Files
platform-external-webrtc/modules/audio_processing/echo_cancellation_bit_exact_unittest.cc
Per Åhgren f204fafdb4 Only create AEC2 when needed
This CL ensures that the AEC2 is only created when needed.
The changes in the CL are bitexact when running AEC2 via
audioproc_f

Bug: webrtc:8671
Change-Id: I5f6d33e45a7031c69ac53098781635c415668e49
Reviewed-on: https://webrtc-review.googlesource.com/c/src/+/129740
Reviewed-by: Sam Zackrisson <saza@webrtc.org>
Commit-Queue: Per Åhgren <peah@webrtc.org>
Cr-Commit-Position: refs/heads/master@{#27772}
2019-04-25 14:01:12 +00:00

355 lines
14 KiB
C++

/*
* Copyright (c) 2016 The WebRTC project authors. All Rights Reserved.
*
* Use of this source code is governed by a BSD-style license
* that can be found in the LICENSE file in the root of the source
* tree. An additional intellectual property rights grant can be found
* in the file PATENTS. All contributing project authors may
* be found in the AUTHORS file in the root of the source tree.
*/
#include <vector>
#include "api/array_view.h"
#include "modules/audio_processing/audio_buffer.h"
#include "modules/audio_processing/echo_cancellation_impl.h"
#include "modules/audio_processing/test/audio_buffer_tools.h"
#include "modules/audio_processing/test/bitexactness_tools.h"
#include "test/gtest.h"
namespace webrtc {
namespace {
const int kNumFramesToProcess = 100;
void SetupComponent(int sample_rate_hz,
EchoCancellationImpl::SuppressionLevel suppression_level,
bool drift_compensation_enabled,
EchoCancellationImpl* echo_canceller) {
echo_canceller->Initialize(sample_rate_hz, 1, 1, 1);
echo_canceller->Enable(true);
echo_canceller->set_suppression_level(suppression_level);
echo_canceller->enable_drift_compensation(drift_compensation_enabled);
Config config;
config.Set<DelayAgnostic>(new DelayAgnostic(true));
config.Set<ExtendedFilter>(new ExtendedFilter(true));
echo_canceller->SetExtraOptions(true, true, false);
}
void ProcessOneFrame(int sample_rate_hz,
int stream_delay_ms,
bool drift_compensation_enabled,
int stream_drift_samples,
AudioBuffer* render_audio_buffer,
AudioBuffer* capture_audio_buffer,
EchoCancellationImpl* echo_canceller) {
if (sample_rate_hz > AudioProcessing::kSampleRate16kHz) {
render_audio_buffer->SplitIntoFrequencyBands();
capture_audio_buffer->SplitIntoFrequencyBands();
}
std::vector<float> render_audio;
EchoCancellationImpl::PackRenderAudioBuffer(
render_audio_buffer, 1, render_audio_buffer->num_channels(),
&render_audio);
echo_canceller->ProcessRenderAudio(render_audio);
if (drift_compensation_enabled) {
echo_canceller->set_stream_drift_samples(stream_drift_samples);
}
echo_canceller->ProcessCaptureAudio(capture_audio_buffer, stream_delay_ms);
if (sample_rate_hz > AudioProcessing::kSampleRate16kHz) {
capture_audio_buffer->MergeFrequencyBands();
}
}
void RunBitexactnessTest(
int sample_rate_hz,
size_t num_channels,
int stream_delay_ms,
bool drift_compensation_enabled,
int stream_drift_samples,
EchoCancellationImpl::SuppressionLevel suppression_level,
bool stream_has_echo_reference,
const rtc::ArrayView<const float>& output_reference) {
EchoCancellationImpl echo_canceller;
SetupComponent(sample_rate_hz, suppression_level, drift_compensation_enabled,
&echo_canceller);
const int samples_per_channel = rtc::CheckedDivExact(sample_rate_hz, 100);
const StreamConfig render_config(sample_rate_hz, num_channels, false);
AudioBuffer render_buffer(
render_config.num_frames(), render_config.num_channels(),
render_config.num_frames(), 1, render_config.num_frames());
test::InputAudioFile render_file(
test::GetApmRenderTestVectorFileName(sample_rate_hz));
std::vector<float> render_input(samples_per_channel * num_channels);
const StreamConfig capture_config(sample_rate_hz, num_channels, false);
AudioBuffer capture_buffer(
capture_config.num_frames(), capture_config.num_channels(),
capture_config.num_frames(), 1, capture_config.num_frames());
test::InputAudioFile capture_file(
test::GetApmCaptureTestVectorFileName(sample_rate_hz));
std::vector<float> capture_input(samples_per_channel * num_channels);
for (int frame_no = 0; frame_no < kNumFramesToProcess; ++frame_no) {
ReadFloatSamplesFromStereoFile(samples_per_channel, num_channels,
&render_file, render_input);
ReadFloatSamplesFromStereoFile(samples_per_channel, num_channels,
&capture_file, capture_input);
test::CopyVectorToAudioBuffer(render_config, render_input, &render_buffer);
test::CopyVectorToAudioBuffer(capture_config, capture_input,
&capture_buffer);
ProcessOneFrame(sample_rate_hz, stream_delay_ms, drift_compensation_enabled,
stream_drift_samples, &render_buffer, &capture_buffer,
&echo_canceller);
}
// Extract and verify the test results.
