlowReturn
gst_audio_decoder_tester_handle_frame (GstAudioDecoder * dec,
GstBuffer * buffer)
{
GstAudioDecoderTester *tester = (GstAudioDecoderTester *) dec;
guint8 *data;
gint size;
GstMapInfo map;
GstBuffer *output_buffer;
if (buffer == NULL)
return GST_FLOW_OK;
if (tester->setoutputformat_on_decoding) {
GstCaps *caps;
GstAudioInfo info;
caps = gst_caps_new_simple ("audio/x-raw", "format", G_TYPE_STRING, "S32LE",
"channels", G_TYPE_INT, 2, "rate", G_TYPE_INT, 44100,
"layout", G_TYPE_STRING, "interleaved", NULL);
gst_audio_info_from_caps (&info, caps);
gst_caps_unref (caps);
gst_audio_decoder_set_output_format (dec, &info);
}
gst_buffer_map (buffer, &map, GST_MAP_READ);
/* the output is SE32LE stereo 44100 Hz */
size = 2 * 4;
g_assert (size == sizeof (guint64));
data = g_malloc0 (size);
memcpy (data, map.data, sizeof (guint64));
output_buffer = gst_buffer_new_wrapped (data, size);
gst_buffer_unmap (buffer, &map);
if (tester->output_too_many_frames) {
return gst_audio_decoder_finish_frame (dec, output_buffer, 2);
} else {
return gst_audio_decoder_finish_frame (dec, output_buffer, 1);
}
}
static void
gst_audio_decoder_tester_class_init (GstAudioDecoderTesterClass * klass)
{
GstElementClass *element_class = GST_ELEMENT_CLASS (klass);
GstAudioDecoderClass *audiosink_class = GST_AUDIO_DECODER_CLASS (klass);
static GstStaticPadTemplate sink_templ = GST_STATIC_PAD_TEMPLATE ("sink",
GST_PAD_SINK, GST_PAD_ALWAYS,
GST_STATIC_CAPS ("audio/x-test-custom"));
static GstStaticPadTemplate src_templ = GST_STATIC_PAD_TEMPLATE ("src",
GST_PAD_SRC, GST_PAD_ALWAYS,
GST_STATIC_CAPS ("audio/x-raw"));
gst_element_class_add_pad_template (element_class,
gst_static_pad_template_get (&sink_templ));
gst_element_class_add_pad_template (element_class,
gst_static_pad_template_get (&src_templ));
gst_element_class_set_metadata (element_class,
"AudioDecoderTester", "Decoder/Audio", "yep", "me");
audiosink_class->start = gst_audio_decoder_tester_start;
audiosink_class->stop = gst_audio_decoder_tester_stop;
audiosink_class->flush = gst_audio_decoder_tester_flush;
audiosink_class->handle_frame = gst_audio_decoder_tester_handle_frame;
audiosink_class->set_format = gst_audio_decoder_tester_set_format;
}
static void
gst_audio_decoder_tester_init (GstAudioDecoderTester * tester)
{
}
static gboolean
_mysinkpad_event (GstPad * pad, GstObject * parent, GstEvent * event)
{
events = g_list_append (events, event);
return TRUE;
}
static void
setup_audiodecodertester (void)
{
static GstStaticPadTemplate sinktemplate = GST_STATIC_PAD_TEMPLATE ("sink",
GST_PAD_SINK,
GST_PAD_ALWAYS,
GST_STATIC_CAPS ("audio/x-raw, format=(string)S32LE, "
"rate=(int)[1, 320000], channels=(int)[1, 32],"
"layout=(string)interleaved")
);
static GstStaticPadTemplate srctemplate = GST_STATIC_PAD_TEMPLATE ("src",
GST_PAD_SRC,
GST_PAD_ALWAYS,
GST_STATIC_CAPS ("audio/x-test-custom")
);
dec = g_object_new (GST_AUDIO_DECODER_TESTER_TYPE, NULL);
mysrcpad = gst_check_setup_src_pad (dec, &srctemplate);
mysinkpad = gst_check_setup_sink_pad (dec, &sinktemplate);
gst_pad_set_event_function (mysinkpad, _mysinkpad_event);
}
