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Removing some warnings from the APM build with -Wall -Wextra. Also cleaning up the unit test a bit.

Browse Source 
Review URL: http://webrtc-codereview.appspot.com/38002

git-svn-id: http://webrtc.googlecode.com/svn/trunk@90 4adac7df-926f-26a2-2b94-8c16560cd09d
This commit is contained in:
ajm@google.com
2011-06-16 00:04:40 +00:00
parent ff72b0d8f3
commit a6f54fd726
6 changed files with 234 additions and 163 deletions
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4
common_audio/vad/main/interface/webrtc_vad.h
View File

@ -34,10 +34,10 @@ extern "C"
* - version : Pointer to a buffer where the version info will
* be stored.
* Input:
* - size_in_bytes : Size of the buffer.
* - size_bytes : Size of the buffer.
*
*/
WebRtc_Word16 WebRtcVad_get_version(char *version, int size_in_bytes);
WebRtc_Word16 WebRtcVad_get_version(char *version, size_t size_bytes);
/****************************************************************************
* WebRtcVad_AssignSize(...)

47
common_audio/vad/main/source/vad_const.h
View File

@ -18,39 +18,42 @@
#include "typedefs.h"
// TODO(ajm): give these internal-linkage by moving to the appropriate file
// where possible, and otherwise tag with WebRtcVad_.
// Spectrum Weighting
WEBRTC_EXTERN const WebRtc_Word16 kSpectrumWeight[];
WEBRTC_EXTERN const WebRtc_Word16 kCompVar;
extern const WebRtc_Word16 kSpectrumWeight[];
extern const WebRtc_Word16 kCompVar;
// Logarithm constant
WEBRTC_EXTERN const WebRtc_Word16 kLogConst;
WEBRTC_EXTERN const WebRtc_Word16 kLog10Const;
extern const WebRtc_Word16 kLogConst;
extern const WebRtc_Word16 kLog10Const;
// Q15
WEBRTC_EXTERN const WebRtc_Word16 kNoiseUpdateConst;
WEBRTC_EXTERN const WebRtc_Word16 kSpeechUpdateConst;
extern const WebRtc_Word16 kNoiseUpdateConst;
extern const WebRtc_Word16 kSpeechUpdateConst;
// Q8
WEBRTC_EXTERN const WebRtc_Word16 kBackEta;
extern const WebRtc_Word16 kBackEta;
// Coefficients used by WebRtcVad_HpOutput, Q14
WEBRTC_EXTERN const WebRtc_Word16 kHpZeroCoefs[];
WEBRTC_EXTERN const WebRtc_Word16 kHpPoleCoefs[];
extern const WebRtc_Word16 kHpZeroCoefs[];
extern const WebRtc_Word16 kHpPoleCoefs[];
// Allpass filter coefficients, upper and lower, in Q15 resp. Q13
WEBRTC_EXTERN const WebRtc_Word16 kAllPassCoefsQ15[];
WEBRTC_EXTERN const WebRtc_Word16 kAllPassCoefsQ13[];
extern const WebRtc_Word16 kAllPassCoefsQ15[];
extern const WebRtc_Word16 kAllPassCoefsQ13[];
// Minimum difference between the two models, Q5
WEBRTC_EXTERN const WebRtc_Word16 kMinimumDifference[];
extern const WebRtc_Word16 kMinimumDifference[];
// Maximum value when updating the speech model, Q7
WEBRTC_EXTERN const WebRtc_Word16 kMaximumSpeech[];
extern const WebRtc_Word16 kMaximumSpeech[];
// Minimum value for mean value
WEBRTC_EXTERN const WebRtc_Word16 kMinimumMean[];
extern const WebRtc_Word16 kMinimumMean[];
// Upper limit of mean value for noise model, Q7
WEBRTC_EXTERN const WebRtc_Word16 kMaximumNoise[];
extern const WebRtc_Word16 kMaximumNoise[];
// Adjustment for division with two in WebRtcVad_SplitFilter
WEBRTC_EXTERN const WebRtc_Word16 kOffsetVector[];
extern const WebRtc_Word16 kOffsetVector[];
// Start values for the Gaussian models, Q7
WEBRTC_EXTERN const WebRtc_Word16 kNoiseDataWeights[];
WEBRTC_EXTERN const WebRtc_Word16 kSpeechDataWeights[];