std::vector<float> capture_output;
test::ExtractVectorFromAudioBuffer(capture_config, &capture_buffer,
&capture_output);
EXPECT_EQ(stream_has_echo_reference, echo_canceller.stream_has_echo());
// Compare the output with the reference. Only the first values of the output
// from last frame processed are compared in order not having to specify all
// preceeding frames as testvectors. As the algorithm being tested has a
// memory, testing only the last frame implicitly also tests the preceeding
// frames.
const float kElementErrorBound = 1.0f / 32768.0f;
EXPECT_TRUE(test::VerifyDeinterleavedArray(
capture_config.num_frames(), capture_config.num_channels(),
output_reference, capture_output, kElementErrorBound));
}
const bool kStreamHasEchoReference = true;
} // namespace
// TODO(peah): Activate all these tests for ARM and ARM64 once the issue on the
// Chromium ARM and ARM64 boths have been identified. This is tracked in the
// issue https://bugs.chromium.org/p/webrtc/issues/detail?id=5711.
#if !(defined(WEBRTC_ARCH_ARM64) || defined(WEBRTC_ARCH_ARM) || \
defined(WEBRTC_ANDROID))
TEST(EchoCancellationBitExactnessTest,
Mono8kHz_HighLevel_NoDrift_StreamDelay0) {
#else
TEST(EchoCancellationBitExactnessTest,
DISABLED_Mono8kHz_HighLevel_NoDrift_StreamDelay0) {
#endif
const float kOutputReference[] = {-0.000646f, -0.001525f, 0.002688f};
RunBitexactnessTest(8000, 1, 0, false, 0,
EchoCancellationImpl::SuppressionLevel::kHighSuppression,
kStreamHasEchoReference, kOutputReference);
}
#if !(defined(WEBRTC_ARCH_ARM64) || defined(WEBRTC_ARCH_ARM) || \
defined(WEBRTC_ANDROID))
TEST(EchoCancellationBitExactnessTest,
Mono16kHz_HighLevel_NoDrift_StreamDelay0) {
#else
TEST(EchoCancellationBitExactnessTest,
DISABLED_Mono16kHz_HighLevel_NoDrift_StreamDelay0) {
#endif
const float kOutputReference[] = {0.000055f, 0.000421f, 0.001149f};
RunBitexactnessTest(16000, 1, 0, false, 0,
EchoCancellationImpl::SuppressionLevel::kHighSuppression,
kStreamHasEchoReference, kOutputReference);
}
#if !(defined(WEBRTC_ARCH_ARM64) || defined(WEBRTC_ARCH_ARM) || \
defined(WEBRTC_ANDROID))
TEST(EchoCancellationBitExactnessTest,
Mono32kHz_HighLevel_NoDrift_StreamDelay0) {
#else
TEST(EchoCancellationBitExactnessTest,
DISABLED_Mono32kHz_HighLevel_NoDrift_StreamDelay0) {
#endif
const float kOutputReference[] = {-0.000671f, 0.000061f, -0.000031f};
RunBitexactnessTest(32000, 1, 0, false, 0,
EchoCancellationImpl::SuppressionLevel::kHighSuppression,
kStreamHasEchoReference, kOutputReference);
}
#if !(defined(WEBRTC_ARCH_ARM64) || defined(WEBRTC_ARCH_ARM) || \
defined(WEBRTC_ANDROID))
TEST(EchoCancellationBitExactnessTest,
Mono48kHz_HighLevel_NoDrift_StreamDelay0) {
#else
TEST(EchoCancellationBitExactnessTest,
DISABLED_Mono48kHz_HighLevel_NoDrift_StreamDelay0) {
#endif
const float kOutputReference[] = {-0.001403f, -0.001411f, -0.000755f};
