| Index: webrtc/media/engine/webrtcvoiceengine.cc
|
| diff --git a/webrtc/media/engine/webrtcvoiceengine.cc b/webrtc/media/engine/webrtcvoiceengine.cc
|
| index f96c8e741e04ad94075ad69add97fab351230e01..af3d3525c96197829c550b67012d32f85d97ff4f 100644
|
| --- a/webrtc/media/engine/webrtcvoiceengine.cc
|
| +++ b/webrtc/media/engine/webrtcvoiceengine.cc
|
| @@ -158,7 +158,7 @@ bool FindCodec(const std::vector<AudioCodec>& codecs,
|
| AudioCodec* found_codec) {
|
| for (const AudioCodec& c : codecs) {
|
| if (c.Matches(codec)) {
|
| - if (found_codec != NULL) {
|
| + if (found_codec != nullptr) {
|
| *found_codec = c;
|
| }
|
| return true;
|
| @@ -2661,7 +2661,7 @@ void WebRtcVoiceMediaChannel::SetRawAudioSink(
|
| std::unique_ptr<webrtc::AudioSinkInterface> sink) {
|
| RTC_DCHECK(worker_thread_checker_.CalledOnValidThread());
|
| LOG(LS_VERBOSE) << "WebRtcVoiceMediaChannel::SetRawAudioSink: ssrc:" << ssrc
|
| - << " " << (sink ? "(ptr)" : "NULL");
|
| + << " " << (sink ? "(ptr)" : "null");
|
| if (ssrc == 0) {
|
| if (default_recv_ssrc_ != -1) {
|
| std::unique_ptr<webrtc::AudioSinkInterface> proxy_sink(
|
|
|