WEBRTC_EXTERN const WebRtc_Word16 kNoiseDataMeans[];
WEBRTC_EXTERN const WebRtc_Word16 kSpeechDataMeans[];
WEBRTC_EXTERN const WebRtc_Word16 kNoiseDataStds[];
WEBRTC_EXTERN const WebRtc_Word16 kSpeechDataStds[];
extern const WebRtc_Word16 kNoiseDataWeights[];
extern const WebRtc_Word16 kSpeechDataWeights[];
extern const WebRtc_Word16 kNoiseDataMeans[];
extern const WebRtc_Word16 kSpeechDataMeans[];
extern const WebRtc_Word16 kNoiseDataStds[];
extern const WebRtc_Word16 kSpeechDataStds[];
#endif // WEBRTC_VAD_CONST_H_

4
common_audio/vad/main/source/webrtc_vad.c
View File

@ -22,7 +22,7 @@
static const int kInitCheck = 42;
WebRtc_Word16 WebRtcVad_get_version(char *version, int length_bytes)
WebRtc_Word16 WebRtcVad_get_version(char *version, size_t size_bytes)
{
const char my_version[] = "VAD 1.2.0";
@ -31,7 +31,7 @@ WebRtc_Word16 WebRtcVad_get_version(char *version, int length_bytes)
return -1;
}
if (length_bytes < sizeof(my_version))
if (size_bytes < sizeof(my_version))
{
return -1;
}

14
modules/audio_processing/main/source/level_estimator_impl.cc
View File

@ -86,7 +86,7 @@ LevelEstimatorImpl::LevelEstimatorImpl(const AudioProcessingImpl* apm)
LevelEstimatorImpl::~LevelEstimatorImpl() {}
int LevelEstimatorImpl::AnalyzeReverseStream(AudioBuffer* audio) {
int LevelEstimatorImpl::AnalyzeReverseStream(AudioBuffer* /*audio*/) {
return apm_->kUnsupportedComponentError;
/*if (!is_component_enabled()) {
return apm_->kNoError;
@ -95,7 +95,7 @@ int LevelEstimatorImpl::AnalyzeReverseStream(AudioBuffer* audio) {
return EstimateLevel(audio, static_cast<Handle*>(handle(1)));*/
}
int LevelEstimatorImpl::ProcessCaptureAudio(AudioBuffer* audio) {
int LevelEstimatorImpl::ProcessCaptureAudio(AudioBuffer* /*audio*/) {
return apm_->kUnsupportedComponentError;
/*if (!is_component_enabled()) {
return apm_->kNoError;
@ -104,7 +104,7 @@ int LevelEstimatorImpl::ProcessCaptureAudio(AudioBuffer* audio) {
return EstimateLevel(audio, static_cast<Handle*>(handle(0)));*/
}
int LevelEstimatorImpl::Enable(bool enable) {
int LevelEstimatorImpl::Enable(bool /*enable*/) {
CriticalSectionScoped crit_scoped(*apm_->crit());
return apm_->kUnsupportedComponentError;
//return EnableComponent(enable);
@ -114,8 +114,8 @@ bool LevelEstimatorImpl::is_enabled() const {
return is_component_enabled();
}
int LevelEstimatorImpl::GetMetrics(LevelEstimator::Metrics* metrics,
LevelEstimator::Metrics* reverse_metrics) {
int LevelEstimatorImpl::GetMetrics(LevelEstimator::Metrics* /*metrics*/,
LevelEstimator::Metrics* /*reverse_metrics*/) {
return apm_->kUnsupportedComponentError;
/*if (!is_component_enabled()) {
return apm_->kNotEnabledError;
@ -152,12 +152,12 @@ void* LevelEstimatorImpl::CreateHandle() const {
return handle;
}
int LevelEstimatorImpl::DestroyHandle(void* handle) const {
int LevelEstimatorImpl::DestroyHandle(void* /*handle*/) const {
return apm_->kUnsupportedComponentError;
//return FreeLvlEst(static_cast<Handle*>(handle));
}
int LevelEstimatorImpl::InitializeHandle(void* handle) const {
int LevelEstimatorImpl::InitializeHandle(void* /*handle*/) const {
return apm_->kUnsupportedComponentError;
/*const double kIntervalSeconds = 1.5;
return InitLvlEst(static_cast<Handle*>(handle),

324
modules/audio_processing/main/test/unit_test/unit_test.cc
View File

@ -21,6 +21,10 @@
namespace webrtc {
namespace {
// If false, this will write out a new statistics file.