RunBitexactnessTest(48000, 1, 0, false, 0,
EchoCancellationImpl::SuppressionLevel::kHighSuppression,
kStreamHasEchoReference, kOutputReference);
}
#if !(defined(WEBRTC_ARCH_ARM64) || defined(WEBRTC_ARCH_ARM) || \
defined(WEBRTC_ANDROID))
TEST(EchoCancellationBitExactnessTest,
Mono16kHz_LowLevel_NoDrift_StreamDelay0) {
#else
TEST(EchoCancellationBitExactnessTest,
DISABLED_Mono16kHz_LowLevel_NoDrift_StreamDelay0) {
#endif
#if defined(WEBRTC_MAC)
const float kOutputReference[] = {-0.000145f, 0.000179f, 0.000917f};
#else
const float kOutputReference[] = {-0.000009f, 0.000363f, 0.001094f};
#endif
RunBitexactnessTest(16000, 1, 0, false, 0,
EchoCancellationImpl::SuppressionLevel::kLowSuppression,
kStreamHasEchoReference, kOutputReference);
}
#if !(defined(WEBRTC_ARCH_ARM64) || defined(WEBRTC_ARCH_ARM) || \
defined(WEBRTC_ANDROID))
TEST(EchoCancellationBitExactnessTest,
Mono16kHz_ModerateLevel_NoDrift_StreamDelay0) {
#else
TEST(EchoCancellationBitExactnessTest,
DISABLED_Mono16kHz_ModerateLevel_NoDrift_StreamDelay0) {
#endif
const float kOutputReference[] = {0.000055f, 0.000421f, 0.001149f};
RunBitexactnessTest(
16000, 1, 0, false, 0,
EchoCancellationImpl::SuppressionLevel::kModerateSuppression,
kStreamHasEchoReference, kOutputReference);
}
#if !(defined(WEBRTC_ARCH_ARM64) || defined(WEBRTC_ARCH_ARM) || \
defined(WEBRTC_ANDROID))
TEST(EchoCancellationBitExactnessTest,
Mono16kHz_HighLevel_NoDrift_StreamDelay10) {
#else
TEST(EchoCancellationBitExactnessTest,
DISABLED_Mono16kHz_HighLevel_NoDrift_StreamDelay10) {
#endif
const float kOutputReference[] = {0.000055f, 0.000421f, 0.001149f};
RunBitexactnessTest(16000, 1, 10, false, 0,
EchoCancellationImpl::SuppressionLevel::kHighSuppression,
kStreamHasEchoReference, kOutputReference);
}
#if !(defined(WEBRTC_ARCH_ARM64) || defined(WEBRTC_ARCH_ARM) || \
defined(WEBRTC_ANDROID))
TEST(EchoCancellationBitExactnessTest,
Mono16kHz_HighLevel_NoDrift_StreamDelay20) {
#else
TEST(EchoCancellationBitExactnessTest,
DISABLED_Mono16kHz_HighLevel_NoDrift_StreamDelay20) {
#endif
const float kOutputReference[] = {0.000055f, 0.000421f, 0.001149f};
RunBitexactnessTest(16000, 1, 20, false, 0,
EchoCancellationImpl::SuppressionLevel::kHighSuppression,
kStreamHasEchoReference, kOutputReference);
}
#if !(defined(WEBRTC_ARCH_ARM64) || defined(WEBRTC_ARCH_ARM) || \
defined(WEBRTC_ANDROID))
TEST(EchoCancellationBitExactnessTest,
Mono16kHz_HighLevel_Drift0_StreamDelay0) {
#else
TEST(EchoCancellationBitExactnessTest,
DISABLED_Mono16kHz_HighLevel_Drift0_StreamDelay0) {
#endif
const float kOutputReference[] = {0.000055f, 0.000421f, 0.001149f};
RunBitexactnessTest(16000, 1, 0, true, 0,
EchoCancellationImpl::SuppressionLevel::kHighSuppression,
kStreamHasEchoReference, kOutputReference);
}
#if !(defined(WEBRTC_ARCH_ARM64) || defined(WEBRTC_ARCH_ARM) || \
defined(WEBRTC_ANDROID))
