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https://gitlab.freedesktop.org/gstreamer/gstreamer.git
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474 lines
13 KiB
C
474 lines
13 KiB
C
/* GStreamer
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*
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* Copyright (C) 2018 Collabora Ltd.
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* Author: Nicolas Dufresne <nicolas.dufresne@collabora.com>
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* Copyright (C) 2019 Pexip
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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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#ifdef HAVE_CONFIG_H
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#include "config.h"
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#endif
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#include <gst/rtp/gstrtpbuffer.h>
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#include <gst/rtp/gstrtcpbuffer.h>
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#include <gst/check/gstcheck.h>
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#include <gst/check/gstharness.h>
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#ifdef HAVE_VALGRIND
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# include <valgrind/valgrind.h>
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#else
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# define RUNNING_ON_VALGRIND 0
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#endif
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#define TEST_BUF_CLOCK_RATE 8000
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#define TEST_BUF_PT 0
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#define TEST_BUF_SSRC 0x01BADBAD
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#define TEST_BUF_MS 20
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#define TEST_BUF_DURATION (TEST_BUF_MS * GST_MSECOND)
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#define TEST_BUF_SIZE (64000 * TEST_BUF_MS / 1000)
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#define TEST_RTP_TS_DURATION (TEST_BUF_CLOCK_RATE * TEST_BUF_MS / 1000)
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static GstCaps *
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generate_caps (void)
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{
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return gst_caps_new_simple ("application/x-rtp",
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"media", G_TYPE_STRING, "audio",
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"clock-rate", G_TYPE_INT, TEST_BUF_CLOCK_RATE, NULL);
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}
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static GstBuffer *
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create_buffer (guint seq_num, guint32 ssrc)
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{
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GstBuffer *buf;
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guint8 *payload;
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guint i;
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GstClockTime dts = seq_num * TEST_BUF_DURATION;
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guint32 rtp_ts = seq_num * TEST_RTP_TS_DURATION;
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GstRTPBuffer rtp = GST_RTP_BUFFER_INIT;
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buf = gst_rtp_buffer_new_allocate (TEST_BUF_SIZE, 0, 0);
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GST_BUFFER_DTS (buf) = dts;
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gst_rtp_buffer_map (buf, GST_MAP_READWRITE, &rtp);
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gst_rtp_buffer_set_payload_type (&rtp, TEST_BUF_PT);
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gst_rtp_buffer_set_seq (&rtp, seq_num);
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gst_rtp_buffer_set_timestamp (&rtp, rtp_ts);
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gst_rtp_buffer_set_ssrc (&rtp, ssrc);
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payload = gst_rtp_buffer_get_payload (&rtp);
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for (i = 0; i < TEST_BUF_SIZE; i++)
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payload[i] = 0xff;
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gst_rtp_buffer_unmap (&rtp);
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return buf;
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}
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typedef struct
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{
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GstHarness *rtp_sink;
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GstHarness *rtcp_sink;
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GstHarness *rtp_src;
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GstHarness *rtcp_src;
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} TestContext;
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static void
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rtpssrcdemux_pad_added (G_GNUC_UNUSED GstElement * demux, GstPad * src_pad,
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TestContext * ctx)
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{
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GstHarness *h;
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h = gst_harness_new_with_element (ctx->rtp_sink->element, NULL,
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GST_PAD_NAME (src_pad));
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/* FIXME We should also check that pads have current caps, but this is not
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* currently the case as both pads are created when the first pad receive a
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* buffer. If the other pad is not linked, you'll get a pad without caps.
