gstreamer/subprojects/gst-plugins-base/tests/check/libs/audioencoder.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_AUDIO_RATE 44100
#define TEST_AUDIO_CHANNELS 2
#define TEST_AUDIO_FORMAT "S16LE"
#define GST_AUDIO_ENCODER_TESTER_TYPE gst_audio_encoder_tester_get_type()
static GType gst_audio_encoder_tester_get_type (void);
typedef struct _GstAudioEncoderTester GstAudioEncoderTester;
typedef struct _GstAudioEncoderTesterClass GstAudioEncoderTesterClass;
struct _GstAudioEncoderTester
{
GstAudioEncoder parent;
};
struct _GstAudioEncoderTesterClass
{
GstAudioEncoderClass parent_class;
};
G_DEFINE_TYPE (GstAudioEncoderTester, gst_audio_encoder_tester,
GST_TYPE_AUDIO_ENCODER);
static gboolean
gst_audio_encoder_tester_start (GstAudioEncoder * enc)
{
return TRUE;
}
static gboolean
gst_audio_encoder_tester_stop (GstAudioEncoder * enc)
{
return TRUE;
}
static gboolean
gst_audio_encoder_tester_set_format (GstAudioEncoder * enc, GstAudioInfo * info)
{
GstCaps *caps;
caps = gst_caps_new_simple ("audio/x-test-custom", "rate", G_TYPE_INT,
TEST_AUDIO_RATE, "channels", G_TYPE_INT, TEST_AUDIO_CHANNELS, NULL);
gst_audio_encoder_set_output_format (enc, caps);
gst_caps_unref (caps);
return TRUE;
}
static GstFlowReturn
gst_audio_encoder_tester_handle_frame (GstAudioEncoder * enc,
GstBuffer * buffer)
{
guint8 *data;
GstMapInfo map;
guint64 input_num;
GstBuffer *output_buffer;
if (buffer == NULL)
return GST_FLOW_OK;
gst_buffer_map (buffer, &map, GST_MAP_READ);
input_num = *((guint64 *) map.data);
gst_buffer_unmap (buffer, &map);
data = g_malloc (sizeof (guint64));
*(guint64 *) data = input_num;
output_buffer = gst_buffer_new_wrapped (data, sizeof (guint64));
GST_BUFFER_PTS (output_buffer) = GST_BUFFER_PTS (buffer);
GST_BUFFER_DURATION (output_buffer) = GST_BUFFER_DURATION (buffer);
return gst_audio_encoder_finish_frame (enc, output_buffer, TEST_AUDIO_RATE);
}
static void
gst_audio_encoder_tester_class_init (GstAudioEncoderTesterClass * klass)
{
GstElementClass *element_class = GST_ELEMENT_CLASS (klass);
GstAudioEncoderClass *audioencoder_class = GST_AUDIO_ENCODER_CLASS (klass);
static GstStaticPadTemplate sink_templ = GST_STATIC_PAD_TEMPLATE ("sink",
GST_PAD_SINK, GST_PAD_ALWAYS,
GST_STATIC_CAPS ("audio/x-raw"));
static GstStaticPadTemplate src_templ = GST_STATIC_PAD_TEMPLATE ("src",
GST_PAD_SRC, GST_PAD_ALWAYS,
GST_STATIC_CAPS ("audio/x-test-custom"));
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,
"AudioEncoderTester", "Encoder/Audio", "yep", "me");
audioencoder_class->start = gst_audio_encoder_tester_start;
audioencoder_class->stop = gst_audio_encoder_tester_stop;
audioencoder_class->handle_frame = gst_audio_encoder_tester_handle_frame;
audioencoder_class->set_format = gst_audio_encoder_tester_set_format;
}
static void
gst_audio_encoder_tester_init (GstAudioEncoderTester * tester)
{
}
static GstHarness *
setup_audioencodertester (void)
{
GstHarness *h;
GstElement *enc;
static GstStaticPadTemplate sinktemplate = GST_STATIC_PAD_TEMPLATE ("sink",
GST_PAD_SINK,
GST_PAD_ALWAYS,
GST_STATIC_CAPS ("audio/x-test-custom")
);
static GstStaticPadTemplate srctemplate = GST_STATIC_PAD_TEMPLATE ("src",
GST_PAD_SRC,
GST_PAD_ALWAYS,
GST_STATIC_CAPS ("audio/x-raw")
);
enc = g_object_new (GST_AUDIO_ENCODER_TESTER_TYPE, NULL);
h = gst_harness_new_full (enc, &srctemplate, "sink", &sinktemplate, "src");
gst_harness_set_src_caps (h,
gst_caps_new_simple ("audio/x-raw",
"rate", G_TYPE_INT, TEST_AUDIO_RATE,
"channels", G_TYPE_INT, TEST_AUDIO_CHANNELS,
"format", G_TYPE_STRING, TEST_AUDIO_FORMAT,
"layout", G_TYPE_STRING, "interleaved", NULL));
gst_object_unref (enc);
return h;
}
static GstBuffer *
create_test_buffer (guint64 num)
{
GstBuffer *buffer;
guint64 *data;
gsize size;
guint64 samples;
samples = TEST_AUDIO_RATE;
size = 2 * 2 * samples;
data = g_malloc0 (size);
*data = num;
buffer = gst_buffer_new_wrapped (data, size);
GST_BUFFER_PTS (buffer) = num * GST_SECOND;
GST_BUFFER_DURATION (buffer) = GST_SECOND;
return buffer;
}
