| Index: webrtc/modules/rtp_rtcp/source/nack_rtx_unittest.cc
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| diff --git a/webrtc/modules/rtp_rtcp/source/nack_rtx_unittest.cc b/webrtc/modules/rtp_rtcp/source/nack_rtx_unittest.cc
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| index d32d09fab077a5d5c2add7d001d2ec2db626d849..6f13056f5f45ac32714f833e1efc0e0f19170635 100644
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| --- a/webrtc/modules/rtp_rtcp/source/nack_rtx_unittest.cc
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| +++ b/webrtc/modules/rtp_rtcp/source/nack_rtx_unittest.cc
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| @@ -22,6 +22,7 @@
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|  #include "webrtc/modules/rtp_rtcp/interface/rtp_receiver.h"
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|  #include "webrtc/modules/rtp_rtcp/interface/rtp_rtcp.h"
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|  #include "webrtc/modules/rtp_rtcp/interface/rtp_rtcp_defines.h"
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| +#include "webrtc/transport.h"
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|  
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|  using namespace webrtc;
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|  
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| @@ -95,7 +96,7 @@ class RtxLoopBackTransport : public webrtc::Transport {
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|      packet_loss_ = 0;
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|    }
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|  
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| -  int SendPacket(const void* data, size_t len) override {
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| +  bool SendRtp(const uint8_t* data, size_t len) override {
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|      count_++;
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|      const unsigned char* ptr = static_cast<const unsigned  char*>(data);
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|      uint32_t ssrc = (ptr[8] << 24) + (ptr[9] << 16) + (ptr[10] << 8) + ptr[11];
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| @@ -110,7 +111,7 @@ class RtxLoopBackTransport : public webrtc::Transport {
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|      RTPHeader header;
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|      rtc::scoped_ptr<RtpHeaderParser> parser(RtpHeaderParser::Create());
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|      if (!parser->Parse(ptr, len, &header)) {
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| -      return -1;
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| +      return false;
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|      }
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|  
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|      if (!rtp_payload_registry_->IsRtx(header)) {
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| @@ -121,11 +122,11 @@ class RtxLoopBackTransport : public webrtc::Transport {
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|      }
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|      if (packet_loss_ > 0) {
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|        if ((count_ % packet_loss_) == 0) {
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| -        return static_cast<int>(len);
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| +        return true;
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|        }
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|      } else if (count_ >= consecutive_drop_start_ &&
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|                 count_ < consecutive_drop_end_) {
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| -      return static_cast<int>(len);
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| +      return true;
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|      }
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|      if (rtp_payload_registry_->IsRtx(header)) {
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|        // Remove the RTX header and parse the original RTP header.
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| @@ -133,7 +134,7 @@ class RtxLoopBackTransport : public webrtc::Transport {
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|            &restored_packet_ptr, ptr, &packet_length, rtp_receiver_->SSRC(),
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|            header));
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|        if (!parser->Parse(restored_packet_ptr, packet_length, &header)) {
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| -        return -1;
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| +        return false;
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|        }
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|      } else {
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|        rtp_payload_registry_->SetIncomingPayloadType(header);
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| @@ -144,21 +145,18 @@ class RtxLoopBackTransport : public webrtc::Transport {
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|      PayloadUnion payload_specific;
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|      if (!rtp_payload_registry_->GetPayloadSpecifics(header.payloadType,
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|                                                      &payload_specific)) {
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| -      return -1;
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| +      return false;
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|      }
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|      if (!rtp_receiver_->IncomingRtpPacket(header, restored_packet_ptr,
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|                                            packet_length, payload_specific,
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|                                            true)) {
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| -      return -1;
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| +      return false;
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|      }
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| -    return static_cast<int>(len);
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| +    return true;
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|    }
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|  
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| -  int SendRTCPPacket(const void* data, size_t len) override {
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| -    if (module_->IncomingRtcpPacket((const uint8_t*)data, len) == 0) {
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| -      return static_cast<int>(len);
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| -    }
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| -    return -1;
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| +  bool SendRtcp(const uint8_t* data, size_t len) override {
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| +    return module_->IncomingRtcpPacket((const uint8_t*)data, len) == 0;
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|    }
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|    int count_;
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|    int packet_loss_;
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| 
 |