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https://github.com/danog/libtgvoip.git
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5caaaafa42
I'm now using the entire audio processing module from WebRTC as opposed to individual DSP algorithms pulled from there before. Seems to work better this way.
51 lines
2.0 KiB
C++
51 lines
2.0 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. If the delay confidence is reset, the reset
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// behavior is as if the call is restarted.
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virtual void Reset(bool reset_delay_confidence) = 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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