Allow extracting the linear AEC output
This CL enables extracting the linear AEC output,
allowing for more straightforward
testing/development.
Bug: b/140823178
Change-Id: I14f7934008d87066b35500466cb6e6d96f811688
Reviewed-on: https://webrtc-review.googlesource.com/c/src/+/153672
Commit-Queue: Per Åhgren <peah@webrtc.org>
Reviewed-by: Gustaf Ullberg <gustaf@webrtc.org>
Cr-Commit-Position: refs/heads/master@{#29789}
diff --git a/api/audio/echo_canceller3_config.h b/api/audio/echo_canceller3_config.h
index 3387526..c8de797 100644
--- a/api/audio/echo_canceller3_config.h
+++ b/api/audio/echo_canceller3_config.h
@@ -79,6 +79,7 @@
bool conservative_initial_phase = false;
bool enable_shadow_filter_output_usage = true;
bool use_linear_filter = true;
+ bool export_linear_aec_output = false;
} filter;
struct Erle {
@@ -185,8 +186,6 @@
} high_bands_suppression;
float floor_first_increase = 0.00001f;
- bool enforce_transparent = false;
- bool enforce_empty_higher_bands = false;
} suppressor;
};
} // namespace webrtc
diff --git a/api/audio/echo_canceller3_config_json.cc b/api/audio/echo_canceller3_config_json.cc
index c6ee708..ab051bd 100644
--- a/api/audio/echo_canceller3_config_json.cc
+++ b/api/audio/echo_canceller3_config_json.cc
@@ -190,6 +190,8 @@
ReadParam(section, "enable_shadow_filter_output_usage",
&cfg.filter.enable_shadow_filter_output_usage);
ReadParam(section, "use_linear_filter", &cfg.filter.use_linear_filter);
+ ReadParam(section, "export_linear_aec_output",
+ &cfg.filter.export_linear_aec_output);
}
if (rtc::GetValueFromJsonObject(aec3_root, "erle", §ion)) {
@@ -314,10 +316,6 @@
ReadParam(section, "floor_first_increase",
&cfg.suppressor.floor_first_increase);
- ReadParam(section, "enforce_transparent",
- &cfg.suppressor.enforce_transparent);
- ReadParam(section, "enforce_empty_higher_bands",
- &cfg.suppressor.enforce_empty_higher_bands);
}
}
@@ -408,7 +406,12 @@
ost << "\"conservative_initial_phase\": "
<< (config.filter.conservative_initial_phase ? "true" : "false") << ",";
ost << "\"enable_shadow_filter_output_usage\": "
- << (config.filter.enable_shadow_filter_output_usage ? "true" : "false");
+ << (config.filter.enable_shadow_filter_output_usage ? "true" : "false")
+ << ",";
+ ost << "\"use_linear_filter\": "
+ << (config.filter.use_linear_filter ? "true" : "false") << ",";
+ ost << "\"export_linear_aec_output\": "
+ << (config.filter.export_linear_aec_output ? "true" : "false");
ost << "},";
@@ -545,12 +548,7 @@
ost << "\"max_gain_during_echo\": "
<< config.suppressor.high_bands_suppression.max_gain_during_echo;
ost << "},";
- ost << "\"floor_first_increase\": " << config.suppressor.floor_first_increase
- << ",";
- ost << "\"enforce_transparent\": "
- << (config.suppressor.enforce_transparent ? "true" : "false") << ",";
- ost << "\"enforce_empty_higher_bands\": "
- << (config.suppressor.enforce_empty_higher_bands ? "true" : "false");
+ ost << "\"floor_first_increase\": " << config.suppressor.floor_first_increase;
ost << "}";
ost << "}";
ost << "}";
diff --git a/api/audio/echo_control.h b/api/audio/echo_control.h
index de80f50..b63f123 100644
--- a/api/audio/echo_control.h
+++ b/api/audio/echo_control.h
@@ -31,6 +31,12 @@
// Processes the capture signal in order to remove the echo.
virtual void ProcessCapture(AudioBuffer* capture, bool echo_path_change) = 0;
+ // As above, but also returns the linear filter output.
+ // TODO(peah): Make pure virtual.
+ virtual void ProcessCapture(AudioBuffer* capture,
+ AudioBuffer* linear_output,
+ bool level_change) {}
+
struct Metrics {
double echo_return_loss;
double echo_return_loss_enhancement;
diff --git a/modules/audio_processing/aec3/aec3_common.h b/modules/audio_processing/aec3/aec3_common.h
index bf554e3..d778e50 100644
--- a/modules/audio_processing/aec3/aec3_common.h
+++ b/modules/audio_processing/aec3/aec3_common.h
@@ -42,7 +42,8 @@
constexpr int kRenderTransferQueueSizeFrames = 100;
constexpr size_t kMaxNumBands = 3;
-constexpr size_t kSubFrameLength = 80;
+constexpr size_t kFrameSize = 160;
+constexpr size_t kSubFrameLength = kFrameSize / 2;
constexpr size_t kBlockSize = kFftLengthBy2;
constexpr size_t kBlockSizeLog2 = kFftLengthBy2Log2;
diff --git a/modules/audio_processing/aec3/block_processor.cc b/modules/audio_processing/aec3/block_processor.cc
index 8942570..bda2589 100644
--- a/modules/audio_processing/aec3/block_processor.cc
+++ b/modules/audio_processing/aec3/block_processor.cc
@@ -52,6 +52,7 @@
void ProcessCapture(
bool echo_path_gain_change,
bool capture_signal_saturation,
+ std::vector<std::vector<std::vector<float>>>* linear_output,
std::vector<std::vector<std::vector<float>>>* capture_block) override;
void BufferRender(
@@ -105,6 +106,7 @@
void BlockProcessorImpl::ProcessCapture(
bool echo_path_gain_change,
bool capture_signal_saturation,
+ std::vector<std::vector<std::vector<float>>>* linear_output,
std::vector<std::vector<std::vector<float>>>* capture_block) {
RTC_DCHECK(capture_block);
RTC_DCHECK_EQ(NumBandsForRate(sample_rate_hz_), capture_block->size());
@@ -191,7 +193,7 @@
if (has_delay_estimator || render_buffer_->HasReceivedBufferDelay()) {
echo_remover_->ProcessCapture(
echo_path_variability, capture_signal_saturation, estimated_delay_,
- render_buffer_->GetRenderBuffer(), capture_block);
+ render_buffer_->GetRenderBuffer(), linear_output, capture_block);
}
// Update the metrics.
