diff options
Diffstat (limited to 'webrtc/modules/audio_processing/test/audio_processing_unittest.cc')
-rw-r--r-- | webrtc/modules/audio_processing/test/audio_processing_unittest.cc | 128 |
1 files changed, 60 insertions, 68 deletions
diff --git a/webrtc/modules/audio_processing/test/audio_processing_unittest.cc b/webrtc/modules/audio_processing/test/audio_processing_unittest.cc index d4bb8aa513..6eae1e5b94 100644 --- a/webrtc/modules/audio_processing/test/audio_processing_unittest.cc +++ b/webrtc/modules/audio_processing/test/audio_processing_unittest.cc @@ -14,6 +14,7 @@ #include <limits> #include <queue> +#include "webrtc/base/arraysize.h" #include "webrtc/base/scoped_ptr.h" #include "webrtc/common_audio/include/audio_util.h" #include "webrtc/common_audio/resampler/include/push_resampler.h" @@ -49,11 +50,8 @@ namespace { // file. This is the typical case. When the file should be updated, it can // be set to true with the command-line switch --write_ref_data. bool write_ref_data = false; -const int kChannels[] = {1, 2}; -const size_t kChannelsSize = sizeof(kChannels) / sizeof(*kChannels); - +const google::protobuf::int32 kChannels[] = {1, 2}; const int kSampleRates[] = {8000, 16000, 32000, 48000}; -const size_t kSampleRatesSize = sizeof(kSampleRates) / sizeof(*kSampleRates); #if defined(WEBRTC_AUDIOPROC_FIXED_PROFILE) // AECM doesn't support super-wb. @@ -61,8 +59,6 @@ const int kProcessSampleRates[] = {8000, 16000}; #elif defined(WEBRTC_AUDIOPROC_FLOAT_PROFILE) const int kProcessSampleRates[] = {8000, 16000, 32000, 48000}; #endif -const size_t kProcessSampleRatesSize = sizeof(kProcessSampleRates) / - sizeof(*kProcessSampleRates); enum StreamDirection { kForward = 0, kReverse }; @@ -96,7 +92,7 @@ int TotalChannelsFromLayout(AudioProcessing::ChannelLayout layout) { return 3; } assert(false); - return -1; + return 0; } int TruncateToMultipleOf10(int value) { @@ -104,25 +100,25 @@ int TruncateToMultipleOf10(int value) { } void MixStereoToMono(const float* stereo, float* mono, - int samples_per_channel) { - for (int i = 0; i < samples_per_channel; ++i) + size_t samples_per_channel) { + for (size_t i = 0; i < samples_per_channel; ++i) mono[i] = (stereo[i * 2] + stereo[i * 2 + 1]) / 2; } void MixStereoToMono(const int16_t* stereo, int16_t* mono, - int samples_per_channel) { - for (int i = 0; i < samples_per_channel; ++i) + size_t samples_per_channel) { + for (size_t i = 0; i < samples_per_channel; ++i) mono[i] = (stereo[i * 2] + stereo[i * 2 + 1]) >> 1; } -void CopyLeftToRightChannel(int16_t* stereo, int samples_per_channel) { - for (int i = 0; i < samples_per_channel; i++) { +void CopyLeftToRightChannel(int16_t* stereo, size_t samples_per_channel) { + for (size_t i = 0; i < samples_per_channel; i++) { stereo[i * 2 + 1] = stereo[i * 2]; } } -void VerifyChannelsAreEqual(int16_t* stereo, int samples_per_channel) { - for (int i = 0; i < samples_per_channel; i++) { +void VerifyChannelsAreEqual(int16_t* stereo, size_t samples_per_channel) { + for (size_t i = 0; i < samples_per_channel; i++) { EXPECT_EQ(stereo[i * 2 + 1], stereo[i * 2]); } } @@ -195,9 +191,9 @@ T AbsValue(T a) { } int16_t MaxAudioFrame(const AudioFrame& frame) { - const int length = frame.samples_per_channel_ * frame.num_channels_; + const size_t length = frame.samples_per_channel_ * frame.num_channels_; int16_t max_data = AbsValue(frame.data_[0]); - for (int i = 1; i < length; i++) { + for (size_t i = 1; i < length; i++) { max_data = std::max(max_data, AbsValue(frame.data_[i])); } @@ -898,7 +894,7 @@ TEST_F(ApmTest, SampleRatesInt) { EXPECT_EQ(apm_->kBadSampleRateError, ProcessStreamChooser(kIntFormat)); // Testing valid sample rates int fs[] = {8000, 16000, 32000, 48000}; - for (size_t i = 0; i < sizeof(fs) / sizeof(*fs); i++) { + for (size_t i = 0; i < arraysize(fs); i++) { SetContainerFormat(fs[i], 2, frame_, &float_cb_); EXPECT_NOERR(ProcessStreamChooser(kIntFormat)); } @@ -917,7 +913,7 @@ TEST_F(ApmTest, EchoCancellation) { EchoCancellation::kModerateSuppression, EchoCancellation::kHighSuppression, }; - for (size_t i = 0; i < sizeof(level)/sizeof(*level); i++) { + for (size_t i = 0; i < arraysize(level); i++) { EXPECT_EQ(apm_->kNoError, apm_->echo_cancellation()->set_suppression_level(level[i])); EXPECT_EQ(level[i], @@ -994,7 +990,7 @@ TEST_F(ApmTest, DISABLED_EchoCancellationReportsCorrectDelays) { // Test a couple of corner cases and verify that the estimated delay is // within a valid region (set to +-1.5 blocks). Note that these cases are // sampling frequency dependent. - for (size_t i = 0; i < kProcessSampleRatesSize; i++) { + for (size_t i = 0; i < arraysize(kProcessSampleRates); i++) { Init(kProcessSampleRates[i], kProcessSampleRates[i], kProcessSampleRates[i], @@ -1066,7 +1062,7 @@ TEST_F(ApmTest, EchoControlMobile) { EchoControlMobile::kSpeakerphone, EchoControlMobile::kLoudSpeakerphone, }; - for (size_t i = 0; i < sizeof(mode)/sizeof(*mode); i++) { + for (size_t i = 0; i < arraysize(mode); i++) { EXPECT_EQ(apm_->kNoError, apm_->echo_control_mobile()->set_routing_mode(mode[i])); EXPECT_EQ(mode[i], @@ -1131,7 +1127,7 @@ TEST_F(ApmTest, GainControl) { GainControl::kAdaptiveDigital, GainControl::kFixedDigital }; - for (size_t i = 0; i < sizeof(mode)/sizeof(*mode); i++) { + for (size_t i = 0; i < arraysize(mode); i++) { EXPECT_EQ(apm_->kNoError, apm_->gain_control()->set_mode(mode[i])); EXPECT_EQ(mode[i], apm_->gain_control()->mode()); @@ -1147,7 +1143,7 @@ TEST_F(ApmTest, GainControl) { apm_->gain_control()->target_level_dbfs())); int level_dbfs[] = {0, 6, 31}; - for (size_t i = 0; i < sizeof(level_dbfs)/sizeof(*level_dbfs); i++) { + for (size_t i = 0; i < arraysize(level_dbfs); i++) { EXPECT_EQ(apm_->kNoError, apm_->gain_control()->set_target_level_dbfs(level_dbfs[i])); EXPECT_EQ(level_dbfs[i], apm_->gain_control()->target_level_dbfs()); @@ -1165,7 +1161,7 @@ TEST_F(ApmTest, GainControl) { apm_->gain_control()->compression_gain_db())); int gain_db[] = {0, 10, 90}; - for (size_t i = 0; i < sizeof(gain_db)/sizeof(*gain_db); i++) { + for (size_t i = 0; i < arraysize(gain_db); i++) { EXPECT_EQ(apm_->kNoError, apm_->gain_control()->set_compression_gain_db(gain_db[i])); EXPECT_EQ(gain_db[i], apm_->gain_control()->compression_gain_db()); @@ -1196,14 +1192,14 @@ TEST_F(ApmTest, GainControl) { apm_->gain_control()->analog_level_maximum())); int min_level[] = {0, 255, 1024}; - for (size_t i = 0; i < sizeof(min_level)/sizeof(*min_level); i++) { + for (size_t i = 0; i < arraysize(min_level); i++) { EXPECT_EQ(apm_->kNoError, apm_->gain_control()->set_analog_level_limits(min_level[i], 1024)); EXPECT_EQ(min_level[i], apm_->gain_control()->analog_level_minimum()); } int max_level[] = {0, 1024, 65535}; - for (size_t i = 0; i < sizeof(min_level)/sizeof(*min_level); i++) { + for (size_t i = 0; i < arraysize(min_level); i++) { EXPECT_EQ(apm_->kNoError, apm_->gain_control()->set_analog_level_limits(0, max_level[i])); EXPECT_EQ(max_level[i], apm_->gain_control()->analog_level_maximum()); @@ -1242,7 +1238,7 @@ void