/*
 *  Copyright 2024 The WebRTC project authors. All Rights Reserved.
 *
 *  Use of this source code is governed by a BSD-style license
 *  that can be found in the LICENSE file in the root of the source
 *  tree. An additional intellectual property rights grant can be found
 *  in the file PATENTS.  All contributing project authors may
 *  be found in the AUTHORS file in the root of the source tree.
 */

// This file contains tests that verify that congestion control options
// are correctly negotiated in the SDP offer/answer.

#include <memory>
#include <string>
#include <vector>

#include "absl/strings/str_cat.h"
#include "api/jsep.h"
#include "api/media_types.h"
#include "api/peer_connection_interface.h"
#include "api/rtp_parameters.h"
#include "api/rtp_transceiver_direction.h"
#include "api/test/rtc_error_matchers.h"
#include "pc/test/integration_test_helpers.h"
#include "test/gmock.h"
#include "test/gtest.h"
#include "test/wait_until.h"

namespace webrtc {

using ::testing::Eq;
using ::testing::Field;
using ::testing::Gt;
using ::testing::HasSubstr;
using ::testing::IsTrue;
using ::testing::Not;

class PeerConnectionCongestionControlTest
    : public PeerConnectionIntegrationBaseTest {
 public:
  PeerConnectionCongestionControlTest()
      : PeerConnectionIntegrationBaseTest(SdpSemantics::kUnifiedPlan) {}
};

TEST_F(PeerConnectionCongestionControlTest, OfferContainsCcfbIfEnabled) {
  SetFieldTrials("WebRTC-RFC8888CongestionControlFeedback/Enabled/");
  ASSERT_TRUE(CreatePeerConnectionWrappers());
  caller()->AddAudioVideoTracks();
  std::unique_ptr<SessionDescriptionInterface> offer =
      caller()->CreateOfferAndWait();
  std::string offer_str = absl::StrCat(*offer);
  EXPECT_THAT(offer_str, HasSubstr("a=rtcp-fb:* ack ccfb\r\n"));
}

TEST_F(PeerConnectionCongestionControlTest, ReceiveOfferSetsCcfbFlag) {
  SetFieldTrials("WebRTC-RFC8888CongestionControlFeedback/Enabled/");
  ASSERT_TRUE(CreatePeerConnectionWrappers());
  ConnectFakeSignalingForSdpOnly();
  caller()->AddAudioVideoTracks();
  caller()->CreateAndSetAndSignalOffer();
  ASSERT_THAT(WaitUntil([&] { return SignalingStateStable(); }, IsTrue()),
              IsRtcOk());
  {
    // Check that the callee parsed it.
    auto parsed_contents =
        callee()->pc()->remote_description()->description()->contents();
    EXPECT_FALSE(parsed_contents.empty());
    for (const auto& content : parsed_contents) {
      EXPECT_TRUE(content.media_description()->rtcp_fb_ack_ccfb());
    }
  }

  {
    // Check that the caller also parsed it.
    auto parsed_contents =
        caller()->pc()->remote_description()->description()->contents();
    EXPECT_FALSE(parsed_contents.empty());
    for (const auto& content : parsed_contents) {
      EXPECT_TRUE(content.media_description()->rtcp_fb_ack_ccfb());
    }
  }
  // Check that the answer does not contain transport-cc
  std::string answer_str = absl::StrCat(*caller()->pc()->remote_description());
  EXPECT_THAT(answer_str, Not(HasSubstr("transport-cc")));
}

TEST_F(PeerConnectionCongestionControlTest, NegotiatingCcfbRemovesTsn) {
  SetFieldTrials("WebRTC-RFC8888CongestionControlFeedback/Enabled/");
  ASSERT_TRUE(CreatePeerConnectionWrappers());
  ConnectFakeSignalingForSdpOnly();
  callee()->AddVideoTrack();
  // Add transceivers to caller in order to accomodate reception
  caller()->pc()->AddTransceiver(MediaType::VIDEO);
  auto parameters = caller()->pc()->GetSenders()[0]->GetParameters();
  caller()->CreateAndSetAndSignalOffer();
  ASSERT_THAT(WaitUntil([&] { return SignalingStateStable(); }, IsTrue()),
              IsRtcOk());

