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In order to eliminate the WebRTC Subtree mirror in Chromium, WebRTC is moving the content of the src/webrtc directory up to the src/ directory. NOPRESUBMIT=true NOTREECHECKS=true NOTRY=true TBR=tommi@webrtc.org Bug: chromium:611808 Change-Id: Iac59c5b51b950f174119565bac87955a7994bc38 Reviewed-on: https://webrtc-review.googlesource.com/1560 Commit-Queue: Mirko Bonadei <mbonadei@webrtc.org> Reviewed-by: Henrik Kjellander <kjellander@webrtc.org> Cr-Commit-Position: refs/heads/master@{#19845}
146 lines
4.9 KiB
C++
146 lines
4.9 KiB
C++
/*
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* Copyright (c) 2017 The WebRTC project authors. All Rights Reserved.
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*
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* Use of this source code is governed by a BSD-style license
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* that can be found in the LICENSE file in the root of the source
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* tree. An additional intellectual property rights grant can be found
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* in the file PATENTS. All contributing project authors may
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* be found in the AUTHORS file in the root of the source tree.
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*/
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#include "webrtc/modules/audio_processing/aec3/render_delay_controller.h"
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#include <algorithm>
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#include <memory>
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#include <string>
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#include <vector>
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#include "webrtc/modules/audio_processing/aec3/aec3_common.h"
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#include "webrtc/modules/audio_processing/aec3/echo_path_delay_estimator.h"
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#include "webrtc/modules/audio_processing/aec3/render_delay_controller_metrics.h"
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#include "webrtc/modules/audio_processing/include/audio_processing.h"
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#include "webrtc/rtc_base/atomicops.h"
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#include "webrtc/rtc_base/constructormagic.h"
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namespace webrtc {
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namespace {
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class RenderDelayControllerImpl final : public RenderDelayController {
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public:
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RenderDelayControllerImpl(
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const AudioProcessing::Config::EchoCanceller3& config,
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int sample_rate_hz);
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~RenderDelayControllerImpl() override;
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void Reset() override;
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void SetDelay(size_t render_delay) override;
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size_t GetDelay(const DownsampledRenderBuffer& render_buffer,
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rtc::ArrayView<const float> capture) override;
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rtc::Optional<size_t> AlignmentHeadroomSamples() const override {
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return headroom_samples_;
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}
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private:
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static int instance_count_;
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std::unique_ptr<ApmDataDumper> data_dumper_;
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size_t delay_ = kMinEchoPathDelayBlocks;
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EchoPathDelayEstimator delay_estimator_;
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size_t blocks_since_last_delay_estimate_ = 300000;
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int echo_path_delay_samples_ = kMinEchoPathDelayBlocks * kBlockSize;
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size_t align_call_counter_ = 0;
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rtc::Optional<size_t> headroom_samples_;
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RenderDelayControllerMetrics metrics_;
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RTC_DISALLOW_IMPLICIT_CONSTRUCTORS(RenderDelayControllerImpl);
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};
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size_t ComputeNewBufferDelay(size_t current_delay,
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size_t echo_path_delay_samples) {
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// The below division is not exact and the truncation is intended.
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const int echo_path_delay_blocks = echo_path_delay_samples / kBlockSize;
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constexpr int kDelayHeadroomBlocks = 1;
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// Compute the buffer delay increase required to achieve the desired latency.
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size_t new_delay = std::max(echo_path_delay_blocks - kDelayHeadroomBlocks, 0);
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// Add hysteresis.
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if (new_delay == current_delay + 1) {
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new_delay = current_delay;
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}
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return new_delay;
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}
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int RenderDelayControllerImpl::instance_count_ = 0;
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RenderDelayControllerImpl::RenderDelayControllerImpl(
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const AudioProcessing::Config::EchoCanceller3& config,
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int sample_rate_hz)
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: data_dumper_(
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new ApmDataDumper(rtc::AtomicOps::Increment(&instance_count_))),
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delay_estimator_(data_dumper_.get(), config) {
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RTC_DCHECK(ValidFullBandRate(sample_rate_hz));
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}
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RenderDelayControllerImpl::~RenderDelayControllerImpl() = default;
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void RenderDelayControllerImpl::Reset() {
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delay_ = kMinEchoPathDelayBlocks;
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blocks_since_last_delay_estimate_ = 300000;
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echo_path_delay_samples_ = delay_ * kBlockSize;
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align_call_counter_ = 0;
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headroom_samples_ = rtc::Optional<size_t>();
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delay_estimator_.Reset();
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}
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void RenderDelayControllerImpl::SetDelay(size_t render_delay) {
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if (delay_ != render_delay) {
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// If a the delay set does not match the actual delay, reset the delay
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// controller.
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Reset();
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delay_ = render_delay;
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}
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}
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size_t RenderDelayControllerImpl::GetDelay(
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const DownsampledRenderBuffer& render_buffer,
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rtc::ArrayView<const float> capture) {
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RTC_DCHECK_EQ(kBlockSize, capture.size());
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++align_call_counter_;
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rtc::Optional<size_t> echo_path_delay_samples =
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delay_estimator_.EstimateDelay(render_buffer, capture);
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if (echo_path_delay_samples) {
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blocks_since_last_delay_estimate_ = 0;
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echo_path_delay_samples_ = *echo_path_delay_samples;
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// Compute and set new render delay buffer delay.
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const size_t new_delay =
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ComputeNewBufferDelay(delay_, echo_path_delay_samples_);
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if (align_call_counter_ > kNumBlocksPerSecond) {
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delay_ = new_delay;
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// Update render delay buffer headroom.
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const int headroom = echo_path_delay_samples_ - delay_ * kBlockSize;
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RTC_DCHECK_LE(0, headroom);
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headroom_samples_ = rtc::Optional<size_t>(headroom);
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}
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}
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metrics_.Update(echo_path_delay_samples, delay_);
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data_dumper_->DumpRaw("aec3_render_delay_controller_delay", 1,
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&echo_path_delay_samples_);
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data_dumper_->DumpRaw("aec3_render_delay_controller_buffer_delay", delay_);
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return delay_;
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}
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} // namespace
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RenderDelayController* RenderDelayController::Create(
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const AudioProcessing::Config::EchoCanceller3& config,
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int sample_rate_hz) {
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return new RenderDelayControllerImpl(config, sample_rate_hz);
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}
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} // namespace webrtc
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