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https://gitlab.freedesktop.org/gstreamer/gstreamer.git
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d9aaa15a30
There is a use-case for a server to re-payload opus going through it. Problem was that the payloader requires channels in the caps, but this is not something the depayloader can parse out of the stream, meaning caps-negotiation would fail. Removing the requirement of channels in the template-caps fixes this.
92 lines
3.2 KiB
C
92 lines
3.2 KiB
C
/* GStreamer
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*
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* Copyright (C) 2020 Pexip AS
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* @author Havard Graff <havard@pexip.com>
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*
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* This library is free software; you can redistribute it and/or
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* modify it under the terms of the GNU Library General Public
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* License as published by the Free Software Foundation; either
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* version 2 of the License, or (at your option) any later version.
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*
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* This library is distributed in the hope that it will be useful,
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* but WITHOUT ANY WARRANTY; without even the implied warranty of
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* MERCHANTABILITY or FITNESS FOR A PARTICULAR PURPOSE. See the GNU
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* Library General Public License for more details.
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*
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* You should have received a copy of the GNU Library General Public
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* License along with this library; if not, write to the
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* Free Software Foundation, Inc., 51 Franklin St, Fifth Floor,
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* Boston, MA 02110-1301, USA.
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*/
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#include <gst/check/check.h>
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#include <gst/check/gstharness.h>
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#include <gst/rtp/gstrtpbuffer.h>
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#define buffer_from_array(a) gst_buffer_new_wrapped (g_memdup (a, G_N_ELEMENTS (a)), G_N_ELEMENTS (a));
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static guint8 opus_data[] = {
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0xf8, 0xb5, 0x0e, 0x7d, 0x91, 0xcc, 0x05, 0x82,
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0x75, 0x72, 0x48, 0xbd, 0xd3, 0x22, 0x24, 0x2e,
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0x59, 0x63, 0xf8, 0xff, 0x5d, 0x59, 0x27, 0xd8,
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0xad, 0x4b, 0xe8, 0xd7, 0xfa, 0x99, 0xaa, 0x46,
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0xb4, 0xf6, 0x29, 0x16, 0x21, 0x86, 0x2a, 0xb5,
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0x83, 0x7d, 0x3a, 0xce, 0xb3, 0xee, 0x37, 0x3b,
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0xf7, 0xb5, 0x03, 0xe7, 0x13, 0x3b, 0xf6, 0x90,
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0x06, 0xea, 0x79, 0xbe, 0x89, 0xc3, 0x2b, 0x1f,
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0x7f, 0x88, 0x5e, 0xe0, 0xe1, 0x88, 0x59, 0x47,
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0x11, 0x10, 0x94, 0xab, 0x5d, 0xa6, 0x3f, 0x5d,
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0xa7, 0xd7, 0x0e, 0x7d, 0x07, 0x85, 0x0d, 0x2f,
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0x7b, 0x3f, 0xf7, 0xc1, 0x8c, 0xb2, 0xda, 0xac,
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0x79, 0x15, 0xda, 0xc7, 0xd2, 0x6e, 0xcc, 0x88,
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0x61, 0x29, 0xcd, 0x78, 0xf4, 0x6d, 0x1b, 0xa6,
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0xe6, 0xd1, 0x7c, 0x76, 0xc2, 0x86, 0x78, 0x3c,
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0xc2, 0x2e, 0x26, 0xd4, 0xdf, 0x7f, 0x3b, 0x98,
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0x7a, 0x7c, 0xbe, 0x1a, 0x17, 0xd2, 0x2d, 0xa5,
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0x90, 0x2a, 0x1b, 0x0b, 0x43, 0x65, 0x63, 0x37,
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0xe5, 0x0d, 0x5c, 0x9c, 0x6c, 0x38, 0xef, 0x2a,
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0xe8, 0x49, 0x47, 0x05, 0x6d, 0x83, 0xcf, 0x6d,
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};
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GST_START_TEST (test_pay_to_depay)
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{
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GstHarness *h = gst_harness_new_parse ("rtpopuspay ! rtpopusdepay");
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GstBuffer *buf = buffer_from_array (opus_data);
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gst_harness_set_src_caps_str (h, "audio/x-opus,channel-mapping-family=0");
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fail_unless_equals_int (GST_FLOW_OK, gst_harness_push (h, buf));
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gst_buffer_unref (gst_harness_pull (h));
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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_depay_to_pay)
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{
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GstHarness *h = gst_harness_new_parse ("rtpopusdepay ! rtpopuspay");
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guint8 opus_rtp_buf[] = {
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0x80, 0x60, 0x54, 0xfd, 0x3b, 0x5a, 0x93, 0xf9, 0x1c, 0x33, 0x2b, 0xbb,
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};
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GstBuffer *buf = buffer_from_array (opus_rtp_buf);
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gst_harness_set_src_caps_str (h,
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"application/x-rtp,encoding-name=OPUS,media=audio,clock-rate=48000,payload=96");
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fail_unless_equals_int (GST_FLOW_OK, gst_harness_push (h, buf));
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gst_buffer_unref (gst_harness_pull (h));
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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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rtpopus_suite (void)
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{
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Suite *s = suite_create ("rtpopus");
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TCase *tc_chain;
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suite_add_tcase (s, (tc_chain = tcase_create ("rtpopus")));
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tcase_add_test (tc_chain, test_pay_to_depay);
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tcase_add_test (tc_chain, test_depay_to_pay);
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return s;
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}
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GST_CHECK_MAIN (rtpopus);
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