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https://gitlab.freedesktop.org/gstreamer/gstreamer.git
synced 2025-02-02 12:32:29 +00:00
tests: fix indentation
This commit is contained in:
parent
567afdd4d3
commit
e6d188967a
5 changed files with 80 additions and 62 deletions
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@ -336,7 +336,7 @@ GST_START_TEST (test_speex_streamable)
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GST_END_TEST;
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static void
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check_buf_type_timestamp (GstBuffer *buf, gint packet_type, gint timestamp)
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check_buf_type_timestamp (GstBuffer * buf, gint packet_type, gint timestamp)
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{
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GstMapInfo map = GST_MAP_INFO_INIT;
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gst_buffer_map (buf, &map, GST_MAP_READ);
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@ -346,7 +346,7 @@ 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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GST_START_TEST(test_increasing_timestamp_when_pts_none)
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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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@ -376,29 +376,26 @@ GST_START_TEST(test_increasing_timestamp_when_pts_none)
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gst_pad_link (video_src, video_sink);
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audio_caps = gst_caps_new_simple ("audio/x-speex",
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"rate", G_TYPE_INT, 16000,
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"channels", G_TYPE_INT, 1,
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NULL);
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"rate", G_TYPE_INT, 16000, "channels", G_TYPE_INT, 1, NULL);
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gst_harness_set_src_caps (audio_q, audio_caps);
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video_caps = gst_caps_new_simple ("video/x-h264",
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"stream-format", G_TYPE_STRING, "avc",
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NULL);
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"stream-format", G_TYPE_STRING, "avc", NULL);
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gst_harness_set_src_caps (video_q, video_caps);
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/* Push audio + video + audio with increasing DTS, but PTS for video is
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* GST_CLOCK_TIME_NONE
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*/
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buf = gst_buffer_new();
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buf = gst_buffer_new ();
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GST_BUFFER_DTS (buf) = timestamp * GST_MSECOND + base_time;
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GST_BUFFER_PTS (buf) = timestamp * GST_MSECOND + base_time;
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gst_harness_push (audio_q, buf);
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buf = gst_buffer_new();
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buf = gst_buffer_new ();
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GST_BUFFER_DTS (buf) = (timestamp + 1) * GST_MSECOND + base_time;
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GST_BUFFER_PTS (buf) = GST_CLOCK_TIME_NONE;
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gst_harness_push (video_q, buf);
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buf = gst_buffer_new();
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buf = gst_buffer_new ();
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GST_BUFFER_DTS (buf) = (timestamp + 2) * GST_MSECOND + base_time;
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GST_BUFFER_PTS (buf) = (timestamp + 2) * GST_MSECOND + base_time;
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gst_harness_push (audio_q, buf);
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@ -422,6 +419,7 @@ GST_START_TEST(test_increasing_timestamp_when_pts_none)
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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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@ -133,9 +133,10 @@ GST_START_TEST (test_h263pay_mode_b_snow)
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if (!have_element ("avenc_h263"))
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return;
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h = gst_harness_new_parse (
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"avenc_h263 rtp-payload-size=1 ! rtph263pay mtu=1350 ");
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gst_harness_add_src_parse (h, "videotestsrc pattern=snow is-live=1 ! "
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h = gst_harness_new_parse
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("avenc_h263 rtp-payload-size=1 ! rtph263pay mtu=1350 ");
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gst_harness_add_src_parse (h,
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"videotestsrc pattern=snow is-live=1 ! "
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"capsfilter caps=\"video/x-raw,format=I420,width=176,height=144\"", TRUE);
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for (i = 0; i < frames; i++)
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@ -144,6 +145,7 @@ GST_START_TEST (test_h263pay_mode_b_snow)
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gst_harness_teardown (h);
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}
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GST_END_TEST;
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/* gst_rtp_buffer_get_payload() may return a copy of the payload. This test
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@ -155,11 +157,14 @@ GST_START_TEST (test_h263pdepay_fragmented_memory_non_writable_buffer)
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GstBuffer *header_buf, *payload_buf, *buf;
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GstRTPBuffer rtp = GST_RTP_BUFFER_INIT;
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guint8 header[] = {
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0x04, 0x00 };
