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flvmux: Test to verify flvmux handles DTS with GST_CLOCK_TIME NONE

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https://bugzilla.gnome.org/show_bug.cgi?id=762207
This commit is contained in:
David Buchmann 2016-03-04 09:42:44 +01:00 committed by Sebastian Dröge
parent da5c8a954c
commit 2b8b5f2246
1 changed files with 92 additions and 0 deletions
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92
tests/check/elements/flvmux.c
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@ -1,6 +1,8 @@
/* GStreamer unit tests for flvmux
*
* Copyright (C) 2009 Tim-Philipp Müller <tim centricular net>
* Copyright (C) 2016 Havard Graff <havard@pexip.com>
* Copyright (C) 2016 David Buchmann <david@pexip.com>
*
* This library is free software; you can redistribute it and/or
* modify it under the terms of the GNU Library General Public
@ -333,6 +335,95 @@ GST_START_TEST (test_speex_streamable)
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)
{
@ -351,6 +442,7 @@ flvmux_suite (void)
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;
}