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| 1 /* | |
| 2 * Copyright (c) 2016 The WebRTC project authors. All Rights Reserved. | |
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hlundin-webrtc
2017/01/18 13:08:50
2017
peah-webrtc
2017/01/19 15:33:07
Done.
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| 3 * | |
| 4 * Use of this source code is governed by a BSD-style license | |
| 5 * that can be found in the LICENSE file in the root of the source | |
| 6 * tree. An additional intellectual property rights grant can be found | |
| 7 * in the file PATENTS. All contributing project authors may | |
| 8 * be found in the AUTHORS file in the root of the source tree. | |
| 9 */ | |
| 10 | |
| 11 #ifndef WEBRTC_MODULES_AUDIO_PROCESSING_AEC3_RENDER_DELAY_CONTROLLER_H_ | |
| 12 #define WEBRTC_MODULES_AUDIO_PROCESSING_AEC3_RENDER_DELAY_CONTROLLER_H_ | |
| 13 | |
| 14 #include "webrtc/base/array_view.h" | |
| 15 #include "webrtc/base/optional.h" | |
| 16 #include "webrtc/modules/audio_processing/aec3/render_delay_buffer.h" | |
| 17 #include "webrtc/modules/audio_processing/logging/apm_data_dumper.h" | |
| 18 | |
| 19 namespace webrtc { | |
| 20 | |
| 21 // Class for aligning the render and capture signal using an RenderDelayBuffer. | |
| 22 class RenderDelayController { | |
| 23 public: | |
| 24 static RenderDelayController* Create( | |
| 25 ApmDataDumper* data_dumper, | |
| 26 int sample_rate_hz, | |
| 27 const RenderDelayBuffer& render_delay_buffer); | |
| 28 virtual ~RenderDelayController() = default; | |
| 29 | |
| 30 // Aligns the render buffer content with the capture signal. | |
| 31 virtual size_t SelectDelay(rtc::ArrayView<const float> capture) = 0; | |
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hlundin-webrtc
2017/01/18 13:08:50
Select sounds a bit like the caller can select a d
peah-webrtc
2017/01/19 15:33:07
I agree. I changed it to GetDelay. I like that bet
hlundin-webrtc
2017/01/20 09:31:44
Acknowledged.
aleloi
2017/01/20 14:45:37
GetDelay sounds fine.
peah-webrtc
2017/01/23 14:16:39
Acknowledged.
peah-webrtc
2017/01/23 14:16:39
Acknowledged.
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| 32 | |
| 33 // Analyses the render signal and returns false if there is a buffer overrun. | |
| 34 virtual bool AnalyzeRender(rtc::ArrayView<const float> render) = 0; | |
| 35 | |
| 36 // Returns an approximate value for the headroom in the buffer alignment | |
| 37 // expressed in samples. | |
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hlundin-webrtc
2017/01/18 13:08:50
When does it return empty?
peah-webrtc
2017/01/19 15:33:07
What it does is to return the alignment headroom t
hlundin-webrtc
2017/01/20 09:31:44
I think some of this explanation should go into th
peah-webrtc
2017/01/23 14:16:39
I changed the name of the method to achieve part o
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| 38 virtual rtc::Optional<size_t> AlignmentHeadroom() const = 0; | |
| 39 }; | |
| 40 } // namespace webrtc | |
| 41 | |
| 42 #endif // WEBRTC_MODULES_AUDIO_PROCESSING_AEC3_RENDER_DELAY_CONTROLLER_H_ | |
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