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0x04, 0x00
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};
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guint8 payload[] = {
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0x80, 0x02, 0x1c, 0xb8, 0x01, 0x00, 0x11, 0xe0, 0x44, 0xc4 };
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0x80, 0x02, 0x1c, 0xb8, 0x01, 0x00, 0x11, 0xe0, 0x44, 0xc4
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};
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guint8 frame[] = {
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0x00, 0x00, 0x80, 0x02, 0x1c, 0xb8, 0x01, 0x00, 0x11, 0xe0, 0x44, 0xc4 };
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0x00, 0x00, 0x80, 0x02, 0x1c, 0xb8, 0x01, 0x00, 0x11, 0xe0, 0x44, 0xc4
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};
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h = gst_harness_new ("rtph263pdepay");
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gst_harness_set_src_caps_str (h, "application/x-rtp, media=video, "
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@ -185,22 +190,26 @@ GST_START_TEST (test_h263pdepay_fragmented_memory_non_writable_buffer)
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gst_harness_teardown (h);
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}
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GST_END_TEST;
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/* gst_rtp_buffer_get_payload() may return a copy of the payload. This test
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* makes sure that the rtph263pdepay also produces the correct output in this
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* case. */
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GST_START_TEST (test_h263pdepay_fragmented_memory_non_writable_buffer_split_frame)
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{
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GST_START_TEST
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(test_h263pdepay_fragmented_memory_non_writable_buffer_split_frame) {
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GstHarness *h;
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GstBuffer *header_buf, *payload_buf, *buf;
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GstRTPBuffer rtp = GST_RTP_BUFFER_INIT;
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guint8 header[] = {
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0x04, 0x00 };
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0x04, 0x00
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};
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guint8 payload[] = {
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0x80, 0x02, 0x1c, 0xb8, 0x01, 0x00, 0x11, 0xe0, 0x44, 0xc4 };
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0x80, 0x02, 0x1c, 0xb8, 0x01, 0x00, 0x11, 0xe0, 0x44, 0xc4
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};
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guint8 frame[] = {
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0x00, 0x00, 0x80, 0x02, 0x1c, 0xb8, 0x01, 0x00, 0x11, 0xe0, 0x44, 0xc4 };
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0x00, 0x00, 0x80, 0x02, 0x1c, 0xb8, 0x01, 0x00, 0x11, 0xe0, 0x44, 0xc4
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};
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h = gst_harness_new ("rtph263pdepay");
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gst_harness_set_src_caps_str (h, "application/x-rtp, media=video, "
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@ -244,6 +253,7 @@ GST_START_TEST (test_h263pdepay_fragmented_memory_non_writable_buffer_split_fram
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gst_harness_teardown (h);
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}
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GST_END_TEST;
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GST_START_TEST (test_h263pdepay_dont_push_empty_frame)
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@ -271,6 +281,7 @@ GST_START_TEST (test_h263pdepay_dont_push_empty_frame)
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gst_harness_teardown (h);
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}
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GST_END_TEST;
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static Suite *
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@ -288,8 +299,10 @@ rtph263_suite (void)
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tcase_add_test (tc_chain, test_h263pay_mode_b_snow);
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suite_add_tcase (s, (tc_chain = tcase_create ("h263pdepay")));
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tcase_add_test (tc_chain, test_h263pdepay_fragmented_memory_non_writable_buffer);
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tcase_add_test (tc_chain, test_h263pdepay_fragmented_memory_non_writable_buffer_split_frame);
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tcase_add_test (tc_chain,
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test_h263pdepay_fragmented_memory_non_writable_buffer);
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tcase_add_test (tc_chain,
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test_h263pdepay_fragmented_memory_non_writable_buffer_split_frame);
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tcase_add_test (tc_chain, test_h263pdepay_dont_push_empty_frame);
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return s;
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@ -1158,16 +1158,16 @@ GST_START_TEST (test_rtx_two_missing)
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/*
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The expected sequence of buffers is this:
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____ ____ ____ ____
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____ ____ ____ ____
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... | 10 | | 11 | | 12 | | 13 |
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–––– –––– –––– ––––
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200ms 220ms 240ms 260ms
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–––– –––– –––– ––––
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200ms 220ms 240ms 260ms
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But instead we get this:
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____ _ _ _ _ ____
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____ _ _ _ _ ____
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... | 10 | | | | | | 13 |
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–––– - - - - ––––
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200ms 260ms
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–––– - - - - ––––
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200ms 260ms
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Now it is important to note that the next thing that happens is that
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the RTX timeout for packet 11 will happen at time 230ms, so we crank
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@ -1193,18 +1193,18 @@ GST_START_TEST (test_rtx_two_missing)
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/*
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This will estimate the dts on the two missing packets to:
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____ ____
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____ ____
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... | 11 | | 12 | ...
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–––– ––––
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220ms 240ms
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–––– ––––
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220ms 240ms
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And given their regular interspacing of 20ms, it will schedule two RTX
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timers for them like so:
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____ ____
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____ ____
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... | 11 | | 12 | ...
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–––– ––––
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230ms 250ms
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–––– ––––
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230ms 250ms
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There are however two problems, packet 11 we have already sent one RTX for
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and its timeout is currently at 270ms, so we should not tamper with that,
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@ -1259,16 +1259,16 @@ GST_START_TEST (text_rtx_two_missing_early)
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/*
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The expected sequence of buffers is this:
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___ ___ ___ ___ ___
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___ ___ ___ ___ ___
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| 0 | | 1 | | 2 | | 3 | | 4 |
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––– ––– ––– ––– –––
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0ms 20ms 40ms 60ms 80ms
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––– ––– ––– ––– –––
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0ms 20ms 40ms 60ms 80ms
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But instead we get this:
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___ ___ _ _ _ _ ___
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___ ___ _ _ _ _ ___
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| 0 | | 1 | | | | | | 4 |
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––– ––– – – – – –––
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0ms 20ms 41ms
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––– ––– – – – – –––
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0ms 20ms 41ms
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*/
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@ -1282,26 +1282,26 @@ GST_START_TEST (text_rtx_two_missing_early)
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With the now calculated packet-spacing of (41-20) / 3 = 7,
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giving us these expected times:
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___ ___ ___ ___ ___
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___ ___ ___ ___ ___
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| 0 | | 1 | | 2 | | 3 | | 4 |
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––– ––– ––– ––– –––
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0ms 20ms 27ms 34ms 41ms
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––– ––– ––– ––– –––
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0ms 20ms 27ms 34ms 41ms
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For RTX, the inital RTX-timeouts for the missing buffers are
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the expected arrival time + half the packet-spacing time, like this:
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___ ___
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___ ___
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| 2 | | 3 |
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––– –––
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50ms 70ms
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––– –––
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50ms 70ms
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But since we have re-calculated the estimated arrival-time
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of these buffers, we have to adjust the RTX timeout as well,
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and we use the original delay (packet-spacing / 2) = 10ms,
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and add it on:
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___ ___
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___ ___
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| 2 | | 3 |
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––– –––
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37ms 44ms
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––– –––
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37ms 44ms
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Also note that the first RTX request is now scheduled for a
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time that is prior to NOW (37ms < 41ms), so it will be sent straight
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@ -1796,7 +1796,8 @@ GST_START_TEST (test_rtx_duplicate_packet_updates_rtx_stats)
