/* GStreamer
 *
 * Copyright (C) 2018 Collabora Ltd.
 *               Author: Nicolas Dufresne <nicolas.dufresne@collabora.com>
 *
 * This library is free software; you can redistribute it and/or
 * modify it under the terms of the GNU Library General Public
 * License as published by the Free Software Foundation; either
 * version 2 of the License, or (at your option) any later version.
 *
 * This library is distributed in the hope that it will be useful,
 * but WITHOUT ANY WARRANTY; without even the implied warranty of
 * MERCHANTABILITY or FITNESS FOR A PARTICULAR PURPOSE.  See the GNU
 * Library General Public License for more details.
 *
 * You should have received a copy of the GNU Library General Public
 * License along with this library; if not, write to the
 * Free Software Foundation, Inc., 51 Franklin St, Fifth Floor,
 * Boston, MA 02110-1301, USA.
 */
#include <gst/rtp/gstrtpbuffer.h>
#include <gst/check/gstcheck.h>
#include <gst/check/gstharness.h>

#define TEST_BUF_CLOCK_RATE 8000
#define TEST_BUF_PT 0
#define TEST_BUF_SSRC 0x01BADBAD
#define TEST_BUF_MS  20
#define TEST_BUF_DURATION (TEST_BUF_MS * GST_MSECOND)
#define TEST_BUF_SIZE (64000 * TEST_BUF_MS / 1000)
#define TEST_RTP_TS_DURATION (TEST_BUF_CLOCK_RATE * TEST_BUF_MS / 1000)

static GstCaps *
generate_caps (void)
{
  return gst_caps_new_simple ("application/x-rtp",
      "media", G_TYPE_STRING, "audio",
      "clock-rate", G_TYPE_INT, TEST_BUF_CLOCK_RATE, NULL);
}

static GstBuffer *
create_buffer (guint seq_num, guint32 ssrc)
{
  GstBuffer *buf;
  guint8 *payload;
  guint i;
  GstClockTime dts = seq_num * TEST_BUF_DURATION;
  guint32 rtp_ts = seq_num * TEST_RTP_TS_DURATION;
  GstRTPBuffer rtp = GST_RTP_BUFFER_INIT;

  buf = gst_rtp_buffer_new_allocate (TEST_BUF_SIZE, 0, 0);
  GST_BUFFER_DTS (buf) = dts;

  gst_rtp_buffer_map (buf, GST_MAP_READWRITE, &rtp);
  gst_rtp_buffer_set_payload_type (&rtp, TEST_BUF_PT);
  gst_rtp_buffer_set_seq (&rtp, seq_num);
  gst_rtp_buffer_set_timestamp (&rtp, rtp_ts);
  gst_rtp_buffer_set_ssrc (&rtp, ssrc);

  payload = gst_rtp_buffer_get_payload (&rtp);
  for (i = 0; i < TEST_BUF_SIZE; i++)
    payload[i] = 0xff;

  gst_rtp_buffer_unmap (&rtp);

  return buf;
}

typedef struct
{
  GstHarness *rtp_sink;
  GstHarness *rtcp_sink;
  GstHarness *rtp_src;
  GstHarness *rtcp_src;
} TestContext;

static void
rtpssrcdemux_pad_added (G_GNUC_UNUSED GstElement * demux, GstPad * src_pad,
    TestContext * ctx)
{
  GstHarness *h;

  h = gst_harness_new_with_element (ctx->rtp_sink->element, NULL,
      GST_PAD_NAME (src_pad));

  /* FIXME We should also check that pads have current caps, but this is not
   * currently the case as both pads are created when the first pad receive a
   * buffer. If the other pad is not linked, you'll get a pad without caps.
   * Changing this implies not having both pads on 'on-new-ssrc' which would
   * break rtpbin assumption. */

  if (g_str_has_prefix (GST_PAD_NAME (src_pad), "src_")) {
    g_assert (ctx->rtp_src == NULL);
    ctx->rtp_src = h;
  } else if (g_str_has_prefix (GST_PAD_NAME (src_pad), "rtcp_src_")) {
    g_assert (ctx->rtcp_src == NULL);
    ctx->rtcp_src = h;
  } else {
    g_assert_not_reached ();
  }
}

GST_START_TEST (test_event_forwarding)
{
  TestContext ctx = { NULL, };
  GstHarness *h;
  GstEvent *event;
  GstCaps *caps;
  GstStructure *s;
  guint ssrc;

  ctx.rtp_sink = h = gst_harness_new_with_padnames ("rtpssrcdemux", "sink",
      NULL);
  g_signal_connect (h->element, "pad_added",
      G_CALLBACK (rtpssrcdemux_pad_added), &ctx);

  ctx.rtcp_sink = gst_harness_new_with_element (h->element, "rtcp_sink", NULL);

  gst_harness_set_src_caps (h, generate_caps ());
  gst_harness_push (h, create_buffer (0, TEST_BUF_SSRC));

  g_assert (ctx.rtp_src);
  g_assert (ctx.rtcp_src);

  gst_harness_push_event (h, gst_event_new_eos ());

  /* We expect stream-start/caps/segment/eos */
  g_assert_cmpint (gst_harness_events_in_queue (ctx.rtp_src), ==, 4);

  event = gst_harness_pull_event (ctx.rtp_src);
  g_assert_cmpint (event->type, ==, GST_EVENT_STREAM_START);
  gst_event_unref (event);

  event = gst_harness_pull_event (ctx.rtp_src);
  g_assert_cmpint (event->type, ==, GST_EVENT_CAPS);
  gst_event_parse_caps (event, &caps);
  s = gst_caps_get_structure (caps, 0);
  g_assert (gst_structure_has_field (s, "ssrc"));
  g_assert (gst_structure_get_uint (s, "ssrc", &ssrc));
  g_assert_cmpuint (ssrc, ==, TEST_BUF_SSRC);
  gst_event_unref (event);

  event = gst_harness_pull_event (ctx.rtp_src);
  g_assert_cmpint (event->type, ==, GST_EVENT_SEGMENT);
  gst_event_unref (event);

  event = gst_harness_pull_event (ctx.rtp_src);
  g_assert_cmpint (event->type, ==, GST_EVENT_EOS);
  gst_event_unref (event);

  /* We pushed on the RTP pad, no events should have reached the RTCP pad */
  g_assert_cmpint (gst_harness_events_in_queue (ctx.rtcp_src), ==, 0);

  /* push EOS on the rtcp sink pad, to make sure it EOS properly, the harness
   * will create the missing stream-start */
  gst_harness_push_event (ctx.rtcp_sink, gst_event_new_eos ());

  g_assert_cmpint (gst_harness_events_in_queue (ctx.rtp_src), ==, 0);
  g_assert_cmpint (gst_harness_events_in_queue (ctx.rtcp_src), ==, 2);

  event = gst_harness_pull_event (ctx.rtcp_src);
  g_assert_cmpint (event->type, ==, GST_EVENT_STREAM_START);
  gst_event_unref (event);

  event = gst_harness_pull_event (ctx.rtcp_src);
  g_assert_cmpint (event->type, ==, GST_EVENT_EOS);
  gst_event_unref (event);

  gst_harness_teardown (ctx.rtp_src);
  gst_harness_teardown (ctx.rtcp_src);
  gst_harness_teardown (ctx.rtcp_sink);
  gst_harness_teardown (ctx.rtp_sink);
}

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);

  return s;
}

GST_CHECK_MAIN (rtpssrcdemux);