diff --git a/modules/audio_processing/aec3/block_processor.h b/modules/audio_processing/aec3/block_processor.h
index 755444a..9bb0cf1 100644
--- a/modules/audio_processing/aec3/block_processor.h
+++ b/modules/audio_processing/aec3/block_processor.h
@@ -59,6 +59,7 @@
virtual void ProcessCapture(
bool echo_path_gain_change,
bool capture_signal_saturation,
+ std::vector<std::vector<std::vector<float>>>* linear_output,
std::vector<std::vector<std::vector<float>>>* capture_block) = 0;
// Buffers a block of render data supplied by a FrameBlocker object.
diff --git a/modules/audio_processing/aec3/block_processor_unittest.cc b/modules/audio_processing/aec3/block_processor_unittest.cc
index 9c315e1..2b928e8 100644
--- a/modules/audio_processing/aec3/block_processor_unittest.cc
+++ b/modules/audio_processing/aec3/block_processor_unittest.cc
@@ -48,7 +48,7 @@
std::vector<float>(kBlockSize, 1000.f)));
for (int k = 0; k < num_iterations; ++k) {
block_processor->BufferRender(block);
- block_processor->ProcessCapture(false, false, &block);
+ block_processor->ProcessCapture(false, false, nullptr, &block);
block_processor->UpdateEchoLeakageStatus(false);
}
}
@@ -81,7 +81,8 @@
std::vector<std::vector<float>>(kNumRenderChannels,
std::vector<float>(kBlockSize - 1, 0.f)));
- EXPECT_DEATH(block_processor->ProcessCapture(false, false, &block), "");
+ EXPECT_DEATH(block_processor->ProcessCapture(false, false, nullptr, &block),
+ "");
}
void RunRenderNumBandsVerificationTest(int sample_rate_hz) {
@@ -117,7 +118,8 @@
std::vector<std::vector<float>>(kNumRenderChannels,
std::vector<float>(kBlockSize, 0.f)));
- EXPECT_DEATH(block_processor->ProcessCapture(false, false, &block), "");
+ EXPECT_DEATH(block_processor->ProcessCapture(false, false, nullptr, &block),
+ "");
}
#endif
@@ -172,7 +174,7 @@
RandomizeSampleVector(&random_generator, render_block[0][0]);
signal_delay_buffer.Delay(render_block[0][0], capture_block[0][0]);
block_processor->BufferRender(render_block);
- block_processor->ProcessCapture(false, false, &capture_block);
+ block_processor->ProcessCapture(false, false, nullptr, &capture_block);
}
}
}
@@ -207,7 +209,7 @@
.WillRepeatedly(Return(0));
EXPECT_CALL(*render_delay_controller_mock, GetDelay(_, _, _))
.Times(kNumBlocks);
- EXPECT_CALL(*echo_remover_mock, ProcessCapture(_, _, _, _, _))
+ EXPECT_CALL(*echo_remover_mock, ProcessCapture(_, _, _, _, _, _))
.Times(kNumBlocks);
EXPECT_CALL(*echo_remover_mock, UpdateEchoLeakageStatus(_))
.Times(kNumBlocks);
@@ -230,7 +232,7 @@
RandomizeSampleVector(&random_generator, render_block[0][0]);
signal_delay_buffer.Delay(render_block[0][0], capture_block[0][0]);
block_processor->BufferRender(render_block);
- block_processor->ProcessCapture(false, false, &capture_block);
+ block_processor->ProcessCapture(false, false, nullptr, &capture_block);
block_processor->UpdateEchoLeakageStatus(false);
}
}
@@ -284,7 +286,7 @@
TEST(BlockProcessor, NullProcessCaptureParameter) {
EXPECT_DEATH(std::unique_ptr<BlockProcessor>(
BlockProcessor::Create(EchoCanceller3Config(), 16000, 1, 1))
- ->ProcessCapture(false, false, nullptr),
+ ->ProcessCapture(false, false, nullptr, nullptr),
"");
}
diff --git a/modules/audio_processing/aec3/echo_canceller3.cc b/modules/audio_processing/aec3/echo_canceller3.cc
index 2b50e61..a68ae01 100644
--- a/modules/audio_processing/aec3/echo_canceller3.cc
+++ b/modules/audio_processing/aec3/echo_canceller3.cc
@@ -16,6 +16,7 @@
#include "modules/audio_processing/high_pass_filter.h"
#include "modules/audio_processing/logging/apm_data_dumper.h"
#include "rtc_base/atomic_ops.h"
+#include "rtc_base/logging.h"
#include "system_wrappers/include/field_trial.h"
namespace webrtc {
@@ -87,28 +88,52 @@
}
void ProcessCaptureFrameContent(
+ AudioBuffer* linear_output,
AudioBuffer* capture,
bool level_change,
bool saturated_microphone_signal,
size_t sub_frame_index,
FrameBlocker* capture_blocker,
+ BlockFramer* linear_output_framer,
BlockFramer* output_framer,
BlockProcessor* block_processor,
- std::vector<std::vector<std::vector<float>>>* block,
- std::vector<std::vector<rtc::ArrayView<float>>>* sub_frame_view) {
- FillSubFrameView(capture, sub_frame_index, sub_frame_view);
- capture_blocker->InsertSubFrameAndExtractBlock(*sub_frame_view, block);
+ std::vector<std::vector<std::vector<float>>>* linear_output_block,
+ std::vector<std::vector<rtc::ArrayView<float>>>*
+ linear_output_sub_frame_view,
+ std::vector<std::vector<std::vector<float>>>* capture_block,
+ std::vector<std::vector<rtc::ArrayView<float>>>* capture_sub_frame_view) {
+ FillSubFrameView(capture, sub_frame_index, capture_sub_frame_view);
+
+ if (linear_output) {
+ RTC_DCHECK(linear_output_framer);
+ RTC_DCHECK(linear_output_block);
+ RTC_DCHECK(linear_output_sub_frame_view);
+ FillSubFrameView(linear_output, sub_frame_index,
+ linear_output_sub_frame_view);
+ }
+
+ capture_blocker->InsertSubFrameAndExtractBlock(*capture_sub_frame_view,
+ capture_block);
block_processor->ProcessCapture(level_change, saturated_microphone_signal,
- block);
- output_framer->InsertBlockAndExtractSubFrame(*block, sub_frame_view);
+ linear_output_block, capture_block);
+ output_framer->InsertBlockAndExtractSubFrame(*capture_block,
+ capture_sub_frame_view);
+
+ if (linear_output) {
+ RTC_DCHECK(linear_output_framer);
+ linear_output_framer->InsertBlockAndExtractSubFrame(