ApmTest::RunQuantizedVolumeDoesNotGetStuckTest(int sample_rate) { // Verifies that despite volume slider quantization, the AGC can continue to // increase its volume. TEST_F(ApmTest, QuantizedVolumeDoesNotGetStuck) { - for (size_t i = 0; i < kSampleRatesSize; ++i) { + for (size_t i = 0; i < arraysize(kSampleRates); ++i) { RunQuantizedVolumeDoesNotGetStuckTest(kSampleRates[i]); } } @@ -1287,7 +1283,7 @@ void ApmTest::RunManualVolumeChangeIsPossibleTest(int sample_rate) { } TEST_F(ApmTest, ManualVolumeChangeIsPossible) { - for (size_t i = 0; i < kSampleRatesSize; ++i) { + for (size_t i = 0; i < arraysize(kSampleRates); ++i) { RunManualVolumeChangeIsPossibleTest(kSampleRates[i]); } } @@ -1295,11 +1291,11 @@ TEST_F(ApmTest, ManualVolumeChangeIsPossible) { #if !defined(WEBRTC_ANDROID) && !defined(WEBRTC_IOS) TEST_F(ApmTest, AgcOnlyAdaptsWhenTargetSignalIsPresent) { const int kSampleRateHz = 16000; - const int kSamplesPerChannel = - AudioProcessing::kChunkSizeMs * kSampleRateHz / 1000; + const size_t kSamplesPerChannel = + static_cast<size_t>(AudioProcessing::kChunkSizeMs * kSampleRateHz / 1000); const int kNumInputChannels = 2; const int kNumOutputChannels = 1; - const int kNumChunks = 700; + const size_t kNumChunks = 700; const float kScaleFactor = 0.25f; Config config; std::vector<webrtc::Point> geometry; @@ -1313,8 +1309,8 @@ TEST_F(ApmTest, AgcOnlyAdaptsWhenTargetSignalIsPresent) { EXPECT_EQ(kNoErr, apm->gain_control()->Enable(true)); ChannelBuffer<float> src_buf(kSamplesPerChannel, kNumInputChannels); ChannelBuffer<float> dest_buf(kSamplesPerChannel, kNumOutputChannels); - const int max_length = kSamplesPerChannel * std::max(kNumInputChannels, - kNumOutputChannels); + const size_t max_length = kSamplesPerChannel * std::max(kNumInputChannels, + kNumOutputChannels); rtc::scoped_ptr<int16_t[]> int_data(new int16_t[max_length]); rtc::scoped_ptr<float[]> float_data(new float[max_length]); std::string filename = ResourceFilePath("far", kSampleRateHz); @@ -1326,13 +1322,13 @@ TEST_F(ApmTest, AgcOnlyAdaptsWhenTargetSignalIsPresent) { bool is_target = false; EXPECT_CALL(*beamformer, is_target_present()) .WillRepeatedly(testing::ReturnPointee(&is_target)); - for (int i = 0; i < kNumChunks; ++i) { + for (size_t i = 0; i < kNumChunks; ++i) { ASSERT_TRUE(ReadChunk(far_file, int_data.get(), float_data.get(), &src_buf)); for (int j = 0; j < kNumInputChannels; ++j) { - for (int k = 0; k < kSamplesPerChannel; ++k) { + for (size_t k = 0; k < kSamplesPerChannel; ++k) { src_buf.channels()[j][k] *= kScaleFactor; } } @@ -1351,13 +1347,13 @@ TEST_F(ApmTest, AgcOnlyAdaptsWhenTargetSignalIsPresent) { apm->gain_control()->compression_gain_db()); rewind(far_file); is_target = true; - for (int i = 0; i < kNumChunks; ++i) { + for (size_t i = 0; i < kNumChunks; ++i) { ASSERT_TRUE(ReadChunk(far_file, int_data.get(), float_data.get(), &src_buf)); for (int j = 0; j < kNumInputChannels; ++j) { - for (int k = 0; k < kSamplesPerChannel; ++k) { + for (size_t k = 0; k < kSamplesPerChannel; ++k) { src_buf.channels()[j][k] *= kScaleFactor; } } @@ -1386,7 +1382,7 @@ TEST_F(ApmTest, NoiseSuppression) { NoiseSuppression::kHigh, NoiseSuppression::kVeryHigh }; - for (size_t i = 0; i < sizeof(level)/sizeof(*level); i++) { + for (size_t i = 0; i < arraysize(level); i++) { EXPECT_EQ(apm_->kNoError, apm_->noise_suppression()->set_level(level[i])); EXPECT_EQ(level[i], apm_->noise_suppression()->level()); @@ -1488,7 +1484,7 @@ TEST_F(ApmTest, VoiceDetection) { VoiceDetection::kModerateLikelihood, VoiceDetection::kHighLikelihood }; - for (size_t i = 0; i < sizeof(likelihood)/sizeof(*likelihood); i++) { + for (size_t i = 0; i < arraysize(likelihood); i++) { EXPECT_EQ(apm_->kNoError, apm_->voice_detection()->set_likelihood(likelihood[i])); EXPECT_EQ(likelihood[i], apm_->voice_detection()->likelihood()); @@ -1520,7 +1516,7 @@ TEST_F(ApmTest, VoiceDetection) { AudioFrame::kVadPassive, AudioFrame::kVadUnknown }; - for (size_t i = 0; i < sizeof(activity)/sizeof(*activity); i++) { + for (size_t i = 0; i < arraysize(activity); i++) { frame_->vad_activity_ = activity[i]; EXPECT_EQ(apm_->kNoError, apm_->ProcessStream(frame_)); EXPECT_EQ(activity[i], frame_->vad_activity_); @@ -1546,7 +1542,7 @@ TEST_F(ApmTest, AllProcessingDisabledByDefault) { } TEST_F(ApmTest, NoProcessingWhenAllComponentsDisabled) { - for (size_t i = 0; i < kSampleRatesSize; i++) { + for (size_t i = 0; i < arraysize(kSampleRates); i++) { Init(kSampleRates[i], kSampleRates[i], kSampleRates[i], 2, 2, 2, false); SetFrameTo(frame_, 1000, 2000); AudioFrame frame_copy; @@ -1598,7 +1594,7 @@ TEST_F(ApmTest, NoProcessingWhenAllComponentsDisabledFloat) { TEST_F(ApmTest, IdenticalInputChannelsResultInIdenticalOutputChannels) { EnableAllComponents(); - for (size_t i = 0; i < kProcessSampleRatesSize; i++) { + for (size_t i = 0; i < arraysize(kProcessSampleRates); i++) { Init(kProcessSampleRates[i], kProcessSampleRates[i], kProcessSampleRates[i], @@ -1937,8 +1933,8 @@ TEST_F(ApmTest, FloatAndIntInterfacesGiveSimilarResults) { const int num_render_channels = test->num_reverse_channels(); const int num_input_channels = test->num_input_channels(); const int num_output_channels = test->num_output_channels(); - const int samples_per_channel = test->sample_rate() * - AudioProcessing::kChunkSizeMs / 1000; + const size_t samples_per_channel = static_cast<size_t>( + test->sample_rate() * AudioProcessing::kChunkSizeMs / 1000); Init(test->sample_rate(), test->sample_rate(), test->sample_rate(), num_input_channels, num_output_channels, num_render_channels, true); @@ -2030,9 +2026,9 @@ TEST_F(ApmTest, Process) { OpenFileAndReadMessage(ref_filename_, &ref_data); } else { // Write the desired tests to the protobuf reference file. - for (size_t i = 0; i < kChannelsSize; i++) { - for (size_t j = 0; j < kChannelsSize; j++) { - for (size_t l = 0; l < kProcessSampleRatesSize; l++) { + for (size_t i = 0; i < arraysize(kChannels); i++) { + for (size_t j = 0; j < arraysize(kChannels); j++) { + for (size_t l = 0; l < arraysize(kProcessSampleRates); l++) { audioproc::Test* test = ref_data.add_test(); test->set_num_reverse_channels(kChannels[i]); test->set_num_input_channels(kChannels[j]); @@ -2259,12 +2255,11 @@ TEST_F(ApmTest, NoErrorsWithKeyboardChannel) { {AudioProcessing::kStereoAndKeyboard, AudioProcessing::kMono}, {AudioProcessing::kStereoAndKeyboard, AudioProcessing::kStereo}, }; - size_t channel_format_size = sizeof(cf) / sizeof(*cf); rtc::scoped_ptr<AudioProcessing> ap(AudioProcessing::Create()); // Enable one component just to ensure some processing takes place. ap->noise_suppression()->Enable(true); - for (size_t i = 0; i < channel_format_size; ++i) { + for (size_t i = 0; i < arraysize(cf); ++i) { const int in_rate = 44100; const int out_rate = 48000; ChannelBuffer<float> in_cb(SamplesFromRate(in_rate), @@ -2291,7 +2286,7 @@ TEST_F(ApmTest, NoErrorsWithKeyboardChannel) { // error results to the supplied accumulators. void UpdateBestSNR(const float* ref, const float* test, - int length, + size_t