| 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;
|
| }
|
|
|