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Removed versions in module/audio_processing and common_audio/vad.

Browse Source 
Affected vad_unittest only.
In addition changed to correct header guards.
Review URL: https://webrtc-codereview.appspot.com/367019

git-svn-id: http://webrtc.googlecode.com/svn/trunk@1567 4adac7df-926f-26a2-2b94-8c16560cd09d
This commit is contained in:
bjornv@webrtc.org
2012-01-30 09:39:08 +00:00
parent 2fd1e1e40d
commit 0c6f931420
28 changed files with 62 additions and 341 deletions
Download Patch File Download Diff File Expand all files Collapse all files

20
src/common_audio/vad/include/webrtc_vad.h
View File

@ -13,8 +13,8 @@
* This header file includes the VAD API calls. Specific function calls are given below.
*/
#ifndef WEBRTC_VAD_WEBRTC_VAD_H_
#define WEBRTC_VAD_WEBRTC_VAD_H_
#ifndef WEBRTC_COMMON_AUDIO_VAD_INCLUDE_WEBRTC_VAD_H_
#define WEBRTC_COMMON_AUDIO_VAD_INCLUDE_WEBRTC_VAD_H_
#include "typedefs.h"
@ -25,20 +25,6 @@ extern "C"
{
#endif
/****************************************************************************
* WebRtcVad_get_version(...)
*
* This function returns the version number of the code.
*
* Output:
* - version : Pointer to a buffer where the version info will
* be stored.
* Input:
* - size_bytes : Size of the buffer.
*
*/
WebRtc_Word16 WebRtcVad_get_version(char *version, size_t size_bytes);
/****************************************************************************
* WebRtcVad_AssignSize(...)
*
@ -156,4 +142,4 @@ WebRtc_Word16 WebRtcVad_Process(VadInst *vad_inst,
}
#endif
#endif // WEBRTC_VAD_WEBRTC_VAD_H_
#endif // WEBRTC_COMMON_AUDIO_VAD_INCLUDE_WEBRTC_VAD_H_

8
src/common_audio/vad/vad_unittest.cc
View File

@ -1,5 +1,5 @@
/*
* Copyright (c) 2011 The WebRTC project authors. All Rights Reserved.
* 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
@ -61,12 +61,6 @@ TEST_F(VadTest, ApiTest) {
speech[i] = (i * i);
}
// WebRtcVad_get_version() tests
char version[32];
EXPECT_EQ(-1, WebRtcVad_get_version(NULL, sizeof(version)));
EXPECT_EQ(-1, WebRtcVad_get_version(version, 1));
EXPECT_EQ(0, WebRtcVad_get_version(version, sizeof(version)));
// Null instance tests
EXPECT_EQ(-1, WebRtcVad_Create(NULL));
EXPECT_EQ(-1, WebRtcVad_Init(NULL));

18
src/common_audio/vad/webrtc_vad.c
View File

@ -22,24 +22,6 @@
static const int kInitCheck = 42;
WebRtc_Word16 WebRtcVad_get_version(char *version, size_t size_bytes)
{
const char my_version[] = "VAD 1.2.0";
if (version == NULL)
{
return -1;
}
if (size_bytes < sizeof(my_version))
{
return -1;
}
memcpy(version, my_version, sizeof(my_version));
return 0;
}
WebRtc_Word16 WebRtcVad_AssignSize(int *size_in_bytes)
{
*size_in_bytes = sizeof(VadInstT) * 2 / sizeof(WebRtc_Word16);

19
src/modules/audio_processing/aec/echo_cancellation.c
View File

@ -1,5 +1,5 @@
/*
* Copyright (c) 2011 The WebRTC project authors. All Rights Reserved.
* 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
@ -867,23 +867,6 @@ int WebRtcAec_GetDelayMetrics(void* handle, int* median, int* std) {
return 0;
}
WebRtc_Word32 WebRtcAec_get_version(WebRtc_Word8 *versionStr, WebRtc_Word16 len)
{
const char version[] = "AEC 2.5.0";
const short versionLen = (short)strlen(version) + 1; // +1 for null-termination
if (versionStr == NULL) {
return -1;
}
if (versionLen > len) {
return -1;
}
strncpy(versionStr, version, versionLen);
return 0;
}
WebRtc_Word32 WebRtcAec_get_error_code(void *aecInst)
{
aecpc_t *aecpc = aecInst;

