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Add tests for downmixing and no processing.

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BUG=issue419
TEST=audioproc_unittest

Review URL: https://webrtc-codereview.appspot.com/532001

git-svn-id: http://webrtc.googlecode.com/svn/trunk@2154 4adac7df-926f-26a2-2b94-8c16560cd09d
This commit is contained in:
andrew@webrtc.org 2012-05-02 00:04:10 +00:00
parent 63ea5ef5da
commit ecac9b721e
1 changed files with 60 additions and 12 deletions
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72
src/modules/audio_processing/test/unit_test.cc
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@ -50,6 +50,11 @@ namespace {
// be set to true with the command-line switch --write_ref_data.
bool write_ref_data = false;
const int kSampleRates[] = {8000, 16000, 32000};
const size_t kSampleRatesSize = sizeof(kSampleRates) / sizeof(*kSampleRates);
const int kChannels[] = {1, 2};
const size_t kChannelsSize = sizeof(kChannels) / sizeof(*kChannels);
class ApmTest : public ::testing::Test {
protected:
ApmTest();
@ -250,6 +255,14 @@ void SetFrameTo(AudioFrame* frame, int16_t value) {
}
}
void SetFrameTo(AudioFrame* frame, int16_t left, int16_t right) {
ASSERT_EQ(2, frame->_audioChannel);
for (int i = 0; i < frame->_payloadDataLengthInSamples * 2; i += 2) {
frame->_payloadData[i] = left;
frame->_payloadData[i + 1] = right;
}
}
int16_t MaxAudioFrame(const AudioFrame& frame) {
const int length = frame._payloadDataLengthInSamples * frame._audioChannel;
int16_t max = AbsValue(frame._payloadData[0]);
@ -961,6 +974,42 @@ TEST_F(ApmTest, VoiceDetection) {
// TODO(bjornv): Add tests for streamed voice; stream_has_voice()
}
TEST_F(ApmTest, VerifyDownMixing) {
for (size_t i = 0; i < kSampleRatesSize; i++) {
Init(kSampleRates[i], 2, 2, 1, false);
SetFrameTo(frame_, 1000, 2000);
AudioFrame mono_frame;
mono_frame._payloadDataLengthInSamples =
frame_->_payloadDataLengthInSamples;
mono_frame._audioChannel = 1;
SetFrameTo(&mono_frame, 1500);
EXPECT_EQ(apm_->kNoError, apm_->ProcessStream(frame_));
EXPECT_TRUE(FrameDataAreEqual(*frame_, mono_frame));
}
}
TEST_F(ApmTest, AllProcessingDisabledByDefault) {
EXPECT_FALSE(apm_->echo_cancellation()->is_enabled());
EXPECT_FALSE(apm_->echo_control_mobile()->is_enabled());
EXPECT_FALSE(apm_->gain_control()->is_enabled());
EXPECT_FALSE(apm_->high_pass_filter()->is_enabled());
EXPECT_FALSE(apm_->level_estimator()->is_enabled());
EXPECT_FALSE(apm_->noise_suppression()->is_enabled());
EXPECT_FALSE(apm_->voice_detection()->is_enabled());
}
TEST_F(ApmTest, NoProcessingWhenAllComponentsDisabled) {
for (size_t i = 0; i < kSampleRatesSize; i++) {
Init(kSampleRates[i], 2, 2, 2, false);
SetFrameTo(frame_, 1000, 2000);
AudioFrame frame_copy = *frame_;
for (int j = 0; j < 1000; j++) {
EXPECT_EQ(apm_->kNoError, apm_->ProcessStream(frame_));
EXPECT_TRUE(FrameDataAreEqual(*frame_, frame_copy));
}
}
}
TEST_F(ApmTest, SplittingFilter) {
// Verify the filter is not active through undistorted audio when:
// 1. No components are enabled...
@ -1081,25 +1130,24 @@ TEST_F(ApmTest, Process) {
ReadMessageLiteFromFile(ref_filename_, &ref_data);
} else {
// Write the desired tests to the protobuf reference file.
const int channels[] = {1, 2};
const size_t channels_size = sizeof(channels) / sizeof(*channels);
#if defined(WEBRTC_AUDIOPROC_FIXED_PROFILE)
// AECM doesn't support super-wb.
const int sample_rates[] = {8000, 16000};
const int kProcessSampleRates[] = {8000, 16000};
#elif defined(WEBRTC_AUDIOPROC_FLOAT_PROFILE)
const int sample_rates[] = {8000, 16000, 32000};
const int kProcessSampleRates[] = {8000, 16000, 32000};
#endif
const size_t sample_rates_size = sizeof(sample_rates) / sizeof(*sample_rates);
for (size_t i = 0; i < channels_size; i++) {
for (size_t j = 0; j < channels_size; j++) {
const size_t kProcessSampleRatesSize = sizeof(kProcessSampleRates) /
sizeof(*kProcessSampleRates);
for (size_t i = 0; i < kChannelsSize; i++) {
for (size_t j = 0; j < kChannelsSize; j++) {
// We can't have more output than input channels.
for (size_t k = 0; k <= j; k++) {
for (size_t l = 0; l < sample_rates_size; l++) {
for (size_t l = 0; l < kProcessSampleRatesSize; l++) {
webrtc::audioproc::Test* test = ref_data.add_test();
test->set_num_reverse_channels(channels[i]);
test->set_num_input_channels(channels[j]);
test->set_num_output_channels(channels[k]);
test->set_sample_rate(sample_rates[l]);
test->set_num_reverse_channels(kChannels[i]);
test->set_num_input_channels(kChannels[j]);
test->set_num_output_channels(kChannels[k]);
test->set_sample_rate(kProcessSampleRates[l]);
}
}
}