// For usual testing we normally want to read the file.
const bool kReadStatFile = true;
struct ThreadData {
ThreadData(int thread_num_, AudioProcessing* ap_)
: thread_num(thread_num_),
@ -104,19 +108,33 @@ ApmTest::ApmTest()
far_file_(NULL),
near_file_(NULL),
stat_file_(NULL),
read_stat_(true) {
}
frame_(NULL),
reverse_frame_(NULL) {}
void ApmTest::SetUp() {
apm_ = AudioProcessing::Create(0);
ASSERT_TRUE(apm_ != NULL);
frame_ = new AudioFrame();
reverse_frame_ = new AudioFrame();
ASSERT_EQ(apm_->kNoError, apm_->set_sample_rate_hz(32000));
ASSERT_EQ(apm_->kNoError, apm_->set_num_channels(2, 2));
ASSERT_EQ(apm_->kNoError, apm_->set_num_reverse_channels(2));
frame_->_payloadDataLengthInSamples = 320;
frame_->_audioChannel = 2;
frame_->_frequencyInHz = 32000;
reverse_frame_->_payloadDataLengthInSamples = 320;
reverse_frame_->_audioChannel = 2;
reverse_frame_->_frequencyInHz = 32000;
far_file_ = fopen("aecFar.pcm", "rb");
ASSERT_TRUE(far_file_ != NULL) << "Cannot read source file aecFar.pcm\n";
near_file_ = fopen("aecNear.pcm", "rb");
ASSERT_TRUE(near_file_ != NULL) << "Cannot read source file aecNear.pcm\n";
if (read_stat_) {
if (kReadStatFile) {
stat_file_ = fopen("statData.dat", "rb");
ASSERT_TRUE(stat_file_ != NULL) <<
"Cannot write to source file statData.dat\n";
@ -124,15 +142,27 @@ void ApmTest::SetUp() {
}
void ApmTest::TearDown() {
if (far_file_ != NULL) {
if (frame_) {
delete frame_;
}
frame_ = NULL;
if (reverse_frame_) {
delete reverse_frame_;
}
reverse_frame_ = NULL;
if (far_file_) {
ASSERT_EQ(0, fclose(far_file_));
}
far_file_ = NULL;
if (near_file_ != NULL) {
if (near_file_) {
ASSERT_EQ(0, fclose(near_file_));
}
near_file_ = NULL;
if (stat_file_ != NULL) {
if (stat_file_) {
ASSERT_EQ(0, fclose(stat_file_));
}
stat_file_ = NULL;
@ -143,7 +173,7 @@ void ApmTest::TearDown() {
apm_ = NULL;
}
TEST_F(ApmTest, Deadlock) {
/*TEST_F(ApmTest, Deadlock) {
const int num_threads = 16;
std::vector<ThreadWrapper*> threads(num_threads);
std::vector<ThreadData*> thread_data(num_threads);
@ -177,44 +207,111 @@ TEST_F(ApmTest, Deadlock) {
delete thread_data[i];
thread_data[i] = NULL;
}
}*/
TEST_F(ApmTest, StreamParameters) {
// No errors when the components are disabled.
