mirror of
https://github.com/mollyim/webrtc.git
synced 2025-05-14 06:10:40 +01:00
ff571c60a9
Internal counters in the RenderDelayBuffer can slip out of sync with external counters, leading to buffer misalignment. This CL gives the RenderDelayBuffer an opportunity to update its counters. Tested: Passes: modules_unittests --gtest_filter=BlockProcessor.* Fails as expected due to new unit test: modules_unittests --gtest_filter=BlockProcessor.* --force_fieldtrials="WebRTC-Aec3RenderBufferCallCounterUpdateKillSwitch/Enabled/" audioproc_f with default AEC settings has been verified to be bit-exact on a large number of aecdumps. Bug: webrtc:11803 Change-Id: I9363b834c8c8c934add0335013df60bf131da4bc Reviewed-on: https://webrtc-review.googlesource.com/c/src/+/180126 Reviewed-by: Per Åhgren <peah@webrtc.org> Commit-Queue: Sam Zackrisson <saza@webrtc.org> Cr-Commit-Position: refs/heads/master@{#31795}
86 lines
2.9 KiB
C++
86 lines
2.9 KiB
C++
/*
|
|
* Copyright (c) 2017 The WebRTC project authors. All Rights Reserved.
|
|
*
|
|
* Use of this source code is governed by a BSD-style license
|
|
* that can be found in the LICENSE file in the root of the source
|
|
* tree. An additional intellectual property rights grant can be found
|
|
* in the file PATENTS. All contributing project authors may
|
|
* be found in the AUTHORS file in the root of the source tree.
|
|
*/
|
|
|
|
#ifndef MODULES_AUDIO_PROCESSING_AEC3_RENDER_DELAY_BUFFER_H_
|
|
#define MODULES_AUDIO_PROCESSING_AEC3_RENDER_DELAY_BUFFER_H_
|
|
|
|
#include <stddef.h>
|
|
|
|
#include <vector>
|
|
|
|
#include "api/audio/echo_canceller3_config.h"
|
|
#include "modules/audio_processing/aec3/downsampled_render_buffer.h"
|
|
#include "modules/audio_processing/aec3/render_buffer.h"
|
|
|
|
namespace webrtc {
|
|
|
|
// Class for buffering the incoming render blocks such that these may be
|
|
// extracted with a specified delay.
|
|
class RenderDelayBuffer {
|
|
public:
|
|
enum class BufferingEvent {
|
|
kNone,
|
|
kRenderUnderrun,
|
|
kRenderOverrun,
|
|
kApiCallSkew
|
|
};
|
|
|
|
static RenderDelayBuffer* Create(const EchoCanceller3Config& config,
|
|
int sample_rate_hz,
|
|
size_t num_render_channels);
|
|
virtual ~RenderDelayBuffer() = default;
|
|
|
|
// Resets the buffer alignment.
|
|
virtual void Reset() = 0;
|
|
|
|
// Inserts a block into the buffer.
|
|
virtual BufferingEvent Insert(
|
|
const std::vector<std::vector<std::vector<float>>>& block) = 0;
|
|
|
|
// Updates the buffers one step based on the specified buffer delay. Returns
|
|
// an enum indicating whether there was a special event that occurred.
|
|
virtual BufferingEvent PrepareCaptureProcessing() = 0;
|
|
|
|
// Called on capture blocks where PrepareCaptureProcessing is not called.
|
|
virtual void HandleSkippedCaptureProcessing() = 0;
|
|
|
|
// Sets the buffer delay and returns a bool indicating whether the delay
|
|
// changed.
|
|
virtual bool AlignFromDelay(size_t delay) = 0;
|
|
|
|
// Sets the buffer delay from the most recently reported external delay.
|
|
virtual void AlignFromExternalDelay() = 0;
|
|
|
|
// Gets the buffer delay.
|
|
virtual size_t Delay() const = 0;
|
|
|
|
// Gets the buffer delay.
|
|
virtual size_t MaxDelay() const = 0;
|
|
|
|
// Returns the render buffer for the echo remover.
|
|
virtual RenderBuffer* GetRenderBuffer() = 0;
|
|
|
|
// Returns the downsampled render buffer.
|
|
virtual const DownsampledRenderBuffer& GetDownsampledRenderBuffer() const = 0;
|
|
|
|
// Returns the maximum non calusal offset that can occur in the delay buffer.
|
|
static int DelayEstimatorOffset(const EchoCanceller3Config& config);
|
|
|
|
// Provides an optional external estimate of the audio buffer delay.
|
|
virtual void SetAudioBufferDelay(int delay_ms) = 0;
|
|
|
|
// Returns whether an external delay estimate has been reported via
|
|
// SetAudioBufferDelay.
|
|
virtual bool HasReceivedBufferDelay() = 0;
|
|
};
|
|
|
|
} // namespace webrtc
|
|
|
|
#endif // MODULES_AUDIO_PROCESSING_AEC3_RENDER_DELAY_BUFFER_H_
|