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 d95c9636d99c2c3bbd6e3df9b77f2246770ccb4f..d2559d6712db34e461a9302e9a853c2a7f41ee52 100644 |
--- a/webrtc/modules/rtp_rtcp/source/nack_rtx_unittest.cc |
+++ b/webrtc/modules/rtp_rtcp/source/nack_rtx_unittest.cc |
@@ -105,10 +105,7 @@ class RtxLoopBackTransport : public webrtc::Transport { |
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[1500]; |
RTPHeader header; |
rtc::scoped_ptr<RtpHeaderParser> parser(RtpHeaderParser::Create()); |
if (!parser->Parse(ptr, len, &header)) { |
@@ -136,21 +133,19 @@ class RtxLoopBackTransport : public webrtc::Transport { |
if (!parser->Parse(restored_packet, packet_length, &header)) { |
return false; |
} |
+ ptr = restored_packet; |
} else { |
rtp_payload_registry_->SetIncomingPayloadType(header); |
} |
- const uint8_t* restored_packet_payload = |
- restored_packet + header.headerLength; |
- packet_length -= header.headerLength; |
PayloadUnion payload_specific; |
if (!rtp_payload_registry_->GetPayloadSpecifics(header.payloadType, |
&payload_specific)) { |
return false; |
} |
- if (!rtp_receiver_->IncomingRtpPacket(header, restored_packet_payload, |
- packet_length, payload_specific, |
- true)) { |
+ if (!rtp_receiver_->IncomingRtpPacket(header, ptr + header.headerLength, |
+ packet_length - header.headerLength, |
+ payload_specific, true)) { |
return false; |
} |
return true; |