EXPECT_EQ(apm_->kNoError,
apm_->ProcessStream(frame_));
// Missing agc level
EXPECT_EQ(apm_->kNoError, apm_->Initialize());
EXPECT_EQ(apm_->kNoError, apm_->gain_control()->Enable(true));
EXPECT_EQ(apm_->kStreamParameterNotSetError,
apm_->ProcessStream(frame_));
EXPECT_EQ(apm_->kNoError, apm_->set_stream_delay_ms(100));
EXPECT_EQ(apm_->kNoError,
apm_->echo_cancellation()->set_stream_drift_samples(0));
EXPECT_EQ(apm_->kStreamParameterNotSetError,
apm_->ProcessStream(frame_));
EXPECT_EQ(apm_->kNoError, apm_->gain_control()->Enable(false));
// Missing delay
EXPECT_EQ(apm_->kNoError, apm_->Initialize());
EXPECT_EQ(apm_->kNoError, apm_->echo_cancellation()->Enable(true));
EXPECT_EQ(apm_->kStreamParameterNotSetError,
apm_->ProcessStream(frame_));
EXPECT_EQ(apm_->kNoError, apm_->gain_control()->Enable(true));
EXPECT_EQ(apm_->kNoError,
apm_->echo_cancellation()->set_stream_drift_samples(0));
EXPECT_EQ(apm_->kNoError,
apm_->gain_control()->set_stream_analog_level(127));
EXPECT_EQ(apm_->kStreamParameterNotSetError,
apm_->ProcessStream(frame_));
EXPECT_EQ(apm_->kNoError, apm_->gain_control()->Enable(false));
// Missing drift
EXPECT_EQ(apm_->kNoError, apm_->Initialize());
EXPECT_EQ(apm_->kNoError,
apm_->echo_cancellation()->enable_drift_compensation(true));
EXPECT_EQ(apm_->kStreamParameterNotSetError,
apm_->ProcessStream(frame_));
EXPECT_EQ(apm_->kNoError, apm_->gain_control()->Enable(true));
EXPECT_EQ(apm_->kNoError, apm_->set_stream_delay_ms(100));
EXPECT_EQ(apm_->kNoError,
apm_->gain_control()->set_stream_analog_level(127));
EXPECT_EQ(apm_->kStreamParameterNotSetError,
apm_->ProcessStream(frame_));
// No stream parameters
EXPECT_EQ(apm_->kNoError, apm_->Initialize());
EXPECT_EQ(apm_->kNoError,
apm_->AnalyzeReverseStream(reverse_frame_));
EXPECT_EQ(apm_->kStreamParameterNotSetError,
apm_->ProcessStream(frame_));
// All there
EXPECT_EQ(apm_->kNoError, apm_->gain_control()->Enable(true));
EXPECT_EQ(apm_->kNoError, apm_->Initialize());
EXPECT_EQ(apm_->kNoError, apm_->set_stream_delay_ms(100));
EXPECT_EQ(apm_->kNoError,
apm_->echo_cancellation()->set_stream_drift_samples(0));
EXPECT_EQ(apm_->kNoError,
apm_->gain_control()->set_stream_analog_level(127));
EXPECT_EQ(apm_->kNoError, apm_->ProcessStream(frame_));
}
TEST_F(ApmTest, Channels) {
// Testing number of invalid channels
EXPECT_EQ(apm_->kBadParameterError, apm_->set_num_channels(0, 1));
EXPECT_EQ(apm_->kBadParameterError, apm_->set_num_channels(1, 0));
EXPECT_EQ(apm_->kBadParameterError, apm_->set_num_channels(3, 1));
EXPECT_EQ(apm_->kBadParameterError, apm_->set_num_channels(1, 3));
EXPECT_EQ(apm_->kBadParameterError, apm_->set_num_reverse_channels(0));
EXPECT_EQ(apm_->kBadParameterError, apm_->set_num_reverse_channels(3));
