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Unified Diff: webrtc/modules/audio_processing/test/audio_processing_unittest.cc

Issue 1774553002: Drop the 16kHz sample rate restriction on AECM and zero out higher bands (Closed) Base URL: https://chromium.googlesource.com/external/webrtc.git@master
Patch Set: Fix error message Created 4 years, 9 months ago
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« webrtc/modules/audio_processing/echo_control_mobile_impl.cc ('K') | « webrtc/modules/audio_processing/include/audio_processing.h ('k') | webrtc/voice_engine/transmit_mixer.cc » ('j') | no next file with comments »
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Index: webrtc/modules/audio_processing/test/audio_processing_unittest.cc
diff --git a/webrtc/modules/audio_processing/test/audio_processing_unittest.cc b/webrtc/modules/audio_processing/test/audio_processing_unittest.cc
index 53f667d50d7815287ae1b873606598bd02120149..ad4bf72387b3f33eb6193fd2fe93ea3081dd9f35 100644
--- a/webrtc/modules/audio_processing/test/audio_processing_unittest.cc
+++ b/webrtc/modules/audio_processing/test/audio_processing_unittest.cc
@@ -54,12 +54,7 @@ bool write_ref_data = false;
const google::protobuf::int32 kChannels[] = {1, 2};
const int kSampleRates[] = {8000, 16000, 32000, 48000};
-#if defined(WEBRTC_AUDIOPROC_FIXED_PROFILE)
-// AECM doesn't support super-wb.
-const int kProcessSampleRates[] = {8000, 16000};
-#elif defined(WEBRTC_AUDIOPROC_FLOAT_PROFILE)
const int kProcessSampleRates[] = {8000, 16000, 32000, 48000};
-#endif
enum StreamDirection { kForward = 0, kReverse };
@@ -435,11 +430,7 @@ void ApmTest::SetUp() {
frame_ = new AudioFrame();
revframe_ = new AudioFrame();
-#if defined(WEBRTC_AUDIOPROC_FIXED_PROFILE)
- Init(16000, 16000, 16000, 2, 2, 2, false);
-#else
Init(32000, 32000, 32000, 2, 2, 2, false);
-#endif
}
void ApmTest::TearDown() {
@@ -1039,18 +1030,6 @@ TEST_F(ApmTest, DISABLED_EchoCancellationReportsCorrectDelays) {
}
TEST_F(ApmTest, EchoControlMobile) {
- // AECM won't use super-wideband.
- SetFrameSampleRate(frame_, 32000);
- EXPECT_NOERR(apm_->ProcessStream(frame_));
- EXPECT_EQ(apm_->kBadSampleRateError,
- apm_->echo_control_mobile()->Enable(true));
- SetFrameSampleRate(frame_, 16000);
- EXPECT_NOERR(apm_->ProcessStream(frame_));
- EXPECT_EQ(apm_->kNoError,
- apm_->echo_control_mobile()->Enable(true));
- SetFrameSampleRate(frame_, 32000);
- EXPECT_EQ(apm_->kUnsupportedComponentError, apm_->ProcessStream(frame_));
-
// Turn AECM on (and AEC off)
Init(16000, 16000, 16000, 2, 2, 2, false);
EXPECT_EQ(apm_->kNoError, apm_->echo_control_mobile()->Enable(true));
@@ -1974,6 +1953,7 @@ TEST_F(ApmTest, FloatAndIntInterfacesGiveSimilarResults) {
num_input_channels);
int analog_level = 127;
+ size_t num_bad_chunks = 0;
while (ReadFrame(far_file_, revframe_, revfloat_cb_.get()) &&
ReadFrame(near_file_, frame_, float_cb_.get())) {
frame_->vad_activity_ = AudioFrame::kVadUnknown;
@@ -2012,18 +1992,13 @@ TEST_F(ApmTest, FloatAndIntInterfacesGiveSimilarResults) {
float snr = ComputeSNR(output_int16.channels()[j],
output_cb.channels()[j],
samples_per_channel, &variance);
- #if defined(WEBRTC_AUDIOPROC_FIXED_PROFILE)
- // There are a few chunks in the fixed-point profile that give low SNR.
- // Listening confirmed the difference is acceptable.
- const float kVarianceThreshold = 150;
- const float kSNRThreshold = 10;
- #else
+
const float kVarianceThreshold = 20;
const float kSNRThreshold = 20;
- #endif
+
// Skip frames with low energy.
- if (sqrt(variance) > kVarianceThreshold) {
- EXPECT_LT(kSNRThreshold, snr);
+ if (sqrt(variance) > kVarianceThreshold && snr < kSNRThreshold) {
+ ++num_bad_chunks;
}
}
@@ -2039,6 +2014,16 @@ TEST_F(ApmTest, FloatAndIntInterfacesGiveSimilarResults) {
// Reset in case of downmixing.
frame_->num_channels_ = static_cast<size_t>(test->num_input_channels());
}
+
+#if defined(WEBRTC_AUDIOPROC_FLOAT_PROFILE)
+ const size_t kMaxNumBadChunks = 0;
+#elif defined(WEBRTC_AUDIOPROC_FIXED_PROFILE)
+ // There are a few chunks in the fixed-point profile that give low SNR.
+ // Listening confirmed the difference is acceptable.