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* Changing this implies not having both pads on 'on-new-ssrc' which would
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* break rtpbin assumption. */
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if (g_str_has_prefix (GST_PAD_NAME (src_pad), "src_")) {
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g_assert (ctx->rtp_src == NULL);
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ctx->rtp_src = h;
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} else if (g_str_has_prefix (GST_PAD_NAME (src_pad), "rtcp_src_")) {
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g_assert (ctx->rtcp_src == NULL);
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ctx->rtcp_src = h;
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} else {
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g_assert_not_reached ();
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}
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}
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GST_START_TEST (test_event_forwarding)
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{
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TestContext ctx = { NULL, NULL, NULL, NULL };
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GstHarness *h;
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GstEvent *event;
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GstCaps *caps;
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GstStructure *s;
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guint ssrc;
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ctx.rtp_sink = h = gst_harness_new_with_padnames ("rtpssrcdemux", "sink",
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NULL);
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g_signal_connect (h->element, "pad_added",
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G_CALLBACK (rtpssrcdemux_pad_added), &ctx);
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ctx.rtcp_sink = gst_harness_new_with_element (h->element, "rtcp_sink", NULL);
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gst_harness_set_src_caps (h, generate_caps ());
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gst_harness_push (h, create_buffer (0, TEST_BUF_SSRC));
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g_assert (ctx.rtp_src);
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g_assert (ctx.rtcp_src);
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gst_harness_push_event (h, gst_event_new_eos ());
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/* We expect stream-start/caps/segment/eos */
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g_assert_cmpint (gst_harness_events_in_queue (ctx.rtp_src), ==, 4);
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event = gst_harness_pull_event (ctx.rtp_src);
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g_assert_cmpint (event->type, ==, GST_EVENT_STREAM_START);
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gst_event_unref (event);
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event = gst_harness_pull_event (ctx.rtp_src);
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g_assert_cmpint (event->type, ==, GST_EVENT_CAPS);
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gst_event_parse_caps (event, &caps);
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s = gst_caps_get_structure (caps, 0);
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g_assert (gst_structure_has_field (s, "ssrc"));
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g_assert (gst_structure_get_uint (s, "ssrc", &ssrc));
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g_assert_cmpuint (ssrc, ==, TEST_BUF_SSRC);
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gst_event_unref (event);
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event = gst_harness_pull_event (ctx.rtp_src);
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g_assert_cmpint (event->type, ==, GST_EVENT_SEGMENT);
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gst_event_unref (event);
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event = gst_harness_pull_event (ctx.rtp_src);
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g_assert_cmpint (event->type, ==, GST_EVENT_EOS);
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gst_event_unref (event);
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/* We pushed on the RTP pad, no events should have reached the RTCP pad */
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g_assert_cmpint (gst_harness_events_in_queue (ctx.rtcp_src), ==, 0);
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/* push EOS on the rtcp sink pad, to make sure it EOS properly, the harness
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* will create the missing stream-start */
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gst_harness_push_event (ctx.rtcp_sink, gst_event_new_eos ());
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g_assert_cmpint (gst_harness_events_in_queue (ctx.rtp_src), ==, 0);
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g_assert_cmpint (gst_harness_events_in_queue (ctx.rtcp_src), ==, 1);
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event = gst_harness_pull_event (ctx.rtcp_src);
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g_assert_cmpint (event->type, ==, GST_EVENT_EOS);
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gst_event_unref (event);
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gst_harness_teardown (ctx.rtp_src);
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gst_harness_teardown (ctx.rtcp_src);
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gst_harness_teardown (ctx.rtcp_sink);
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gst_harness_teardown (ctx.rtp_sink);
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}
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GST_END_TEST;
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typedef struct
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{
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gint ready;
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GMutex mutex;
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GCond cond;
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} LockTestContext;
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static void
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new_ssrc_pad_cb (G_GNUC_UNUSED GstElement * element, G_GNUC_UNUSED guint ssrc,
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G_GNUC_UNUSED GstPad * pad, LockTestContext * ctx)
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{
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g_message ("Signalling ready");
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g_atomic_int_set (&ctx->ready, 1);
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g_message ("Waiting no more ready");
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while (g_atomic_int_get (&ctx->ready))
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g_usleep (G_USEC_PER_SEC / 100);
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g_mutex_lock (&ctx->mutex);
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g_mutex_unlock (&ctx->mutex);
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}
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static gpointer
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push_buffer_func (gpointer user_data)
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{
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GstHarness *h = user_data;
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gst_harness_push (h, create_buffer (0, 0xdeadbeef));
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return NULL;
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}