static void
cleanup_audiodecodertest (void)
{
gst_pad_set_active (mysrcpad, FALSE);
gst_pad_set_active (mysinkpad, FALSE);
gst_check_teardown_src_pad (dec);
gst_check_teardown_sink_pad (dec);
gst_check_teardown_element (dec);
}
static GstBuffer *
create_test_buffer (guint64 num)
{
GstBuffer *buffer;
guint64 *data = g_malloc (sizeof (guint64));
*data = num;
buffer = gst_buffer_new_wrapped (data, sizeof (guint64));
GST_BUFFER_PTS (buffer) =
gst_util_uint64_scale_round (num, GST_SECOND, TEST_MSECS_PER_SAMPLE);
GST_BUFFER_DURATION (buffer) =
gst_util_uint64_scale_round (1, GST_SECOND, TEST_MSECS_PER_SAMPLE);
return buffer;
}
static void
send_startup_events (void)
{
GstCaps *caps;
fail_unless (gst_pad_push_event (mysrcpad,
gst_event_new_stream_start ("randomvalue")));
/* push caps */
caps =
gst_caps_new_simple ("audio/x-test-custom", "channels", G_TYPE_INT, 2,
"rate", G_TYPE_INT, 44100, NULL);
fail_unless (gst_pad_push_event (mysrcpad, gst_event_new_caps (caps)));
}
#define NUM_BUFFERS 1000
GST_START_TEST (audiodecoder_playback)
{
GstSegment segment;
GstBuffer *buffer;
guint64 i;
setup_audiodecodertester ();
gst_pad_set_active (mysrcpad, TRUE);
gst_element_set_state (dec, GST_STATE_PLAYING);
gst_pad_set_active (mysinkpad, TRUE);
send_startup_events ();
/* push a new segment */
gst_segment_init (&segment, GST_FORMAT_TIME);
fail_unless (gst_pad_push_event (mysrcpad, gst_event_new_segment (&segment)));
/* push buffers, the data is actually a number so we can track them */
for (i = 0; i < NUM_BUFFERS; i++) {
GstMapInfo map;
guint64 num;
buffer = create_test_buffer (i);
fail_unless (gst_pad_push (mysrcpad, buffer) == GST_FLOW_OK);
/* check that buffer was received by our source pad */
buffer = buffers->data;
gst_buffer_map (buffer, &map, GST_MAP_READ);
num = *(guint64 *) map.data;
fail_unless_equals_uint64 (i, num);
fail_unless_equals_uint64 (GST_BUFFER_PTS (buffer),
gst_util_uint64_scale_round (i, GST_SECOND, TEST_MSECS_PER_SAMPLE));
fail_unless_equals_uint64 (GST_BUFFER_DURATION (buffer),
gst_util_uint64_scale_round (1, GST_SECOND, TEST_MSECS_PER_SAMPLE));
gst_buffer_unmap (buffer, &map);
gst_buffer_unref (buffer);
buffers = g_list_delete_link (buffers, buffers);
}
fail_unless (gst_pad_push_event (mysrcpad, gst_event_new_eos ()));
fail_unless (buffers == NULL);
cleanup_audiodecodertest ();
}
GST_END_TEST;
static void
check_audiodecoder_negotiation (void)
{
gboolean received_caps = FALSE;
GList *iter;
for (iter = events; iter; iter = g_list_next (iter)) {
GstEvent *event = iter->data;
if (GST_EVENT_TYPE (event) == GST_EVENT_CAPS) {
GstCaps *caps;
GstStructure *structure;
gint channels;
gint rate;
gst_event_parse_caps (event, &caps);
structure = gst_caps_get_structure (caps, 0);
fail_unless (gst_structure_get_int (structure, "rate", &rate));
fail_unless (gst_structure_get_int (structure, "channels", &channels));
fail_unless (rate == 44100, "%d != %d", rate, 44100);
fail_unless (channels == 2, "%d != %d", channels, 2);
received_caps = TRUE;
break;
}
}
fail_unless (received_caps);
}
GST_START_TEST (audiodecoder_negotiation_with_buffer)
{
GstSegment segment;