+ *linear_output_block, linear_output_sub_frame_view);
+ }
}
void ProcessRemainingCaptureFrameContent(
bool level_change,
bool saturated_microphone_signal,
FrameBlocker* capture_blocker,
+ BlockFramer* linear_output_framer,
BlockFramer* output_framer,
BlockProcessor* block_processor,
+ std::vector<std::vector<std::vector<float>>>* linear_output_block,
std::vector<std::vector<std::vector<float>>>* block) {
if (!capture_blocker->IsBlockAvailable()) {
return;
@@ -116,8 +141,13 @@
capture_blocker->ExtractBlock(block);
block_processor->ProcessCapture(level_change, saturated_microphone_signal,
- block);
+ linear_output_block, block);
output_framer->InsertBlock(*block);
+
+ if (linear_output_framer) {
+ RTC_DCHECK(linear_output_block);
+ linear_output_framer->InsertBlock(*linear_output_block);
+ }
}
void BufferRenderFrameContent(
@@ -295,12 +325,24 @@
RTC_DCHECK_EQ(num_bands_, std::max(sample_rate_hz_, 16000) / 16000);
RTC_DCHECK_GE(kMaxNumBands, num_bands_);
+
+ if (config_.filter.export_linear_aec_output) {
+ linear_output_framer_.reset(new BlockFramer(1, num_capture_channels_));
+ linear_output_block_ =
+ std::make_unique<std::vector<std::vector<std::vector<float>>>>(
+ 1, std::vector<std::vector<float>>(
+ num_capture_channels_, std::vector<float>(kBlockSize, 0.f)));
+ linear_output_sub_frame_view_ =
+ std::vector<std::vector<rtc::ArrayView<float>>>(
+ 1, std::vector<rtc::ArrayView<float>>(num_capture_channels_));
+ }
}
EchoCanceller3::~EchoCanceller3() = default;
void EchoCanceller3::AnalyzeRender(const AudioBuffer& render) {
RTC_DCHECK_RUNS_SERIALIZED(&render_race_checker_);
+
RTC_DCHECK_EQ(render.num_channels(), num_render_channels_);
data_dumper_->DumpRaw("aec3_call_order",
static_cast<int>(EchoCanceller3ApiCall::kRender));
@@ -312,7 +354,6 @@
RTC_DCHECK_RUNS_SERIALIZED(&capture_race_checker_);
data_dumper_->DumpWav("aec3_capture_analyze_input", capture.num_frames(),
capture.channels_const()[0], sample_rate_hz_, 1);
-
saturated_microphone_signal_ = false;
for (size_t channel = 0; channel < capture.num_channels(); ++channel) {
saturated_microphone_signal_ |=
@@ -325,6 +366,12 @@
}
void EchoCanceller3::ProcessCapture(AudioBuffer* capture, bool level_change) {
+ ProcessCapture(capture, nullptr, level_change);
+}
+
+void EchoCanceller3::ProcessCapture(AudioBuffer* capture,
+ AudioBuffer* linear_output,
+ bool level_change) {
RTC_DCHECK_RUNS_SERIALIZED(&capture_race_checker_);
RTC_DCHECK(capture);
RTC_DCHECK_EQ(num_bands_, capture->num_bands());
@@ -333,6 +380,12 @@
data_dumper_->DumpRaw("aec3_call_order",
static_cast<int>(EchoCanceller3ApiCall::kCapture));
+ if (linear_output && !linear_output_framer_) {
+ RTC_LOG(LS_ERROR) << "Trying to retrieve the linear AEC output without "
+ "properly configuring AEC3.";
+ RTC_NOTREACHED();
+ }
+
// Report capture call in the metrics and periodically update API call
// metrics.
api_call_metrics_.ReportCaptureCall();
@@ -349,19 +402,24 @@
EmptyRenderQueue();
- ProcessCaptureFrameContent(capture, level_change,
+ ProcessCaptureFrameContent(linear_output, capture, level_change,
saturated_microphone_signal_, 0, &capture_blocker_,
- &output_framer_, block_processor_.get(),
- &capture_block_, &capture_sub_frame_view_);
+ linear_output_framer_.get(), &output_framer_,
+ block_processor_.get(), linear_output_block_.get(),
+ &linear_output_sub_frame_view_, &capture_block_,
+ &capture_sub_frame_view_);
- ProcessCaptureFrameContent(capture, level_change,
+ ProcessCaptureFrameContent(linear_output, capture, level_change,
saturated_microphone_signal_, 1, &capture_blocker_,
- &output_framer_, block_processor_.get(),
- &capture_block_, &capture_sub_frame_view_);
+ linear_output_framer_.get(), &output_framer_,
+ block_processor_.get(), linear_output_block_.get(),
+ &linear_output_sub_frame_view_, &capture_block_,
+ &capture_sub_frame_view_);
ProcessRemainingCaptureFrameContent(
level_change, saturated_microphone_signal_, &capture_blocker_,
- &output_framer_, block_processor_.get(), &capture_block_);
+ linear_output_framer_.get(), &output_framer_, block_processor_.get(),
+ linear_output_block_.get(), &capture_block_);
data_dumper_->DumpWav("aec3_capture_output", AudioBuffer::kSplitBandSize,
&capture->split_bands(0)[0][0], 16000, 1);
diff --git a/modules/audio_processing/aec3/echo_canceller3.h b/modules/audio_processing/aec3/echo_canceller3.h
index ce36cc1..a828d57 100644
--- a/modules/audio_processing/aec3/echo_canceller3.h
+++ b/modules/audio_processing/aec3/echo_canceller3.h
@@ -70,8 +70,6 @@
// Main class for the echo canceller3.
// It does 4 things:
// -Receives 10 ms frames of band-split audio.
-// -Optionally applies an anti-hum (high-pass) filter on the
-// received signals.
// -Provides the lower level echo canceller functionality with
// blocks of 64 samples of audio data.
// -Partially handles the jitter in the render and capture API
@@ -106,6 +104,10 @@
// Processes the split-band domain capture signal in order to remove any echo
// present in the signal.
void ProcessCapture(AudioBuffer* capture, bool level_change) override;
+ // As above, but also returns the linear filter output.
+ void ProcessCapture(AudioBuffer* capture,
+ AudioBuffer* linear_output,
+ bool level_change) override;
// Collect current metrics from the echo canceller.
Metrics GetMetrics() const override;
// Provides an optional external estimate of the audio buffer delay.