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GST_START_TEST (test_oob_event_locking)
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{
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GstHarness *h = gst_harness_new_with_padnames ("rtpssrcdemux", "sink", NULL);
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LockTestContext ctx;
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GThread *thread;
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memset (&ctx, 0, sizeof (LockTestContext));
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g_mutex_init (&ctx.mutex);
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g_cond_init (&ctx.cond);
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gst_harness_set_src_caps_str (h, "application/x-rtp");
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g_signal_connect (h->element,
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"new-ssrc-pad", G_CALLBACK (new_ssrc_pad_cb), &ctx);
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thread = g_thread_new ("streaming-thread", push_buffer_func, h);
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g_mutex_lock (&ctx.mutex);
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g_message ("Waiting for ready");
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while (!g_atomic_int_get (&ctx.ready))
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g_usleep (G_USEC_PER_SEC / 100);
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g_message ("Signal no more ready");
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g_atomic_int_set (&ctx.ready, 0);
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gst_harness_push_event (h,
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gst_event_new_custom (GST_EVENT_CUSTOM_DOWNSTREAM_OOB, NULL));
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g_mutex_unlock (&ctx.mutex);
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g_thread_join (thread);
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g_mutex_clear (&ctx.mutex);
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g_cond_clear (&ctx.cond);
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gst_harness_teardown (h);
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}
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GST_END_TEST;
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static void
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new_ssrc_pad_found (GstElement * element, G_GNUC_UNUSED guint ssrc,
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GstPad * pad, GSList ** src_h)
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{
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GstHarness *h = gst_harness_new_with_element (element, NULL, NULL);
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gst_harness_add_element_src_pad (h, pad);
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*src_h = g_slist_prepend (*src_h, h);
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}
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GST_START_TEST (test_rtpssrcdemux_max_streams)
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{
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GstHarness *h = gst_harness_new_with_padnames ("rtpssrcdemux", "sink", NULL);
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GSList *src_h = NULL;
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gint i;
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g_object_set (h->element, "max-streams", 64, NULL);
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gst_harness_set_src_caps_str (h, "application/x-rtp");
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g_signal_connect (h->element,
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"new-ssrc-pad", (GCallback) new_ssrc_pad_found, &src_h);
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gst_harness_play (h);
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for (i = 0; i < 128; ++i) {
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fail_unless_equals_int (GST_FLOW_OK,
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gst_harness_push (h, create_buffer (0, i)));
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}
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fail_unless_equals_int (g_slist_length (src_h), 64);
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g_slist_free_full (src_h, (GDestroyNotify) gst_harness_teardown);
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gst_harness_teardown (h);
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}
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GST_END_TEST;
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static void
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new_rtcp_ssrc_pad_found (GstElement * element, guint ssrc,
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G_GNUC_UNUSED GstPad * rtp_pad, GSList ** src_h)
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{
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GstHarness *h;
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gchar *name;
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name = g_strdup_printf ("rtcp_src_%u", ssrc);
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h = gst_harness_new_with_element (element, NULL, name);
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g_free (name);
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*src_h = g_slist_prepend (*src_h, h);
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}
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GST_START_TEST (test_rtpssrcdemux_rtcp_app)
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{
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GstHarness *h =
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gst_harness_new_with_padnames ("rtpssrcdemux", "rtcp_sink", NULL);
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GSList *src_h = NULL;
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guint8 rtcp_app_pkt[] = { 0x81, 0xcc, 0x00, 0x05, 0x00, 0x00, 0x5d, 0xaf,
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0x20, 0x20, 0x20, 0x20, 0x21, 0x02, 0x00, 0x0a,
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0x00, 0x00, 0x5d, 0xaf, 0x00, 0x00, 0x16, 0x03
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};
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gst_harness_set_src_caps_str (h, "application/x-rtcp");
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g_signal_connect (h->element,
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"new-ssrc-pad", (GCallback) new_rtcp_ssrc_pad_found, &src_h);
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gst_harness_play (h);
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fail_unless_equals_int (GST_FLOW_OK,
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gst_harness_push (h, gst_buffer_new_wrapped_full (0, rtcp_app_pkt,
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sizeof rtcp_app_pkt, 0, sizeof rtcp_app_pkt, NULL, NULL)));
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fail_unless_equals_int (g_slist_length (src_h), 1);
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g_slist_free_full (src_h, (GDestroyNotify) gst_harness_teardown);
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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_rtpssrcdemux_invalid_rtp)
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{
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GstHarness *h = gst_harness_new_with_padnames ("rtpssrcdemux", "sink", NULL);
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guint8 bad_pkt[] = {
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0x01, 0x02, 0x03
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};
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gst_harness_set_src_caps_str (h, "application/x-rtp");
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gst_harness_play (h);
