/* GStreamer * * Copyright (C) 2014 Samsung Electronics. All rights reserved. * Author: Thiago Santos * * 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 #include #include #include static GstPad *mysrcpad, *mysinkpad; static GstElement *enc; static GList *events = NULL; #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_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, "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 gboolean _mysinkpad_event (GstPad * pad, GstObject * parent, GstEvent * event) { events = g_list_append (events, event); return TRUE; } static void setup_audioencodertester (void) { 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); mysrcpad = gst_check_setup_src_pad (enc, &srctemplate); mysinkpad = gst_check_setup_sink_pad (enc, &sinktemplate); gst_pad_set_event_function (mysinkpad, _mysinkpad_event); } static void cleanup_audioencodertest (void) { gst_pad_set_active (mysrcpad, FALSE); gst_pad_set_active (mysinkpad, FALSE); gst_element_set_state (enc, GST_STATE_NULL); gst_check_teardown_src_pad (enc); gst_check_teardown_sink_pad (enc); gst_check_teardown_element (enc); g_list_free_full (events, (GDestroyNotify) gst_event_unref); events = NULL; } 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; } 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-raw", "rate", G_TYPE_INT, TEST_AUDIO_RATE, "channels", G_TYPE_INT, TEST_AUDIO_CHANNELS, "format", G_TYPE_STRING, "S16LE", "layout", G_TYPE_STRING, "interleaved", NULL); fail_unless (gst_pad_push_event (mysrcpad, gst_event_new_caps (caps))); gst_caps_unref (caps); } #define NUM_BUFFERS 100 GST_START_TEST (audioencoder_playback) { GstSegment segment; GstBuffer *buffer; guint64 i; GList *iter; setup_audioencodertester (); gst_pad_set_active (mysrcpad, TRUE); gst_element_set_state (enc, 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++) { buffer = create_test_buffer (i); fail_unless (gst_pad_push (mysrcpad, buffer) == GST_FLOW_OK); } fail_unless (gst_pad_push_event (mysrcpad, gst_event_new_eos ())); /* check that all buffers were received by our source pad */ fail_unless (g_list_length (buffers) == NUM_BUFFERS); i = 0; for (iter = buffers; iter; iter = g_list_next (iter)) { GstMapInfo map; guint64 num; buffer = iter->data; 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); i++; } g_list_free_full (buffers, (GDestroyNotify) gst_buffer_unref); buffers = NULL; cleanup_audioencodertest (); } GST_END_TEST; /* make sure tags sent right before eos are pushed */ GST_START_TEST (audioencoder_tags_before_eos) { GstSegment segment; GstBuffer *buffer; GstTagList *tags; setup_audioencodertester (); gst_pad_set_active (mysrcpad, TRUE); gst_element_set_state (enc, 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 buffer */ buffer = create_test_buffer (0); fail_unless (gst_pad_push (mysrcpad, buffer) == GST_FLOW_OK); /* clean received events list */ g_list_free_full (events, (GDestroyNotify) gst_event_unref); events = NULL; /* push a tag event */ tags = gst_tag_list_new (GST_TAG_COMMENT, "test-comment", NULL); fail_unless (gst_pad_push_event (mysrcpad, gst_event_new_tag (tags))); fail_unless (gst_pad_push_event (mysrcpad, gst_event_new_eos ())); /* check that the tag was received */ { GstEvent *tag_event = events->data; 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); } g_list_free_full (buffers, (GDestroyNotify) gst_buffer_unref); buffers = NULL; g_list_free_full (events, (GDestroyNotify) gst_event_unref); events = NULL; cleanup_audioencodertest (); } GST_END_TEST; /* make sure events sent right before eos are pushed */ GST_START_TEST (audioencoder_events_before_eos) { GstSegment segment; GstBuffer *buffer; GstMessage *msg; setup_audioencodertester (); gst_pad_set_active (mysrcpad, TRUE); gst_element_set_state (enc, 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 buffer */ buffer = create_test_buffer (0); fail_unless (gst_pad_push (mysrcpad, buffer) == GST_FLOW_OK); /* clean received events list */ g_list_free_full (events, (GDestroyNotify) gst_event_unref); events = NULL; /* push a serialized event */ 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 ("sink-test", msg))); gst_message_unref (msg); fail_unless (gst_pad_push_event (mysrcpad, gst_event_new_eos ())); /* check that the tag was received */ { GstEvent *msg_event = events->data; 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); } g_list_free_full (buffers, (GDestroyNotify) gst_buffer_unref); buffers = NULL; g_list_free_full (events, (GDestroyNotify) gst_event_unref); events = NULL; cleanup_audioencodertest (); } 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); return s; } GST_CHECK_MAIN (gst_audioencoder);