| Index: talk/app/webrtc/rtpsender.cc
|
| diff --git a/talk/app/webrtc/rtpsender.cc b/talk/app/webrtc/rtpsender.cc
|
| index c0d23a0503cc08fb5a7fd8b301f8bd99bf076bd8..91e484b733c6263211402878875ff40030d53675 100644
|
| --- a/talk/app/webrtc/rtpsender.cc
|
| +++ b/talk/app/webrtc/rtpsender.cc
|
| @@ -44,7 +44,7 @@ LocalAudioSinkAdapter::~LocalAudioSinkAdapter() {
|
| void LocalAudioSinkAdapter::OnData(const void* audio_data,
|
| int bits_per_sample,
|
| int sample_rate,
|
| - int number_of_channels,
|
| + size_t number_of_channels,
|
| size_t number_of_frames) {
|
| rtc::CritScope lock(&lock_);
|
| if (sink_) {
|
|
|