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- /*
- * 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_CONTROLLER_H_
- #define MODULES_AUDIO_PROCESSING_AEC3_RENDER_DELAY_CONTROLLER_H_
- #include "absl/types/optional.h"
- #include "api/array_view.h"
- #include "api/audio/echo_canceller3_config.h"
- #include "modules/audio_processing/aec3/delay_estimate.h"
- #include "modules/audio_processing/aec3/downsampled_render_buffer.h"
- #include "modules/audio_processing/aec3/render_delay_buffer.h"
- #include "modules/audio_processing/logging/apm_data_dumper.h"
- namespace webrtc {
- // Class for aligning the render and capture signal using a RenderDelayBuffer.
- class RenderDelayController {
- public:
- static RenderDelayController* Create(const EchoCanceller3Config& config,
- int sample_rate_hz,
- size_t num_capture_channels);
- virtual ~RenderDelayController() = default;
- // Resets the delay controller. If the delay confidence is reset, the reset
- // behavior is as if the call is restarted.
- virtual void Reset(bool reset_delay_confidence) = 0;
- // Logs a render call.
- virtual void LogRenderCall() = 0;
- // Aligns the render buffer content with the capture signal.
- virtual absl::optional<DelayEstimate> GetDelay(
- const DownsampledRenderBuffer& render_buffer,
- size_t render_delay_buffer_delay,
- const std::vector<std::vector<float>>& capture) = 0;
- // Returns true if clockdrift has been detected.
- virtual bool HasClockdrift() const = 0;
- };
- } // namespace webrtc
- #endif // MODULES_AUDIO_PROCESSING_AEC3_RENDER_DELAY_CONTROLLER_H_
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