Unify Transport and newapi::Transport 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"
+#include "webrtc/transport.h"
 
 using namespace webrtc;
 
 const int kVideoNackListSize = 30;
 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;
@@ skipping 53 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(const void* data, size_t len) override {
+  bool SendRtp(const uint8_t* 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};
     uint8_t* restored_packet_ptr = restored_packet;
     RTPHeader header;
     rtc::scoped_ptr<RtpHeaderParser> parser(RtpHeaderParser::Create());
     if (!parser->Parse(ptr, len, &header)) {
-      return -1;
+      return false;
     }
 
     if (!rtp_payload_registry_->IsRtx(header)) {
       // Don't store retransmitted packets since we compare it to the list
       // created by the receiver.
       expected_sequence_numbers_.insert(expected_sequence_numbers_.end(),
                                         sequence_number);
     }
     if (packet_loss_ > 0) {
       if ((count_ % packet_loss_) == 0) {
-        return static_cast<int>(len);
+        return true;
       }
     } else if (count_ >= consecutive_drop_start_ &&
                count_ < consecutive_drop_end_) {
-      return static_cast<int>(len);
+      return true;
     }
     if (rtp_payload_registry_->IsRtx(header)) {
       // Remove the RTX header and parse the original RTP header.
       EXPECT_TRUE(rtp_payload_registry_->RestoreOriginalPacket(
           &restored_packet_ptr, ptr, &packet_length, rtp_receiver_->SSRC(),
           header));
       if (!parser->Parse(restored_packet_ptr, packet_length, &header)) {
-        return -1;
+        return false;
       }
     } else {
       rtp_payload_registry_->SetIncomingPayloadType(header);
     }
 
     restored_packet_ptr += header.headerLength;
     packet_length -= header.headerLength;
     PayloadUnion payload_specific;
     if (!rtp_payload_registry_->GetPayloadSpecifics(header.payloadType,
                                                     &payload_specific)) {
-      return -1;
+      return false;
     }
     if (!rtp_receiver_->IncomingRtpPacket(header, restored_packet_ptr,
                                           packet_length, payload_specific,
                                           true)) {
-      return -1;
+      return false;
     }
-    return static_cast<int>(len);
+    return true;
   }
 
-  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;
+  bool SendRtcp(const uint8_t* data, size_t len) override {
+    return module_->IncomingRtcpPacket((const uint8_t*)data, len) == 0;
   }
   int count_;
   int packet_loss_;
   int consecutive_drop_start_;
   int consecutive_drop_end_;
   uint32_t rtx_ssrc_;
   int count_rtx_ssrc_;
   RTPPayloadRegistry* rtp_payload_registry_;
   RtpReceiver* rtp_receiver_;
   RtpRtcp* module_;
@@ skipping 172 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());
 }