gstreamer/subprojects/gst-plugins-base/tests/check/libs/audiodecoder.c

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/* 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/check/gstharness.h>
#include <gst/audio/audio.h>
#include <gst/app/app.h>
#define TEST_MSECS_PER_SAMPLE 44100
#define RESTRICTED_CAPS_RATE 44100
#define RESTRICTED_CAPS_CHANNELS 6
static GstStaticPadTemplate sinktemplate_restricted =
GST_STATIC_PAD_TEMPLATE ("sink",
GST_PAD_SINK,
GST_PAD_ALWAYS,
GST_STATIC_CAPS ("audio/x-raw, rate=(int)44100, channels=(int)6")
);
static GstStaticPadTemplate sinktemplate_with_range =
GST_STATIC_PAD_TEMPLATE ("sink",
GST_PAD_SINK,
GST_PAD_ALWAYS,
GST_STATIC_CAPS ("audio/x-raw, rate=(int)[1,44100], channels=(int)[1,6]")
);
static GstStaticPadTemplate sinktemplate_default =
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_default =
GST_STATIC_PAD_TEMPLATE ("src",
GST_PAD_SRC,
GST_PAD_ALWAYS,
GST_STATIC_CAPS ("audio/x-test-custom")
);
#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;
gboolean delay_decoding;
GstBuffer *prev_buf;
};
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)
{
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GstAudioDecoderTester *tester = (GstAudioDecoderTester *) dec;
if (tester->prev_buf) {
gst_buffer_unref (tester->prev_buf);
tester->prev_buf = NULL;
}
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;
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;
GstFlowReturn ret = GST_FLOW_OK;
gboolean do_plc = gst_audio_decoder_get_plc (dec) &&
gst_audio_decoder_get_plc_aware (dec);
if (buffer == NULL || (!do_plc && gst_buffer_get_size (buffer) == 0))
return GST_FLOW_OK;
gst_buffer_ref (buffer);
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);
}
if ((tester->delay_decoding && tester->prev_buf != NULL) ||
!tester->delay_decoding) {
gsize buf_num = tester->delay_decoding ? 2 : 1;
gint i;
for (i = 0; i != buf_num; ++i) {
GstBuffer *cur_buf = buf_num == 1 || i != 0 ? buffer : tester->prev_buf;
gst_buffer_map (cur_buf, &map, GST_MAP_READ);
/* the output is SE32LE stereo 44100 Hz */
size = 2 * 4;
g_assert (size == sizeof (guint64));
data = g_malloc0 (size);
if (map.size) {
g_assert_cmpint (map.size, >=, sizeof (guint64));
memcpy (data, map.data, sizeof (guint64));
}
output_buffer = gst_buffer_new_wrapped (data, size);
gst_buffer_unmap (cur_buf, &map);
if (tester->output_too_many_frames) {
ret = gst_audio_decoder_finish_frame (dec, output_buffer, 2);
} else {
ret = gst_audio_decoder_finish_frame (dec, output_buffer, 1);
}
if (ret != GST_FLOW_OK)
break;
}
tester->delay_decoding = FALSE;
}
if (tester->prev_buf)
gst_buffer_unref (tester->prev_buf);
tester->prev_buf = NULL;
if (tester->delay_decoding)
tester->prev_buf = buffer;
else
gst_buffer_unref (buffer);
return ret;
}
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_static_pad_template (element_class, &sink_templ);
gst_element_class_add_static_pad_template (element_class, &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 GstHarness *
setup_audiodecodertester (GstStaticPadTemplate * sinktemplate,
GstStaticPadTemplate * srctemplate)
{
GstHarness *h;
GstElement *dec;
if (sinktemplate == NULL)
sinktemplate = &sinktemplate_default;
if (srctemplate == NULL)
srctemplate = &srctemplate_default;
dec = g_object_new (GST_AUDIO_DECODER_TESTER_TYPE, NULL);
h = gst_harness_new_full (dec, srctemplate, "sink", sinktemplate, "src");
gst_harness_set_src_caps (h,
gst_caps_new_simple ("audio/x-test-custom",
"channels", G_TYPE_INT, 2, "rate", G_TYPE_INT, 44100, NULL));
gst_object_unref (dec);
return h;
}
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;
}
#define NUM_BUFFERS 10
GST_START_TEST (audiodecoder_playback)
{
GstBuffer *buffer;
guint64 i;
GstHarness *h = setup_audiodecodertester (NULL, NULL);
