| /* 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 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; |
| }; |
| |
| 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; |
| |
| 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 (GstStaticPadTemplate * sinktemplate, |
| GstStaticPadTemplate * srctemplate) |
| { |
| if (sinktemplate == NULL) |
| sinktemplate = &sinktemplate_default; |
| if (srctemplate == NULL) |
| srctemplate = &srctemplate_default; |
| |
| 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))); |
| gst_caps_unref (caps); |
| } |
| |
| #define NUM_BUFFERS 1000 |
| GST_START_TEST (audiodecoder_playback) |
| { |
| GstSegment segment; |
| GstBuffer *buffer; |
| guint64 i; |
| |
| setup_audiodecodertester (NULL, NULL); |
| |
| 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 (NULL, NULL); |
| |
| 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 (NULL, NULL); |
| |
| 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 (NULL, NULL); |
| |
| ((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; |
| guint i; |
| GList *events_iter; |
| GstMessage *msg; |
| |
| setup_audiodecodertester (NULL, NULL); |
| |
| 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 = g_list_last (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 (events, (GDestroyNotify) gst_event_unref); |
| events = NULL; |
| |
| g_list_free_full (buffers, (GDestroyNotify) gst_buffer_unref); |
| buffers = NULL; |
| |
| gst_element_set_state (dec, GST_STATE_NULL); |
| cleanup_audiodecodertest (); |
| } |
| |
| /* An element should always push its segment before sending EOS */ |
| GST_START_TEST (audiodecoder_eos_events_no_buffers) |
| { |
| GstSegment segment; |
| setup_audiodecodertester (NULL, NULL); |
| |
| gst_pad_set_active (mysrcpad, TRUE); |
| gst_element_set_state (dec, GST_STATE_PLAYING); |
| gst_pad_set_active (mysinkpad, TRUE); |
| send_startup_events (); |
| |
| gst_segment_init (&segment, GST_FORMAT_TIME); |
| fail_unless (gst_pad_push_event (mysrcpad, gst_event_new_segment (&segment))); |
| fail_unless (gst_pad_push_event (mysrcpad, gst_event_new_eos ())); |
| |
| fail_unless (GST_PAD_IS_EOS (mysinkpad)); |
| |
| { |
| GstEvent *segment_event = |
| gst_pad_get_sticky_event (mysinkpad, GST_EVENT_SEGMENT, 0); |
| fail_unless (segment_event != NULL); |
| gst_event_unref (segment_event); |
| } |
| |
| gst_element_set_state (dec, GST_STATE_NULL); |
| cleanup_audiodecodertest (); |
| } |
| |
| GST_END_TEST; |
| |
| 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 (NULL, NULL); |
| |
| 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 (NULL, NULL); |
| |
| ((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; |
| |
| GST_START_TEST (audiodecoder_query_caps_with_fixed_caps_peer) |
| { |
| GstCaps *caps; |
| GstCaps *filter; |
| GstStructure *structure; |
| gint rate, channels; |
| |
| setup_audiodecodertester (&sinktemplate_restricted, NULL); |
| |
| gst_pad_set_active (mysrcpad, TRUE); |
| gst_element_set_state (dec, GST_STATE_PLAYING); |
| gst_pad_set_active (mysinkpad, TRUE); |
| |
| caps = gst_pad_peer_query_caps (mysrcpad, 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 (mysrcpad, filter); |
| fail_unless (caps != NULL); |
| fail_unless (gst_caps_is_empty (caps)); |
| gst_caps_unref (caps); |
| gst_caps_unref (filter); |
| |
| cleanup_audiodecodertest (); |
| } |
| |
| 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; |
| |
| setup_audiodecodertester (&sinktemplate_with_range, NULL); |
| |
| gst_pad_set_active (mysrcpad, TRUE); |
| gst_element_set_state (dec, GST_STATE_PLAYING); |
| gst_pad_set_active (mysinkpad, TRUE); |
| |
| caps = gst_pad_peer_query_caps (mysrcpad, 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 (mysrcpad, 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 (mysrcpad, filter); |
| fail_unless (caps != NULL); |
| fail_unless (gst_caps_is_empty (caps)); |
| gst_caps_unref (caps); |
| gst_caps_unref (filter); |
| |
| cleanup_audiodecodertest (); |
| } |
| |
| 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; |
| |
| setup_audiodecodertester (&sinktemplate_restricted, NULL); |
| |
| klass = GST_AUDIO_DECODER_CLASS (GST_AUDIO_DECODER_GET_CLASS (dec)); |
| klass->getcaps = _custom_audio_decoder_getcaps; |
| |
| gst_pad_set_active (mysrcpad, TRUE); |
| gst_element_set_state (dec, GST_STATE_PLAYING); |
| gst_pad_set_active (mysinkpad, TRUE); |
| |
| caps = gst_pad_peer_query_caps (mysrcpad, 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); |
| |
| 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_eos_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); |
| |
| 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); |
| |
| return s; |
| } |
| |
| GST_CHECK_MAIN (gst_audiodecoder); |