| Index: webrtc/modules/rtp_rtcp/source/rtp_format_h264.cc
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| diff --git a/webrtc/modules/rtp_rtcp/source/rtp_format_h264.cc b/webrtc/modules/rtp_rtcp/source/rtp_format_h264.cc
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| index 092ac1cc4989137d520fa7938728819ee343630c..5aa7e27d2391e75ede72cba6c74bf318937121bc 100644
 | 
| --- a/webrtc/modules/rtp_rtcp/source/rtp_format_h264.cc
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| +++ b/webrtc/modules/rtp_rtcp/source/rtp_format_h264.cc
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| @@ -389,15 +389,6 @@ void RtpPacketizerH264::NextFragmentPacket(RtpPacketToSend* rtp_packet) {
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|    packets_.pop();
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|  }
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|  
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| -ProtectionType RtpPacketizerH264::GetProtectionType() {
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| -  return kProtectedPacket;
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| -}
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| -
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| -StorageType RtpPacketizerH264::GetStorageType(
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| -    uint32_t retransmission_settings) {
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| -  return kAllowRetransmission;
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| -}
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| -
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|  std::string RtpPacketizerH264::ToString() {
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|    return "RtpPacketizerH264";
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|  }
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| 
 |