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92ea95e34a
In https://webrtc-review.googlesource.com/c/src/+/1560 we moved WebRTC from src/webrtc to src/ (in order to preserve an healthy git history). This CL takes care of fixing header guards, #include paths, etc... NOPRESUBMIT=true NOTREECHECKS=true NOTRY=true TBR=tommi@webrtc.org Bug: chromium:611808 Change-Id: Iea91618212bee0af16aa3f05071eab8f93706578 Reviewed-on: https://webrtc-review.googlesource.com/1561 Reviewed-by: Mirko Bonadei <mbonadei@webrtc.org> Reviewed-by: Henrik Kjellander <kjellander@webrtc.org> Commit-Queue: Mirko Bonadei <mbonadei@webrtc.org> Cr-Commit-Position: refs/heads/master@{#19846}
39 lines
1.4 KiB
C++
39 lines
1.4 KiB
C++
/*
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* Copyright (c) 2012 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_AEC_AEC_RESAMPLER_H_
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#define MODULES_AUDIO_PROCESSING_AEC_AEC_RESAMPLER_H_
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#include "modules/audio_processing/aec/aec_core.h"
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namespace webrtc {
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enum { kResamplingDelay = 1 };
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enum { kResamplerBufferSize = FRAME_LEN * 4 };
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// Unless otherwise specified, functions return 0 on success and -1 on error.
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void* WebRtcAec_CreateResampler(); // Returns NULL on error.
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int WebRtcAec_InitResampler(void* resampInst, int deviceSampleRateHz);
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void WebRtcAec_FreeResampler(void* resampInst);
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// Estimates skew from raw measurement.
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int WebRtcAec_GetSkew(void* resampInst, int rawSkew, float* skewEst);
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// Resamples input using linear interpolation.
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void WebRtcAec_ResampleLinear(void* resampInst,
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const float* inspeech,
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size_t size,
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float skew,
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float* outspeech,
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size_t* size_out);
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} // namespace webrtc
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#endif // MODULES_AUDIO_PROCESSING_AEC_AEC_RESAMPLER_H_
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