| Index: webrtc/modules/audio_processing/voice_detection_unittest.cc
|
| diff --git a/webrtc/modules/audio_processing/voice_detection_unittest.cc b/webrtc/modules/audio_processing/voice_detection_unittest.cc
|
| new file mode 100644
|
| index 0000000000000000000000000000000000000000..791bd55500b9844935ca83f102777cf4e9ca61bf
|
| --- /dev/null
|
| +++ b/webrtc/modules/audio_processing/voice_detection_unittest.cc
|
| @@ -0,0 +1,123 @@
|
| +/*
|
| + * Copyright (c) 2016 The WebRTC project authors. All Rights Reserved.
|
| + *
|
| + * Use of this source code is governed by a BSD-style license
|
| + * that can be found in the LICENSE file in the root of the source
|
| + * tree. An additional intellectual property rights grant can be found
|
| + * in the file PATENTS. All contributing project authors may
|
| + * be found in the AUTHORS file in the root of the source tree.
|
| + */
|
| +#include <vector>
|
| +
|
| +#include "testing/gtest/include/gtest/gtest.h"
|
| +#include "webrtc/base/array_view.h"
|
| +#include "webrtc/modules/audio_processing/audio_buffer.h"
|
| +#include "webrtc/modules/audio_processing/voice_detection_impl.h"
|
| +#include "webrtc/modules/audio_processing/test/audio_buffer_tools.h"
|
| +#include "webrtc/modules/audio_processing/test/bitexactness_tools.h"
|
| +
|
| +namespace webrtc {
|
| +namespace {
|
| +
|
| +const int kNumFramesToProcess = 1000;
|
| +
|
| +// Process one frame of data and produce the output.
|
| +void ProcessOneFrame(int sample_rate_hz,
|
| + AudioBuffer* audio_buffer,
|
| + VoiceDetectionImpl* voice_detection) {
|
| + if (sample_rate_hz > AudioProcessing::kSampleRate16kHz) {
|
| + audio_buffer->SplitIntoFrequencyBands();
|
| + }
|
| +
|
| + voice_detection->ProcessCaptureAudio(audio_buffer);
|
| +}
|
| +
|
| +// Processes a specified amount of frames, verifies the results and reports
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| +// any errors.
|
| +void RunBitexactnessTest(int sample_rate_hz,
|
| + size_t num_channels,
|
| + int frame_size_ms_reference,
|
| + bool stream_has_voice_reference,
|
| + VoiceDetection::Likelihood likelihood_reference) {
|
| + rtc::CriticalSection crit_capture;
|
| + VoiceDetectionImpl voice_detection(&crit_capture);
|
| + voice_detection.Initialize(sample_rate_hz > 16000 ? 16000 : sample_rate_hz);
|
| + voice_detection.Enable(true);
|
| +
|
| + int samples_per_channel = rtc::CheckedDivExact(sample_rate_hz, 100);
|
| + const StreamConfig capture_config(sample_rate_hz, num_channels, false);
|
| + AudioBuffer capture_buffer(
|
| + capture_config.num_frames(), capture_config.num_channels(),
|
| + capture_config.num_frames(), capture_config.num_channels(),
|
| + capture_config.num_frames());
|
| + test::InputAudioFile capture_file(
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| + test::GetApmCaptureTestVectorFileName(sample_rate_hz));
|
| + std::vector<float> capture_input(samples_per_channel * num_channels);
|
| + for (int frame_no = 0; frame_no < kNumFramesToProcess; ++frame_no) {
|
| + ReadFloatSamplesFromStereoFile(samples_per_channel, num_channels,
|
| + &capture_file, capture_input);
|
| +
|
| + test::CopyVectorToAudioBuffer(capture_config, capture_input,
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| + &capture_buffer);
|
| +
|
| + ProcessOneFrame(sample_rate_hz, &capture_buffer, &voice_detection);
|
| + }
|
| +
|
| + int frame_size_ms = voice_detection.frame_size_ms();
|
| + bool stream_has_voice = voice_detection.stream_has_voice();
|
| + VoiceDetection::Likelihood likelihood = voice_detection.likelihood();
|
| +
|
| + // Compare the outputs to the references.
|
| + EXPECT_EQ(frame_size_ms_reference, frame_size_ms);
|
| + EXPECT_EQ(stream_has_voice_reference, stream_has_voice);
|
| + EXPECT_EQ(likelihood_reference, likelihood);
|
| +}
|
| +
|
| +const int kFrameSizeMsReference = 10;
|
| +const bool kStreamHasVoiceReference = true;
|
| +const VoiceDetection::Likelihood kLikelihoodReference =
|
| + VoiceDetection::kLowLikelihood;
|
| +
|
| +} // namespace
|
| +
|
| +TEST(VoiceDetectionBitExactnessTest, Mono8kHz) {
|
| + RunBitexactnessTest(8000, 1, kFrameSizeMsReference, kStreamHasVoiceReference,
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| + kLikelihoodReference);
|
| +}
|
| +
|
| +TEST(VoiceDetectionBitExactnessTest, Mono16kHz) {
|
| + RunBitexactnessTest(16000, 1, kFrameSizeMsReference, kStreamHasVoiceReference,
|
| + kLikelihoodReference);
|
| +}
|
| +
|
| +TEST(VoiceDetectionBitExactnessTest, Mono32kHz) {
|
| + RunBitexactnessTest(32000, 1, kFrameSizeMsReference, kStreamHasVoiceReference,
|
| + kLikelihoodReference);
|
| +}
|
| +
|
| +TEST(VoiceDetectionBitExactnessTest, Mono48kHz) {
|
| + RunBitexactnessTest(48000, 1, kFrameSizeMsReference, kStreamHasVoiceReference,
|
| + kLikelihoodReference);
|
| +}
|
| +
|
| +TEST(VoiceDetectionBitExactnessTest, Stereo8kHz) {
|
| + RunBitexactnessTest(8000, 2, kFrameSizeMsReference, kStreamHasVoiceReference,
|
| + kLikelihoodReference);
|
| +}
|
| +
|
| +TEST(VoiceDetectionBitExactnessTest, Stereo16kHz) {
|
| + RunBitexactnessTest(16000, 2, kFrameSizeMsReference, kStreamHasVoiceReference,
|
| + kLikelihoodReference);
|
| +}
|
| +
|
| +TEST(VoiceDetectionBitExactnessTest, Stereo32kHz) {
|
| + RunBitexactnessTest(32000, 2, kFrameSizeMsReference, kStreamHasVoiceReference,
|
| + kLikelihoodReference);
|
| +}
|
| +
|
| +TEST(VoiceDetectionBitExactnessTest, Stereo48kHz) {
|
| + RunBitexactnessTest(48000, 2, kFrameSizeMsReference, kStreamHasVoiceReference,
|
| + kLikelihoodReference);
|
| +}
|
| +
|
| +} // namespace webrtc
|
|
|