/* push buffers, the data is actually a number so we can track them */
for (i = 0; i < NUM_BUFFERS; i++) {
GstMapInfo map;
guint64 num;
fail_unless (gst_harness_push (h, create_test_buffer (i)) == GST_FLOW_OK);
/* check that buffer was received by our source pad */
buffer = gst_harness_pull (h);
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);
}
fail_unless (gst_harness_push_event (h, gst_event_new_eos ()));
fail_unless_equals_int (0, gst_harness_buffers_in_queue (h));
gst_harness_teardown (h);
}
GST_END_TEST;
static void
check_audiodecoder_negotiation (GstHarness * h)
{
gboolean received_caps = FALSE;
guint i;
guint events_received = gst_harness_events_received (h);
for (i = 0; i < events_received; i++) {
GstEvent *event = gst_harness_pull_event (h);
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;
gst_event_unref (event);
break;
}
gst_event_unref (event);
}
fail_unless (received_caps);
}
GST_START_TEST (audiodecoder_negotiation_with_buffer)
{
GstHarness *h = setup_audiodecodertester (NULL, NULL);
/* push a buffer event to force audiodecoder to push a caps event */
fail_unless (gst_harness_push (h, create_test_buffer (0)) == GST_FLOW_OK);
check_audiodecoder_negotiation (h);
gst_harness_teardown (h);
}
GST_END_TEST;
GST_START_TEST (audiodecoder_negotiation_with_gap_event)
{
GstHarness *h = setup_audiodecodertester (NULL, NULL);
/* push a gap event to force audiodecoder to push a caps event */
fail_unless (gst_harness_push_event (h, gst_event_new_gap (0, GST_SECOND)));
fail_unless_equals_int (0, gst_harness_buffers_in_queue (h));
check_audiodecoder_negotiation (h);
gst_harness_teardown (h);
}
GST_END_TEST;
GST_START_TEST (audiodecoder_delayed_negotiation_with_gap_event)
{
GstHarness *h = setup_audiodecodertester (NULL, NULL);
((GstAudioDecoderTester *) h->element)->setoutputformat_on_decoding = TRUE;
/* push a gap event to force audiodecoder to push a caps event */
fail_unless (gst_harness_push_event (h, gst_event_new_gap (0, GST_SECOND)));
fail_unless_equals_int (0, gst_harness_buffers_in_queue (h));
check_audiodecoder_negotiation (h);
gst_harness_teardown (h);
}
GST_END_TEST;
/* make sure that the segment event is pushed before the gap */
GST_START_TEST (audiodecoder_first_data_is_gap)
{
GstHarness *h = setup_audiodecodertester (NULL, NULL);
/* push a gap */
fail_unless (gst_harness_push_event (h, gst_event_new_gap (0, GST_SECOND)));
/* make sure the usual events have been received */
{
GstEvent *sstart = gst_harness_pull_event (h);
fail_unless (GST_EVENT_TYPE (sstart) == GST_EVENT_STREAM_START);
gst_event_unref (sstart);
}
{
GstEvent *caps_event = gst_harness_pull_event (h);
fail_unless (GST_EVENT_TYPE (caps_event) == GST_EVENT_CAPS);
gst_event_unref (caps_event);
}
{
GstEvent *segment_event = gst_harness_pull_event (h);
fail_unless (GST_EVENT_TYPE (segment_event) == GST_EVENT_SEGMENT);
gst_event_unref (segment_event);
}
/* Make sure the gap was pushed */
{
GstEvent *gap = gst_harness_pull_event (h);
fail_unless (GST_EVENT_TYPE (gap) == GST_EVENT_GAP);
gst_event_unref (gap);
}
fail_unless_equals_int (0, gst_harness_events_in_queue (h));
gst_harness_teardown (h);
}
GST_END_TEST;
/*
*/
static void
_audiodecoder_flush_events (gboolean send_buffers)
{
guint i;
GstMessage *msg;
GstHarness *h = setup_audiodecodertester (NULL, NULL);
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_harness_push_event (h, gst_event_new_tag (tags)));
} else {
fail_unless (gst_harness_push (h,
create_test_buffer (i)) == GST_FLOW_OK);
}
}
} else {
/* push sticky event */
GstTagList *tags;
tags = gst_tag_list_new (GST_TAG_TRACK_NUMBER, 0, NULL);
fail_unless (gst_harness_push_event (h, gst_event_new_tag (tags)));
}
msg = gst_message_new_element (GST_OBJECT (h->element),
gst_structure_new_empty ("test"));