// Testing number of valid channels
for (int i = 1; i < 3; i++) {
for (int j = 1; j < 3; j++) {
if (j > i) {
EXPECT_EQ(apm_->kBadParameterError, apm_->set_num_channels(i, j));
} else {
EXPECT_EQ(apm_->kNoError, apm_->set_num_channels(i, j));
EXPECT_EQ(j, apm_->num_output_channels());
}
}
EXPECT_EQ(i, apm_->num_input_channels());
EXPECT_EQ(apm_->kNoError, apm_->set_num_reverse_channels(i));
EXPECT_EQ(i, apm_->num_reverse_channels());
}
}
TEST_F(ApmTest, SampleRates) {
// Testing invalid sample rates
EXPECT_EQ(apm_->kBadParameterError, apm_->set_sample_rate_hz(10000));
// Testing valid sample rates
int fs[] = {8000, 16000, 32000};
for (size_t i = 0; i < sizeof(fs) / sizeof(*fs); i++) {
EXPECT_EQ(apm_->kNoError, apm_->set_sample_rate_hz(fs[i]));
EXPECT_EQ(fs[i], apm_->sample_rate_hz());
}
}
TEST_F(ApmTest, Process) {
if (!read_stat_) {
if (!kReadStatFile) {
stat_file_ = fopen("statData.dat", "wb");
ASSERT_TRUE(stat_file_ != NULL)
<< "Cannot write to source file statData.dat\n";
}
// Testing number of invalid channels
ASSERT_EQ(apm_->kBadParameterError, apm_->set_num_channels(0, 1));
ASSERT_EQ(apm_->kBadParameterError, apm_->set_num_channels(1, 0));
ASSERT_EQ(apm_->kBadParameterError, apm_->set_num_channels(3, 1));
ASSERT_EQ(apm_->kBadParameterError, apm_->set_num_channels(1, 3));
ASSERT_EQ(apm_->kBadParameterError, apm_->set_num_reverse_channels(0));
ASSERT_EQ(apm_->kBadParameterError, apm_->set_num_reverse_channels(3));
// Testing number of valid channels
for (int ii = 1; ii < 3; ++ii) {
for (int jj = 1; jj < 3; ++jj) {
if (jj > ii ) {
ASSERT_EQ(apm_->kBadParameterError, apm_->set_num_channels(ii, jj));
} else {
ASSERT_EQ(apm_->kNoError, apm_->set_num_channels(ii, jj));
ASSERT_EQ(jj, apm_->num_output_channels());
}
}
ASSERT_EQ(ii, apm_->num_input_channels());
ASSERT_EQ(apm_->kNoError, apm_->set_num_reverse_channels(ii));
ASSERT_EQ(ii, apm_->num_reverse_channels());
}
// Testing invalid sample rates
ASSERT_EQ(apm_->kBadParameterError, apm_->set_sample_rate_hz(10000));
// Testing valid sample rates
int fs[] = {8000, 16000, 32000};
for (int ii = 0; ii < sizeof(fs)/sizeof(*fs); ++ii) {
ASSERT_EQ(apm_->kNoError, apm_->set_sample_rate_hz(fs[ii]));
ASSERT_EQ(fs[ii], apm_->sample_rate_hz());
}
AudioFrame render_audio;
AudioFrame capture_audio;
@ -279,82 +376,44 @@ TEST_F(ApmTest, Process) {
apm_->voice_detection()->stream_has_voice();
}
// Wrong order tests
// No stream parameter
EXPECT_EQ(apm_->kNoError,
apm_->AnalyzeReverseStream(&render_audio));
EXPECT_EQ(apm_->kStreamParameterNotSetError,
apm_->ProcessStream(&capture_audio));
// Missing agc level
EXPECT_EQ(apm_->kNoError, apm_->Initialize());
EXPECT_EQ(apm_->kNoError, apm_->set_stream_delay_ms(100));
EXPECT_EQ(apm_->kNoError,