#define NUM_BUFFERS 100
GST_START_TEST (audioencoder_playback)
{
GstBuffer *buffer;
guint64 i;
guint buffers_available;
GstHarness *h = setup_audioencodertester ();
/* push buffers, the data is actually a number so we can track them */
for (i = 0; i < NUM_BUFFERS; i++) {
fail_unless (gst_harness_push (h, create_test_buffer (i)) == GST_FLOW_OK);
}
fail_unless (gst_harness_push_event (h, gst_event_new_eos ()));
/* check that all buffers were received by our source pad */
buffers_available = gst_harness_buffers_in_queue (h);
fail_unless_equals_int (NUM_BUFFERS, buffers_available);
for (i = 0; i < buffers_available; i++) {
GstMapInfo map;
guint64 num;
buffer = gst_harness_pull (h);
gst_buffer_map (buffer, &map, GST_MAP_READ);
num = *(guint64 *) map.data;
fail_unless (i == num);
fail_unless (GST_BUFFER_PTS (buffer) == i * GST_SECOND);
fail_unless (GST_BUFFER_DURATION (buffer) == GST_SECOND);
gst_buffer_unmap (buffer, &map);
gst_buffer_unref (buffer);
}
gst_harness_teardown (h);
}
GST_END_TEST;
GST_START_TEST (audioencoder_flush_events)
{
guint i;
GstHarness *h = setup_audioencodertester ();
/* 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);
}
}
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);
}
{
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);
}
/* 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);
}
gst_harness_teardown (h);
}
GST_END_TEST;
/* make sure tags sent right before eos are pushed */
GST_START_TEST (audioencoder_tags_before_eos)
{
GstTagList *tags;
GstEvent *event;
GstHarness *h = setup_audioencodertester ();
/* push buffer */
fail_unless (gst_harness_push (h, create_test_buffer (0)) == GST_FLOW_OK);
/* clean received events list */
while ((event = gst_harness_try_pull_event (h)))
gst_event_unref (event);
/* push a tag event */
tags = gst_tag_list_new (GST_TAG_COMMENT, "test-comment", NULL);
fail_unless (gst_harness_push_event (h, gst_event_new_tag (tags)));
fail_unless (gst_harness_push_event (h, gst_event_new_eos ()));
/* check that the tag was received */
{
GstEvent *tag_event = gst_harness_pull_event (h);
gchar *str;
fail_unless (GST_EVENT_TYPE (tag_event) == GST_EVENT_TAG);
gst_event_parse_tag (tag_event, &tags);
fail_unless (gst_tag_list_get_string (tags, GST_TAG_COMMENT, &str));
fail_unless (strcmp (str, "test-comment") == 0);
g_free (str);
gst_event_unref (tag_event);
}
gst_harness_teardown (h);
}
GST_END_TEST;
/* make sure events sent right before eos are pushed */
GST_START_TEST (audioencoder_events_before_eos)
{
GstMessage *msg;
GstEvent *event;
GstHarness *h = setup_audioencodertester ();
/* push buffer */
fail_unless (gst_harness_push (h, create_test_buffer (0)) == GST_FLOW_OK);
/* clean received events list */
while ((event = gst_harness_try_pull_event (h)))
gst_event_unref (event);
/* push a serialized event */
msg = gst_message_new_element (GST_OBJECT (h->element),
gst_structure_new_empty ("test"));
fail_unless (gst_harness_push_event (h,
gst_event_new_sink_message ("sink-test", msg)));
gst_message_unref (msg);
fail_unless (gst_harness_push_event (h, gst_event_new_eos ()));
/* check that the tag was received */
{
GstEvent *msg_event = gst_harness_pull_event (h);
const GstStructure *structure;
fail_unless (GST_EVENT_TYPE (msg_event) == GST_EVENT_SINK_MESSAGE);
fail_unless (gst_event_has_name (msg_event, "sink-test"));
gst_event_parse_sink_message (msg_event, &msg);
structure = gst_message_get_structure (msg);
fail_unless (gst_structure_has_name (structure, "test"));
gst_message_unref (msg);
gst_event_unref (msg_event);
}
gst_harness_teardown (h);
}
GST_END_TEST;
static Suite *
gst_audioencoder_suite (void)
{
Suite *s = suite_create ("GstAudioEncoder");
TCase *tc = tcase_create ("general");
suite_add_tcase (s, tc);
tcase_add_test (tc, audioencoder_playback);
tcase_add_test (tc, audioencoder_tags_before_eos);
tcase_add_test (tc, audioencoder_events_before_eos);
tcase_add_test (tc, audioencoder_flush_events);
return s;
}
GST_CHECK_MAIN (gst_audioencoder);