+ const size_t kMaxNumBadChunks = 60;
+#endif
+ EXPECT_LE(num_bad_chunks, kMaxNumBadChunks);
+
rewind(far_file_);
rewind(near_file_);
}
@@ -2560,11 +2545,6 @@ TEST_P(AudioProcessingTest, Formats) {
} else {
ref_rate = 8000;
}
-#ifdef WEBRTC_AUDIOPROC_FIXED_PROFILE
- if (file_direction == kForward) {
- ref_rate = std::min(ref_rate, 16000);
- }
-#endif
FILE* out_file = fopen(
OutputFilePath("out", input_rate_, output_rate_, reverse_input_rate_,
reverse_output_rate_, cf[i].num_input,
@@ -2716,12 +2696,12 @@ INSTANTIATE_TEST_CASE_P(
INSTANTIATE_TEST_CASE_P(
CommonFormats,
AudioProcessingTest,
- testing::Values(std::tr1::make_tuple(48000, 48000, 48000, 48000, 20, 0),
- std::tr1::make_tuple(48000, 48000, 32000, 48000, 20, 30),
- std::tr1::make_tuple(48000, 48000, 16000, 48000, 20, 20),
- std::tr1::make_tuple(48000, 44100, 48000, 44100, 15, 20),
- std::tr1::make_tuple(48000, 44100, 32000, 44100, 15, 15),
- std::tr1::make_tuple(48000, 44100, 16000, 44100, 15, 15),
+ testing::Values(std::tr1::make_tuple(48000, 48000, 48000, 48000, 0, 0),
+ std::tr1::make_tuple(48000, 48000, 32000, 48000, 40, 30),
+ std::tr1::make_tuple(48000, 48000, 16000, 48000, 40, 20),
+ std::tr1::make_tuple(48000, 44100, 48000, 44100, 25, 20),
+ std::tr1::make_tuple(48000, 44100, 32000, 44100, 25, 15),
+ std::tr1::make_tuple(48000, 44100, 16000, 44100, 25, 15),
std::tr1::make_tuple(48000, 32000, 48000, 32000, 20, 35),
std::tr1::make_tuple(48000, 32000, 32000, 32000, 20, 0),
std::tr1::make_tuple(48000, 32000, 16000, 32000, 20, 20),
@@ -2729,9 +2709,9 @@ INSTANTIATE_TEST_CASE_P(
std::tr1::make_tuple(48000, 16000, 32000, 16000, 20, 20),
std::tr1::make_tuple(48000, 16000, 16000, 16000, 20, 0),
- std::tr1::make_tuple(44100, 48000, 48000, 48000, 20, 0),
- std::tr1::make_tuple(44100, 48000, 32000, 48000, 20, 30),
- std::tr1::make_tuple(44100, 48000, 16000, 48000, 20, 20),
+ std::tr1::make_tuple(44100, 48000, 48000, 48000, 15, 0),
+ std::tr1::make_tuple(44100, 48000, 32000, 48000, 15, 30),
+ std::tr1::make_tuple(44100, 48000, 16000, 48000, 15, 20),
std::tr1::make_tuple(44100, 44100, 48000, 44100, 15, 20),
std::tr1::make_tuple(44100, 44100, 32000, 44100, 15, 15),
std::tr1::make_tuple(44100, 44100, 16000, 44100, 15, 15),
@@ -2742,15 +2722,15 @@ INSTANTIATE_TEST_CASE_P(
std::tr1::make_tuple(44100, 16000, 32000, 16000, 20, 20),
std::tr1::make_tuple(44100, 16000, 16000, 16000, 20, 0),
- std::tr1::make_tuple(32000, 48000, 48000, 48000, 20, 0),
- std::tr1::make_tuple(32000, 48000, 32000, 48000, 20, 30),
- std::tr1::make_tuple(32000, 48000, 16000, 48000, 20, 20),
- std::tr1::make_tuple(32000, 44100, 48000, 44100, 15, 20),
- std::tr1::make_tuple(32000, 44100, 32000, 44100, 15, 15),
- std::tr1::make_tuple(32000, 44100, 16000, 44100, 15, 15),
- std::tr1::make_tuple(32000, 32000, 48000, 32000, 20, 35),
- std::tr1::make_tuple(32000, 32000, 32000, 32000, 20, 0),
- std::tr1::make_tuple(32000, 32000, 16000, 32000, 20, 20),
+ std::tr1::make_tuple(32000, 48000, 48000, 48000, 35, 0),
+ std::tr1::make_tuple(32000, 48000, 32000, 48000, 65, 30),
+ std::tr1::make_tuple(32000, 48000, 16000, 48000, 40, 20),
+ std::tr1::make_tuple(32000, 44100, 48000, 44100, 20, 20),
+ std::tr1::make_tuple(32000, 44100, 32000, 44100, 20, 15),
+ std::tr1::make_tuple(32000, 44100, 16000, 44100, 20, 15),
+ std::tr1::make_tuple(32000, 32000, 48000, 32000, 35, 35),
+ std::tr1::make_tuple(32000, 32000, 32000, 32000, 0, 0),
+ std::tr1::make_tuple(32000, 32000, 16000, 32000, 40, 20),
std::tr1::make_tuple(32000, 16000, 48000, 16000, 20, 20),
std::tr1::make_tuple(32000, 16000, 32000, 16000, 20, 20),
std::tr1::make_tuple(32000, 16000, 16000, 16000, 20, 0),
« webrtc/modules/audio_processing/echo_control_mobile_impl.cc ('K') | « webrtc/modules/audio_processing/include/audio_processing.h ('k') | webrtc/voice_engine/transmit_mixer.cc » ('j') | no next file with comments »

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