length, int expected_delay, double* variance_acc, double* sq_error_acc) { @@ -2303,7 +2298,7 @@ void UpdateBestSNR(const float* ref, ++delay) { double sq_error = 0; double variance = 0; - for (int i = 0; i < length - delay; ++i) { + for (size_t i = 0; i < length - delay; ++i) { double error = test[i + delay] - ref[i]; sq_error += error * error; variance += ref[i] * ref[i]; @@ -2355,14 +2350,10 @@ class AudioProcessingTest static void SetUpTestCase() { // Create all needed output reference files. const int kNativeRates[] = {8000, 16000, 32000, 48000}; - const size_t kNativeRatesSize = - sizeof(kNativeRates) / sizeof(*kNativeRates); const int kNumChannels[] = {1, 2}; - const size_t kNumChannelsSize = - sizeof(kNumChannels) / sizeof(*kNumChannels); - for (size_t i = 0; i < kNativeRatesSize; ++i) { - for (size_t j = 0; j < kNumChannelsSize; ++j) { - for (size_t k = 0; k < kNumChannelsSize; ++k) { + for (size_t i = 0; i < arraysize(kNativeRates); ++i) { + for (size_t j = 0; j < arraysize(kNumChannels); ++j) { + for (size_t k = 0; k < arraysize(kNumChannels); ++k) { // The reference files always have matching input and output channels. ProcessFormat(kNativeRates[i], kNativeRates[i], kNativeRates[i], kNativeRates[i], kNumChannels[j], kNumChannels[j], @@ -2461,18 +2452,19 @@ class AudioProcessingTest // Dump forward output to file. Interleave(out_cb.channels(), out_cb.num_frames(), out_cb.num_channels(), float_data.get()); - int out_length = out_cb.num_channels() * out_cb.num_frames(); + size_t out_length = out_cb.num_channels() * out_cb.num_frames(); - ASSERT_EQ(static_cast<size_t>(out_length), + ASSERT_EQ(out_length, fwrite(float_data.get(), sizeof(float_data[0]), out_length, out_file)); // Dump reverse output to file. Interleave(rev_out_cb.channels(), rev_out_cb.num_frames(), rev_out_cb.num_channels(), float_data.get()); - int rev_out_length = rev_out_cb.num_channels() * rev_out_cb.num_frames(); + size_t rev_out_length = + rev_out_cb.num_channels() * rev_out_cb.num_frames(); - ASSERT_EQ(static_cast<size_t>(rev_out_length), + ASSERT_EQ(rev_out_length, fwrite(float_data.get(), sizeof(float_data[0]), rev_out_length, rev_out_file)); @@ -2508,9 +2500,8 @@ TEST_P(AudioProcessingTest, Formats) { {2, 2, 1, 1}, {2, 2, 2, 2}, }; - size_t channel_format_size = sizeof(cf) / sizeof(*cf); - for (size_t i = 0; i < channel_format_size; ++i) { + for (size_t i = 0; i < arraysize(cf); ++i) { ProcessFormat(input_rate_, output_rate_, reverse_input_rate_, reverse_output_rate_, cf[i].num_input, cf[i].num_output, cf[i].num_reverse_input, cf[i].num_reverse_output, "out"); @@ -2560,8 +2551,8 @@ TEST_P(AudioProcessingTest, Formats) { ASSERT_TRUE(out_file != NULL); ASSERT_TRUE(ref_file != NULL); - const int ref_length = SamplesFromRate(ref_rate) * out_num; - const int out_length = SamplesFromRate(out_rate) * out_num; + const size_t ref_length = SamplesFromRate(ref_rate) * out_num; + const size_t out_length = SamplesFromRate(out_rate) * out_num; // Data from the reference file. rtc::scoped_ptr<float[]> ref_data(new float[ref_length]); // Data from the output file. @@ -2601,8 +2592,9 @@ TEST_P(AudioProcessingTest, Formats) { if (out_rate != ref_rate) { // Resample the output back to its internal processing rate if // necssary. - ASSERT_EQ(ref_length, resampler.Resample(out_ptr, out_length, - cmp_data.get(), ref_length)); + ASSERT_EQ(ref_length, + static_cast<size_t>(resampler.Resample( + out_ptr, out_length, cmp_data.get(), ref_length))); out_ptr = cmp_data.get(); } |