Reland of Delete class SSRCDatabase, and its global ssrc registry.

webrtc/modules/rtp_rtcp/source/rtp_sender_unittest.cc

 /*
  *  Copyright (c) 2012 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.
  */
 
@@ skipping 325 matching lines @@
   ASSERT_EQ(1u, transport_.sent_packets_.size());
   // Verify padding packet timestamp.
   EXPECT_EQ(kTimestamp, transport_.last_sent_packet().Timestamp());
 }
 
 TEST_F(RtpSenderTestWithoutPacer, SendsPacketsWithTransportSequenceNumber) {
   rtp_sender_.reset(new RTPSender(
       false, &fake_clock_, &transport_, nullptr, nullptr, &seq_num_allocator_,
       &feedback_observer_, nullptr, nullptr, nullptr, &mock_rtc_event_log_,
       &send_packet_observer_, &retransmission_rate_limiter_, nullptr));
+  rtp_sender_->SetSSRC(kSsrc);
   EXPECT_EQ(0, rtp_sender_->RegisterRtpHeaderExtension(
                    kRtpExtensionTransportSequenceNumber,
                    kTransportSequenceNumberExtensionId));
 
   EXPECT_CALL(seq_num_allocator_, AllocateSequenceNumber())
       .WillOnce(testing::Return(kTransportSequenceNumber));
   EXPECT_CALL(send_packet_observer_,
               OnSendPacket(kTransportSequenceNumber, _, _))
       .Times(1);
   EXPECT_CALL(feedback_observer_,
@@ skipping 597 matching lines @@
 
     uint32_t num_calls_;
     uint32_t ssrc_;
     FrameCounts frame_counts_;
   } callback;
 
   rtp_sender_.reset(
       new RTPSender(false, &fake_clock_, &transport_, &mock_paced_sender_,
                     nullptr, nullptr, nullptr, nullptr, &callback, nullptr,
                     nullptr, nullptr, &retransmission_rate_limiter_, nullptr));
-
+  rtp_sender_->SetSSRC(kSsrc);
   char payload_name[RTP_PAYLOAD_NAME_SIZE] = "GENERIC";
   const uint8_t payload_type = 127;
   ASSERT_EQ(0, rtp_sender_->RegisterPayload(payload_name, payload_type, 90000,
                                             0, 1500));
   uint8_t payload[] = {47, 11, 32, 93, 89};
   rtp_sender_->SetStorePacketsStatus(true, 1);
   uint32_t ssrc = rtp_sender_->SSRC();
 
   EXPECT_CALL(mock_paced_sender_, InsertPacket(_, _, _, _, _, _))
       .Times(::testing::AtLeast(2));
@@ skipping 41 matching lines @@
 
     uint32_t num_calls_;
     uint32_t ssrc_;
     uint32_t total_bitrate_;
     uint32_t retransmit_bitrate_;
   } callback;
   rtp_sender_.reset(new RTPSender(false, &fake_clock_, &transport_, nullptr,
                                   nullptr, nullptr, nullptr, &callback, nullptr,
                                   nullptr, nullptr, nullptr,
                                   &retransmission_rate_limiter_, nullptr));
+  rtp_sender_->SetSSRC(kSsrc);
 
   // Simulate kNumPackets sent with kPacketInterval ms intervals, with the
   // number of packets selected so that we fill (but don't overflow) the one
   // second averaging window.
   const uint32_t kWindowSizeMs = 1000;
   const uint32_t kPacketInterval = 20;
   const uint32_t kNumPackets =
       (kWindowSizeMs - kPacketInterval) / kPacketInterval;
   // Overhead = 12 bytes RTP header + 1 byte generic header.
   const uint32_t kPacketOverhead = 13;
@@ skipping 38 matching lines @@
 class RtpSenderAudioTest : public RtpSenderTest {
  protected:
   RtpSenderAudioTest() {}
 
   void SetUp() override {
     payload_ = kAudioPayload;
     rtp_sender_.reset(new RTPSender(true, &fake_clock_, &transport_, nullptr,
                                     nullptr, nullptr, nullptr, nullptr, nullptr,
                                     nullptr, nullptr, nullptr,
                                     &retransmission_rate_limiter_, nullptr));
+    rtp_sender_->SetSSRC(kSsrc);
     rtp_sender_->SetSequenceNumber(kSeqNum);
   }
 };
 
