Index: webrtc/modules/rtp_rtcp/source/rtp_receiver_video.cc |
diff --git a/webrtc/modules/rtp_rtcp/source/rtp_receiver_video.cc b/webrtc/modules/rtp_rtcp/source/rtp_receiver_video.cc |
index d6c5e5c29b6ac4667f80681c0df5322d645747a1..622806c0cff325c713386cab659abcecd0627014 100644 |
--- a/webrtc/modules/rtp_rtcp/source/rtp_receiver_video.cc |
+++ b/webrtc/modules/rtp_rtcp/source/rtp_receiver_video.cc |
@@ -65,9 +65,10 @@ int32_t RTPReceiverVideo::ParseRtpPacket(WebRtcRTPHeader* rtp_header, |
const size_t payload_data_length = |
payload_length - rtp_header->header.paddingLength; |
- if (payload == NULL || payload_data_length == 0) { |
- return data_callback_->OnReceivedPayloadData(NULL, 0, rtp_header) == 0 ? 0 |
- : -1; |
+ if (payload == nullptr || payload_data_length == 0) { |
+ return data_callback_->OnReceivedPayloadData(nullptr, 0, rtp_header) == 0 |
+ ? 0 |
+ : -1; |
} |
if (first_packet_received_()) { |
@@ -77,7 +78,7 @@ int32_t RTPReceiverVideo::ParseRtpPacket(WebRtcRTPHeader* rtp_header, |
// We are not allowed to hold a critical section when calling below functions. |
std::unique_ptr<RtpDepacketizer> depacketizer( |
RtpDepacketizer::Create(rtp_header->type.Video.codec)); |
- if (depacketizer.get() == NULL) { |
+ if (depacketizer.get() == nullptr) { |
LOG(LS_ERROR) << "Failed to create depacketizer."; |
return -1; |
} |