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Diffstat (limited to 'third_party/webrtc/src/webrtc/modules/audio_processing/aec/aec_resampler.h')
-rw-r--r-- | third_party/webrtc/src/webrtc/modules/audio_processing/aec/aec_resampler.h | 39 |
1 files changed, 39 insertions, 0 deletions
diff --git a/third_party/webrtc/src/webrtc/modules/audio_processing/aec/aec_resampler.h b/third_party/webrtc/src/webrtc/modules/audio_processing/aec/aec_resampler.h new file mode 100644 index 00000000..a5002c15 --- /dev/null +++ b/third_party/webrtc/src/webrtc/modules/audio_processing/aec/aec_resampler.h @@ -0,0 +1,39 @@ +/* + * 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_ |