GstBuffer *buffer;
setup_audiodecodertester ();
gst_pad_set_active (mysrcpad, TRUE);
gst_element_set_state (dec, GST_STATE_PLAYING);
gst_pad_set_active (mysinkpad, TRUE);
send_startup_events ();
/* push a new segment */
gst_segment_init (&segment, GST_FORMAT_TIME);
fail_unless (gst_pad_push_event (mysrcpad, gst_event_new_segment (&segment)));
/* push a buffer event to force audiodecoder to push a caps event */
buffer = create_test_buffer (0);
fail_unless (gst_pad_push (mysrcpad, buffer) == GST_FLOW_OK);
check_audiodecoder_negotiation ();
cleanup_audiodecodertest ();
g_list_free_full (buffers, (GDestroyNotify) gst_buffer_unref);
}
GST_END_TEST;
GST_START_TEST (audiodecoder_negotiation_with_gap_event)
{
GstSegment segment;
setup_audiodecodertester ();
gst_pad_set_active (mysrcpad, TRUE);
gst_element_set_state (dec, GST_STATE_PLAYING);
gst_pad_set_active (mysinkpad, TRUE);
send_startup_events ();
/* push a new segment */
gst_segment_init (&segment, GST_FORMAT_TIME);
fail_unless (gst_pad_push_event (mysrcpad, gst_event_new_segment (&segment)));
/* push a gap event to force audiodecoder to push a caps event */
fail_unless (gst_pad_push_event (mysrcpad, gst_event_new_gap (0,
GST_SECOND)));
fail_unless (buffers == NULL);
check_audiodecoder_negotiation ();
cleanup_audiodecodertest ();
}
GST_END_TEST;
GST_START_TEST (audiodecoder_delayed_negotiation_with_gap_event)
{
GstSegment segment;
setup_audiodecodertester ();
((GstAudioDecoderTester *) dec)->setoutputformat_on_decoding = TRUE;
gst_pad_set_active (mysrcpad, TRUE);
gst_element_set_state (dec, GST_STATE_PLAYING);
gst_pad_set_active (mysinkpad, TRUE);
send_startup_events ();
/* push a new segment */
gst_segment_init (&segment, GST_FORMAT_TIME);
fail_unless (gst_pad_push_event (mysrcpad, gst_event_new_segment (&segment)));
/* push a gap event to force audiodecoder to push a caps event */
fail_unless (gst_pad_push_event (mysrcpad, gst_event_new_gap (0,
GST_SECOND)));
fail_unless (buffers == NULL);
check_audiodecoder_negotiation ();
cleanup_audiodecodertest ();
}
GST_END_TEST;
static void
_audiodecoder_flush_events (gboolean send_buffers)
{
GstSegment segment;
GstBuffer *buffer;
guint64 i;
GList *events_iter;
GstMessage *msg;
setup_audiodecodertester ();
gst_pad_set_active (mysrcpad, TRUE);
gst_element_set_state (dec, GST_STATE_PLAYING);
gst_pad_set_active (mysinkpad, TRUE);
send_startup_events ();
/* push a new segment */
gst_segment_init (&segment, GST_FORMAT_TIME);
fail_unless (gst_pad_push_event (mysrcpad, gst_event_new_segment (&segment)));
if (send_buffers) {
/* push buffers, the data is actually a number so we can track them */
for (i = 0; i < NUM_BUFFERS; i++) {
if (i % 10 == 0) {
GstTagList *tags;
tags = gst_tag_list_new (GST_TAG_TRACK_NUMBER, i, NULL);
fail_unless (gst_pad_push_event (mysrcpad, gst_event_new_tag (tags)));
} else {
buffer = create_test_buffer (i);
fail_unless (gst_pad_push (mysrcpad, buffer) == GST_FLOW_OK);
}
}
} else {
/* push sticky event */
GstTagList *tags;
tags = gst_tag_list_new (GST_TAG_TRACK_NUMBER, 0, NULL);
fail_unless (gst_pad_push_event (mysrcpad, gst_event_new_tag (tags)));