 TEST_F(RtpSenderTestWithoutPacer, StreamDataCountersCallbacks) {
   class TestCallback : public StreamDataCountersCallback {
    public:
     TestCallback() : StreamDataCountersCallback(), ssrc_(0), counters_() {}
     virtual ~TestCallback() {}
 
@@ skipping 340 matching lines @@
   MOCK_METHOD1(OnOverheadChanged, void(size_t overhead_bytes_per_packet));
 };
 }  // namespace
 
 TEST_F(RtpSenderTest, OnOverheadChanged) {
   MockOverheadObserver mock_overhead_observer;
   rtp_sender_.reset(
       new RTPSender(false, &fake_clock_, &transport_, nullptr, nullptr, nullptr,
                     nullptr, nullptr, nullptr, nullptr, nullptr, nullptr,
                     &retransmission_rate_limiter_, &mock_overhead_observer));
+  rtp_sender_->SetSSRC(kSsrc);
 
   // RTP overhead is 12B.
   EXPECT_CALL(mock_overhead_observer, OnOverheadChanged(12)).Times(1);
   SendGenericPayload();
 
   rtp_sender_->RegisterRtpHeaderExtension(kRtpExtensionTransmissionTimeOffset,
                                           kTransmissionTimeOffsetExtensionId);
 
   // TransmissionTimeOffset extension has a size of 8B.
   // 12B + 8B = 20B
   EXPECT_CALL(mock_overhead_observer, OnOverheadChanged(20)).Times(1);
   SendGenericPayload();
 }
 
 TEST_F(RtpSenderTest, DoesNotUpdateOverheadOnEqualSize) {
   MockOverheadObserver mock_overhead_observer;
   rtp_sender_.reset(
       new RTPSender(false, &fake_clock_, &transport_, nullptr, nullptr, nullptr,
                     nullptr, nullptr, nullptr, nullptr, nullptr, nullptr,
                     &retransmission_rate_limiter_, &mock_overhead_observer));
+  rtp_sender_->SetSSRC(kSsrc);
 
   EXPECT_CALL(mock_overhead_observer, OnOverheadChanged(_)).Times(1);
   SendGenericPayload();
   SendGenericPayload();
 }
 
 TEST_F(RtpSenderTest, AddOverheadToTransportFeedbackObserver) {
   constexpr int kRtpOverheadBytesPerPacket = 12 + 8;
   test::ScopedFieldTrials override_field_trials(
       "WebRTC-SendSideBwe-WithOverhead/Enabled/");
   testing::NiceMock<MockOverheadObserver> mock_overhead_observer;
   rtp_sender_.reset(new RTPSender(
       false, &fake_clock_, &transport_, nullptr, nullptr, &seq_num_allocator_,
       &feedback_observer_, nullptr, nullptr, nullptr, &mock_rtc_event_log_,
       nullptr, &retransmission_rate_limiter_, &mock_overhead_observer));
+  rtp_sender_->SetSSRC(kSsrc);
   EXPECT_EQ(0, rtp_sender_->RegisterRtpHeaderExtension(
                    kRtpExtensionTransportSequenceNumber,
                    kTransportSequenceNumberExtensionId));
   EXPECT_CALL(seq_num_allocator_, AllocateSequenceNumber())
       .WillOnce(testing::Return(kTransportSequenceNumber));
   EXPECT_CALL(feedback_observer_,
               AddPacket(kTransportSequenceNumber,
                         sizeof(kPayloadData) + kGenericHeaderLength +
                             kRtpOverheadBytesPerPacket,
                         PacedPacketInfo::kNotAProbe))
@@ skipping 25 matching lines @@
 
   // Requested padding size is too small, will send a larger one.
   const size_t kMinPaddingSize = 50;
   EXPECT_CALL(transport, SendRtp(_, kMinPaddingSize + kRtpHeaderSize, _))
       .WillOnce(testing::Return(true));
   EXPECT_EQ(kMinPaddingSize,
             rtp_sender_->TimeToSendPadding(kMinPaddingSize - 5,
                                            PacedPacketInfo::kNotAProbe));
 }
 }  // namespace webrtc