@@ -149,6 +151,8 @@
const int num_bands_;
const size_t num_render_channels_;
const size_t num_capture_channels_;
+ std::unique_ptr<BlockFramer> linear_output_framer_
+ RTC_GUARDED_BY(capture_race_checker_);
BlockFramer output_framer_ RTC_GUARDED_BY(capture_race_checker_);
FrameBlocker capture_blocker_ RTC_GUARDED_BY(capture_race_checker_);
FrameBlocker render_blocker_ RTC_GUARDED_BY(capture_race_checker_);
@@ -163,10 +167,14 @@
false;
std::vector<std::vector<std::vector<float>>> render_block_
RTC_GUARDED_BY(capture_race_checker_);
+ std::unique_ptr<std::vector<std::vector<std::vector<float>>>>
+ linear_output_block_ RTC_GUARDED_BY(capture_race_checker_);
std::vector<std::vector<std::vector<float>>> capture_block_
RTC_GUARDED_BY(capture_race_checker_);
std::vector<std::vector<rtc::ArrayView<float>>> render_sub_frame_view_
RTC_GUARDED_BY(capture_race_checker_);
+ std::vector<std::vector<rtc::ArrayView<float>>> linear_output_sub_frame_view_
+ RTC_GUARDED_BY(capture_race_checker_);
std::vector<std::vector<rtc::ArrayView<float>>> capture_sub_frame_view_
RTC_GUARDED_BY(capture_race_checker_);
BlockDelayBuffer block_delay_buffer_ RTC_GUARDED_BY(capture_race_checker_);
diff --git a/modules/audio_processing/aec3/echo_canceller3_unittest.cc b/modules/audio_processing/aec3/echo_canceller3_unittest.cc
index e7b9ddb..8d9199c 100644
--- a/modules/audio_processing/aec3/echo_canceller3_unittest.cc
+++ b/modules/audio_processing/aec3/echo_canceller3_unittest.cc
@@ -112,6 +112,7 @@
void ProcessCapture(
bool level_change,
bool saturated_microphone_signal,
+ std::vector<std::vector<std::vector<float>>>* linear_output,
std::vector<std::vector<std::vector<float>>>* capture_block) override {}
void BufferRender(
@@ -137,6 +138,7 @@
void ProcessCapture(
bool level_change,
bool saturated_microphone_signal,
+ std::vector<std::vector<std::vector<float>>>* linear_output,
std::vector<std::vector<std::vector<float>>>* capture_block) override {
std::vector<std::vector<std::vector<float>>> render_block =
received_render_blocks_.front();
@@ -267,17 +269,17 @@
switch (echo_path_change_test_variant) {
case EchoPathChangeTestVariant::kNone:
- EXPECT_CALL(*block_processor_mock, ProcessCapture(false, _, _))
+ EXPECT_CALL(*block_processor_mock, ProcessCapture(false, _, _, _))
.Times(kExpectedNumBlocksToProcess);
break;
case EchoPathChangeTestVariant::kOneSticky:
- EXPECT_CALL(*block_processor_mock, ProcessCapture(true, _, _))
+ EXPECT_CALL(*block_processor_mock, ProcessCapture(true, _, _, _))
.Times(kExpectedNumBlocksToProcess);
break;
case EchoPathChangeTestVariant::kOneNonSticky:
- EXPECT_CALL(*block_processor_mock, ProcessCapture(true, _, _))
+ EXPECT_CALL(*block_processor_mock, ProcessCapture(true, _, _, _))
.Times(kNumFullBlocksPerFrame);
- EXPECT_CALL(*block_processor_mock, ProcessCapture(false, _, _))
+ EXPECT_CALL(*block_processor_mock, ProcessCapture(false, _, _, _))
.Times(kExpectedNumBlocksToProcess - kNumFullBlocksPerFrame);
break;
}
@@ -338,7 +340,7 @@
new StrictMock<webrtc::test::MockBlockProcessor>());
EXPECT_CALL(*block_processor_mock, BufferRender(_))
.Times(kExpectedNumBlocksToProcess);
- EXPECT_CALL(*block_processor_mock, ProcessCapture(_, _, _))
+ EXPECT_CALL(*block_processor_mock, ProcessCapture(_, _, _, _))
.Times(kExpectedNumBlocksToProcess);
switch (leakage_report_variant) {
@@ -429,21 +431,21 @@
switch (saturation_variant) {
case SaturationTestVariant::kNone:
- EXPECT_CALL(*block_processor_mock, ProcessCapture(_, false, _))
+ EXPECT_CALL(*block_processor_mock, ProcessCapture(_, false, _, _))
.Times(kExpectedNumBlocksToProcess);
break;
case SaturationTestVariant::kOneNegative: {
::testing::InSequence s;
- EXPECT_CALL(*block_processor_mock, ProcessCapture(_, true, _))
+ EXPECT_CALL(*block_processor_mock, ProcessCapture(_, true, _, _))
.Times(kNumFullBlocksPerFrame);
- EXPECT_CALL(*block_processor_mock, ProcessCapture(_, false, _))
+ EXPECT_CALL(*block_processor_mock, ProcessCapture(_, false, _, _))
.Times(kExpectedNumBlocksToProcess - kNumFullBlocksPerFrame);
} break;
case SaturationTestVariant::kOnePositive: {
::testing::InSequence s;
- EXPECT_CALL(*block_processor_mock, ProcessCapture(_, true, _))
+ EXPECT_CALL(*block_processor_mock, ProcessCapture(_, true, _, _))
.Times(kNumFullBlocksPerFrame);
- EXPECT_CALL(*block_processor_mock, ProcessCapture(_, false, _))
+ EXPECT_CALL(*block_processor_mock, ProcessCapture(_, false, _, _))
.Times(kExpectedNumBlocksToProcess - kNumFullBlocksPerFrame);
} break;
}
diff --git a/modules/audio_processing/aec3/echo_remover.cc b/modules/audio_processing/aec3/echo_remover.cc
index bf68f36..89ba736 100644
--- a/modules/audio_processing/aec3/echo_remover.cc
+++ b/modules/audio_processing/aec3/echo_remover.cc
@@ -123,6 +123,7 @@
bool capture_signal_saturation,
const absl::optional<DelayEstimate>& external_delay,
RenderBuffer* render_buffer,
+ std::vector<std::vector<std::vector<float>>>* linear_output,
std::vector<std::vector<std::vector<float>>>* capture) override;
// Updates the status on whether echo leakage is detected in the output of the
@@ -235,6 +236,7 @@
bool capture_signal_saturation,
const absl::optional<DelayEstimate>& external_delay,
RenderBuffer* render_buffer,
+ std::vector<std::vector<std::vector<float>>>* linear_output,
std::vector<std::vector<std::vector<float>>>* capture) {
++block_counter_;
const std::vector<std::vector<std::vector<float>>>& x =
@@ -367,6 +369,16 @@
E[ch].Spectrum(optimization_, E2[ch]);
}
+ // Optionally return the linear filter output.
