mirror of
https://gitlab.freedesktop.org/gstreamer/gstreamer.git
synced 2024-11-24 10:41:04 +00:00
tests: flvmux: Add test for rollover timestamp
The timestamps that exceed uint32 maximum value should be handled to rollover.
This commit is contained in:
parent
e836640bd5
commit
9feb35638a
1 changed files with 107 additions and 2 deletions
|
@ -935,8 +935,8 @@ GST_END_TEST;
|
|||
typedef struct
|
||||
{
|
||||
guint media_type;
|
||||
gint ts; /* timestamp in ms */
|
||||
gint rt; /* running_time in ms */
|
||||
guint64 ts; /* timestamp in ms */
|
||||
guint64 rt; /* running_time in ms */
|
||||
} InputData;
|
||||
|
||||
GST_START_TEST (test_incrementing_timestamps)
|
||||
|
@ -1034,6 +1034,110 @@ GST_START_TEST (test_incrementing_timestamps)
|
|||
|
||||
GST_END_TEST;
|
||||
|
||||
GST_START_TEST (test_rollover_timestamps)
|
||||
{
|
||||
GstPad *audio_sink, *video_sink, *audio_src, *video_src;
|
||||
GstHarness *h, *audio, *video, *audio_q, *video_q;
|
||||
GstTestClock *tclock;
|
||||
guint i;
|
||||
guint64 rollover_pts = (guint64) G_MAXUINT32 + 100;
|
||||
InputData input[] = {
|
||||
{AUDIO, 0, 1}
|
||||
,
|
||||
{VIDEO, 0, 2}
|
||||
,
|
||||
{VIDEO, (guint64) G_MAXUINT32 - 100, (guint64) G_MAXUINT32 - 99}
|
||||
,
|
||||
{AUDIO, (guint64) G_MAXUINT32 - 95, (guint64) G_MAXUINT32 - 90}
|
||||
,
|
||||
{AUDIO, rollover_pts, (guint64) G_MAXUINT32 + 110}
|
||||
,
|
||||
};
|
||||
|
||||
/* setup flvmuxer with queues in front */
|
||||
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);
|
||||
g_object_set (h->element, "streamable", TRUE, NULL);
|
||||
|
||||
gst_harness_set_src_caps_str (audio_q,
|
||||
"audio/mpeg, mpegversion=(int)4, "
|
||||
"rate=(int)44100, channels=(int)1, "
|
||||
"stream-format=(string)raw, codec_data=(buffer)1208");
|
||||
|
||||
gst_harness_set_src_caps_str (video_q,
|
||||
"video/x-h264, stream-format=(string)avc, alignment=(string)au, "
|
||||
"codec_data=(buffer)0142c00dffe1000d6742c00d95a0507c807844235001000468ce3c80");
|
||||
|
||||
tclock = gst_harness_get_testclock (h);
|
||||
|
||||
for (i = 0; i < G_N_ELEMENTS (input); i++) {
|
||||
InputData *d = &input[i];
|
||||
GstBuffer *buf = gst_buffer_new ();
|
||||
GstClockTime now = d->rt * GST_MSECOND;
|
||||
GstClockID pending, res;
|
||||
|
||||
GST_BUFFER_DTS (buf) = GST_BUFFER_PTS (buf) = d->ts * GST_MSECOND;
|
||||
GST_DEBUG ("Push media=%u, pts=%" G_GUINT64_FORMAT " (%" GST_TIME_FORMAT
|
||||
")", d->media_type, d->ts, GST_TIME_ARGS (GST_BUFFER_PTS (buf)));
|
||||
gst_test_clock_set_time (tclock, now);
|
||||
|
||||
if (d->media_type == AUDIO)
|
||||
gst_harness_push (audio_q, buf);
|
||||
else
|
||||
gst_harness_push (video_q, buf);
|
||||
|
||||
gst_test_clock_wait_for_next_pending_id (tclock, &pending);
|
||||
res = gst_test_clock_process_next_clock_id (tclock);
|
||||
gst_clock_id_unref (pending);
|
||||
gst_clock_id_unref (res);
|
||||
}
|
||||
|
||||
/* pull the flv metadata */
|
||||
gst_buffer_unref (gst_harness_pull (h));
|
||||
gst_buffer_unref (gst_harness_pull (h));
|
||||
gst_buffer_unref (gst_harness_pull (h));
|
||||
gst_buffer_unref (gst_harness_pull (h));
|
||||
|
||||
/* verify rollover pts in the flvheader is handled */
|
||||
for (i = 0; i < G_N_ELEMENTS (input); i++) {
|
||||
GstBuffer *buf = gst_harness_pull (h);
|
||||
GstMapInfo map;
|
||||
guint32 pts, pts_ext;
|
||||
gst_buffer_map (buf, &map, GST_MAP_READ);
|
||||
pts = GST_READ_UINT24_BE (map.data + 4);
|
||||
pts_ext = GST_READ_UINT8 (map.data + 7);
|
||||
pts |= pts_ext << 24;
|
||||
GST_DEBUG ("media=%u, pts=%u (%" GST_TIME_FORMAT ")",
|
||||
map.data[0], pts, GST_TIME_ARGS (pts * GST_MSECOND));
|
||||
fail_unless (pts == (guint32) input[i].ts);
|
||||
gst_buffer_unmap (buf, &map);
|
||||
gst_buffer_unref (buf);
|
||||
}
|
||||
|
||||
/* teardown */
|
||||
gst_object_unref (tclock);
|
||||
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)
|
||||
{
|
||||
|
@ -1059,6 +1163,7 @@ flvmux_suite (void)
|
|||
tcase_add_test (tc_chain, test_audio_caps_change_streamable_single);
|
||||
tcase_add_test (tc_chain, test_video_caps_change_streamable_single);
|
||||
tcase_add_test (tc_chain, test_incrementing_timestamps);
|
||||
tcase_add_test (tc_chain, test_rollover_timestamps);
|
||||
|
||||
return s;
|
||||
}
|
||||
|
|
Loading…
Reference in a new issue