}
msg =
gst_message_new_element (GST_OBJECT (mysrcpad),
gst_structure_new_empty ("test"));
fail_unless (gst_pad_push_event (mysrcpad,
gst_event_new_sink_message ("test", msg)));
gst_message_unref (msg);
fail_unless (gst_pad_push_event (mysrcpad, gst_event_new_eos ()));
events_iter = events;
/* make sure the usual events have been received */
{
GstEvent *sstart = events_iter->data;
fail_unless (GST_EVENT_TYPE (sstart) == GST_EVENT_STREAM_START);
events_iter = g_list_next (events_iter);
}
if (send_buffers) {
{
GstEvent *caps_event = events_iter->data;
fail_unless (GST_EVENT_TYPE (caps_event) == GST_EVENT_CAPS);
events_iter = g_list_next (events_iter);
}
{
GstEvent *segment_event = events_iter->data;
fail_unless (GST_EVENT_TYPE (segment_event) == GST_EVENT_SEGMENT);
events_iter = g_list_next (events_iter);
}
for (int i=0; i< NUM_BUFFERS / 10; i++)
{
GstEvent *tag_event = events_iter->data;
fail_unless (GST_EVENT_TYPE (tag_event) == GST_EVENT_TAG);
events_iter = g_list_next (events_iter);
}
}
{
GstEvent *eos_event = events_iter->data;
fail_unless (GST_EVENT_TYPE (eos_event) == GST_EVENT_EOS);
events_iter = g_list_next (events_iter);
}
/* check that EOS was received */
fail_unless (GST_PAD_IS_EOS (mysrcpad));
fail_unless (gst_pad_push_event (mysrcpad, gst_event_new_flush_start ()));
fail_unless (GST_PAD_IS_EOS (mysrcpad));
/* Check that we have tags */
{
GstEvent *tags = gst_pad_get_sticky_event (mysrcpad, GST_EVENT_TAG, 0);
fail_unless (tags != NULL);
gst_event_unref (tags);
}
/* Check that we still have a segment set */
{
GstEvent *segment =
gst_pad_get_sticky_event (mysrcpad, GST_EVENT_SEGMENT, 0);
fail_unless (segment != NULL);
gst_event_unref (segment);
}
fail_unless (gst_pad_push_event (mysrcpad, gst_event_new_flush_stop (TRUE)));
fail_if (GST_PAD_IS_EOS (mysrcpad));
/* Check that the segment was flushed on FLUSH_STOP */
{
GstEvent *segment =
gst_pad_get_sticky_event (mysrcpad, GST_EVENT_SEGMENT, 0);
fail_unless (segment == NULL);
}
/* Check the tags were not lost on FLUSH_STOP */
{
GstEvent *tags = gst_pad_get_sticky_event (mysrcpad, GST_EVENT_TAG, 0);
fail_unless (tags != NULL);
gst_event_unref (tags);
}
g_list_free_full (buffers, (GDestroyNotify) gst_buffer_unref);
buffers = NULL;
gst_element_set_state (dec, GST_STATE_NULL);
cleanup_audiodecodertest ();
}
GST_START_TEST (audiodecoder_flush_events_no_buffers)
{
_audiodecoder_flush_events (FALSE);
}
GST_END_TEST;
GST_START_TEST (audiodecoder_flush_events)
{
_audiodecoder_flush_events (TRUE);
}
GST_END_TEST;
GST_START_TEST (audiodecoder_buffer_after_segment)
{
GstSegment segment;
GstBuffer *buffer;
guint64 i;
GstClockTime pos;
setup_audiodecodertester ();
gst_pad_set_active (mysrcpad, TRUE);
gst_element_set_state (dec, GST_STATE_PLAYING);
gst_pad_set_active (mysinkpad, TRUE);
send_startup_events ();
/* push a new segment */
gst_segment_init (&segment, GST_FORMAT_TIME);
segment.stop = GST_SECOND;
fail_unless (gst_pad_push_event (mysrcpad, gst_event_new_segment (&segment)));
/* push buffers, the data is actually a number so we can track them */
i = 0;
pos = 0;
while (pos < GST_SECOND) {