fail_unless (gst_harness_push_event (h,
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gst_event_new_sink_message ("test", msg)));
gst_message_unref (msg);
fail_unless (gst_harness_push_event (h, gst_event_new_eos ()));
/* make sure the usual events have been received */
{
GstEvent *sstart = gst_harness_pull_event (h);
fail_unless (GST_EVENT_TYPE (sstart) == GST_EVENT_STREAM_START);
gst_event_unref (sstart);
}
if (send_buffers) {
{
GstEvent *caps_event = gst_harness_pull_event (h);
fail_unless (GST_EVENT_TYPE (caps_event) == GST_EVENT_CAPS);
gst_event_unref (caps_event);
}
{
GstEvent *segment_event = gst_harness_pull_event (h);
fail_unless (GST_EVENT_TYPE (segment_event) == GST_EVENT_SEGMENT);
gst_event_unref (segment_event);
}
for (i = 0; i < NUM_BUFFERS / 10; i++) {
GstEvent *tag_event = gst_harness_pull_event (h);
fail_unless (GST_EVENT_TYPE (tag_event) == GST_EVENT_TAG);
gst_event_unref (tag_event);
}
} else {
{
GstEvent *segment_event = gst_harness_pull_event (h);
fail_unless (GST_EVENT_TYPE (segment_event) == GST_EVENT_SEGMENT);
gst_event_unref (segment_event);
}
{
GstEvent *tag_event = gst_harness_pull_event (h);
fail_unless (GST_EVENT_TYPE (tag_event) == GST_EVENT_TAG);
gst_event_unref (tag_event);
}
}
{
GstEvent *sink_msg_event = gst_harness_pull_event (h);
fail_unless (GST_EVENT_TYPE (sink_msg_event) == GST_EVENT_SINK_MESSAGE);
gst_event_unref (sink_msg_event);
}
{
GstEvent *eos_event = gst_harness_pull_event (h);
fail_unless (GST_EVENT_TYPE (eos_event) == GST_EVENT_EOS);
gst_event_unref (eos_event);
}
/* check that EOS was received */
fail_unless (GST_PAD_IS_EOS (h->srcpad));
fail_unless (gst_harness_push_event (h, gst_event_new_flush_start ()));
fail_unless (GST_PAD_IS_EOS (h->srcpad));
/* Check that we have tags */
{
GstEvent *tags = gst_pad_get_sticky_event (h->srcpad, 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 (h->srcpad, GST_EVENT_SEGMENT, 0);
fail_unless (segment != NULL);
gst_event_unref (segment);
}
fail_unless (gst_harness_push_event (h, gst_event_new_flush_stop (TRUE)));
fail_if (GST_PAD_IS_EOS (h->srcpad));
/* Check that the segment was flushed on FLUSH_STOP */
{
GstEvent *segment =
gst_pad_get_sticky_event (h->srcpad, GST_EVENT_SEGMENT, 0);
fail_unless (segment == NULL);
}
/* Check the tags were not lost on FLUSH_STOP */
{
GstEvent *tags = gst_pad_get_sticky_event (h->srcpad, GST_EVENT_TAG, 0);
fail_unless (tags != NULL);
gst_event_unref (tags);
}
if (send_buffers) {
fail_unless_equals_int (NUM_BUFFERS - NUM_BUFFERS / 10,
gst_harness_buffers_in_queue (h));
} else {
fail_unless_equals_int (0, gst_harness_buffers_in_queue (h));
}
fail_unless_equals_int (2, gst_harness_events_in_queue (h));
gst_harness_teardown (h);
}
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;
/* An element should always push its segment before sending EOS */
GST_START_TEST (audiodecoder_eos_events_no_buffers)
{
GstHarness *h = setup_audiodecodertester (NULL, NULL);
fail_unless (gst_harness_push_event (h, gst_event_new_eos ()));
fail_unless (GST_PAD_IS_EOS (h->sinkpad));
{
GstEvent *segment_event =
gst_pad_get_sticky_event (h->sinkpad, GST_EVENT_SEGMENT, 0);
fail_unless (segment_event != NULL);
gst_event_unref (segment_event);
}
gst_harness_teardown (h);
}
GST_END_TEST;
GST_START_TEST (audiodecoder_buffer_after_segment)
{
GstSegment segment;
GstBuffer *buffer;
guint64 i;
GstClockTime pos;
#define SEGMENT_STOP (GST_MSECOND * 10)
GstHarness *h = setup_audiodecodertester (NULL, NULL);
/* push a new segment */
gst_segment_init (&segment, GST_FORMAT_TIME);
segment.stop = SEGMENT_STOP;
fail_unless (gst_harness_push_event (h, gst_event_new_segment (&segment)));
/* push buffers, the data is actually a number so we can track them */
i = 0;
pos = 0;
while (pos < SEGMENT_STOP) {
