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Unified Diff: webrtc/modules/rtp_rtcp/source/rtp_fec_unittest.cc

Issue 2275443002: Minor fixes in FEC and RtpSender{,Video}. (pt. 2) (Closed) Base URL: https://chromium.googlesource.com/external/webrtc.git@header_reader_writer-pt1-move_copy_column
Patch Set: Rebase. Created 4 years, 3 months ago
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Index: webrtc/modules/rtp_rtcp/source/rtp_fec_unittest.cc
diff --git a/webrtc/modules/rtp_rtcp/source/rtp_fec_unittest.cc b/webrtc/modules/rtp_rtcp/source/rtp_fec_unittest.cc
index 012bc561e73d22f1c48cfcfc93280bd62b57127e..1bb1d6543cc14cbff76dae0ba5c5e5770f37fbb6 100644
--- a/webrtc/modules/rtp_rtcp/source/rtp_fec_unittest.cc
+++ b/webrtc/modules/rtp_rtcp/source/rtp_fec_unittest.cc
@@ -876,7 +876,7 @@ int RtpFecTest::ConstructMediaPacketsSeqNum(int num_media_packets,
constexpr uint32_t kMinPacketSize = kRtpHeaderSize;
const uint32_t kMaxPacketSize = IP_PACKET_SIZE - kRtpHeaderSize -
kTransportOverhead -
- ForwardErrorCorrection::PacketOverhead();
+ fec_.MaxPacketOverhead();
media_packet->length = random_.Rand(kMinPacketSize, kMaxPacketSize);
// Generate random values for the first 2 bytes
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