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Removed the file echo_cancellation_internal.h and moved

Browse Source 
the file content to echo_cancellation.h.

The purpose of this CL is to simplify upcoming AEC algorithm
changes.

The changes should be bitexact.

BUG=webrtc:5298, webrtc:5201

Review-Url: https://codereview.webrtc.org/1947743004
Cr-Commit-Position: refs/heads/master@{#12638}
This commit is contained in:
peah 2016-05-05 13:34:29 -07:00 committed by Commit bot
parent cf5b37cc46
commit 44c8a373a5
6 changed files with 54 additions and 77 deletions
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1
webrtc/modules/audio_processing/BUILD.gn
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@ -31,7 +31,6 @@ source_set("audio_processing") {
"aec/aec_resampler.h",
"aec/echo_cancellation.cc",
"aec/echo_cancellation.h",
"aec/echo_cancellation_internal.h",
"aecm/aecm_core.cc",
"aecm/aecm_core.h",
"aecm/echo_control_mobile.cc",

1
webrtc/modules/audio_processing/aec/echo_cancellation.cc
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@ -23,7 +23,6 @@ extern "C" {
}
#include "webrtc/modules/audio_processing/aec/aec_core.h"
#include "webrtc/modules/audio_processing/aec/aec_resampler.h"
#include "webrtc/modules/audio_processing/aec/echo_cancellation_internal.h"
#include "webrtc/modules/audio_processing/logging/apm_data_dumper.h"
#include "webrtc/typedefs.h"

54
webrtc/modules/audio_processing/aec/echo_cancellation.h
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@ -11,8 +11,14 @@
#ifndef WEBRTC_MODULES_AUDIO_PROCESSING_AEC_ECHO_CANCELLATION_H_
#define WEBRTC_MODULES_AUDIO_PROCESSING_AEC_ECHO_CANCELLATION_H_
#include <memory>
#include <stddef.h>
extern "C" {
#include "webrtc/common_audio/ring_buffer.h"
}
#include "webrtc/modules/audio_processing/aec/aec_core.h"
#include "webrtc/typedefs.h"
namespace webrtc {
@ -56,6 +62,54 @@ typedef struct {
struct AecCore;
class ApmDataDumper;
typedef struct Aec {
std::unique_ptr<ApmDataDumper> data_dumper;
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;
// 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 farend_started;
// Aec instance counter.
static int instance_count;
AecCore* aec;
} Aec;
/*
* Allocates the memory needed by the AEC. The memory needs to be initialized
* separately using the WebRtcAec_Init() function. Returns a pointer to the

73
webrtc/modules/audio_processing/aec/echo_cancellation_internal.h
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@ -1,73 +0,0 @@
/*
* 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 <memory>
extern "C" {
#include "webrtc/common_audio/ring_buffer.h"
}
#include "webrtc/modules/audio_processing/aec/aec_core.h"
namespace webrtc {
class ApmDataDumper;
typedef struct Aec {
std::unique_ptr<ApmDataDumper> data_dumper;
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;
// 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 farend_started;
// Aec instance counter.
static int instance_count;
AecCore* aec;
} Aec;
} // namespace webrtc
#endif // WEBRTC_MODULES_AUDIO_PROCESSING_AEC_ECHO_CANCELLATION_INTERNAL_H_

1
webrtc/modules/audio_processing/aec/system_delay_unittest.cc
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@ -10,7 +10,6 @@
#include "testing/gtest/include/gtest/gtest.h"
#include "webrtc/modules/audio_processing/aec/aec_core.h"
#include "webrtc/modules/audio_processing/aec/echo_cancellation_internal.h"
#include "webrtc/modules/audio_processing/aec/echo_cancellation.h"
#include "webrtc/typedefs.h"
namespace webrtc {

1
webrtc/modules/audio_processing/audio_processing.gypi
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@ -41,7 +41,6 @@
'aec/aec_resampler.cc',
'aec/aec_resampler.h',
'aec/echo_cancellation.cc',
'aec/echo_cancellation_internal.h',
'aec/echo_cancellation.h',
'aecm/aecm_core.cc',
'aecm/aecm_core.h',