22
src/modules/audio_processing/aec/include/echo_cancellation.h
View File

@ -8,8 +8,8 @@
* be found in the AUTHORS file in the root of the source tree.
*/
#ifndef WEBRTC_MODULES_AUDIO_PROCESSING_AEC_MAIN_INTERFACE_ECHO_CANCELLATION_H_
#define WEBRTC_MODULES_AUDIO_PROCESSING_AEC_MAIN_INTERFACE_ECHO_CANCELLATION_H_
#ifndef WEBRTC_MODULES_AUDIO_PROCESSING_AEC_INCLUDE_ECHO_CANCELLATION_H_
#define WEBRTC_MODULES_AUDIO_PROCESSING_AEC_INCLUDE_ECHO_CANCELLATION_H_
#include "typedefs.h"
@ -256,23 +256,7 @@ int WebRtcAec_GetDelayMetrics(void* handle, int* median, int* std);
*/
WebRtc_Word32 WebRtcAec_get_error_code(void *aecInst);
/*
* Gets a version string.
*
* Inputs Description
* -------------------------------------------------------------------
* char *versionStr Pointer to a string array
* WebRtc_Word16 len The maximum length of the string
*
* Outputs Description
* -------------------------------------------------------------------
* WebRtc_Word8 *versionStr Pointer to a string array
* WebRtc_Word32 return 0: OK
* -1: error
*/
WebRtc_Word32 WebRtcAec_get_version(WebRtc_Word8 *versionStr, WebRtc_Word16 len);
#ifdef __cplusplus
}
#endif
#endif /* WEBRTC_MODULES_AUDIO_PROCESSING_AEC_MAIN_INTERFACE_ECHO_CANCELLATION_H_ */
#endif // WEBRTC_MODULES_AUDIO_PROCESSING_AEC_INCLUDE_ECHO_CANCELLATION_H_

21
src/modules/audio_processing/aecm/echo_control_mobile.c
View File

@ -1,5 +1,5 @@
/*
* Copyright (c) 2011 The WebRTC project authors. All Rights Reserved.
* 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
@ -702,25 +702,6 @@ size_t WebRtcAecm_echo_path_size_bytes()
return (PART_LEN1 * sizeof(WebRtc_Word16));
}
WebRtc_Word32 WebRtcAecm_get_version(WebRtc_Word8 *versionStr, WebRtc_Word16 len)
{
const char version[] = "AECM 1.2.0";
const short versionLen = (short)strlen(version) + 1; // +1 for null-termination
if (versionStr == NULL)
{
return -1;
}
if (versionLen > len)
{
return -1;
}
strncpy(versionStr, version, versionLen);
return 0;
}
WebRtc_Word32 WebRtcAecm_get_error_code(void *aecmInst)
{
aecmob_t *aecm = aecmInst;

23
src/modules/audio_processing/aecm/include/echo_control_mobile.h
View File

@ -8,8 +8,8 @@
* be found in the AUTHORS file in the root of the source tree.
*/
#ifndef WEBRTC_MODULES_AUDIO_PROCESSING_AECM_MAIN_INTERFACE_ECHO_CONTROL_MOBILE_H_
#define WEBRTC_MODULES_AUDIO_PROCESSING_AECM_MAIN_INTERFACE_ECHO_CONTROL_MOBILE_H_
#ifndef WEBRTC_MODULES_AUDIO_PROCESSING_AECM_INCLUDE_ECHO_CONTROL_MOBILE_H_
#define WEBRTC_MODULES_AUDIO_PROCESSING_AECM_INCLUDE_ECHO_CONTROL_MOBILE_H_
#include "typedefs.h"
@ -227,24 +227,7 @@ size_t WebRtcAecm_echo_path_size_bytes();
*/
WebRtc_Word32 WebRtcAecm_get_error_code(void *aecmInst);
/*
* Gets a version string
*
* Inputs Description
* -------------------------------------------------------------------
* char *versionStr Pointer to a string array
* WebRtc_Word16 len The maximum length of the string
*
* Outputs Description
* -------------------------------------------------------------------
* WebRtc_Word8 *versionStr Pointer to a string array
* WebRtc_Word32 return 0: OK
* -1: error
*/
WebRtc_Word32 WebRtcAecm_get_version(WebRtc_Word8 *versionStr,
WebRtc_Word16 len);
#ifdef __cplusplus
}
#endif
#endif /* WEBRTC_MODULES_AUDIO_PROCESSING_AECM_MAIN_INTERFACE_ECHO_CONTROL_MOBILE_H_ */
#endif // WEBRTC_MODULES_AUDIO_PROCESSING_AECM_INCLUDE_ECHO_CONTROL_MOBILE_H_

