| Index: webrtc/modules/rtp_rtcp/source/nack_rtx_unittest.cc
|
| diff --git a/webrtc/modules/rtp_rtcp/source/nack_rtx_unittest.cc b/webrtc/modules/rtp_rtcp/source/nack_rtx_unittest.cc
|
| index d32d09fab077a5d5c2add7d001d2ec2db626d849..6f13056f5f45ac32714f833e1efc0e0f19170635 100644
|
| --- a/webrtc/modules/rtp_rtcp/source/nack_rtx_unittest.cc
|
| +++ b/webrtc/modules/rtp_rtcp/source/nack_rtx_unittest.cc
|
| @@ -22,6 +22,7 @@
|
| #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;
|
|
|
| @@ -95,7 +96,7 @@ class RtxLoopBackTransport : public webrtc::Transport {
|
| 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];
|
| @@ -110,7 +111,7 @@ class RtxLoopBackTransport : public webrtc::Transport {
|
| 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)) {
|
| @@ -121,11 +122,11 @@ class RtxLoopBackTransport : public webrtc::Transport {
|
| }
|
| 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.
|
| @@ -133,7 +134,7 @@ class RtxLoopBackTransport : public webrtc::Transport {
|
| &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);
|
| @@ -144,21 +145,18 @@ class RtxLoopBackTransport : public webrtc::Transport {
|
| 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_;
|
|
|