GstMapInfo map;
guint64 num;
buffer = create_test_buffer (i);
pos = GST_BUFFER_TIMESTAMP (buffer) + GST_BUFFER_DURATION (buffer);
fail_unless (gst_harness_push (h, buffer) == GST_FLOW_OK);
/* check that buffer was received by our source pad */
buffer = gst_harness_pull (h);
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);
i++;
}
/* this buffer is after the segment */
buffer = create_test_buffer (i++);
fail_unless (gst_harness_push (h, buffer) == GST_FLOW_EOS);
fail_unless (gst_harness_push_event (h, gst_event_new_eos ()));
fail_unless_equals_int (0, gst_harness_buffers_in_queue (h));
gst_harness_teardown (h);
}
GST_END_TEST;
GST_START_TEST (audiodecoder_output_too_many_frames)
{
GstBuffer *buffer;
guint64 i;
GstHarness *h = setup_audiodecodertester (NULL, NULL);
((GstAudioDecoderTester *) h->element)->output_too_many_frames = TRUE;
/* push buffers, the data is actually a number so we can track them */
for (i = 0; i < 3; i++) {
GstMapInfo map;
guint64 num;
fail_unless (gst_harness_push (h, create_test_buffer (i)) == GST_FLOW_OK);
/* check that buffer was received by our source pad */
buffer = gst_harness_pull (h);
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);
}
fail_unless (gst_harness_push_event (h, gst_event_new_eos ()));
fail_unless_equals_int (0, gst_harness_buffers_in_queue (h));
gst_harness_teardown (h);
}
GST_END_TEST;
GST_START_TEST (audiodecoder_query_caps_with_fixed_caps_peer)
{
GstCaps *caps;
GstCaps *filter;
GstStructure *structure;
gint rate, channels;
GstHarness *h = setup_audiodecodertester (&sinktemplate_restricted, NULL);
caps = gst_pad_peer_query_caps (h->srcpad, NULL);
fail_unless (caps != NULL);
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));
/* match our restricted caps values */
fail_unless (channels == RESTRICTED_CAPS_CHANNELS);
fail_unless (rate == RESTRICTED_CAPS_RATE);
gst_caps_unref (caps);
filter = gst_caps_new_simple ("audio/x-custom-test", "rate", G_TYPE_INT,
10000, "channels", G_TYPE_INT, 12, NULL);
caps = gst_pad_peer_query_caps (h->srcpad, filter);
fail_unless (caps != NULL);
fail_unless (gst_caps_is_empty (caps));
gst_caps_unref (caps);
gst_caps_unref (filter);
gst_harness_teardown (h);
}
GST_END_TEST;
static void
_get_int_range (GstStructure * s, const gchar * field, gint * min_v,
gint * max_v)
{
const GValue *value;
value = gst_structure_get_value (s, field);
fail_unless (value != NULL);
fail_unless (GST_VALUE_HOLDS_INT_RANGE (value));
*min_v = gst_value_get_int_range_min (value);
*max_v = gst_value_get_int_range_max (value);
}
GST_START_TEST (audiodecoder_query_caps_with_range_caps_peer)
{
GstCaps *caps;
GstCaps *filter;
GstStructure *structure;
gint rate, channels;
gint rate_min, channels_min;
gint rate_max, channels_max;
GstHarness *h = setup_audiodecodertester (&sinktemplate_with_range, NULL);
caps = gst_pad_peer_query_caps (h->srcpad, NULL);
fail_unless (caps != NULL);
structure = gst_caps_get_structure (caps, 0);
_get_int_range (structure, "rate", &rate_min, &rate_max);
_get_int_range (structure, "channels", &channels_min, &channels_max);
fail_unless (rate_min == 1);
fail_unless (rate_max == RESTRICTED_CAPS_RATE);
fail_unless (channels_min == 1);
fail_unless (channels_max == RESTRICTED_CAPS_CHANNELS);
gst_caps_unref (caps);
/* query with a fixed filter */
filter = gst_caps_new_simple ("audio/x-test-custom", "rate", G_TYPE_INT,
RESTRICTED_CAPS_RATE, "channels", G_TYPE_INT, RESTRICTED_CAPS_CHANNELS,
NULL);
caps = gst_pad_peer_query_caps (h->srcpad, filter);
fail_unless (caps != NULL);
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 == RESTRICTED_CAPS_RATE);
fail_unless (channels == RESTRICTED_CAPS_CHANNELS);
gst_caps_unref (caps);
gst_caps_unref (filter);
/* query with a fixed filter that will lead to empty result */