+ if (linear_output) {
+ RTC_DCHECK_GE(1, linear_output->size());
+ RTC_DCHECK_EQ(num_capture_channels_, linear_output[0].size());
+ for (size_t ch = 0; ch < num_capture_channels_; ++ch) {
+ RTC_DCHECK_EQ(kBlockSize, (*linear_output)[0][ch].size());
+ std::copy(e[ch].begin(), e[ch].end(), (*linear_output)[0][ch].begin());
+ }
+ }
+
// Update the AEC state information.
aec_state_.Update(external_delay, subtractor_.FilterFrequencyResponses(),
subtractor_.FilterImpulseResponses(), *render_buffer, E2,
diff --git a/modules/audio_processing/aec3/echo_remover.h b/modules/audio_processing/aec3/echo_remover.h
index 6098a68..ef41646 100644
--- a/modules/audio_processing/aec3/echo_remover.h
+++ b/modules/audio_processing/aec3/echo_remover.h
@@ -42,6 +42,7 @@
bool capture_signal_saturation,
const absl::optional<DelayEstimate>& external_delay,
RenderBuffer* render_buffer,
+ std::vector<std::vector<std::vector<float>>>* linear_output,
std::vector<std::vector<std::vector<float>>>* capture) = 0;
// Updates the status on whether echo leakage is detected in the output of the
diff --git a/modules/audio_processing/aec3/echo_remover_unittest.cc b/modules/audio_processing/aec3/echo_remover_unittest.cc
index 15d0913..d79993a 100644
--- a/modules/audio_processing/aec3/echo_remover_unittest.cc
+++ b/modules/audio_processing/aec3/echo_remover_unittest.cc
@@ -73,9 +73,9 @@
render_buffer->Insert(render);
render_buffer->PrepareCaptureProcessing();
- remover->ProcessCapture(echo_path_variability,
- k % 2 == 0 ? true : false, delay_estimate,
- render_buffer->GetRenderBuffer(), &capture);
+ remover->ProcessCapture(
+ echo_path_variability, k % 2 == 0 ? true : false, delay_estimate,
+ render_buffer->GetRenderBuffer(), nullptr, &capture);
}
}
}
@@ -107,10 +107,10 @@
1, std::vector<float>(kBlockSize - 1, 0.f)));
EchoPathVariability echo_path_variability(
false, EchoPathVariability::DelayAdjustment::kNone, false);
- EXPECT_DEATH(
- remover->ProcessCapture(echo_path_variability, false, delay_estimate,
- render_buffer->GetRenderBuffer(), &capture),
- "");
+ EXPECT_DEATH(remover->ProcessCapture(
+ echo_path_variability, false, delay_estimate,
+ render_buffer->GetRenderBuffer(), nullptr, &capture),
+ "");
}
}
@@ -131,10 +131,10 @@
std::vector<float>(kBlockSize, 0.f)));
EchoPathVariability echo_path_variability(
false, EchoPathVariability::DelayAdjustment::kNone, false);
- EXPECT_DEATH(
- remover->ProcessCapture(echo_path_variability, false, delay_estimate,
- render_buffer->GetRenderBuffer(), &capture),
- "");
+ EXPECT_DEATH(remover->ProcessCapture(
+ echo_path_variability, false, delay_estimate,
+ render_buffer->GetRenderBuffer(), nullptr, &capture),
+ "");
}
}
@@ -147,10 +147,10 @@
RenderDelayBuffer::Create(EchoCanceller3Config(), 16000, 1));
EchoPathVariability echo_path_variability(
false, EchoPathVariability::DelayAdjustment::kNone, false);
- EXPECT_DEATH(
- remover->ProcessCapture(echo_path_variability, false, delay_estimate,
- render_buffer->GetRenderBuffer(), nullptr),
- "");
+ EXPECT_DEATH(remover->ProcessCapture(
+ echo_path_variability, false, delay_estimate,
+ render_buffer->GetRenderBuffer(), nullptr, nullptr),
+ "");
}
#endif
@@ -222,7 +222,8 @@
render_buffer->PrepareCaptureProcessing();
remover->ProcessCapture(echo_path_variability, false, delay_estimate,
- render_buffer->GetRenderBuffer(), &y);
+ render_buffer->GetRenderBuffer(), nullptr,
+ &y);
if (k > kNumBlocksToProcess / 2) {
output_energy = std::inner_product(y[0][0].begin(), y[0][0].end(),
diff --git a/modules/audio_processing/aec3/mock/mock_block_processor.h b/modules/audio_processing/aec3/mock/mock_block_processor.h
index 634d26e..e9a95c8 100644
--- a/modules/audio_processing/aec3/mock/mock_block_processor.h
+++ b/modules/audio_processing/aec3/mock/mock_block_processor.h
@@ -24,10 +24,11 @@
MockBlockProcessor();
virtual ~MockBlockProcessor();
- MOCK_METHOD3(
+ MOCK_METHOD4(
ProcessCapture,
void(bool level_change,
bool saturated_microphone_signal,
+ std::vector<std::vector<std::vector<float>>>* linear_output,
std::vector<std::vector<std::vector<float>>>* capture_block));
MOCK_METHOD1(BufferRender,
void(const std::vector<std::vector<std::vector<float>>>& block));
diff --git a/modules/audio_processing/aec3/mock/mock_echo_remover.h b/modules/audio_processing/aec3/mock/mock_echo_remover.h
index f8dd348..6c580f3 100644
--- a/modules/audio_processing/aec3/mock/mock_echo_remover.h
+++ b/modules/audio_processing/aec3/mock/mock_echo_remover.h
@@ -27,11 +27,12 @@
MockEchoRemover();
virtual ~MockEchoRemover();
- MOCK_METHOD5(ProcessCapture,
+ MOCK_METHOD6(ProcessCapture,
void(EchoPathVariability echo_path_variability,
bool capture_signal_saturation,
const absl::optional<DelayEstimate>& delay_estimate,
RenderBuffer* render_buffer,
+ std::vector<std::vector<std::vector<float>>>* linear_output,
std::vector<std::vector<std::vector<float>>>* capture));
MOCK_CONST_METHOD0(Delay, absl::optional<int>());
MOCK_METHOD1(UpdateEchoLeakageStatus, void(bool leakage_detected));
diff --git a/modules/audio_processing/aec3/suppression_gain.cc b/modules/audio_processing/aec3/suppression_gain.cc
index d1ef326..bd7a3d6 100644
--- a/modules/audio_processing/aec3/suppression_gain.cc
+++ b/modules/audio_processing/aec3/suppression_gain.cc
@@ -343,13 +343,6 @@
std::array<float, kFftLengthBy2Plus1>* low_band_gain) {
RTC_DCHECK(high_bands_gain);
RTC_DCHECK(low_band_gain);
- const auto& cfg = config_.suppressor;
-
- if (cfg.enforce_transparent) {
- low_band_gain->fill(1.f);
- *high_bands_gain = cfg.enforce_empty_higher_bands ? 0.f : 1.f;
- return;
- }
// Update the nearend state selection.
dominant_nearend_detector_.Update(nearend_spectrum, residual_echo_spectrum,
@@ -360,11 +353,6 @@
LowerBandGain(low_noise_render, aec_state, nearend_spectrum,
residual_echo_spectrum, comfort_noise_spectrum, low_band_gain);
- if (cfg.enforce_empty_higher_bands) {
- *high_bands_gain = 0.f;
- return;
- }
-
// Compute the gain for the upper bands.
