/* GStreamer unit tests for flvmux * * Copyright (C) 2009 Tim-Philipp Müller * Copyright (C) 2016 Havard Graff * Copyright (C) 2016 David Buchmann * * 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 #ifdef HAVE_VALGRIND # include #endif #include #include #include static GstBusSyncReply error_cb (GstBus * bus, GstMessage * msg, gpointer user_data) { if (GST_MESSAGE_TYPE (msg) == GST_MESSAGE_ERROR) { GError *err = NULL; gchar *dbg = NULL; gst_message_parse_error (msg, &err, &dbg); g_error ("ERROR: %s\n%s\n", err->message, dbg); } return GST_BUS_PASS; } static void handoff_cb (GstElement * element, GstBuffer * buf, GstPad * pad, gint * p_counter) { *p_counter += 1; GST_LOG ("counter = %d", *p_counter); } static void mux_pcm_audio (guint num_buffers, guint repeat) { GstElement *src, *sink, *flvmux, *conv, *pipeline; GstPad *sinkpad, *srcpad; gint counter; GST_LOG ("num_buffers = %u", num_buffers); pipeline = gst_pipeline_new ("pipeline"); fail_unless (pipeline != NULL, "Failed to create pipeline!"); /* kids, don't use a sync handler for this at home, really; we do because * we just want to abort and nothing else */ gst_bus_set_sync_handler (GST_ELEMENT_BUS (pipeline), error_cb, NULL, NULL); src = gst_element_factory_make ("audiotestsrc", "audiotestsrc"); fail_unless (src != NULL, "Failed to create 'audiotestsrc' element!"); g_object_set (src, "num-buffers", num_buffers, NULL); conv = gst_element_factory_make ("audioconvert", "audioconvert"); fail_unless (conv != NULL, "Failed to create 'audioconvert' element!"); flvmux = gst_element_factory_make ("flvmux", "flvmux"); fail_unless (flvmux != NULL, "Failed to create 'flvmux' element!"); sink = gst_element_factory_make ("fakesink", "fakesink"); fail_unless (sink != NULL, "Failed to create 'fakesink' element!"); g_object_set (sink, "signal-handoffs", TRUE, NULL); g_signal_connect (sink, "handoff", G_CALLBACK (handoff_cb), &counter); gst_bin_add_many (GST_BIN (pipeline), src, conv, flvmux, sink, NULL); fail_unless (gst_element_link (src, conv)); fail_unless (gst_element_link (flvmux, sink)); /* now link the elements */ sinkpad = gst_element_get_request_pad (flvmux, "audio"); fail_unless (sinkpad != NULL, "Could not get audio request pad"); srcpad = gst_element_get_static_pad (conv, "src"); fail_unless (srcpad != NULL, "Could not get audioconvert's source pad"); fail_unless_equals_int (gst_pad_link (srcpad, sinkpad), GST_PAD_LINK_OK); gst_object_unref (srcpad); gst_object_unref (sinkpad); do { GstStateChangeReturn state_ret; GstMessage *msg; GST_LOG ("repeat=%d", repeat); counter = 0; state_ret = gst_element_set_state (pipeline, GST_STATE_PAUSED); fail_unless (state_ret != GST_STATE_CHANGE_FAILURE); if (state_ret == GST_STATE_CHANGE_ASYNC) { GST_LOG ("waiting for pipeline to reach PAUSED state"); state_ret = gst_element_get_state (pipeline, NULL, NULL, -1); fail_unless_equals_int (state_ret, GST_STATE_CHANGE_SUCCESS); } GST_LOG ("PAUSED, let's do the rest of it"); state_ret = gst_element_set_state (pipeline, GST_STATE_PLAYING); fail_unless (state_ret != GST_STATE_CHANGE_FAILURE); msg = gst_bus_poll (GST_ELEMENT_BUS (pipeline), GST_MESSAGE_EOS, -1); fail_unless (msg != NULL, "Expected EOS message on bus!"); GST_LOG ("EOS"); gst_message_unref (msg); /* should have some output */ fail_unless (counter > 2); fail_unless_equals_int (gst_element_set_state (pipeline, GST_STATE_NULL), GST_STATE_CHANGE_SUCCESS); /* repeat = test re-usability */ --repeat; } while (repeat > 0); gst_object_unref (pipeline); } GST_START_TEST (test_index_writing) { /* note: there's a magic 128 value in flvmux when doing index writing */ if ((__i__ % 33) == 1) mux_pcm_audio (__i__, 2); } GST_END_TEST; static GstBuffer * create_buffer (guint8 * data, gsize size, GstClockTime timestamp, GstClockTime duration) { GstBuffer *buf = gst_buffer_new_wrapped_full (GST_MEMORY_FLAG_READONLY, data, size, 0, size, NULL, NULL); GST_BUFFER_PTS (buf) = timestamp; GST_BUFFER_DTS (buf) = timestamp; GST_BUFFER_DURATION (buf) = duration; GST_BUFFER_OFFSET (buf) = 0; GST_BUFFER_OFFSET_END (buf) = 0; return buf; } GST_START_TEST (test_speex_streamable) { GstBuffer *buf; GstMapInfo map = GST_MAP_INFO_INIT; guint8 header0[] = { 0x53, 0x70, 0x65, 0x65, 0x78, 0x20, 0x20, 0x20, 0x31, 0x2e, 0x32, 0x72, 0x63, 0x31, 0x00, 0x00, 0x00, 0x00, 0x00, 0x00, 0x00, 0x00, 0x00, 0x00, 0x00, 0x00, 0x00, 0x00, 0x01, 0x00, 0x00, 0x00, 0x50, 0x00, 0x00, 0x00, 0x80, 0x3e, 0x00, 0x00, 0x01, 0x00, 0x00, 0x00, 0x04, 0x00, 0x00, 0x00, 0x01, 0x00, 0x00, 0x00, 0xff, 0xff, 0xff, 0xff, 0x40, 0x01, 0x00, 0x00, 0x00, 0x00, 0x00, 0x00, 0x01, 0x00, 0x00, 0x00, 0x00, 0x00, 0x00, 0x00, 0x00, 0x00, 0x00, 0x00, 0x00, 0x00, 0x00, 0x00 }; guint8 header1[] = { 0x1f, 0x00, 0x00, 0x00, 0x45, 0x6e, 0x63, 0x6f, 0x64, 0x65, 0x64, 0x20, 0x77, 0x69, 0x74, 0x68, 0x20, 0x47, 0x53, 0x74, 0x72, 0x65, 0x61, 0x6d, 0x65, 0x72, 0x20, 0x53, 0x70, 0x65, 0x65, 0x78, 0x65, 0x6e, 0x63, 0x00, 0x00, 0x00, 0x00, 0x01 }; guint8 buffer[] = { 0x36, 0x9d, 0x1b, 0x9a, 0x20, 0x00, 0x01, 0x68, 0xe8, 0xe8, 0xe8, 0xe8, 0xe8, 0xe8, 0xe8, 0x84, 0x00, 0xb4, 0x74, 0x74, 0x74, 0x74, 0x74, 0x74, 0x74, 0x42, 0x00, 0x5a, 0x3a, 0x3a, 0x3a, 0x3a, 0x3a, 0x3a, 0x3a, 0x21, 0x00, 0x2d, 0x1d, 0x1d, 0x1d, 0x1d, 0x1d, 0x1d, 0x1d, 0x1b, 0x3b, 0x60, 0xab, 0xab, 0xab, 0xab, 0xab, 0x0a, 0xba, 0xba, 0xba, 0xba, 0xb0, 0xab, 0xab, 0xab, 0xab, 0xab, 0x0a, 0xba, 0xba, 0xba, 0xba, 0xb7 }; GstCaps *caps = gst_caps_new_simple ("audio/x-speex", "rate", G_TYPE_INT, 16000, "channels", G_TYPE_INT, 1, NULL); const GstClockTime base_time = 123456789; const GstClockTime duration_ms = 20; const GstClockTime duration = duration_ms * GST_MSECOND; GstHarness *h = gst_harness_new_with_padnames ("flvmux", "audio", "src"); gst_harness_set_src_caps (h, caps); g_object_set (h->element, "streamable", 1, NULL); /* push speex header0 */ gst_harness_push (h, create_buffer (header0, sizeof (header0), base_time, 0)); /* push speex header1 */ gst_harness_push (h, create_buffer (header1, sizeof (header1), base_time, 0)); /* push speex data */ gst_harness_push (h, create_buffer (buffer, sizeof (buffer), base_time, duration)); /* push speex data 2 */ gst_harness_push (h, create_buffer (buffer, sizeof (buffer), base_time + duration, duration)); /* pull out stream-start event */ gst_event_unref (gst_harness_pull_event (h)); /* pull out caps event */ gst_event_unref (gst_harness_pull_event (h)); /* pull out segment event and verify we are using GST_FORMAT_TIME */ { GstEvent *event = gst_harness_pull_event (h); const GstSegment *segment; gst_event_parse_segment (event, &segment); fail_unless_equals_int (GST_FORMAT_TIME, segment->format); gst_event_unref (event); } /* pull FLV header buffer */ buf = gst_harness_pull (h); gst_buffer_unref (buf); /* pull Metadata buffer */ buf = gst_harness_pull (h); gst_buffer_unref (buf); /* pull header0 */ buf = gst_harness_pull (h); fail_unless_equals_uint64 (base_time, GST_BUFFER_PTS (buf)); fail_unless_equals_uint64 (base_time, GST_BUFFER_DTS (buf)); gst_buffer_map (buf, &map, GST_MAP_READ); /* 0x08 means it is audio */ fail_unless_equals_int (0x08, map.data[0]); /* timestamp should be starting from 0 */ fail_unless_equals_int (0x00, map.data[6]); /* 0xb2 means Speex, 16000Hz, Mono */ fail_unless_equals_int (0xb2, map.data[11]); /* verify content is intact */ fail_unless_equals_int (0, memcmp (&map.data[12], header0, sizeof (header0))); gst_buffer_unmap (buf, &map); gst_buffer_unref (buf); /* pull header1 */ buf = gst_harness_pull (h); fail_unless_equals_uint64 (base_time, GST_BUFFER_PTS (buf)); fail_unless_equals_uint64 (base_time, GST_BUFFER_DTS (buf)); fail_unless_equals_uint64 (0, GST_BUFFER_DURATION (buf)); gst_buffer_map (buf, &map, GST_MAP_READ); /* 0x08 means it is audio */ fail_unless_equals_int (0x08, map.data[0]); /* timestamp should be starting from 0 */ fail_unless_equals_int (0x00, map.data[6]); /* 0xb2 means Speex, 16000Hz, Mono */ fail_unless_equals_int (0xb2, map.data[11]); /* verify content is intact */ fail_unless_equals_int (0, memcmp (&map.data[12], header1, sizeof (header1))); gst_buffer_unmap (buf, &map); gst_buffer_unref (buf); /* pull data */ buf = gst_harness_pull (h); fail_unless_equals_uint64 (base_time, GST_BUFFER_PTS (buf)); fail_unless_equals_uint64 (base_time, GST_BUFFER_DTS (buf)); fail_unless_equals_uint64 (duration, GST_BUFFER_DURATION (buf)); fail_unless_equals_uint64 (GST_BUFFER_OFFSET_NONE, GST_BUFFER_OFFSET (buf)); fail_unless_equals_uint64 (GST_BUFFER_OFFSET_NONE, GST_BUFFER_OFFSET_END (buf)); gst_buffer_map (buf, &map, GST_MAP_READ); /* 0x08 means it is audio */ fail_unless_equals_int (0x08, map.data[0]); /* timestamp should be starting from 0 */ fail_unless_equals_int (0x00, map.data[6]); /* 0xb2 means Speex, 16000Hz, Mono */ fail_unless_equals_int (0xb2, map.data[11]); /* verify content is intact */ fail_unless_equals_int (0, memcmp (&map.data[12], buffer, sizeof (buffer))); gst_buffer_unmap (buf, &map); gst_buffer_unref (buf); /* pull data */ buf = gst_harness_pull (h); fail_unless_equals_uint64 (base_time + duration, GST_BUFFER_PTS (buf)); fail_unless_equals_uint64 (base_time + duration, GST_BUFFER_DTS (buf)); fail_unless_equals_uint64 (duration, GST_BUFFER_DURATION (buf)); fail_unless_equals_uint64 (GST_BUFFER_OFFSET_NONE, GST_BUFFER_OFFSET (buf)); fail_unless_equals_uint64 (GST_BUFFER_OFFSET_NONE, GST_BUFFER_OFFSET_END (buf)); gst_buffer_map (buf, &map, GST_MAP_READ); /* 0x08 means it is audio */ fail_unless_equals_int (0x08, map.data[0]); /* timestamp should reflect the duration_ms */ fail_unless_equals_int (duration_ms, map.data[6]); /* 0xb2 means Speex, 16000Hz, Mono */ fail_unless_equals_int (0xb2, map.data[11]); /* verify content is intact */ fail_unless_equals_int (0, memcmp (&map.data[12], buffer, sizeof (buffer))); gst_buffer_unmap (buf, &map); gst_buffer_unref (buf); gst_harness_teardown (h); } GST_END_TEST; static void