| Index: webrtc/modules/audio_processing/intelligibility/test/intelligibility_proc.cc
|
| diff --git a/webrtc/modules/audio_processing/intelligibility/test/intelligibility_proc.cc b/webrtc/modules/audio_processing/intelligibility/test/intelligibility_proc.cc
|
| index 4d2f5f4c5d71a1e09a575b57cdc2dc6354f52a7d..757f9d83409333e20a957e2316c60de189527536 100644
|
| --- a/webrtc/modules/audio_processing/intelligibility/test/intelligibility_proc.cc
|
| +++ b/webrtc/modules/audio_processing/intelligibility/test/intelligibility_proc.cc
|
| @@ -23,10 +23,14 @@
|
| #include "gflags/gflags.h"
|
| #include "testing/gtest/include/gtest/gtest.h"
|
| #include "webrtc/base/checks.h"
|
| +#include "webrtc/base/criticalsection.h"
|
| #include "webrtc/common_audio/real_fourier.h"
|
| #include "webrtc/common_audio/wav_file.h"
|
| +#include "webrtc/modules/audio_processing/audio_buffer.h"
|
| +#include "webrtc/modules/audio_processing/include/audio_processing.h"
|
| #include "webrtc/modules/audio_processing/intelligibility/intelligibility_enhancer.h"
|
| #include "webrtc/modules/audio_processing/intelligibility/intelligibility_utils.h"
|
| +#include "webrtc/modules/audio_processing/noise_suppression_impl.h"
|
| #include "webrtc/system_wrappers/include/critical_section_wrapper.h"
|
| #include "webrtc/test/testsupport/fileutils.h"
|
|
|
| @@ -115,6 +119,17 @@ void void_main(int argc, char* argv[]) {
|
| config.analysis_rate = FLAGS_ana_rate;
|
| config.gain_change_limit = FLAGS_gain_limit;
|
| IntelligibilityEnhancer enh(config);
|
| + rtc::CriticalSection crit;
|
| + NoiseSuppressionImpl ns(&crit);
|
| + ns.Initialize(kNumChannels, FLAGS_sample_rate);
|
| + ns.Enable(true);
|
| +
|
| + AudioBuffer capture_audio(fragment_size,
|
| + kNumChannels,
|
| + fragment_size,
|
| + kNumChannels,
|
| + fragment_size);
|
| + StreamConfig stream_config(FLAGS_sample_rate, kNumChannels);
|
|
|
| // Slice the input into smaller chunks, as the APM would do, and feed them
|
| // through the enhancer.
|
| @@ -122,7 +137,10 @@ void void_main(int argc, char* argv[]) {
|
| float* noise_cursor = &noise_fpcm[0];
|
|
|
| for (size_t i = 0; i < samples; i += fragment_size) {
|
| - enh.AnalyzeCaptureAudio(&noise_cursor, FLAGS_sample_rate, kNumChannels);
|
| + capture_audio.CopyFrom(&noise_cursor, stream_config);
|
| + ns.AnalyzeCaptureAudio(&capture_audio);
|
| + ns.ProcessCaptureAudio(&capture_audio);
|
| + enh.SetCaptureNoiseEstimate(ns.noise_estimate());
|
| enh.ProcessRenderAudio(&clear_cursor, FLAGS_sample_rate, kNumChannels);
|
| clear_cursor += fragment_size;
|
| noise_cursor += fragment_size;
|
|
|