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Diffstat (limited to 'third_party/webrtc/src/webrtc/modules/audio_processing/aec/echo_cancellation_internal.h')
-rw-r--r-- | third_party/webrtc/src/webrtc/modules/audio_processing/aec/echo_cancellation_internal.h | 67 |
1 files changed, 67 insertions, 0 deletions
diff --git a/third_party/webrtc/src/webrtc/modules/audio_processing/aec/echo_cancellation_internal.h b/third_party/webrtc/src/webrtc/modules/audio_processing/aec/echo_cancellation_internal.h new file mode 100644 index 00000000..95a6cf33 --- /dev/null +++ b/third_party/webrtc/src/webrtc/modules/audio_processing/aec/echo_cancellation_internal.h @@ -0,0 +1,67 @@ +/* + * 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_ECHO_CANCELLATION_INTERNAL_H_ +#define WEBRTC_MODULES_AUDIO_PROCESSING_AEC_ECHO_CANCELLATION_INTERNAL_H_ + +#include "webrtc/common_audio/ring_buffer.h" +#include "webrtc/modules/audio_processing/aec/aec_core.h" + +typedef struct { + int delayCtr; + int sampFreq; + int splitSampFreq; + int scSampFreq; + float sampFactor; // scSampRate / sampFreq + short skewMode; + int bufSizeStart; + int knownDelay; + int rate_factor; + + short initFlag; // indicates if AEC has been initialized + + // Variables used for averaging far end buffer size + short counter; + int sum; + short firstVal; + short checkBufSizeCtr; + + // Variables used for delay shifts + short msInSndCardBuf; + short filtDelay; // Filtered delay estimate. + int timeForDelayChange; + int startup_phase; + int checkBuffSize; + short lastDelayDiff; + +#ifdef WEBRTC_AEC_DEBUG_DUMP + FILE* bufFile; + FILE* delayFile; + FILE* skewFile; +#endif + + // Structures + void* resampler; + + int skewFrCtr; + int resample; // if the skew is small enough we don't resample + int highSkewCtr; + float skew; + + RingBuffer* far_pre_buf; // Time domain far-end pre-buffer. + + int lastError; + + int farend_started; + + AecCore* aec; +} Aec; + +#endif // WEBRTC_MODULES_AUDIO_PROCESSING_AEC_ECHO_CANCELLATION_INTERNAL_H_ |