Remove channel ids from various interfaces.

webrtc/modules/rtp_rtcp/source/nack_rtx_unittest.cc

 /*
 *  Copyright (c) 2013 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.
 */
 
 #include <algorithm>
 #include <iterator>
 #include <list>
 #include <set>
 
 #include "testing/gtest/include/gtest/gtest.h"
 #include "webrtc/base/scoped_ptr.h"
 #include "webrtc/common_types.h"
 #include "webrtc/modules/rtp_rtcp/interface/receive_statistics.h"
 #include "webrtc/modules/rtp_rtcp/interface/rtp_header_parser.h"
 #include "webrtc/modules/rtp_rtcp/interface/rtp_payload_registry.h"
 #include "webrtc/modules/rtp_rtcp/interface/rtp_receiver.h"
 #include "webrtc/modules/rtp_rtcp/interface/rtp_rtcp.h"
 #include "webrtc/modules/rtp_rtcp/interface/rtp_rtcp_defines.h"
 
 using namespace webrtc;
 
 const int kVideoNackListSize = 30;
-const int kTestId = 123;
 const uint32_t kTestSsrc = 3456;
 const uint16_t kTestSequenceNumber = 2345;
 const uint32_t kTestNumberOfPackets = 1350;
 const int kTestNumberOfRtxPackets = 149;
 const int kNumFrames = 30;
 const int kPayloadType = 123;
 const int kRtxPayloadType = 98;
 
 class VerifyingRtxReceiver : public NullRtpData
 {
@@ skipping 10 matching lines @@
     return 0;
   }
   std::list<uint16_t> sequence_numbers_;
 };
 
 class TestRtpFeedback : public NullRtpFeedback {
  public:
   TestRtpFeedback(RtpRtcp* rtp_rtcp) : rtp_rtcp_(rtp_rtcp) {}
   virtual ~TestRtpFeedback() {}
 
-  void OnIncomingSSRCChanged(const int32_t id, const uint32_t ssrc) override {
+  void OnIncomingSSRCChanged(const uint32_t ssrc) override {
     rtp_rtcp_->SetRemoteSSRC(ssrc);
   }
 
  private:
   RtpRtcp* rtp_rtcp_;
 };
 
 class RtxLoopBackTransport : public webrtc::Transport {
  public:
   explicit RtxLoopBackTransport(uint32_t rtx_ssrc)
@@ skipping 18 matching lines @@
   void DropEveryNthPacket(int n) {
     packet_loss_ = n;
   }
 
   void DropConsecutivePackets(int start, int total) {
     consecutive_drop_start_ = start;
     consecutive_drop_end_ = start + total;
     packet_loss_ = 0;
   }
 
-  int SendPacket(int channel, const void* data, size_t len) override {
+  int SendPacket(const void* data, size_t len) override {
     count_++;
     const unsigned char* ptr = static_cast<const unsigned  char*>(data);
     uint32_t ssrc = (ptr[8] << 24) + (ptr[9] << 16) + (ptr[10] << 8) + ptr[11];
     if (ssrc == rtx_ssrc_) count_rtx_ssrc_++;
     uint16_t sequence_number = (ptr[2] << 8) + ptr[3];
     size_t packet_length = len;
     // TODO(pbos): Figure out why this needs to be initialized. Likely this
     // is hiding a bug either in test setup or other code.
     // https://code.google.com/p/webrtc/issues/detail?id=3183
     uint8_t restored_packet[1500] = {0};
@@ skipping 38 matching lines @@
       return -1;
     }
     if (!rtp_receiver_->IncomingRtpPacket(header, restored_packet_ptr,
                                           packet_length, payload_specific,
                                           true)) {
       return -1;
     }
     return static_cast<int>(len);
   }
 
-  int SendRTCPPacket(int channel, const void* data, size_t len) override {
+  int SendRTCPPacket(const void* data, size_t len) override {
     if (module_->IncomingRtcpPacket((const uint8_t*)data, len) == 0) {
       return static_cast<int>(len);
     }
     return -1;
   }
   int count_;
   int packet_loss_;
   int consecutive_drop_start_;
   int consecutive_drop_end_;
   uint32_t rtx_ssrc_;
@@ skipping 10 matching lines @@
       : rtp_payload_registry_(RTPPayloadStrategy::CreateStrategy(false)),
         rtp_rtcp_module_(NULL),
         transport_(kTestSsrc + 1),
         receiver_(),
         payload_data_length(sizeof(payload_data)),
         fake_clock(123456) {}
   ~RtpRtcpRtxNackTest() {}
 
   void SetUp() override {
     RtpRtcp::Configuration configuration;
-    configuration.id = kTestId;
     configuration.audio = false;
     configuration.clock = &fake_clock;
     receive_statistics_.reset(ReceiveStatistics::Create(&fake_clock));
     configuration.receive_statistics = receive_statistics_.get();
     configuration.outgoing_transport = &transport_;
     rtp_rtcp_module_ = RtpRtcp::CreateRtpRtcp(configuration);
 
     rtp_feedback_.reset(new TestRtpFeedback(rtp_rtcp_module_));
 
     rtp_receiver_.reset(RtpReceiver::CreateVideoReceiver(
-        kTestId, &fake_clock, &receiver_, rtp_feedback_.get(),
+        &fake_clock, &receiver_, rtp_feedback_.get(),
         &rtp_payload_registry_));
 
     rtp_rtcp_module_->SetSSRC(kTestSsrc);
     rtp_rtcp_module_->SetRTCPStatus(kRtcpCompound);
     rtp_receiver_->SetNACKStatus(kNackRtcp);
     rtp_rtcp_module_->SetStorePacketsStatus(true, 600);
     EXPECT_EQ(0, rtp_rtcp_module_->SetSendingStatus(true));
     rtp_rtcp_module_->SetSequenceNumber(kTestSequenceNumber);
     rtp_rtcp_module_->SetStartTimestamp(111111);
 
@@ skipping 135 matching lines @@
 
 TEST_F(RtpRtcpRtxNackTest, RtxNack) {
   RunRtxTest(kRtxRetransmitted, 10);
   EXPECT_EQ(kTestSequenceNumber, *(receiver_.sequence_numbers_.begin()));
   EXPECT_EQ(kTestSequenceNumber + kTestNumberOfPackets - 1,
       *(receiver_.sequence_numbers_.rbegin()));
   EXPECT_EQ(kTestNumberOfPackets, receiver_.sequence_numbers_.size());
   EXPECT_EQ(kTestNumberOfRtxPackets, transport_.count_rtx_ssrc_);
   EXPECT_TRUE(ExpectedPacketsReceived());
 }