| 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;
|
|
|