const absl::optional<int> narrow_peak_band =
render_signal_analyzer.NarrowPeakBand();
diff --git a/modules/audio_processing/audio_processing_impl.cc b/modules/audio_processing/audio_processing_impl.cc
index 59d0c32..fad02a0 100644
--- a/modules/audio_processing/audio_processing_impl.cc
+++ b/modules/audio_processing/audio_processing_impl.cc
@@ -1342,8 +1342,9 @@
submodules_.echo_controller->SetAudioBufferDelay(stream_delay_ms());
}
+ AudioBuffer* linear_aec_buffer = capture_.linear_aec_output.get();
submodules_.echo_controller->ProcessCapture(
- capture_buffer, capture_.echo_path_gain_change);
+ capture_buffer, linear_aec_buffer, capture_.echo_path_gain_change);
} else if (submodules_.echo_cancellation) {
// Ensure that the stream delay was set before the call to the
// AEC ProcessCaptureAudio function.
@@ -1625,6 +1626,31 @@
return retval;
}
+bool AudioProcessingImpl::GetLinearAecOutput(
+ rtc::ArrayView<std::array<float, 160>> linear_output) const {
+ rtc::CritScope cs(&crit_capture_);
+ AudioBuffer* linear_aec_buffer = capture_.linear_aec_output.get();
+
+ RTC_DCHECK(linear_aec_buffer);
+ if (linear_aec_buffer) {
+ RTC_DCHECK_EQ(1, linear_aec_buffer->num_bands());
+ RTC_DCHECK_EQ(linear_output.size(), linear_aec_buffer->num_channels());
+
+ for (size_t ch = 0; ch < linear_aec_buffer->num_channels(); ++ch) {
+ RTC_DCHECK_EQ(linear_output[ch].size(), linear_aec_buffer->num_frames());
+ rtc::ArrayView<const float> channel_view =
+ rtc::ArrayView<const float>(linear_aec_buffer->channels_const()[ch],
+ linear_aec_buffer->num_frames());
+ std::copy(channel_view.begin(), channel_view.end(),
+ linear_output[ch].begin());
+ }
+ return true;
+ }
+ RTC_LOG(LS_ERROR) << "No linear AEC output available";
+ RTC_NOTREACHED();
+ return false;
+}
+
int AudioProcessingImpl::stream_delay_ms() const {
// Used as callback from submodules, hence locking is not allowed.
return capture_nonlocked_.stream_delay_ms;
@@ -1790,6 +1816,16 @@
num_proc_channels());
}
+ // Setup the storage for returning the linear AEC output.
+ if (config_.echo_canceller.export_linear_aec_output) {
+ constexpr int kLinearOutputRateHz = 16000;
+ capture_.linear_aec_output = std::make_unique<AudioBuffer>(
+ kLinearOutputRateHz, num_proc_channels(), kLinearOutputRateHz,
+ num_proc_channels(), kLinearOutputRateHz, num_proc_channels());
+ } else {
+ capture_.linear_aec_output.reset();
+ }
+
capture_nonlocked_.echo_controller_enabled = true;
submodules_.echo_cancellation.reset();
@@ -1801,6 +1837,7 @@
submodules_.echo_controller.reset();
capture_nonlocked_.echo_controller_enabled = false;
+ capture_.linear_aec_output.reset();
if (!config_.echo_canceller.enabled) {
submodules_.echo_cancellation.reset();
diff --git a/modules/audio_processing/audio_processing_impl.h b/modules/audio_processing/audio_processing_impl.h
index e13034f..f7320ac 100644
--- a/modules/audio_processing/audio_processing_impl.h
+++ b/modules/audio_processing/audio_processing_impl.h
@@ -87,6 +87,8 @@
const StreamConfig& input_config,
const StreamConfig& output_config,
float* const* dest) override;
+ bool GetLinearAecOutput(
+ rtc::ArrayView<std::array<float, 160>> linear_output) const override;
void set_output_will_be_muted(bool muted) override;
int set_stream_delay_ms(int delay) override;
void set_delay_offset_ms(int offset) override;
@@ -412,6 +414,7 @@
bool transient_suppressor_enabled;
std::unique_ptr<AudioBuffer> capture_audio;
std::unique_ptr<AudioBuffer> capture_fullband_audio;
+ std::unique_ptr<AudioBuffer> linear_aec_output;
// Only the rate and samples fields of capture_processing_format_ are used
// because the capture processing number of channels is mutable and is
// tracked by the capture_audio_.
diff --git a/modules/audio_processing/audio_processing_impl_unittest.cc b/modules/audio_processing/audio_processing_impl_unittest.cc
index 5707f47..c7e25a9 100644
--- a/modules/audio_processing/audio_processing_impl_unittest.cc
+++ b/modules/audio_processing/audio_processing_impl_unittest.cc
@@ -242,13 +242,13 @@
EXPECT_CALL(*echo_control_mock, AnalyzeCapture(testing::_)).Times(1);
EXPECT_CALL(*echo_control_mock,
- ProcessCapture(NotNull(), /*echo_path_change=*/false))
+ ProcessCapture(NotNull(), testing::_, /*echo_path_change=*/false))
.Times(1);
apm->ProcessStream(&frame);
EXPECT_CALL(*echo_control_mock, AnalyzeCapture(testing::_)).Times(1);
EXPECT_CALL(*echo_control_mock,
- ProcessCapture(NotNull(), /*echo_path_change=*/true))
+ ProcessCapture(NotNull(), testing::_, /*echo_path_change=*/true))
.Times(1);
apm->SetRuntimeSetting(
AudioProcessing::RuntimeSetting::CreateCapturePreGain(2.f));
@@ -286,7 +286,8 @@
const int initial_analog_gain = apm->recommended_stream_analog_level();
EXPECT_CALL(*echo_control_mock, AnalyzeCapture(testing::_)).Times(1);
- EXPECT_CALL(*echo_control_mock, ProcessCapture(NotNull(), false)).Times(1);
+ EXPECT_CALL(*echo_control_mock, ProcessCapture(NotNull(), testing::_, false))
+ .Times(1);
apm->ProcessStream(&frame);
// Force an analog gain change if it did not happen.
