Index: webrtc/modules/audio_processing/test/conversational_speech/multiend_call.cc |
diff --git a/webrtc/modules/audio_processing/test/conversational_speech/multiend_call.cc b/webrtc/modules/audio_processing/test/conversational_speech/multiend_call.cc |
index ad1d9a0c87e90d9dc4c398bc21d120673379c366..624f981ae51f0e22b3928652a875c2e1618812f2 100644 |
--- a/webrtc/modules/audio_processing/test/conversational_speech/multiend_call.cc |
+++ b/webrtc/modules/audio_processing/test/conversational_speech/multiend_call.cc |
@@ -120,9 +120,9 @@ bool MultiEndCall::CheckTiming() { |
// Begin and end timestamps for the current turn. |
int offset_samples = millisecond_to_samples( |
- turn.offset, it->second->sample_rate()); |
- size_t begin_timestamp = last_turn.end + offset_samples; |
- size_t end_timestamp = begin_timestamp + it->second->num_samples(); |
+ turn.offset, it->second->SampleRate()); |
+ std::size_t begin_timestamp = last_turn.end + offset_samples; |
+ std::size_t end_timestamp = begin_timestamp + it->second->NumSamples(); |
LOG(LS_INFO) << "turn #" << turn_index << " " << begin_timestamp |
<< "-" << end_timestamp << " ms"; |