filter = gst_caps_new_simple ("audio/x-test-custom", "rate", G_TYPE_INT,
10000, "channels", G_TYPE_INT, 12, NULL);
caps = gst_pad_peer_query_caps (h->srcpad, filter);
fail_unless (caps != NULL);
fail_unless (gst_caps_is_empty (caps));
gst_caps_unref (caps);
gst_caps_unref (filter);
gst_harness_teardown (h);
}
GST_END_TEST;
#define GETCAPS_CAPS_STR "audio/x-test-custom, somefield=(string)getcaps"
static GstCaps *
_custom_audio_decoder_getcaps (GstAudioDecoder * dec, GstCaps * filter)
{
return gst_caps_from_string (GETCAPS_CAPS_STR);
}
GST_START_TEST (audiodecoder_query_caps_with_custom_getcaps)
{
GstCaps *caps;
GstAudioDecoderClass *klass;
GstCaps *expected_caps;
GstHarness *h = setup_audiodecodertester (&sinktemplate_restricted, NULL);
klass = GST_AUDIO_DECODER_CLASS (GST_AUDIO_DECODER_GET_CLASS (h->element));
klass->getcaps = _custom_audio_decoder_getcaps;
caps = gst_pad_peer_query_caps (h->srcpad, NULL);
fail_unless (caps != NULL);
expected_caps = gst_caps_from_string (GETCAPS_CAPS_STR);
fail_unless (gst_caps_is_equal (expected_caps, caps));
gst_caps_unref (expected_caps);
gst_caps_unref (caps);
gst_harness_teardown (h);
}
GST_END_TEST;
static GstTagList *
pad_get_sticky_tags (GstPad * pad, GstTagScope scope)
{
GstTagList *tags = NULL;
GstEvent *event;
guint i = 0;
do {
event = gst_pad_get_sticky_event (pad, GST_EVENT_TAG, i++);
if (event == NULL)
break;
gst_event_parse_tag (event, &tags);
if (scope == gst_tag_list_get_scope (tags))
tags = gst_tag_list_ref (tags);
else
tags = NULL;
gst_event_unref (event);
}
while (tags == NULL);
return tags;
}
#define tag_list_peek_string(list,tag,p_s) \
gst_tag_list_peek_string_index(list,tag,0,p_s)
/* Check tag transformations and updates */
GST_START_TEST (audiodecoder_tag_handling)
{
GstTagList *global_tags;
GstTagList *tags;
const gchar *s = NULL;
guint u = 0;
GstHarness *h = setup_audiodecodertester (NULL, NULL);
/* =======================================================================
* SCENARIO 0: global tags passthrough; check upstream/decoder tag merging
* ======================================================================= */
/* push some global tags (these should be passed through and not messed with) */
global_tags = gst_tag_list_new (GST_TAG_TITLE, "Global", NULL);
gst_tag_list_set_scope (global_tags, GST_TAG_SCOPE_GLOBAL);
fail_unless (gst_harness_push_event (h,
gst_event_new_tag (gst_tag_list_ref (global_tags))));
/* create some (upstream) stream tags */
tags = gst_tag_list_new (GST_TAG_AUDIO_CODEC, "Upstream Codec",
GST_TAG_DESCRIPTION, "Upstream Description", NULL);
gst_tag_list_set_scope (tags, GST_TAG_SCOPE_STREAM);
fail_unless (gst_harness_push_event (h, gst_event_new_tag (tags)));
tags = NULL;
/* decoder tags: override/add AUDIO_CODEC, BITRATE and MAXIMUM_BITRATE */
{
GstTagList *decoder_tags;
decoder_tags = gst_tag_list_new (GST_TAG_AUDIO_CODEC, "Decoder Codec",
GST_TAG_BITRATE, 250000, GST_TAG_MAXIMUM_BITRATE, 255000, NULL);
gst_audio_decoder_merge_tags (GST_AUDIO_DECODER (h->element),
decoder_tags, GST_TAG_MERGE_REPLACE);
gst_tag_list_unref (decoder_tags);
}
/* push buffer (this will call gst_audio_decoder_merge_tags with the above) */
fail_unless (gst_harness_push (h, create_test_buffer (0)) == GST_FLOW_OK);
gst_buffer_unref (gst_harness_pull (h));
/* check global tags: should not have been tampered with */
tags = pad_get_sticky_tags (h->sinkpad, GST_TAG_SCOPE_GLOBAL);
fail_unless (tags != NULL);
GST_INFO ("global tags: %" GST_PTR_FORMAT, tags);
fail_unless (gst_tag_list_is_equal (tags, global_tags));
gst_tag_list_unref (tags);
/* check merged stream tags */
tags = pad_get_sticky_tags (h->sinkpad, GST_TAG_SCOPE_STREAM);
fail_unless (tags != NULL);
GST_INFO ("stream tags: %" GST_PTR_FORMAT, tags);