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fail_unless_equals_int (GST_FLOW_OK,
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gst_harness_push (h, gst_buffer_new_wrapped_full (0, bad_pkt,
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sizeof bad_pkt, 0, sizeof bad_pkt, NULL, NULL)));
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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_rtpssrcdemux_invalid_rtcp)
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{
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GstHarness *h =
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gst_harness_new_with_padnames ("rtpssrcdemux", "rtcp_sink", NULL);
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guint8 bad_pkt[] = {
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0x01, 0x02, 0x03
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};
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gst_harness_set_src_caps_str (h, "application/x-rtcp");
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gst_harness_play (h);
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fail_unless_equals_int (GST_FLOW_OK,
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gst_harness_push (h, gst_buffer_new_wrapped_full (0, bad_pkt,
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sizeof bad_pkt, 0, sizeof bad_pkt, NULL, NULL)));
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gst_harness_teardown (h);
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}
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GST_END_TEST;
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static GstBuffer *
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generate_rtcp_sr_buffer (guint ssrc)
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{
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GstBuffer *buf;
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GstRTCPBuffer rtcp = GST_RTCP_BUFFER_INIT;
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GstRTCPPacket packet;
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buf = gst_rtcp_buffer_new (1000);
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fail_unless (gst_rtcp_buffer_map (buf, GST_MAP_READWRITE, &rtcp));
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fail_unless (gst_rtcp_buffer_add_packet (&rtcp, GST_RTCP_TYPE_SR, &packet));
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gst_rtcp_packet_sr_set_sender_info (&packet, ssrc, 0, 0, 1, 1);
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gst_rtcp_buffer_unmap (&rtcp);
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return buf;
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}
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typedef struct
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{
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GstHarness *rtp_h;
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GstHarness *rtcp_h;
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} SimulCtx;
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static void
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_simul_ctx_new_ssrc_pad_cb (GstElement * element, guint ssrc,
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GstPad * rtp_pad, SimulCtx * ctx)
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{
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GstPad *rtcp_pad;
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gchar *name;
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gst_harness_add_element_src_pad (ctx->rtp_h, rtp_pad);
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name = g_strdup_printf ("rtcp_src_%u", ssrc);
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rtcp_pad = gst_element_get_static_pad (element, name);
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gst_harness_add_element_src_pad (ctx->rtcp_h, rtcp_pad);
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gst_object_unref (rtcp_pad);
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g_free (name);
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}
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static gpointer
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_simul_ctx_push_rtp_buffers (gpointer user_data)
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{
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SimulCtx *ctx = user_data;
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gst_harness_set_src_caps_str (ctx->rtp_h, "application/x-rtp");
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gst_harness_push (ctx->rtp_h, create_buffer (0, 1111));
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return NULL;
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}
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static gpointer
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_simul_ctx_push_rtcp_buffers (gpointer user_data)
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{
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SimulCtx *ctx = user_data;
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g_usleep (10);
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gst_harness_set_src_caps_str (ctx->rtcp_h, "application/x-rtcp");
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gst_harness_push (ctx->rtcp_h, generate_rtcp_sr_buffer (1111));
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return NULL;
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}
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GST_START_TEST (test_rtp_and_rtcp_arrives_simultaneously)
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{
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guint r;
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guint repeats = 1000;
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if (RUNNING_ON_VALGRIND)
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repeats = 2;
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for (r = 0; r < repeats; r++) {
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SimulCtx ctx;
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GThread *t0, *t1;
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ctx.rtp_h = gst_harness_new_with_padnames ("rtpssrcdemux", "sink", NULL);
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ctx.rtcp_h =
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gst_harness_new_with_element (ctx.rtp_h->element, "rtcp_sink", NULL);
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g_signal_connect (ctx.rtp_h->element,
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"new-ssrc-pad", (GCallback) _simul_ctx_new_ssrc_pad_cb, &ctx);
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t0 = g_thread_new ("push rtp", _simul_ctx_push_rtp_buffers, &ctx);
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t1 = g_thread_new ("push rtcp", _simul_ctx_push_rtcp_buffers, &ctx);
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g_thread_join (t0);
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g_thread_join (t1);
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gst_harness_teardown (ctx.rtp_h);
|
|
gst_harness_teardown (ctx.rtcp_h);
|
|
}
|
|
}
|
|
|
|
GST_END_TEST;
|
|
|
|
static Suite *
|
|
rtpssrcdemux_suite (void)
|
|
{
|
|
Suite *s = suite_create ("rtpssrcdemux");
|
|
TCase *tc_chain = tcase_create ("general");
|
|
|
|
suite_add_tcase (s, tc_chain);
|
|
tcase_add_test (tc_chain, test_event_forwarding);
|
|
tcase_add_test (tc_chain, test_oob_event_locking);
|
|
tcase_add_test (tc_chain, test_rtpssrcdemux_max_streams);
|
|
tcase_add_test (tc_chain, test_rtpssrcdemux_rtcp_app);
|
|
tcase_add_test (tc_chain, test_rtpssrcdemux_invalid_rtp);
|
|
tcase_add_test (tc_chain, test_rtpssrcdemux_invalid_rtcp);
|
|
tcase_add_test (tc_chain, test_rtp_and_rtcp_arrives_simultaneously);
|
|
|
|
return s;
|
|
}
|
|
|
|
GST_CHECK_MAIN (rtpssrcdemux);
|