Add a fixed-point profile to the APM unit test.

It uses fixed-point NS, AECM and adaptive digital AGC. It's selected by enabling "prefer_fixed_point" in common.gypi.
Review URL: http://webrtc-codereview.appspot.com/88009

git-svn-id: http://webrtc.googlecode.com/svn/trunk@266 4adac7df-926f-26a2-2b94-8c16560cd09d
This commit is contained in:
ajm@google.com 2011-07-28 17:34:04 +00:00
parent 11791b23f7
commit 59e41405d1
4 changed files with 44 additions and 15 deletions

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@ -14,6 +14,13 @@
{
'target_name': 'unit_test',
'type': 'executable',
'conditions': [
['prefer_fixed_point==1', {
'defines': ['WEBRTC_APM_UNIT_TEST_FIXED_PROFILE'],
}, {
'defines': ['WEBRTC_APM_UNIT_TEST_FLOAT_PROFILE'],
}],
],
'dependencies': [
'source/apm.gyp:audio_processing',
'../../../system_wrappers/source/system_wrappers.gyp:system_wrappers',
@ -49,7 +56,6 @@
'test/process_test/process_test.cc',
],
},
],
}

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@ -8,7 +8,7 @@
* be found in the AUTHORS file in the root of the source tree.
*/
#include <cstdio>
#include <stdio.h>
#include <gtest/gtest.h>
@ -20,9 +20,9 @@
#endif
#include "event_wrapper.h"
#include "module_common_types.h"
#include "signal_processing_library.h"
#include "thread_wrapper.h"
#include "trace.h"
#include "signal_processing_library.h"
using webrtc::AudioProcessing;
using webrtc::AudioFrame;
@ -37,10 +37,16 @@ using webrtc::EchoControlMobile;
using webrtc::VoiceDetection;
namespace {
// When true, this will compare the output data with the results stored to
// When false, this will compare the output data with the results stored to
// file. This is the typical case. When the file should be updated, it can
// be set to false with the command-line switch --write_output_data.
bool global_read_output_data = true;
// be set to true with the command-line switch --write_output_data.
bool write_output_data = false;
#if defined(WEBRTC_APM_UNIT_TEST_FIXED_PROFILE)
const char kOutputFileName[] = "output_data_fixed.pb";
#elif defined(WEBRTC_APM_UNIT_TEST_FLOAT_PROFILE)
const char kOutputFileName[] = "output_data_float.pb";
#endif
class ApmEnvironment : public ::testing::Environment {
public:
@ -412,15 +418,19 @@ TEST_F(ApmTest, Process) {
GOOGLE_PROTOBUF_VERIFY_VERSION;
audio_processing_unittest::OutputData output_data;
if (global_read_output_data) {
ReadMessageLiteFromFile("output_data.pb", &output_data);
if (!write_output_data) {
ReadMessageLiteFromFile(kOutputFileName, &output_data);
} else {
// We don't have a file; add the required tests to the protobuf.
// TODO(ajm): vary the output channels as well?
const int channels[] = {1, 2};
const size_t channels_size = sizeof(channels) / sizeof(*channels);
#if defined(WEBRTC_APM_UNIT_TEST_FIXED_PROFILE)
// AECM doesn't support super-wb.
const int sample_rates[] = {8000, 16000};
#elif defined(WEBRTC_APM_UNIT_TEST_FLOAT_PROFILE)
const int sample_rates[] = {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++) {
@ -435,6 +445,14 @@ TEST_F(ApmTest, Process) {
}
}
#if defined(WEBRTC_APM_UNIT_TEST_FIXED_PROFILE)
EXPECT_EQ(apm_->kNoError, apm_->set_sample_rate_hz(16000));
EXPECT_EQ(apm_->kNoError, apm_->echo_control_mobile()->Enable(true));
EXPECT_EQ(apm_->kNoError,
apm_->gain_control()->set_mode(GainControl::kAdaptiveDigital));
EXPECT_EQ(apm_->kNoError, apm_->gain_control()->Enable(true));
#elif defined(WEBRTC_APM_UNIT_TEST_FLOAT_PROFILE)
EXPECT_EQ(apm_->kNoError,
apm_->echo_cancellation()->enable_drift_compensation(true));
EXPECT_EQ(apm_->kNoError,
@ -446,6 +464,7 @@ TEST_F(ApmTest, Process) {
EXPECT_EQ(apm_->kNoError,
apm_->gain_control()->set_analog_level_limits(0, 255));
EXPECT_EQ(apm_->kNoError, apm_->gain_control()->Enable(true));
#endif
EXPECT_EQ(apm_->kNoError,
apm_->high_pass_filter()->Enable(true));
@ -564,12 +583,13 @@ TEST_F(ApmTest, Process) {
//EXPECT_EQ(apm_->kNoError,
// apm_->level_estimator()->GetMetrics(&near_metrics,
#if defined(WEBRTC_APM_UNIT_TEST_FLOAT_PROFILE)
EchoCancellation::Metrics echo_metrics;
EXPECT_EQ(apm_->kNoError,
apm_->echo_cancellation()->GetMetrics(&echo_metrics));
#endif
// TODO(ajm): check echo metrics and output audio.
if (global_read_output_data) {
if (!write_output_data) {
EXPECT_EQ(test->has_echo_count(), has_echo_count);
EXPECT_EQ(test->has_voice_count(), has_voice_count);
EXPECT_EQ(test->is_saturated_count(), is_saturated_count);
@ -577,6 +597,7 @@ TEST_F(ApmTest, Process) {
EXPECT_EQ(test->analog_level_average(), analog_level_average);
EXPECT_EQ(test->max_output_average(), max_output_average);
#if defined(WEBRTC_APM_UNIT_TEST_FLOAT_PROFILE)
audio_processing_unittest::Test::EchoMetrics reference =
test->echo_metrics();
TestStats(echo_metrics.residual_echo_return_loss,
@ -587,7 +608,7 @@ TEST_F(ApmTest, Process) {
reference.echo_return_loss_enhancement());
TestStats(echo_metrics.a_nlp,
reference.a_nlp());
#endif
} else {
test->set_has_echo_count(has_echo_count);
test->set_has_voice_count(has_voice_count);
@ -596,6 +617,7 @@ TEST_F(ApmTest, Process) {
test->set_analog_level_average(analog_level_average);
test->set_max_output_average(max_output_average);
#if defined(WEBRTC_APM_UNIT_TEST_FLOAT_PROFILE)
audio_processing_unittest::Test::EchoMetrics* message =
test->mutable_echo_metrics();
WriteStatsMessage(echo_metrics.residual_echo_return_loss,
@ -606,14 +628,15 @@ TEST_F(ApmTest, Process) {
message->mutable_echo_return_loss_enhancement());
WriteStatsMessage(echo_metrics.a_nlp,
message->mutable_a_nlp());
#endif
}
rewind(far_file_);
rewind(near_file_);
}
if (!global_read_output_data) {
WriteMessageLiteToFile("output_data.pb", output_data);
if (write_output_data) {
WriteMessageLiteToFile(kOutputFileName, output_data);
}
google::protobuf::ShutdownProtobufLibrary();
@ -975,7 +998,7 @@ int main(int argc, char** argv) {
for (int i = 1; i < argc; i++) {
if (strcmp(argv[i], "--write_output_data") == 0) {
global_read_output_data = false;
write_output_data = true;
}
}

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