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rtphdrext-rfc6464: Add test for inserting it based on caps
Tests adding the extension based on the caps. Part-of: <https://gitlab.freedesktop.org/gstreamer/gst-plugins-good/-/merge_requests/1058>
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@ -24,6 +24,7 @@
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#include <gst/rtp/rtp.h>
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#include <gst/rtp/rtp.h>
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#include <gst/sdp/gstsdpmessage.h>
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#include <gst/sdp/gstsdpmessage.h>
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#include <gst/audio/audio.h>
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#include <gst/audio/audio.h>
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#include <gst/check/gstharness.h>
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#define URN "urn:ietf:params:rtp-hdrext:ssrc-audio-level"
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#define URN "urn:ietf:params:rtp-hdrext:ssrc-audio-level"
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@ -270,6 +271,38 @@ GST_START_TEST (rtprfc6464_no_meta)
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GST_END_TEST;
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GST_END_TEST;
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GST_START_TEST (rtprfc6464_payloader_depayloader)
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{
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GstHarness *h;
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GstBuffer *b;
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GstFlowReturn fret;
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GstAudioLevelMeta *meta;
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h = gst_harness_new_parse ("rtpL16pay ! "
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"application/x-rtp, extmap-1=(string)< \"\", " URN " , \"vad=on\" >"
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" ! rtpL16depay");
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gst_harness_set_src_caps_str (h, "audio/x-raw, rate=44100, channels=1,"
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" layout=interleaved, format=S16BE");
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b = gst_buffer_new_allocate (NULL, 100, NULL);
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gst_buffer_add_audio_level_meta (b, 12, TRUE);
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fret = gst_harness_push (h, b);
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fail_unless (fret == GST_FLOW_OK);
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b = gst_harness_pull (h);
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meta = gst_buffer_get_audio_level_meta (b);
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fail_unless (meta != NULL);
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fail_unless (meta->level == 12);
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fail_unless (meta->voice_activity == TRUE);
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gst_buffer_unref (b);
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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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static Suite *
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rtprfc6464_suite (void)
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rtprfc6464_suite (void)
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{
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{
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@ -282,6 +315,7 @@ rtprfc6464_suite (void)
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tcase_add_test (tc_chain, rtprfc6464_one_byte);
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tcase_add_test (tc_chain, rtprfc6464_one_byte);
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tcase_add_test (tc_chain, rtprfc6464_two_bytes);
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tcase_add_test (tc_chain, rtprfc6464_two_bytes);
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tcase_add_test (tc_chain, rtprfc6464_no_meta);
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tcase_add_test (tc_chain, rtprfc6464_no_meta);
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tcase_add_test (tc_chain, rtprfc6464_payloader_depayloader);
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return s;
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return s;
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}
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}
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