GstMapInfo map;
guint64 num;
buffer = create_test_buffer (i);
pos = GST_BUFFER_TIMESTAMP (buffer) + GST_BUFFER_DURATION (buffer);
fail_unless (gst_pad_push (mysrcpad, buffer) == GST_FLOW_OK);
/* check that buffer was received by our source pad */
buffer = buffers->data;
gst_buffer_map (buffer, &map, GST_MAP_READ);
num = *(guint64 *) map.data;
fail_unless_equals_uint64 (i, num);
fail_unless_equals_uint64 (GST_BUFFER_PTS (buffer),
gst_util_uint64_scale_round (i, GST_SECOND, TEST_MSECS_PER_SAMPLE));
fail_unless_equals_uint64 (GST_BUFFER_DURATION (buffer),
gst_util_uint64_scale_round (1, GST_SECOND, TEST_MSECS_PER_SAMPLE));
gst_buffer_unmap (buffer, &map);
gst_buffer_unref (buffer);
buffers = g_list_delete_link (buffers, buffers);
i++;
}
/* this buffer is after the segment */
buffer = create_test_buffer (i++);
fail_unless (gst_pad_push (mysrcpad, buffer) == GST_FLOW_EOS);
fail_unless (gst_pad_push_event (mysrcpad, gst_event_new_eos ()));
fail_unless (buffers == NULL);
cleanup_audiodecodertest ();
}
GST_END_TEST;
GST_START_TEST (audiodecoder_output_too_many_frames)
{
GstSegment segment;
GstBuffer *buffer;
guint64 i;
setup_audiodecodertester ();
((GstAudioDecoderTester *) dec)->output_too_many_frames = TRUE;
gst_pad_set_active (mysrcpad, TRUE);
gst_element_set_state (dec, GST_STATE_PLAYING);
gst_pad_set_active (mysinkpad, TRUE);
send_startup_events ();
/* push a new segment */
gst_segment_init (&segment, GST_FORMAT_TIME);
fail_unless (gst_pad_push_event (mysrcpad, gst_event_new_segment (&segment)));
/* push buffers, the data is actually a number so we can track them */
for (i = 0; i < 3; i++) {
GstMapInfo map;
guint64 num;
buffer = create_test_buffer (i);
fail_unless (gst_pad_push (mysrcpad, buffer) == GST_FLOW_OK);
/* check that buffer was received by our source pad */
buffer = buffers->data;
gst_buffer_map (buffer, &map, GST_MAP_READ);
num = *(guint64 *) map.data;
fail_unless_equals_uint64 (i, num);
fail_unless_equals_uint64 (GST_BUFFER_PTS (buffer),
gst_util_uint64_scale_round (i, GST_SECOND, TEST_MSECS_PER_SAMPLE));
fail_unless_equals_uint64 (GST_BUFFER_DURATION (buffer),
gst_util_uint64_scale_round (1, GST_SECOND, TEST_MSECS_PER_SAMPLE));
gst_buffer_unmap (buffer, &map);
gst_buffer_unref (buffer);
buffers = g_list_delete_link (buffers, buffers);
}
fail_unless (gst_pad_push_event (mysrcpad, gst_event_new_eos ()));
fail_unless (buffers == NULL);
cleanup_audiodecodertest ();
}
GST_END_TEST;
static Suite *
gst_audiodecoder_suite (void)
{
Suite *s = suite_create ("GstAudioDecoder");
TCase *tc = tcase_create ("general");
suite_add_tcase (s, tc);
tcase_add_test (tc, audiodecoder_playback);
tcase_add_test (tc, audiodecoder_flush_events_no_buffers);
tcase_add_test (tc, audiodecoder_flush_events);
tcase_add_test (tc, audiodecoder_negotiation_with_buffer);
tcase_add_test (tc, audiodecoder_negotiation_with_gap_event);
tcase_add_test (tc, audiodecoder_delayed_negotiation_with_gap_event);
tcase_add_test (tc, audiodecoder_buffer_after_segment);
tcase_add_test (tc, audiodecoder_output_too_many_frames);
return s;
}
GST_CHECK_MAIN (gst_audiodecoder);