Index: webrtc/modules/rtp_rtcp/source/rtp_sender_audio.cc |
diff --git a/webrtc/modules/rtp_rtcp/source/rtp_sender_audio.cc b/webrtc/modules/rtp_rtcp/source/rtp_sender_audio.cc |
index 0f32d4e7861d2b9454bc5f0d9a465f7c73e7002a..04ecd5e2e677e2dea523abb1ae3707f284255ade 100644 |
--- a/webrtc/modules/rtp_rtcp/source/rtp_sender_audio.cc |
+++ b/webrtc/modules/rtp_rtcp/source/rtp_sender_audio.cc |
@@ -208,7 +208,7 @@ bool RTPSenderAudio::SendAudio(FrameType frame_type, |
} |
return true; |
} |
- if (payload_size == 0 || payload_data == NULL) { |
+ if (payload_size == 0 || payload_data == nullptr) { |
if (frame_type == kEmptyFrame) { |
// we don't send empty audio RTP packets |
// no error since we use it to drive DTMF when we use VAD |