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"rtx-per-packet", G_TYPE_DOUBLE, 1.0,
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"rtx-rtt", G_TYPE_UINT64, (guint64)
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/* Use the rtx-rtt formula. Can be subject to change though. */
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((now - rtx_request_6) + 47 * (now - rtx_request_7)) / 48, NULL)));
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((now - rtx_request_6) + 47 * (now - rtx_request_7)) / 48,
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NULL)));
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gst_object_unref (testclock);
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gst_harness_teardown (h);
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@ -2490,7 +2491,8 @@ rtpjitterbuffer_suite (void)
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tcase_add_loop_test (tc_chain, test_num_late_when_considered_lost_arrives, 0,
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2);
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tcase_add_test (tc_chain, test_reorder_of_non_equidistant_packets);
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tcase_add_test (tc_chain, test_loss_equidistant_spacing_with_parameter_packets);
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tcase_add_test (tc_chain,
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test_loss_equidistant_spacing_with_parameter_packets);
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tcase_add_test (tc_chain, test_rtx_expected_next);
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tcase_add_test (tc_chain, test_rtx_two_missing);
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@ -2500,7 +2502,8 @@ rtpjitterbuffer_suite (void)
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tcase_add_test (tc_chain, test_rtx_buffer_arrives_too_late);
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tcase_add_test (tc_chain, test_rtx_original_buffer_does_not_update_rtx_stats);
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tcase_add_test (tc_chain, test_rtx_duplicate_packet_updates_rtx_stats);
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tcase_add_test (tc_chain, test_rtx_buffer_arrives_after_lost_updates_rtx_stats);
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tcase_add_test (tc_chain,
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test_rtx_buffer_arrives_after_lost_updates_rtx_stats);
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tcase_add_test (tc_chain, test_rtx_rtt_larger_than_retry_timeout);
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tcase_add_test (tc_chain, test_rtx_no_request_if_time_past_retry_period);
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@ -669,25 +669,27 @@ stats_test_cb (GObject * object, GParamSpec * spec, gpointer data)
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*cb_called = TRUE;
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/* We should be able to get a rtpsession property
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without introducing the deadlock */
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without introducing the deadlock */
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g_object_get (object, "num-sources", &num_sources, NULL);
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}
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GST_START_TEST (test_dont_lock_on_stats)
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{
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GstHarness * h_rtcp;
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GstHarness * h_send;
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GstClock * clock = gst_test_clock_new ();
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GstTestClock * testclock = GST_TEST_CLOCK (clock);
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GstHarness *h_rtcp;
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GstHarness *h_send;
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GstClock *clock = gst_test_clock_new ();
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GstTestClock *testclock = GST_TEST_CLOCK (clock);
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gboolean cb_called = FALSE;
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/* use testclock as the systemclock to capture the rtcp thread waits */
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gst_system_clock_set_default (GST_CLOCK (testclock));
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h_rtcp = gst_harness_new_with_padnames (
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"rtpsession", "recv_rtcp_sink", "send_rtcp_src");
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h_send = gst_harness_new_with_element (
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h_rtcp->element, "send_rtp_sink", "send_rtp_src");
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h_rtcp =
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gst_harness_new_with_padnames ("rtpsession", "recv_rtcp_sink",
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"send_rtcp_src");
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h_send =
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gst_harness_new_with_element (h_rtcp->element, "send_rtp_sink",
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"send_rtp_src");
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/* connect to the stats-reporting */
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g_signal_connect (h_rtcp->element, "notify::stats",
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@ -702,6 +704,7 @@ GST_START_TEST (test_dont_lock_on_stats)
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gst_harness_teardown (h_rtcp);
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gst_object_unref (clock);
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}
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GST_END_TEST;
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static void
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@ -93,6 +93,7 @@ GST_START_TEST (test_depay_non_flexible_mode)
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gst_harness_teardown (h);
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}
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GST_END_TEST;
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