dce40cf804
use of int16_t/uint16_t. This is the upshot of a recommendation by henrik.lundin and kwiberg on an original small change ( https://webrtc-codereview.appspot.com/42569004/#ps1 ) to stop using int16_t just because values could fit in it, and is similar in nature to a previous "mass change to use size_t more" ( https://webrtc-codereview.appspot.com/23129004/ ) which also needed to be split up for review but to land all at once, since, like adding "const", such changes tend to cause a lot of transitive effects. This was be reviewed and approved in pieces: https://codereview.webrtc.org/1224093003 https://codereview.webrtc.org/1224123002 https://codereview.webrtc.org/1224163002 https://codereview.webrtc.org/1225133003 https://codereview.webrtc.org/1225173002 https://codereview.webrtc.org/1227163003 https://codereview.webrtc.org/1227203003 https://codereview.webrtc.org/1227213002 https://codereview.webrtc.org/1227893002 https://codereview.webrtc.org/1228793004 https://codereview.webrtc.org/1228803003 https://codereview.webrtc.org/1228823002 https://codereview.webrtc.org/1228823003 https://codereview.webrtc.org/1228843002 https://codereview.webrtc.org/1230693002 https://codereview.webrtc.org/1231713002 The change is being landed as TBR to all the folks who reviewed the above. BUG=chromium:81439 TEST=none R=andrew@webrtc.org, pbos@webrtc.org TBR=aluebs, andrew, asapersson, henrika, hlundin, jan.skoglund, kwiberg, minyue, pbos, pthatcher Review URL: https://codereview.webrtc.org/1230503003 . Cr-Commit-Position: refs/heads/master@{#9768}
40 lines
1.4 KiB
C
40 lines
1.4 KiB
C
/*
|
|
* Copyright (c) 2012 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 WEBRTC_MODULES_AUDIO_PROCESSING_AEC_AEC_RESAMPLER_H_
|
|
#define WEBRTC_MODULES_AUDIO_PROCESSING_AEC_AEC_RESAMPLER_H_
|
|
|
|
#include "webrtc/modules/audio_processing/aec/aec_core.h"
|
|
|
|
enum {
|
|
kResamplingDelay = 1
|
|
};
|
|
enum {
|
|
kResamplerBufferSize = FRAME_LEN * 4
|
|
};
|
|
|
|
// Unless otherwise specified, functions return 0 on success and -1 on error.
|
|
void* WebRtcAec_CreateResampler(); // Returns NULL on error.
|
|
int WebRtcAec_InitResampler(void* resampInst, int deviceSampleRateHz);
|
|
void WebRtcAec_FreeResampler(void* resampInst);
|
|
|
|
// Estimates skew from raw measurement.
|
|
int WebRtcAec_GetSkew(void* resampInst, int rawSkew, float* skewEst);
|
|
|
|
// Resamples input using linear interpolation.
|
|
void WebRtcAec_ResampleLinear(void* resampInst,
|
|
const float* inspeech,
|
|
size_t size,
|
|
float skew,
|
|
float* outspeech,
|
|
size_t* size_out);
|
|
|
|
#endif // WEBRTC_MODULES_AUDIO_PROCESSING_AEC_AEC_RESAMPLER_H_
|