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Correction of the AEC3 underrun behavior and minor other corrections

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This CL corrects the behavior during buffer underruns.

Furthermore, it increases the tolerance for API call jitter, and
removes the minimum value for the comfort noise.

BUG=webrtc:6018

Review-Url: https://codereview.webrtc.org/2787123003
Cr-Commit-Position: refs/heads/master@{#17576}
This commit is contained in:
peah 2017-04-06 16:33:06 -07:00 committed by Commit bot
parent 86afe9d661
commit 4b572d048a
3 changed files with 32 additions and 27 deletions
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2
webrtc/modules/audio_processing/aec3/aec3_common.h
View File

@ -63,7 +63,7 @@ constexpr float kFixedEchoPathGain = 100;
constexpr size_t kRenderDelayBufferSize =
(3 * kDownsampledRenderBufferSize) / (4 * kSubBlockSize);
constexpr size_t kMaxApiCallsJitterBlocks = 10;
constexpr size_t kMaxApiCallsJitterBlocks = 20;
constexpr size_t kRenderTransferQueueSize = kMaxApiCallsJitterBlocks / 2;
static_assert(2 * kRenderTransferQueueSize >= kMaxApiCallsJitterBlocks,
"Requirement to ensure buffer overflow detection");

52
webrtc/modules/audio_processing/aec3/block_processor.cc
View File

@ -101,34 +101,34 @@ void BlockProcessorImpl::ProcessCapture(
// been a render buffer overrun as the buffer alignment may be noncausal.
delay_controller_->Reset();
render_buffer_->Reset();
} else {
// Update the render buffers with new render data, filling the buffers with
// empty blocks when there is no render data available.
render_buffer_underrun = !render_buffer_->UpdateBuffers();
// Compute and and apply the render delay required to achieve proper signal
// alignment.
const size_t old_delay = render_buffer_->Delay();
const size_t new_delay = delay_controller_->GetDelay(
render_buffer_->GetDownsampledRenderBuffer(), (*capture_block)[0]);
render_buffer_->SetDelay(new_delay);
const size_t achieved_delay = render_buffer_->Delay();
// Inform the delay controller of the actually set delay to allow it to
// properly react to a non-feasible delay.
delay_controller_->SetDelay(achieved_delay);
// Remove the echo from the capture signal.
echo_remover_->ProcessCapture(
delay_controller_->AlignmentHeadroomSamples(),
EchoPathVariability(echo_path_gain_change,
old_delay != achieved_delay ||
old_delay != new_delay ||
render_buffer_overrun_occurred_),
capture_signal_saturation, render_buffer_->GetRenderBuffer(),
capture_block);
}
// Update the render buffers with new render data, filling the buffers with
// empty blocks when there is no render data available.
render_buffer_underrun = !render_buffer_->UpdateBuffers();
// Compute and and apply the render delay required to achieve proper signal
// alignment.
const size_t old_delay = render_buffer_->Delay();
const size_t new_delay = delay_controller_->GetDelay(
render_buffer_->GetDownsampledRenderBuffer(), (*capture_block)[0]);
render_buffer_->SetDelay(new_delay);
const size_t achieved_delay = render_buffer_->Delay();
// Inform the delay controller of the actually set delay to allow it to
// properly react to a non-feasible delay.
delay_controller_->SetDelay(achieved_delay);
// Remove the echo from the capture signal.
echo_remover_->ProcessCapture(
delay_controller_->AlignmentHeadroomSamples(),
EchoPathVariability(echo_path_gain_change,
old_delay != achieved_delay ||
old_delay != new_delay ||
render_buffer_overrun_occurred_),
capture_signal_saturation, render_buffer_->GetRenderBuffer(),
capture_block);
// Update the metrics.
metrics_.UpdateCapture(render_buffer_underrun);

5
webrtc/modules/audio_processing/aec3/render_delay_buffer.cc
View File

@ -124,6 +124,11 @@ void RenderDelayBufferImpl::Reset() {
downsampled_render_buffer_.buffer.fill(0.f);
fft_buffer_.Clear();
api_call_jitter_buffer_.Reset();
for (auto& c : buffer_) {
for (auto& b : c) {
std::fill(b.begin(), b.end(), 0.f);
}
}
}
bool RenderDelayBufferImpl::Insert(