Generalized the hysteresis behavior in the AEC3 delay estimator
This CL generalizes the hysteresis behavior on the AEC3 delay estimator
to be two-sided and easier to configure.
Bug: webrtc:8671
Change-Id: Ife21c1511416e32eb3618c81178deefe332ac1e8
Reviewed-on: https://webrtc-review.googlesource.com/39267
Reviewed-by: Gustaf Ullberg <gustaf@webrtc.org>
Commit-Queue: Per Åhgren <peah@webrtc.org>
Cr-Commit-Position: refs/heads/master@{#21604}
diff --git a/modules/audio_processing/aec3/render_delay_controller.cc b/modules/audio_processing/aec3/render_delay_controller.cc
index dfc47f4..16a4477 100644
--- a/modules/audio_processing/aec3/render_delay_controller.cc
+++ b/modules/audio_processing/aec3/render_delay_controller.cc
@@ -39,6 +39,9 @@
private:
static int instance_count_;
std::unique_ptr<ApmDataDumper> data_dumper_;
+ const int delay_headroom_blocks_;
+ const int hysteresis_limit_1_blocks_;
+ const int hysteresis_limit_2_blocks_;
rtc::Optional<size_t> delay_;
EchoPathDelayEstimator delay_estimator_;
size_t align_call_counter_ = 0;
@@ -48,19 +51,30 @@
RTC_DISALLOW_IMPLICIT_CONSTRUCTORS(RenderDelayControllerImpl);
};
-size_t ComputeNewBufferDelay(rtc::Optional<size_t> current_delay,
+size_t ComputeNewBufferDelay(const rtc::Optional<size_t>& current_delay,
+ int delay_headroom_blocks,
+ int hysteresis_limit_1_blocks,
+ int hysteresis_limit_2_blocks,
size_t delay_samples) {
// The below division is not exact and the truncation is intended.
const int echo_path_delay_blocks = delay_samples >> kBlockSizeLog2;
- constexpr int kDelayHeadroomBlocks = 1;
// Compute the buffer delay increase required to achieve the desired latency.
- size_t new_delay = std::max(echo_path_delay_blocks - kDelayHeadroomBlocks, 0);
+ size_t new_delay =
+ std::max(echo_path_delay_blocks - delay_headroom_blocks, 0);
// Add hysteresis.
if (current_delay) {
- if (new_delay == *current_delay + 1) {
- new_delay = *current_delay;
+ if (new_delay > *current_delay) {
+ if (new_delay <= *current_delay + hysteresis_limit_1_blocks) {
+ new_delay = *current_delay;
+ }
+ } else if (new_delay < *current_delay) {
+ size_t hysteresis_limit = std::max(
+ static_cast<int>(*current_delay) - hysteresis_limit_2_blocks, 0);
+ if (new_delay >= hysteresis_limit) {
+ new_delay = *current_delay;
+ }
}
}
@@ -75,6 +89,12 @@
int sample_rate_hz)
: data_dumper_(
new ApmDataDumper(rtc::AtomicOps::Increment(&instance_count_))),
+ delay_headroom_blocks_(
+ static_cast<int>(config.delay.delay_headroom_blocks)),
+ hysteresis_limit_1_blocks_(
+ static_cast<int>(config.delay.hysteresis_limit_1_blocks)),
+ hysteresis_limit_2_blocks_(
+ static_cast<int>(config.delay.hysteresis_limit_2_blocks)),
delay_estimator_(data_dumper_.get(), config),
delay_buf_(kBlockSize * non_causal_offset, 0.f) {
RTC_DCHECK(ValidFullBandRate(sample_rate_hz));
@@ -120,7 +140,9 @@
if (delay_samples) {
// Compute and set new render delay buffer delay.
if (align_call_counter_ > kNumBlocksPerSecond) {
- delay_ = ComputeNewBufferDelay(delay_, static_cast<int>(*delay_samples));
+ delay_ = ComputeNewBufferDelay(
+ delay_, delay_headroom_blocks_, hysteresis_limit_1_blocks_,
+ hysteresis_limit_2_blocks_, static_cast<int>(*delay_samples));
}
metrics_.Update(static_cast<int>(*delay_samples), delay_ ? *delay_ : 0);
diff --git a/modules/audio_processing/include/audio_processing.h b/modules/audio_processing/include/audio_processing.h
index 8951b8c..6f7cda0 100644
--- a/modules/audio_processing/include/audio_processing.h
+++ b/modules/audio_processing/include/audio_processing.h
@@ -1238,6 +1238,9 @@
size_t num_filters = 4;
size_t api_call_jitter_blocks = 26;
size_t min_echo_path_delay_blocks = 5;
+ size_t delay_headroom_blocks = 1;
+ size_t hysteresis_limit_1_blocks = 1;
+ size_t hysteresis_limit_2_blocks = 0;
} delay;
struct Filter {