apm_->echo_cancellation()->set_stream_drift_samples(0));
EXPECT_EQ(apm_->kStreamParameterNotSetError,
apm_->ProcessStream(&capture_audio));
// Missing delay
EXPECT_EQ(apm_->kNoError, apm_->Initialize());
EXPECT_EQ(apm_->kNoError,
apm_->echo_cancellation()->set_stream_drift_samples(0));
EXPECT_EQ(apm_->kNoError,
apm_->gain_control()->set_stream_analog_level(127));
EXPECT_EQ(apm_->kStreamParameterNotSetError,
apm_->ProcessStream(&capture_audio));
// Missing drift
EXPECT_EQ(apm_->kNoError, apm_->Initialize());
EXPECT_EQ(apm_->kNoError, apm_->set_stream_delay_ms(100));
EXPECT_EQ(apm_->kNoError,
apm_->gain_control()->set_stream_analog_level(127));
EXPECT_EQ(apm_->kStreamParameterNotSetError,
apm_->ProcessStream(&capture_audio));
// All there
EXPECT_EQ(apm_->kNoError, apm_->Initialize());
EXPECT_EQ(apm_->kNoError, apm_->set_stream_delay_ms(100));
EXPECT_EQ(apm_->kNoError,
apm_->echo_cancellation()->set_stream_drift_samples(0));
EXPECT_EQ(apm_->kNoError,
apm_->gain_control()->set_stream_analog_level(127));
EXPECT_EQ(apm_->kNoError, apm_->ProcessStream(&capture_audio));
// Test with real audio
// Loop through all possible combinations
// (# reverse channels, # channels, sample rates)
int revCh[] = {1, 2};
int nRevCh = 2;
int rev_ch[] = {1, 2};
int ch[] = {1, 2};
int nCh = 2;
int nFs = (int)(sizeof(fs)/sizeof(*fs));
if (read_stat_) {
fread(&nRevCh, sizeof(int), 1, stat_file_);
fread(revCh, sizeof(int), nRevCh, stat_file_);
fread(&nCh, sizeof(int), 1, stat_file_);
fread(ch, sizeof(int), nCh, stat_file_);
fread(&nFs, sizeof(int), 1, stat_file_);
fread(fs, 1, sizeof(fs), stat_file_);
int fs[] = {8000, 16000, 32000};
size_t rev_ch_size = sizeof(rev_ch) / sizeof(*rev_ch);
size_t ch_size = sizeof(ch) / sizeof(*ch);
size_t fs_size = sizeof(fs) / sizeof(*fs);
if (kReadStatFile) {
fread(&rev_ch_size, sizeof(rev_ch_size), 1, stat_file_);
fread(rev_ch, sizeof(int), rev_ch_size, stat_file_);
fread(&ch_size, sizeof(ch_size), 1, stat_file_);
fread(ch, sizeof(int), ch_size, stat_file_);
fread(&fs_size, sizeof(fs_size), 1, stat_file_);
fread(fs, sizeof(int), fs_size, stat_file_);
} else {
fwrite(&nRevCh, sizeof(int), 1, stat_file_);
fwrite(revCh, sizeof(int), nRevCh, stat_file_);
fwrite(&nCh, sizeof(int), 1, stat_file_);
fwrite(ch, sizeof(int), nCh, stat_file_);
fwrite(&nFs, sizeof(int), 1, stat_file_);
fwrite(fs, 1, sizeof(fs), stat_file_);
fwrite(&rev_ch_size, sizeof(int), 1, stat_file_);
fwrite(rev_ch, sizeof(int), rev_ch_size, stat_file_);
fwrite(&ch_size, sizeof(int), 1, stat_file_);
fwrite(ch, sizeof(int), ch_size, stat_file_);
fwrite(&fs_size, sizeof(int), 1, stat_file_);
fwrite(fs, sizeof(int), fs_size, stat_file_);
}
int testCnt = 0;
for (int iRevCh = 0; iRevCh < nRevCh; ++iRevCh) {
