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flvmux: Test that timestamps are always increasing
Decreasing timestamps break rtmpsink. With contributions from Olivier Crête. https://bugzilla.gnome.org/show_bug.cgi?id=796382
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1 changed files with 106 additions and 2 deletions
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@ -370,10 +370,11 @@ check_buf_type_timestamp (GstBuffer * buf, gint packet_type, gint timestamp)
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gst_buffer_unref (buf);
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}
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static const gint AUDIO = 0x08;
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static const gint VIDEO = 0x09;
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GST_START_TEST (test_increasing_timestamp_when_pts_none)
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{
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const gint AUDIO = 0x08;
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const gint VIDEO = 0x09;
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gint timestamp = 3;
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GstClockTime base_time = 42 * GST_SECOND;
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GstPad *audio_sink, *video_sink, *audio_src, *video_src;
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@ -774,6 +775,108 @@ GST_START_TEST (test_audio_caps_change_streamable)
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GST_END_TEST;
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typedef struct
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{
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guint media_type;
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gint ts; /* timestamp in ms */
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gint rt; /* running_time in ms */
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} InputData;
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GST_START_TEST (test_incrementing_timestamps)
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{
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GstPad *audio_sink, *video_sink, *audio_src, *video_src;
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GstHarness *h, *audio, *video, *audio_q, *video_q;
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GstTestClock *tclock;
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guint i;
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guint32 prev_pts;
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InputData input[] = {
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{AUDIO, 155, 175},
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{VIDEO, 156, 191},
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{VIDEO, 190, 191},
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{AUDIO, 176, 195},
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{AUDIO, 197, 215},
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};
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/* setup flvmuxer with queues in front */
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h = gst_harness_new_with_padnames ("flvmux", NULL, "src");
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audio = gst_harness_new_with_element (h->element, "audio", NULL);
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video = gst_harness_new_with_element (h->element, "video", NULL);
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audio_q = gst_harness_new ("queue");
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video_q = gst_harness_new ("queue");
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audio_sink = GST_PAD_PEER (audio->srcpad);
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video_sink = GST_PAD_PEER (video->srcpad);
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audio_src = GST_PAD_PEER (audio_q->sinkpad);
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video_src = GST_PAD_PEER (video_q->sinkpad);
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gst_pad_unlink (audio->srcpad, audio_sink);
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gst_pad_unlink (video->srcpad, video_sink);
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gst_pad_unlink (audio_src, audio_q->sinkpad);
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gst_pad_unlink (video_src, video_q->sinkpad);
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gst_pad_link (audio_src, audio_sink);
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gst_pad_link (video_src, video_sink);
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g_object_set (h->element, "streamable", TRUE, NULL);
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gst_harness_set_src_caps_str (audio_q,
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"audio/mpeg, mpegversion=(int)4, "
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"rate=(int)44100, channels=(int)1, "
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"stream-format=(string)raw, codec_data=(buffer)1208");
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gst_harness_set_src_caps_str (video_q,
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"video/x-h264, stream-format=(string)avc, alignment=(string)au, "
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"codec_data=(buffer)0142c00dffe1000d6742c00d95a0507c807844235001000468ce3c80");
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tclock = gst_harness_get_testclock (h);
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for (i = 0; i < G_N_ELEMENTS (input); i++) {
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InputData *d = &input[i];
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GstBuffer *buf = gst_buffer_new ();
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GstClockTime now = d->rt * GST_MSECOND;
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GstClockID pending, res;
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GST_BUFFER_DTS (buf) = GST_BUFFER_PTS (buf) = d->ts * GST_MSECOND;
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gst_test_clock_set_time (tclock, now);
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if (d->media_type == AUDIO)
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gst_harness_push (audio_q, buf);
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else
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gst_harness_push (video_q, buf);
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gst_test_clock_wait_for_next_pending_id (tclock, &pending);
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res = gst_test_clock_process_next_clock_id (tclock);
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gst_clock_id_unref (pending);
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gst_clock_id_unref (res);
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}
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/* pull the flv metadata */
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gst_buffer_unref (gst_harness_pull (h));
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gst_buffer_unref (gst_harness_pull (h));
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gst_buffer_unref (gst_harness_pull (h));
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/* verify pts in the flvheader is increasing */
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prev_pts = 0;
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for (i = 0; i < G_N_ELEMENTS (input); i++) {
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GstBuffer *buf = gst_harness_pull (h);
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GstMapInfo map;
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guint32 pts;
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gst_buffer_map (buf, &map, GST_MAP_READ);
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pts = GST_READ_UINT24_BE (map.data + 4);
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GST_DEBUG ("media=%u, pts = %u\n", map.data[0], pts);
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fail_unless (pts >= prev_pts);
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prev_pts = pts;
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gst_buffer_unmap (buf, &map);
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gst_buffer_unref (buf);
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}
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/* teardown */
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gst_object_unref (tclock);
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gst_harness_teardown (h);
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gst_harness_teardown (audio);
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gst_harness_teardown (video);
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gst_harness_teardown (audio_q);
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gst_harness_teardown (video_q);
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}
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GST_END_TEST;
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static Suite *
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flvmux_suite (void)
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{
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@ -796,6 +899,7 @@ flvmux_suite (void)
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tcase_add_test (tc_chain, test_video_caps_late);
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tcase_add_test (tc_chain, test_audio_caps_change_streamable);
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tcase_add_test (tc_chain, test_video_caps_change_streamable);
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tcase_add_test (tc_chain, test_incrementing_timestamps);
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return s;
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}
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