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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.
42 lines
1.4 KiB
C++
42 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 <stddef.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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