/* upstream audio codec should've been replaced with audiodecoder one */
fail_unless (tag_list_peek_string (tags, GST_TAG_AUDIO_CODEC, &s));
fail_unless_equals_string (s, "Decoder Codec");
/* no upstream bitrate, so audiodecoder one should've been added */
fail_unless (gst_tag_list_get_uint (tags, GST_TAG_BITRATE, &u));
fail_unless_equals_int (u, 250000);
/* no upstream maximum-bitrate, so audiodecoder one should've been added */
fail_unless (gst_tag_list_get_uint (tags, GST_TAG_MAXIMUM_BITRATE, &u));
fail_unless_equals_int (u, 255000);
fail_unless (gst_tag_list_get_tag_size (tags, GST_TAG_AUDIO_CODEC) == 1);
fail_unless (gst_tag_list_get_tag_size (tags, GST_TAG_BITRATE) == 1);
fail_unless (gst_tag_list_get_tag_size (tags, GST_TAG_MAXIMUM_BITRATE) == 1);
/* upstream description should've been maintained */
fail_unless (gst_tag_list_get_tag_size (tags, GST_TAG_DESCRIPTION) == 1);
/* and that should be all: AUDIO_CODEC, DESCRIPTION, BITRATE, MAX BITRATE */
fail_unless_equals_int (gst_tag_list_n_tags (tags), 4);
gst_tag_list_unref (tags);
s = NULL;
/* ===================================================================
* SCENARIO 1: upstream sends updated tags, decoder tags stay the same
* =================================================================== */
/* push same upstream stream tags again */
tags = gst_tag_list_new (GST_TAG_AUDIO_CODEC, "Upstream Codec",
GST_TAG_DESCRIPTION, "Upstream Description", NULL);
fail_unless (gst_harness_push_event (h, gst_event_new_tag (tags)));
tags = NULL;
/* decoder tags are still:
* audio-codec = "Decoder Codec", bitrate=250000, maximum-bitrate=255000 */
/* check possibly updated merged stream tags, should be same as before */
tags = pad_get_sticky_tags (h->sinkpad, GST_TAG_SCOPE_STREAM);
fail_unless (tags != NULL);
GST_INFO ("stream tags: %" GST_PTR_FORMAT, tags);
/* upstream audio codec still be the one merge-replaced by the subclass */
fail_unless (tag_list_peek_string (tags, GST_TAG_AUDIO_CODEC, &s));
fail_unless_equals_string (s, "Decoder Codec");
/* no upstream bitrate, so audiodecoder one should've been added */
fail_unless (gst_tag_list_get_uint (tags, GST_TAG_BITRATE, &u));
fail_unless_equals_int (u, 250000);
fail_unless (gst_tag_list_get_tag_size (tags, GST_TAG_AUDIO_CODEC) == 1);
fail_unless (gst_tag_list_get_tag_size (tags, GST_TAG_BITRATE) == 1);
fail_unless (gst_tag_list_get_tag_size (tags, GST_TAG_MAXIMUM_BITRATE) == 1);
/* upstream description should've been maintained */
fail_unless (gst_tag_list_get_tag_size (tags, GST_TAG_DESCRIPTION) == 1);
/* and that should be all: AUDIO_CODEC, DESCRIPTION, BITRATE, MAX BITRATE */
fail_unless_equals_int (gst_tag_list_n_tags (tags), 4);
gst_tag_list_unref (tags);
s = NULL;
/* =============================================================
* SCENARIO 2: decoder updates tags, upstream tags stay the same
* ============================================================= */
/* new decoder tags: override AUDIO_CODEC, update/add BITRATE,
* no MAXIMUM_BITRATE this time (which means it should not appear
* any longer in the output tags now) (bitrate is a different value now) */
{
GstTagList *decoder_tags;
decoder_tags = gst_tag_list_new (GST_TAG_AUDIO_CODEC, "Decoder Codec",
GST_TAG_BITRATE, 275000, NULL);
gst_audio_decoder_merge_tags (GST_AUDIO_DECODER (h->element),
decoder_tags, GST_TAG_MERGE_REPLACE);
gst_tag_list_unref (decoder_tags);
}
/* push another buffer to make decoder update tags */
fail_unless (gst_harness_push (h, create_test_buffer (2)) == GST_FLOW_OK);
gst_buffer_unref (gst_harness_pull (h));
/* check updated merged stream tags, the decoder bits should be different */
tags = pad_get_sticky_tags (h->sinkpad, GST_TAG_SCOPE_STREAM);
fail_unless (tags != NULL);
GST_INFO ("stream tags: %" GST_PTR_FORMAT, tags);