for (int iCh = 0; iCh < nCh; ++iCh) {
for (int iFs = 0; iFs < nFs; ++iFs) {
render_audio._payloadDataLengthInSamples = fs[iFs]/100;
render_audio._audioChannel = revCh[iRevCh];
render_audio._frequencyInHz = fs[iFs];
capture_audio._payloadDataLengthInSamples = fs[iFs]/100;
capture_audio._audioChannel = ch[iCh];
capture_audio._frequencyInHz = fs[iFs];
int test_count = 0;
for (size_t i_rev_ch = 0; i_rev_ch < rev_ch_size; i_rev_ch++) {
for (size_t i_ch = 0; i_ch < ch_size; i_ch++) {
for (size_t i_fs = 0; i_fs < fs_size; i_fs++) {
render_audio._payloadDataLengthInSamples = fs[i_fs] / 100;
render_audio._audioChannel = rev_ch[i_rev_ch];
render_audio._frequencyInHz = fs[i_fs];
capture_audio._payloadDataLengthInSamples = fs[i_fs] / 100;
capture_audio._audioChannel = ch[i_ch];
capture_audio._frequencyInHz = fs[i_fs];
EXPECT_EQ(apm_->kNoError, apm_->Initialize());
ASSERT_EQ(apm_->kNoError, apm_->set_sample_rate_hz(fs[iFs]));
ASSERT_EQ(apm_->kNoError, apm_->set_sample_rate_hz(fs[i_fs]));
ASSERT_EQ(apm_->kNoError,
apm_->set_num_channels(capture_audio._audioChannel,
capture_audio._audioChannel));
@ -384,7 +443,6 @@ TEST_F(ApmTest, Process) {
EXPECT_EQ(apm_->kNoError,
apm_->voice_detection()->Enable(true));
bool runningFiles = true;
int i = 0;
int echo_count = 0;
int vad_count = 0;
int sat_count = 0;
@ -402,21 +460,22 @@ TEST_F(ApmTest, Process) {
EchoCancellation::Metrics echo_metrics_ref_;
while (runningFiles) {
i++;
// Read far end frame
read_count = fread(&tmpData[0],
read_count = fread(tmpData,
sizeof(WebRtc_Word16),
render_audio._payloadDataLengthInSamples * 2,
far_file_);
if (read_count != render_audio._payloadDataLengthInSamples * 2) {
if (read_count !=
static_cast<size_t>
(render_audio._payloadDataLengthInSamples * 2)) {
break; // This is expected.
}
if (render_audio._audioChannel == 1) {
for (int jj = 0; jj < render_audio._payloadDataLengthInSamples;
++jj) {
tmp_int = (static_cast<int>(tmpData[jj * 2]) +
static_cast<int>(tmpData[jj * 2 + 1])) >> 1;
render_audio._payloadData[jj] =
for (int i = 0; i < render_audio._payloadDataLengthInSamples;
i++) {
tmp_int = (static_cast<int>(tmpData[i * 2]) +
static_cast<int>(tmpData[i * 2 + 1])) >> 1;
render_audio._payloadData[i] =
static_cast<WebRtc_Word16>(tmp_int);
}
} else {
@ -432,18 +491,21 @@ TEST_F(ApmTest, Process) {
apm_->gain_control()->set_stream_analog_level(analog_level));
// Read near end frame
read_count = fread(&tmpData[0],
read_count = fread(tmpData,
sizeof(WebRtc_Word16),
capture_audio._payloadDataLengthInSamples * 2,
near_file_);
if (read_count != capture_audio._payloadDataLengthInSamples * 2) {
if (read_count !=
static_cast<size_t>
(capture_audio._payloadDataLengthInSamples * 2)) {
break; // This is expected.
