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This CL was generated by running git ls-files | grep -P "(\.h|\.cc)$" | grep -v 'sdk/' | grep -v 'rtc_base/ssl_' | \ grep -v 'fake_rtc_certificate_generator.h' | grep -v 'modules/audio_device/win/' | \ grep -v 'system_wrappers/source/clock.cc' | grep -v 'rtc_base/trace_event.h' | \ grep -v 'modules/audio_coding/codecs/ilbc/' | grep -v 'screen_capturer_mac.h' | \ grep -v 'spl_inl_mips.h' | grep -v 'data_size_unittest.cc' | grep -v 'timestamp_unittest.cc' \ | xargs clang-format -i ; git cl format Most of these changes are clang-format grouping and reordering includes differently. Bug: webrtc:9340 Change-Id: Ic83ddbc169bfacd21883e381b5181c3dd4fe8a63 Reviewed-on: https://webrtc-review.googlesource.com/c/src/+/144051 Commit-Queue: Jonas Olsson <jonasolsson@webrtc.org> Reviewed-by: Karl Wiberg <kwiberg@webrtc.org> Cr-Commit-Position: refs/heads/master@{#28505}
82 lines
2.7 KiB
C++
82 lines
2.7 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_BUFFER_H_
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#define MODULES_AUDIO_PROCESSING_AEC3_RENDER_DELAY_BUFFER_H_
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#include <stddef.h>
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#include <vector>
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#include "api/audio/echo_canceller3_config.h"
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#include "modules/audio_processing/aec3/downsampled_render_buffer.h"
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#include "modules/audio_processing/aec3/render_buffer.h"
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namespace webrtc {
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// Class for buffering the incoming render blocks such that these may be
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// extracted with a specified delay.
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class RenderDelayBuffer {
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public:
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enum class BufferingEvent {
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kNone,
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kRenderUnderrun,
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kRenderOverrun,
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kApiCallSkew
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};
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static RenderDelayBuffer* Create(const EchoCanceller3Config& config,
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size_t num_bands);
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virtual ~RenderDelayBuffer() = default;
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// Resets the buffer alignment.
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virtual void Reset() = 0;
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// Inserts a block into the buffer.
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virtual BufferingEvent Insert(
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const std::vector<std::vector<float>>& block) = 0;
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// Updates the buffers one step based on the specified buffer delay. Returns
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// an enum indicating whether there was a special event that occurred.
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virtual BufferingEvent PrepareCaptureProcessing() = 0;
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// Sets the buffer delay and returns a bool indicating whether the delay
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// changed.
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virtual bool AlignFromDelay(size_t delay) = 0;
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// Sets the buffer delay from the most recently reported external delay.
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virtual void AlignFromExternalDelay() = 0;
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// Gets the buffer delay.
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virtual size_t Delay() const = 0;
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// Gets the buffer delay.
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virtual size_t MaxDelay() const = 0;
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// Returns the render buffer for the echo remover.
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virtual RenderBuffer* GetRenderBuffer() = 0;
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// Returns the downsampled render buffer.
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virtual const DownsampledRenderBuffer& GetDownsampledRenderBuffer() const = 0;
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// Returns the maximum non calusal offset that can occur in the delay buffer.
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static int DelayEstimatorOffset(const EchoCanceller3Config& config);
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// Provides an optional external estimate of the audio buffer delay.
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virtual void SetAudioBufferDelay(size_t delay_ms) = 0;
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// Returns whether an external delay estimate has been reported via
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// SetAudioBufferDelay.
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virtual bool HasReceivedBufferDelay() = 0;
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};
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} // namespace webrtc
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#endif // MODULES_AUDIO_PROCESSING_AEC3_RENDER_DELAY_BUFFER_H_
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