Index: webrtc/modules/audio_processing/test/audio_processing_simulator.cc |
diff --git a/webrtc/modules/audio_processing/test/audio_processing_simulator.cc b/webrtc/modules/audio_processing/test/audio_processing_simulator.cc |
index 57b03ed4b29cb2ef70c9debf3e8aedc7d1974dba..a34c3ca78ee2922722031c98df0a76e02d7e3269 100644 |
--- a/webrtc/modules/audio_processing/test/audio_processing_simulator.cc |
+++ b/webrtc/modules/audio_processing/test/audio_processing_simulator.cc |
@@ -44,6 +44,10 @@ std::string GetIndexedOutputWavFilename(const std::string& wav_name, |
} // namespace |
+SimulationSettings::SimulationSettings() = default; |
+SimulationSettings::SimulationSettings(const SimulationSettings&) = default; |
+SimulationSettings::~SimulationSettings() = default; |
+ |
void CopyToAudioFrame(const ChannelBuffer<float>& src, AudioFrame* dest) { |
RTC_CHECK_EQ(src.num_channels(), dest->num_channels_); |
RTC_CHECK_EQ(src.num_frames(), dest->samples_per_channel_); |
@@ -55,6 +59,12 @@ void CopyToAudioFrame(const ChannelBuffer<float>& src, AudioFrame* dest) { |
} |
} |
+AudioProcessingSimulator::AudioProcessingSimulator( |
+ const SimulationSettings& settings) |
+ : settings_(settings) {} |
+ |
+AudioProcessingSimulator::~AudioProcessingSimulator() = default; |
+ |
AudioProcessingSimulator::ScopedTimer::~ScopedTimer() { |
int64_t interval = rtc::TimeNanos() - start_time_; |
proc_time_->sum += interval; |