| 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_)) {
|
|
|