  std::vector<RtpHeaderExtensionCapability> negotiated_header_extensions =
      caller()->pc()->GetTransceivers()[0]->GetNegotiatedHeaderExtensions();
  EXPECT_THAT(
      negotiated_header_extensions,
      Not(Contains(
          AllOf(Field("uri", &RtpHeaderExtensionCapability::uri,
                      RtpExtension::kTransportSequenceNumberUri),
                Not(Field("direction", &RtpHeaderExtensionCapability::direction,
                          RtpTransceiverDirection::kStopped))))))
      << " in caller negotiated header extensions";

  parameters = caller()->pc()->GetSenders()[0]->GetParameters();
  EXPECT_THAT(parameters.header_extensions,
              Not(Contains(Field("uri", &RtpExtension::uri,
                                 RtpExtension::kTransportSequenceNumberUri))))
      << " in caller sender parameters";
  parameters = caller()->pc()->GetReceivers()[0]->GetParameters();
  EXPECT_THAT(parameters.header_extensions,
              Not(Contains(Field("uri", &RtpExtension::uri,
                                 RtpExtension::kTransportSequenceNumberUri))))
      << " in caller receiver parameters";
  /* Callee senders are not fixed yet.
     TODO: issues.webrtc.org/383078466 - enable
  parameters = callee()->pc()->GetSenders()[0]->GetParameters();
  EXPECT_THAT(parameters.header_extensions,
              Not(Contains(Field("uri", &RtpExtension::uri,
                                 RtpExtension::kTransportSequenceNumberUri))))
      << " in callee sender parameters";
  */
  parameters = callee()->pc()->GetReceivers()[0]->GetParameters();
  EXPECT_THAT(parameters.header_extensions,
              Not(Contains(Field("uri", &RtpExtension::uri,
                                 RtpExtension::kTransportSequenceNumberUri))))
      << " in callee receiver parameters";
}

TEST_F(PeerConnectionCongestionControlTest, CcfbGetsUsed) {
  SetFieldTrials("WebRTC-RFC8888CongestionControlFeedback/Enabled/");
  ASSERT_TRUE(CreatePeerConnectionWrappers());
  ConnectFakeSignaling();
  caller()->AddAudioVideoTracks();
  caller()->CreateAndSetAndSignalOffer();
  ASSERT_THAT(WaitUntil([&] { return SignalingStateStable(); }, IsTrue()),
              IsRtcOk());
  MediaExpectations media_expectations;
  media_expectations.CalleeExpectsSomeAudio();
  media_expectations.CalleeExpectsSomeVideo();
  ASSERT_TRUE(ExpectNewFrames(media_expectations));
  auto pc_internal = caller()->pc_internal();
  EXPECT_THAT(
      WaitUntil(
          [&] {
            return pc_internal->FeedbackAccordingToRfc8888CountForTesting();
          },
          Gt(0)),
      IsRtcOk());
  // There should be no transport-cc generated.
  EXPECT_THAT(pc_internal->FeedbackAccordingToTransportCcCountForTesting(),
              Eq(0));
}

TEST_F(PeerConnectionCongestionControlTest, TransportCcGetsUsed) {
  SetFieldTrials("WebRTC-RFC8888CongestionControlFeedback/Disabled/");
  ASSERT_TRUE(CreatePeerConnectionWrappers());
  ConnectFakeSignaling();
  caller()->AddAudioVideoTracks();
  caller()->CreateAndSetAndSignalOffer();
  ASSERT_THAT(WaitUntil([&] { return SignalingStateStable(); }, IsTrue()),
              IsRtcOk());
  MediaExpectations media_expectations;
  media_expectations.CalleeExpectsSomeAudio();
  media_expectations.CalleeExpectsSomeVideo();
  ASSERT_TRUE(ExpectNewFrames(media_expectations));
  auto pc_internal = caller()->pc_internal();
  EXPECT_THAT(
      WaitUntil(
          [&] {
            return pc_internal->FeedbackAccordingToTransportCcCountForTesting();
          },
          Gt(0)),
      IsRtcOk());
  // Test that RFC 8888 feedback is NOT generated when field trial disabled.
  EXPECT_THAT(pc_internal->FeedbackAccordingToRfc8888CountForTesting(), Eq(0));
}

}  // namespace webrtc