@@ -295,7 +296,8 @@
}
EXPECT_CALL(*echo_control_mock, AnalyzeCapture(testing::_)).Times(1);
- EXPECT_CALL(*echo_control_mock, ProcessCapture(NotNull(), true)).Times(1);
+ EXPECT_CALL(*echo_control_mock, ProcessCapture(NotNull(), testing::_, true))
+ .Times(1);
apm->ProcessStream(&frame);
}
@@ -326,13 +328,13 @@
EXPECT_CALL(*echo_control_mock, AnalyzeCapture(testing::_)).Times(1);
EXPECT_CALL(*echo_control_mock,
- ProcessCapture(NotNull(), /*echo_path_change=*/false))
+ ProcessCapture(NotNull(), testing::_, /*echo_path_change=*/false))
.Times(1);
apm->ProcessStream(&frame);
EXPECT_CALL(*echo_control_mock, AnalyzeCapture(testing::_)).Times(1);
EXPECT_CALL(*echo_control_mock,
- ProcessCapture(NotNull(), /*echo_path_change=*/false))
+ ProcessCapture(NotNull(), testing::_, /*echo_path_change=*/false))
.Times(1);
apm->SetRuntimeSetting(
AudioProcessing::RuntimeSetting::CreatePlayoutVolumeChange(50));
@@ -340,7 +342,7 @@
EXPECT_CALL(*echo_control_mock, AnalyzeCapture(testing::_)).Times(1);
EXPECT_CALL(*echo_control_mock,
- ProcessCapture(NotNull(), /*echo_path_change=*/false))
+ ProcessCapture(NotNull(), testing::_, /*echo_path_change=*/false))
.Times(1);
apm->SetRuntimeSetting(
AudioProcessing::RuntimeSetting::CreatePlayoutVolumeChange(50));
@@ -348,7 +350,7 @@
EXPECT_CALL(*echo_control_mock, AnalyzeCapture(testing::_)).Times(1);
EXPECT_CALL(*echo_control_mock,
- ProcessCapture(NotNull(), /*echo_path_change=*/true))
+ ProcessCapture(NotNull(), testing::_, /*echo_path_change=*/true))
.Times(1);
apm->SetRuntimeSetting(
AudioProcessing::RuntimeSetting::CreatePlayoutVolumeChange(100));
diff --git a/modules/audio_processing/audio_processing_unittest.cc b/modules/audio_processing/audio_processing_unittest.cc
index 06dbba7..0e17db7 100644
--- a/modules/audio_processing/audio_processing_unittest.cc
+++ b/modules/audio_processing/audio_processing_unittest.cc
@@ -2425,7 +2425,8 @@
auto ec = new test::MockEchoControl();
EXPECT_CALL(*ec, AnalyzeRender(::testing::_)).Times(1);
EXPECT_CALL(*ec, AnalyzeCapture(::testing::_)).Times(2);
- EXPECT_CALL(*ec, ProcessCapture(::testing::_, ::testing::_)).Times(2);
+ EXPECT_CALL(*ec, ProcessCapture(::testing::_, ::testing::_, ::testing::_))
+ .Times(2);
return std::unique_ptr<EchoControl>(ec);
}
diff --git a/modules/audio_processing/include/audio_processing.h b/modules/audio_processing/include/audio_processing.h
index f1242a7..113bd2a 100644
--- a/modules/audio_processing/include/audio_processing.h
+++ b/modules/audio_processing/include/audio_processing.h
@@ -280,6 +280,7 @@
bool legacy_moderate_suppression_level = false;
// Recommended not to use. Will be removed in the future.
bool use_legacy_aec = false;
+ bool export_linear_aec_output = false;
} echo_canceller;
// Enables background noise suppression.
@@ -611,6 +612,13 @@
virtual int AnalyzeReverseStream(const float* const* data,
const StreamConfig& reverse_config) = 0;
+ // Returns the most recently produced 10 ms of the linear AEC output at a rate
+ // of 16 kHz. If there is more than one capture channel, a mono representation
+ // of the input is returned. Returns true/false to indicate whether an output
+ // returned.
+ virtual bool GetLinearAecOutput(
+ rtc::ArrayView<std::array<float, 160>> linear_output) const = 0;
+
// This must be called prior to ProcessStream() if and only if adaptive analog
// gain control is enabled, to pass the current analog level from the audio
// HAL. Must be within the range provided in Config::GainController1.
diff --git a/modules/audio_processing/include/mock_audio_processing.h b/modules/audio_processing/include/mock_audio_processing.h
index 6b12392..0932696 100644
--- a/modules/audio_processing/include/mock_audio_processing.h
+++ b/modules/audio_processing/include/mock_audio_processing.h
@@ -47,6 +47,10 @@
MOCK_METHOD1(AnalyzeCapture, void(AudioBuffer* capture));
MOCK_METHOD2(ProcessCapture,
void(AudioBuffer* capture, bool echo_path_change));
+ MOCK_METHOD3(ProcessCapture,
+ void(AudioBuffer* capture,
+ AudioBuffer* linear_output,
+ bool echo_path_change));
MOCK_CONST_METHOD0(GetMetrics, Metrics());
MOCK_METHOD1(SetAudioBufferDelay, void(int delay_ms));
MOCK_CONST_METHOD0(ActiveProcessing, bool());
@@ -105,6 +109,9 @@
const StreamConfig& input_config,
const StreamConfig& output_config,
float* const* dest));
+ MOCK_CONST_METHOD1(
+ GetLinearAecOutput,
+ bool(rtc::ArrayView<std::array<float, 160>> linear_output));
MOCK_METHOD1(set_stream_delay_ms, int(int delay));
MOCK_CONST_METHOD0(stream_delay_ms, int());
MOCK_CONST_METHOD0(was_stream_delay_set, bool());
diff --git a/modules/audio_processing/test/audio_processing_simulator.cc b/modules/audio_processing/test/audio_processing_simulator.cc
index 7f354a9..38b97ca 100644
--- a/modules/audio_processing/test/audio_processing_simulator.cc
+++ b/modules/audio_processing/test/audio_processing_simulator.cc
@@ -227,6 +227,20 @@
buffer_file_writer_->Write(*out_buf_);
}
+ if (linear_aec_output_file_writer_) {
+ bool output_available = ap_->GetLinearAecOutput(linear_aec_output_buf_);
+ RTC_CHECK(output_available);
+ RTC_CHECK_GT(linear_aec_output_buf_.size(), 0);
+ RTC_CHECK_EQ(linear_aec_output_buf_[0].size(), 160);
+ for (size_t k = 0; k < linear_aec_output_buf_[0].size(); ++k) {
+ for (size_t ch = 0; ch < linear_aec_output_buf_.size(); ++ch) {
+ RTC_CHECK_EQ(linear_aec_output_buf_[ch].size(), 160);
+ linear_aec_output_file_writer_->WriteSamples(
+ &linear_aec_output_buf_[ch][k], 1);
+ }
+ }
+ }
+
if (residual_echo_likelihood_graph_writer_.is_open()) {
auto stats = ap_->GetStatistics(true /*has_remote_tracks*/);
residual_echo_likelihood_graph_writer_