}
if (capture_audio._audioChannel == 1) {
for (int jj = 0; jj < capture_audio._payloadDataLengthInSamples; ++jj) {
tmp_int = (static_cast<int>(tmpData[jj * 2]) +
static_cast<int>(tmpData[jj * 2 + 1])) >> 1;
capture_audio._payloadData[jj] =
for (int i = 0;
i < capture_audio._payloadDataLengthInSamples; i++) {
tmp_int = (static_cast<int>(tmpData[i * 2]) +
static_cast<int>(tmpData[i * 2 + 1])) >> 1;
capture_audio._payloadData[i] =
static_cast<WebRtc_Word16>(tmp_int);
}
} else {
@ -452,14 +514,14 @@ TEST_F(ApmTest, Process) {
sizeof(WebRtc_Word16) * read_count);
}
WebRtc_Word32 tmpF = 0;
for (int jj = 0; jj < read_count; ++jj) {
tmpF = (WebRtc_Word32)capture_audio._payloadData[jj] * sat_gain;
for (size_t i = 0; i < read_count; i++) {
tmpF = (WebRtc_Word32)capture_audio._payloadData[i] * sat_gain;
if (tmpF > WEBRTC_SPL_WORD16_MAX) {
capture_audio._payloadData[jj] = WEBRTC_SPL_WORD16_MAX;
capture_audio._payloadData[i] = WEBRTC_SPL_WORD16_MAX;
} else if (tmpF < WEBRTC_SPL_WORD16_MIN) {
capture_audio._payloadData[jj] = WEBRTC_SPL_WORD16_MIN;
capture_audio._payloadData[i] = WEBRTC_SPL_WORD16_MIN;
} else {
capture_audio._payloadData[jj] = (WebRtc_Word16)tmpF;
capture_audio._payloadData[i] = static_cast<WebRtc_Word16>(tmpF);
}
}
EXPECT_EQ(apm_->kNoError, apm_->ProcessStream(&capture_audio));
@ -485,7 +547,7 @@ TEST_F(ApmTest, Process) {
// TODO(ajm): Perhaps we don't have to check every value? The average
// could be sufficient. Or, how about hashing the output?
if (read_stat_) {
if (kReadStatFile) {
// Read from statData
fread(&echo_count_ref_, 1, sizeof(echo_count), stat_file_);
EXPECT_EQ(echo_count_ref_, echo_count);
@ -607,12 +669,12 @@ TEST_F(ApmTest, Process) {
rewind(far_file_);
rewind(near_file_);
testCnt++;
printf("Loop %d of %d\n", testCnt, nRevCh * nCh * nFs);
test_count++;
printf("Loop %d of %lu\n", test_count, rev_ch_size * ch_size * fs_size);
}
}
}
if (!read_stat_) {
if (!kReadStatFile) {
if (stat_file_ != NULL) {
ASSERT_EQ(0, fclose(stat_file_));
}
@ -679,6 +741,10 @@ TEST_F(ApmTest, EchoCancellation) {
}
TEST_F(ApmTest, EchoControlMobile) {
// AECM won't use super-wideband.
EXPECT_EQ(apm_->kNoError, apm_->set_sample_rate_hz(32000));
EXPECT_EQ(apm_->kBadSampleRateError, apm_->echo_control_mobile()->Enable(true));
EXPECT_EQ(apm_->kNoError, apm_->set_sample_rate_hz(16000));
// Turn AECM on (and AEC off)
EXPECT_EQ(apm_->kNoError, apm_->echo_control_mobile()->Enable(true));
EXPECT_TRUE(apm_->echo_control_mobile()->is_enabled());

4
modules/audio_processing/main/test/unit_test/unit_test.h
View File

@ -16,6 +16,7 @@
namespace webrtc {
class AudioProcessing;
class AudioFrame;
class ApmTest : public ::testing::Test {
protected:
@ -27,7 +28,8 @@ class ApmTest : public ::testing::Test {
FILE* far_file_;
FILE* near_file_;
FILE* stat_file_;
bool read_stat_;
AudioFrame* frame_;
AudioFrame* reverse_frame_;
};
} // namespace webrtc