check_buf_type_timestamp (GstBuffer *buf, gint packet_type, gint timestamp) { GstMapInfo map = GST_MAP_INFO_INIT; gst_buffer_map (buf, &map, GST_MAP_READ); fail_unless_equals_int (packet_type, map.data[0]); fail_unless_equals_int (timestamp, map.data[6]); gst_buffer_unmap (buf, &map); gst_buffer_unref (buf); } GST_START_TEST(test_increasing_timestamp_when_pts_none) { const gint AUDIO = 0x08; const gint VIDEO = 0x09; gint timestamp = 3; GstClockTime base_time = 42 * GST_SECOND; GstPad *audio_sink, *video_sink, *audio_src, *video_src; GstHarness *h, *audio, *video, *audio_q, *video_q; GstCaps *audio_caps, *video_caps; GstBuffer *buf; h = gst_harness_new_with_padnames ("flvmux", NULL, "src"); audio = gst_harness_new_with_element (h->element, "audio", NULL); video = gst_harness_new_with_element (h->element, "video", NULL); audio_q = gst_harness_new ("queue"); video_q = gst_harness_new ("queue"); audio_sink = GST_PAD_PEER (audio->srcpad); video_sink = GST_PAD_PEER (video->srcpad); audio_src = GST_PAD_PEER (audio_q->sinkpad); video_src = GST_PAD_PEER (video_q->sinkpad); gst_pad_unlink (audio->srcpad, audio_sink); gst_pad_unlink (video->srcpad, video_sink); gst_pad_unlink (audio_src, audio_q->sinkpad); gst_pad_unlink (video_src, video_q->sinkpad); gst_pad_link (audio_src, audio_sink); gst_pad_link (video_src, video_sink); audio_caps = gst_caps_new_simple ("audio/x-speex", "rate", G_TYPE_INT, 16000, "channels", G_TYPE_INT, 1, NULL); gst_harness_set_src_caps (audio_q, audio_caps); video_caps = gst_caps_new_simple ("video/x-h264", "stream-format", G_TYPE_STRING, "avc", NULL); gst_harness_set_src_caps (video_q, video_caps); /* Push audio + video + audio with increasing DTS, but PTS for video is * GST_CLOCK_TIME_NONE */ buf = gst_buffer_new(); GST_BUFFER_DTS (buf) = timestamp * GST_MSECOND + base_time; GST_BUFFER_PTS (buf) = timestamp * GST_MSECOND + base_time; gst_harness_push (audio_q, buf); buf = gst_buffer_new(); GST_BUFFER_DTS (buf) = (timestamp + 1) * GST_MSECOND + base_time; GST_BUFFER_PTS (buf) = GST_CLOCK_TIME_NONE; gst_harness_push (video_q, buf); buf = gst_buffer_new(); GST_BUFFER_DTS (buf) = (timestamp + 2) * GST_MSECOND + base_time; GST_BUFFER_PTS (buf) = (timestamp + 2) * GST_MSECOND + base_time; gst_harness_push (audio_q, buf); /* Pull two metadata packets out */ gst_buffer_unref (gst_harness_pull (h)); gst_buffer_unref (gst_harness_pull (h)); /* Check that we receive the packets in monotonically increasing order and * that their timestamps are correct (should start at 0) */ buf = gst_harness_pull (h); check_buf_type_timestamp (buf, AUDIO, 0); buf = gst_harness_pull (h); check_buf_type_timestamp (buf, VIDEO, 1); /* teardown */ gst_harness_teardown (h); gst_harness_teardown (audio); gst_harness_teardown (video); gst_harness_teardown (audio_q); gst_harness_teardown (video_q); } GST_END_TEST; static Suite * flvmux_suite (void) { Suite *s = suite_create ("flvmux"); TCase *tc_chain = tcase_create ("general"); gint loop = 499; suite_add_tcase (s, tc_chain); #ifdef HAVE_VALGRIND if (RUNNING_ON_VALGRIND) { loop = 140; } #endif tcase_add_loop_test (tc_chain, test_index_writing, 1, loop); tcase_add_test (tc_chain, test_speex_streamable); tcase_add_test (tc_chain, test_increasing_timestamp_when_pts_none); return s; } GST_CHECK_MAIN (flvmux)