@@ -342,6 +356,21 @@
settings_.processed_capture_samples);
}
+ if (settings_.linear_aec_output_filename) {
+ std::string filename;
+ if (settings_.store_intermediate_output) {
+ filename = GetIndexedOutputWavFilename(
+ *settings_.linear_aec_output_filename, output_reset_counter_);
+ } else {
+ filename = *settings_.linear_aec_output_filename;
+ }
+
+ linear_aec_output_file_writer_.reset(
+ new WavWriter(filename, 16000, out_config_.num_channels()));
+
+ linear_aec_output_buf_.resize(out_config_.num_channels());
+ }
+
if (settings_.reverse_output_filename) {
std::string filename;
if (settings_.store_intermediate_output) {
@@ -410,6 +439,8 @@
apm_config.echo_canceller.mobile_mode = use_aecm;
apm_config.echo_canceller.use_legacy_aec = use_legacy_aec;
}
+ apm_config.echo_canceller.export_linear_aec_output =
+ !!settings_.linear_aec_output_filename;
RTC_CHECK(!(use_legacy_aec && settings_.aec_settings_filename))
<< "The legacy AEC cannot be configured using settings";
@@ -421,9 +452,14 @@
std::cout << "Reading AEC Parameters from JSON input." << std::endl;
}
cfg = ReadAec3ConfigFromJsonFile(*settings_.aec_settings_filename);
- echo_control_factory.reset(new EchoCanceller3Factory(cfg));
}
+ if (settings_.linear_aec_output_filename) {
+ cfg.filter.export_linear_aec_output = true;
+ }
+
+ echo_control_factory.reset(new EchoCanceller3Factory(cfg));
+
if (settings_.print_aec_parameter_values) {
if (!settings_.use_quiet_output) {
std::cout << "AEC settings:" << std::endl;
diff --git a/modules/audio_processing/test/audio_processing_simulator.h b/modules/audio_processing/test/audio_processing_simulator.h
index bf718b2..8ee2db8 100644
--- a/modules/audio_processing/test/audio_processing_simulator.h
+++ b/modules/audio_processing/test/audio_processing_simulator.h
@@ -47,6 +47,7 @@
absl::optional<std::string> input_filename;
absl::optional<std::string> reverse_input_filename;
absl::optional<std::string> artificial_nearend_filename;
+ absl::optional<std::string> linear_aec_output_filename;
absl::optional<bool> use_aec;
absl::optional<bool> use_aecm;
absl::optional<bool> use_ed; // Residual Echo Detector.
@@ -156,6 +157,7 @@
std::unique_ptr<ChannelBuffer<float>> out_buf_;
std::unique_ptr<ChannelBuffer<float>> reverse_in_buf_;
std::unique_ptr<ChannelBuffer<float>> reverse_out_buf_;
+ std::vector<std::array<float, 160>> linear_aec_output_buf_;
StreamConfig in_config_;
StreamConfig out_config_;
StreamConfig reverse_in_config_;
@@ -178,6 +180,7 @@
std::unique_ptr<ChannelBufferWavWriter> buffer_file_writer_;
std::unique_ptr<ChannelBufferWavWriter> reverse_buffer_file_writer_;
std::unique_ptr<ChannelBufferVectorWriter> buffer_memory_writer_;
+ std::unique_ptr<WavWriter> linear_aec_output_file_writer_;
ApiCallStatistics api_call_statistics_;
std::ofstream residual_echo_likelihood_graph_writer_;
int analog_mic_level_;
diff --git a/modules/audio_processing/test/audioproc_float_impl.cc b/modules/audio_processing/test/audioproc_float_impl.cc
index 3e755b5..8301c4e 100644
--- a/modules/audio_processing/test/audioproc_float_impl.cc
+++ b/modules/audio_processing/test/audioproc_float_impl.cc
@@ -40,6 +40,7 @@
artificial_nearend,
"",
"Artificial nearend wav filename");
+ABSL_FLAG(std::string, linear_aec_output, "", "Linear AEC output wav filename");
ABSL_FLAG(int,
output_num_channels,
kParameterNotSpecifiedValue,
@@ -364,6 +365,8 @@
&settings.reverse_output_filename);
SetSettingIfSpecified(absl::GetFlag(FLAGS_artificial_nearend),
&settings.artificial_nearend_filename);
+ SetSettingIfSpecified(absl::GetFlag(FLAGS_linear_aec_output),
+ &settings.linear_aec_output_filename);
SetSettingIfSpecified(absl::GetFlag(FLAGS_output_num_channels),
&settings.output_num_channels);
SetSettingIfSpecified(absl::GetFlag(FLAGS_reverse_output_num_channels),
@@ -508,6 +511,19 @@
"aec dump input string!\n");
}
+ ReportConditionalErrorAndExit(settings.use_aec && !(*settings.use_aec) &&
+ settings.linear_aec_output_filename,
+ "Error: The linear AEC ouput filename cannot "
+ "be specified without the AEC being active");
+
+ ReportConditionalErrorAndExit(
+ ((settings.use_aec && *settings.use_aec && settings.use_legacy_aec &&
+ *settings.use_legacy_aec) ||
+ (settings.use_aecm && *settings.use_aecm)) &&
+ !!settings.linear_aec_output_filename,
+ "Error: The linear AEC ouput filename cannot be specified when the "
+ "legacy AEC or the AECm are used");
+
ReportConditionalErrorAndExit(
settings.use_aec && *settings.use_aec && settings.use_aecm &&
*settings.use_aecm,
@@ -618,6 +634,11 @@
"Error: --artifical_nearend must be a valid .wav file name.\n");
ReportConditionalErrorAndExit(
+ settings.linear_aec_output_filename &&
+ (!valid_wav_name(*settings.linear_aec_output_filename)),
+ "Error: --linear_aec_output must be a valid .wav file name.\n");
+
+ ReportConditionalErrorAndExit(
WEBRTC_APM_DEBUG_DUMP == 0 && settings.dump_internal_data,
"Error: --dump_data cannot be set without proper build support.\n");
diff --git a/modules/audio_processing/test/echo_control_mock.h b/modules/audio_processing/test/echo_control_mock.h
index c2082c2..95d3be5 100644
--- a/modules/audio_processing/test/echo_control_mock.h
+++ b/modules/audio_processing/test/echo_control_mock.h
@@ -24,6 +24,10 @@
MOCK_METHOD1(AnalyzeCapture, void(AudioBuffer* capture));
MOCK_METHOD2(ProcessCapture,
void(AudioBuffer* capture, bool echo_path_change));
+ MOCK_METHOD3(ProcessCapture,
+ void(AudioBuffer* capture,
+ AudioBuffer* linear_output,
+ bool echo_path_change));
MOCK_CONST_METHOD0(GetMetrics, EchoControl::Metrics());
MOCK_METHOD1(SetAudioBufferDelay, void(int delay_ms));
MOCK_CONST_METHOD0(ActiveProcessing, bool());