mirror of
https://github.com/mollyim/webrtc.git
synced 2025-05-16 15:20:42 +01:00
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}
79 lines
3 KiB
C++
79 lines
3 KiB
C++
/*
|
|
* Copyright (c) 2016 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_BLOCK_PROCESSOR_H_
|
|
#define MODULES_AUDIO_PROCESSING_AEC3_BLOCK_PROCESSOR_H_
|
|
|
|
#include <memory>
|
|
#include <vector>
|
|
|
|
#include "modules/audio_processing/aec3/echo_remover.h"
|
|
#include "modules/audio_processing/aec3/render_delay_buffer.h"
|
|
#include "modules/audio_processing/aec3/render_delay_controller.h"
|
|
|
|
namespace webrtc {
|
|
|
|
// Class for performing echo cancellation on 64 sample blocks of audio data.
|
|
class BlockProcessor {
|
|
public:
|
|
// Create a block processor with the legacy render buffering.
|
|
static BlockProcessor* Create(const EchoCanceller3Config& config,
|
|
int sample_rate_hz);
|
|
// Create a block processor with the new render buffering.
|
|
static BlockProcessor* Create2(const EchoCanceller3Config& config,
|
|
int sample_rate_hz);
|
|
// Only used for testing purposes.
|
|
static BlockProcessor* Create(
|
|
const EchoCanceller3Config& config,
|
|
int sample_rate_hz,
|
|
std::unique_ptr<RenderDelayBuffer> render_buffer);
|
|
static BlockProcessor* Create2(
|
|
const EchoCanceller3Config& config,
|
|
int sample_rate_hz,
|
|
std::unique_ptr<RenderDelayBuffer> render_buffer);
|
|
static BlockProcessor* Create(
|
|
const EchoCanceller3Config& config,
|
|
int sample_rate_hz,
|
|
std::unique_ptr<RenderDelayBuffer> render_buffer,
|
|
std::unique_ptr<RenderDelayController> delay_controller,
|
|
std::unique_ptr<EchoRemover> echo_remover);
|
|
static BlockProcessor* Create2(
|
|
const EchoCanceller3Config& config,
|
|
int sample_rate_hz,
|
|
std::unique_ptr<RenderDelayBuffer> render_buffer,
|
|
std::unique_ptr<RenderDelayController> delay_controller,
|
|
std::unique_ptr<EchoRemover> echo_remover);
|
|
|
|
virtual ~BlockProcessor() = default;
|
|
|
|
// Get current metrics.
|
|
virtual void GetMetrics(EchoControl::Metrics* metrics) const = 0;
|
|
|
|
// Provides an optional external estimate of the audio buffer delay.
|
|
virtual void SetAudioBufferDelay(size_t delay_ms) = 0;
|
|
|
|
// Processes a block of capture data.
|
|
virtual void ProcessCapture(
|
|
bool echo_path_gain_change,
|
|
bool capture_signal_saturation,
|
|
std::vector<std::vector<float>>* capture_block) = 0;
|
|
|
|
// Buffers a block of render data supplied by a FrameBlocker object.
|
|
virtual void BufferRender(
|
|
const std::vector<std::vector<float>>& render_block) = 0;
|
|
|
|
// Reports whether echo leakage has been detected in the echo canceller
|
|
// output.
|
|
virtual void UpdateEchoLeakageStatus(bool leakage_detected) = 0;
|
|
};
|
|
|
|
} // namespace webrtc
|
|
|
|
#endif // MODULES_AUDIO_PROCESSING_AEC3_BLOCK_PROCESSOR_H_
|