Index: webrtc/modules/audio_coding/neteq/tools/rtp_file_source.cc |
diff --git a/webrtc/modules/audio_coding/neteq/tools/rtp_file_source.cc b/webrtc/modules/audio_coding/neteq/tools/rtp_file_source.cc |
index 9ca48e9ea5be0caa5657474e0fff3745bc56ba66..052436efbb6b8a9d031b21f597ef82315c6d8f41 100644 |
--- a/webrtc/modules/audio_coding/neteq/tools/rtp_file_source.cc |
+++ b/webrtc/modules/audio_coding/neteq/tools/rtp_file_source.cc |
@@ -59,7 +59,7 @@ std::unique_ptr<Packet> RtpFileSource::NextPacket() { |
while (true) { |
RtpPacket temp_packet; |
if (!rtp_reader_->NextPacket(&temp_packet)) { |
- return NULL; |
+ return nullptr; |
} |
if (temp_packet.original_length == 0) { |
// May be an RTCP packet. |
@@ -73,7 +73,7 @@ std::unique_ptr<Packet> RtpFileSource::NextPacket() { |
temp_packet.original_length, temp_packet.time_ms, *parser_.get())); |
if (!packet->valid_header()) { |
assert(false); |
- return NULL; |
+ return nullptr; |
} |
if (filter_.test(packet->header().payloadType) || |
(use_ssrc_filter_ && packet->header().ssrc != ssrc_)) { |