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_; |