/* upstream audio codec still replaced by the subclass's (wasn't updated) */
fail_unless (tag_list_peek_string (tags, GST_TAG_AUDIO_CODEC, &s));
fail_unless_equals_string (s, "Decoder Codec");
/* no upstream bitrate, so audiodecoder one should've been added, was updated */
fail_unless (gst_tag_list_get_uint (tags, GST_TAG_BITRATE, &u));
fail_unless_equals_int (u, 275000);
/* no upstream maximum-bitrate, and audiodecoder removed it now */
fail_unless (!gst_tag_list_get_uint (tags, GST_TAG_MAXIMUM_BITRATE, &u));
fail_unless (gst_tag_list_get_tag_size (tags, GST_TAG_AUDIO_CODEC) == 1);
fail_unless (gst_tag_list_get_tag_size (tags, GST_TAG_BITRATE) == 1);
/* upstream description should've been maintained */
fail_unless (gst_tag_list_get_tag_size (tags, GST_TAG_DESCRIPTION) == 1);
/* and that should be all, just AUDIO_CODEC, DESCRIPTION, BITRATE */
fail_unless_equals_int (gst_tag_list_n_tags (tags), 3);
gst_tag_list_unref (tags);
s = NULL;
/* =================================================================
* SCENARIO 3: stream-start event should clear upstream tags
* ================================================================= */
/* also tests if the stream-start event clears the upstream tags */
fail_unless (gst_harness_push_event (h, gst_event_new_stream_start ("x")));
/* push another buffer to make decoder update tags */
fail_unless (gst_harness_push (h, create_test_buffer (3)) == GST_FLOW_OK);
gst_buffer_unref (gst_harness_pull (h));
/* check updated merged stream tags, should be just decoder tags now */
tags = pad_get_sticky_tags (h->sinkpad, GST_TAG_SCOPE_STREAM);
fail_unless (tags != NULL);
GST_INFO ("stream tags: %" GST_PTR_FORMAT, tags);
fail_unless (tag_list_peek_string (tags, GST_TAG_AUDIO_CODEC, &s));
fail_unless_equals_string (s, "Decoder Codec");
fail_unless (gst_tag_list_get_uint (tags, GST_TAG_BITRATE, &u));
fail_unless_equals_int (u, 275000);
/* no upstream maximum-bitrate, and audiodecoder removed it now */
fail_unless (!gst_tag_list_get_uint (tags, GST_TAG_MAXIMUM_BITRATE, &u));
fail_unless (gst_tag_list_get_tag_size (tags, GST_TAG_AUDIO_CODEC) == 1);
fail_unless (gst_tag_list_get_tag_size (tags, GST_TAG_BITRATE) == 1);
/* no more description tag since no more upstream tags */
fail_unless (gst_tag_list_get_tag_size (tags, GST_TAG_DESCRIPTION) == 0);
/* and that should be all, just AUDIO_CODEC, BITRATE */
fail_unless_equals_int (gst_tag_list_n_tags (tags), 2);
gst_tag_list_unref (tags);
s = NULL;
/* clean up */
fail_unless (gst_harness_push_event (h, gst_event_new_eos ()));
fail_unless_equals_int (0, gst_harness_buffers_in_queue (h));
gst_tag_list_unref (global_tags);
gst_harness_teardown (h);
}
GST_END_TEST;
GST_START_TEST (audiodecoder_plc_on_gap_event)
{
/* GstAudioDecoder should not mark the stream DISCOUNT flag when
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concealed audio eliminate discontinuity. More important it should not
mess with the timestamps */
GstClockTime pts;
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GstClockTime dur =
gst_util_uint64_scale_round (1, GST_SECOND, TEST_MSECS_PER_SAMPLE);
GstBuffer *buf;
GstHarness *h = setup_audiodecodertester (NULL, NULL);
gst_audio_decoder_set_plc_aware (GST_AUDIO_DECODER (h->element), TRUE);
gst_audio_decoder_set_plc (GST_AUDIO_DECODER (h->element), TRUE);
pts = gst_util_uint64_scale_round (0, GST_SECOND, TEST_MSECS_PER_SAMPLE);
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gst_harness_push (h, create_test_buffer (0));
buf = gst_harness_pull (h);
fail_unless_equals_int (pts, GST_BUFFER_PTS (buf));
fail_unless_equals_int (dur, GST_BUFFER_DURATION (buf));
fail_unless (GST_BUFFER_FLAG_IS_SET (buf, GST_BUFFER_FLAG_DISCONT));
gst_buffer_unref (buf);
pts = gst_util_uint64_scale_round (1, GST_SECOND, TEST_MSECS_PER_SAMPLE);
