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11539f0b29
This CL simplifies the buffering of render data. Instead of making assumptions about the worst possible platform, it leverages recent improvements in the delay estimator to quickly adapt when the conditions change. Pros: - No capture delay, delay is found ~200 ms faster. - Cleaner code that makes the concept of delay more clear. - Allows for removal of one matched filter because of the jitter headroom removal. Cons: - Delay estimator needs to re-adapt when the call jitter increases. The code can be deactivated by a kill switch. When the kill switch is pulled the CL is bit exact. Bug: webrtc:9726,chromium:895338 Change-Id: Ie2f9c8c5ce5b5a4510b4bdb95db2b970b57cd5d0 Reviewed-on: https://webrtc-review.googlesource.com/c/96920 Commit-Queue: Gustaf Ullberg <gustaf@webrtc.org> Reviewed-by: Per Åhgren <peah@webrtc.org> Cr-Commit-Position: refs/heads/master@{#25169}
49 lines
1.9 KiB
C++
49 lines
1.9 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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#ifndef MODULES_AUDIO_PROCESSING_AEC3_RENDER_DELAY_CONTROLLER_H_
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#define MODULES_AUDIO_PROCESSING_AEC3_RENDER_DELAY_CONTROLLER_H_
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#include "absl/types/optional.h"
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#include "api/array_view.h"
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#include "api/audio/echo_canceller3_config.h"
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#include "modules/audio_processing/aec3/delay_estimate.h"
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#include "modules/audio_processing/aec3/downsampled_render_buffer.h"
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#include "modules/audio_processing/aec3/render_delay_buffer.h"
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#include "modules/audio_processing/logging/apm_data_dumper.h"
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namespace webrtc {
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// Class for aligning the render and capture signal using a RenderDelayBuffer.
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class RenderDelayController {
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public:
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static RenderDelayController* Create(const EchoCanceller3Config& config,
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int non_causal_offset,
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int sample_rate_hz);
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static RenderDelayController* Create2(const EchoCanceller3Config& config,
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int sample_rate_hz);
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virtual ~RenderDelayController() = default;
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// Resets the delay controller.
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virtual void Reset() = 0;
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// Logs a render call.
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virtual void LogRenderCall() = 0;
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// Aligns the render buffer content with the capture signal.
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virtual absl::optional<DelayEstimate> GetDelay(
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const DownsampledRenderBuffer& render_buffer,
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size_t render_delay_buffer_delay,
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const absl::optional<int>& echo_remover_delay,
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rtc::ArrayView<const float> capture) = 0;
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};
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} // namespace webrtc
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#endif // MODULES_AUDIO_PROCESSING_AEC3_RENDER_DELAY_CONTROLLER_H_
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