TEST(EchoCancellationBitExactnessTest,
Mono16kHz_HighLevel_Drift5_StreamDelay0) {
#else
TEST(EchoCancellationBitExactnessTest,
DISABLED_Mono16kHz_HighLevel_Drift5_StreamDelay0) {
#endif
const float kOutputReference[] = {0.000055f, 0.000421f, 0.001149f};
RunBitexactnessTest(16000, 1, 0, true, 5,
EchoCancellationImpl::SuppressionLevel::kHighSuppression,
kStreamHasEchoReference, kOutputReference);
}
#if !(defined(WEBRTC_ARCH_ARM64) || defined(WEBRTC_ARCH_ARM) || \
defined(WEBRTC_ANDROID))
TEST(EchoCancellationBitExactnessTest,
Stereo8kHz_HighLevel_NoDrift_StreamDelay0) {
#else
TEST(EchoCancellationBitExactnessTest,
DISABLED_Stereo8kHz_HighLevel_NoDrift_StreamDelay0) {
#endif
#if defined(WEBRTC_MAC)
const float kOutputReference[] = {-0.000392f, -0.001449f, 0.003004f,
-0.000392f, -0.001449f, 0.003004f};
#else
const float kOutputReference[] = {-0.000464f, -0.001525f, 0.002933f,
-0.000464f, -0.001525f, 0.002933f};
#endif
RunBitexactnessTest(8000, 2, 0, false, 0,
EchoCancellationImpl::SuppressionLevel::kHighSuppression,
kStreamHasEchoReference, kOutputReference);
}
#if !(defined(WEBRTC_ARCH_ARM64) || defined(WEBRTC_ARCH_ARM) || \
defined(WEBRTC_ANDROID))
TEST(EchoCancellationBitExactnessTest,
Stereo16kHz_HighLevel_NoDrift_StreamDelay0) {
#else
TEST(EchoCancellationBitExactnessTest,
DISABLED_Stereo16kHz_HighLevel_NoDrift_StreamDelay0) {
#endif
const float kOutputReference[] = {0.000166f, 0.000735f, 0.000841f,
0.000166f, 0.000735f, 0.000841f};
RunBitexactnessTest(16000, 2, 0, false, 0,
EchoCancellationImpl::SuppressionLevel::kHighSuppression,
kStreamHasEchoReference, kOutputReference);
}
#if !(defined(WEBRTC_ARCH_ARM64) || defined(WEBRTC_ARCH_ARM) || \
defined(WEBRTC_ANDROID))
TEST(EchoCancellationBitExactnessTest,
Stereo32kHz_HighLevel_NoDrift_StreamDelay0) {
#else
TEST(EchoCancellationBitExactnessTest,
DISABLED_Stereo32kHz_HighLevel_NoDrift_StreamDelay0) {
#endif
#if defined(WEBRTC_MAC)
const float kOutputReference[] = {-0.000458f, 0.000244f, 0.000153f,
-0.000458f, 0.000244f, 0.000153f};
#else
const float kOutputReference[] = {-0.000427f, 0.000183f, 0.000183f,
-0.000427f, 0.000183f, 0.000183f};
#endif
RunBitexactnessTest(32000, 2, 0, false, 0,
EchoCancellationImpl::SuppressionLevel::kHighSuppression,
kStreamHasEchoReference, kOutputReference);
}
#if !(defined(WEBRTC_ARCH_ARM64) || defined(WEBRTC_ARCH_ARM) || \
defined(WEBRTC_ANDROID))
TEST(EchoCancellationBitExactnessTest,
Stereo48kHz_HighLevel_NoDrift_StreamDelay0) {
#else
TEST(EchoCancellationBitExactnessTest,
DISABLED_Stereo48kHz_HighLevel_NoDrift_StreamDelay0) {
#endif
const float kOutputReference[] = {-0.001101f, -0.001101f, -0.000449f,
-0.001101f, -0.001101f, -0.000449f};
RunBitexactnessTest(48000, 2, 0, false, 0,
EchoCancellationImpl::SuppressionLevel::kHighSuppression,
kStreamHasEchoReference, kOutputReference);
}
} // namespace webrtc