gst_harness_push_event (h, gst_event_new_gap (pts, dur));
buf = gst_harness_pull (h);
fail_unless_equals_int (pts, GST_BUFFER_PTS (buf));
fail_unless_equals_int (dur, GST_BUFFER_DURATION (buf));
fail_unless (!GST_BUFFER_FLAG_IS_SET (buf, GST_BUFFER_FLAG_DISCONT));
gst_buffer_unref (buf);
pts = gst_util_uint64_scale_round (2, GST_SECOND, TEST_MSECS_PER_SAMPLE);
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buf = create_test_buffer (2);
GST_BUFFER_FLAG_SET (buf, GST_BUFFER_FLAG_DISCONT);
gst_harness_push (h, buf);
buf = gst_harness_pull (h);
fail_unless_equals_int (pts, GST_BUFFER_PTS (buf));
fail_unless_equals_int (dur, GST_BUFFER_DURATION (buf));
fail_unless (!GST_BUFFER_FLAG_IS_SET (buf, GST_BUFFER_FLAG_DISCONT));
gst_buffer_unref (buf);
gst_harness_teardown (h);
}
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GST_END_TEST;
GST_START_TEST (audiodecoder_plc_on_gap_event_with_delay)
{
/* The same thing as in audiodecoder_plc_on_gap_event, but GstAudioDecoder
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subclass delays the decoding
*/
GstClockTime pts0, pts1;
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GstClockTime dur =
gst_util_uint64_scale_round (1, GST_SECOND, TEST_MSECS_PER_SAMPLE);
GstBuffer *buf;
GstHarness *h = setup_audiodecodertester (NULL, NULL);
gst_audio_decoder_set_plc_aware (GST_AUDIO_DECODER (h->element), TRUE);
gst_audio_decoder_set_plc (GST_AUDIO_DECODER (h->element), TRUE);
pts0 = gst_util_uint64_scale_round (0, GST_SECOND, TEST_MSECS_PER_SAMPLE);;
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gst_harness_push (h, create_test_buffer (0));
buf = gst_harness_pull (h);
fail_unless_equals_int (pts0, GST_BUFFER_PTS (buf));
fail_unless_equals_int (dur, GST_BUFFER_DURATION (buf));
fail_unless (GST_BUFFER_FLAG_IS_SET (buf, GST_BUFFER_FLAG_DISCONT));
gst_buffer_unref (buf);
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((GstAudioDecoderTester *) h->element)->delay_decoding = TRUE;
pts0 = gst_util_uint64_scale_round (1, GST_SECOND, TEST_MSECS_PER_SAMPLE);
gst_harness_push_event (h, gst_event_new_gap (pts0, dur));
fail_unless_equals_int (0, gst_harness_buffers_in_queue (h));
pts1 = gst_util_uint64_scale_round (2, GST_SECOND, TEST_MSECS_PER_SAMPLE);
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buf = create_test_buffer (2);
GST_BUFFER_FLAG_SET (buf, GST_BUFFER_FLAG_DISCONT);
gst_harness_push (h, buf);
buf = gst_harness_pull (h);
fail_unless_equals_int (pts0, GST_BUFFER_PTS (buf));
fail_unless_equals_int (dur, GST_BUFFER_DURATION (buf));
fail_unless (!GST_BUFFER_FLAG_IS_SET (buf, GST_BUFFER_FLAG_DISCONT));
gst_buffer_unref (buf);
buf = gst_harness_pull (h);
fail_unless_equals_int (pts1, GST_BUFFER_PTS (buf));
fail_unless_equals_int (dur, GST_BUFFER_DURATION (buf));
fail_unless (!GST_BUFFER_FLAG_IS_SET (buf, GST_BUFFER_FLAG_DISCONT));
gst_buffer_unref (buf);
gst_harness_teardown (h);
}
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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_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_first_data_is_gap);
tcase_add_test (tc, audiodecoder_flush_events_no_buffers);
tcase_add_test (tc, audiodecoder_flush_events);
tcase_add_test (tc, audiodecoder_eos_events_no_buffers);
tcase_add_test (tc, audiodecoder_buffer_after_segment);
tcase_add_test (tc, audiodecoder_output_too_many_frames);
tcase_add_test (tc, audiodecoder_query_caps_with_fixed_caps_peer);
tcase_add_test (tc, audiodecoder_query_caps_with_range_caps_peer);
tcase_add_test (tc, audiodecoder_query_caps_with_custom_getcaps);
tcase_add_test (tc, audiodecoder_tag_handling);
tcase_add_test (tc, audiodecoder_plc_on_gap_event);
tcase_add_test (tc, audiodecoder_plc_on_gap_event_with_delay);
return s;
}
GST_CHECK_MAIN (gst_audiodecoder);