21
src/modules/audio_processing/agc/analog_agc.c
View File

@ -1,5 +1,5 @@
/*
* Copyright (c) 2011 The WebRTC project authors. All Rights Reserved.
* 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
@ -1692,22 +1692,3 @@ int WebRtcAgc_Init(void *agcInst, WebRtc_Word32 minLevel, WebRtc_Word32 maxLevel
return 0;
}
}
int WebRtcAgc_Version(WebRtc_Word8 *versionStr, WebRtc_Word16 length)
{
const WebRtc_Word8 version[] = "AGC 1.7.0";
const WebRtc_Word16 versionLen = (WebRtc_Word16)strlen(version) + 1;
if (versionStr == NULL)
{
return -1;
}
if (versionLen > length)
{
return -1;
}
strncpy(versionStr, version, versionLen);
return 0;
}

20
src/modules/audio_processing/agc/include/gain_control.h
View File

@ -8,8 +8,8 @@
* be found in the AUTHORS file in the root of the source tree.
*/
#ifndef WEBRTC_MODULES_AUDIO_PROCESSING_AGC_MAIN_INTERFACE_GAIN_CONTROL_H_
#define WEBRTC_MODULES_AUDIO_PROCESSING_AGC_MAIN_INTERFACE_GAIN_CONTROL_H_
#ifndef WEBRTC_MODULES_AUDIO_PROCESSING_AGC_INCLUDE_GAIN_CONTROL_H_
#define WEBRTC_MODULES_AUDIO_PROCESSING_AGC_INCLUDE_GAIN_CONTROL_H_
#include "typedefs.h"
@ -252,22 +252,8 @@ int WebRtcAgc_Init(void *agcInst,
WebRtc_Word16 agcMode,
WebRtc_UWord32 fs);
/*
* This function returns a text string containing the version.
*
* Input:
* - length : Length of the char array pointed to by version
* Output:
* - version : Pointer to a char array of to which the version
* : string will be copied.
*
* Return value : 0 - OK
* -1 - Error
*/
int WebRtcAgc_Version(WebRtc_Word8 *versionStr, WebRtc_Word16 length);
#if defined(__cplusplus)
}
#endif
#endif // WEBRTC_MODULES_AUDIO_PROCESSING_AGC_MAIN_INTERFACE_GAIN_CONTROL_H_
#endif // WEBRTC_MODULES_AUDIO_PROCESSING_AGC_INCLUDE_GAIN_CONTROL_H_

11
src/modules/audio_processing/echo_cancellation_impl.cc
View File

@ -1,5 +1,5 @@
/*
* Copyright (c) 2011 The WebRTC project authors. All Rights Reserved.
* 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
@ -328,15 +328,6 @@ int EchoCancellationImpl::Initialize() {
return apm_->kNoError;
}
int EchoCancellationImpl::get_version(char* version,
int version_len_bytes) const {
if (WebRtcAec_get_version(version, version_len_bytes) != 0) {
return apm_->kBadParameterError;
}
return apm_->kNoError;
}
void* EchoCancellationImpl::CreateHandle() const {
Handle* handle = NULL;
if (WebRtcAec_Create(&handle) != apm_->kNoError) {

9
src/modules/audio_processing/echo_cancellation_impl.h
View File

@ -1,5 +1,5 @@
/*
* Copyright (c) 2011 The WebRTC project authors. All Rights Reserved.
* 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
@ -8,8 +8,8 @@
* be found in the AUTHORS file in the root of the source tree.
*/
#ifndef WEBRTC_MODULES_AUDIO_PROCESSING_MAIN_SOURCE_ECHO_CANCELLATION_IMPL_H_
#define WEBRTC_MODULES_AUDIO_PROCESSING_MAIN_SOURCE_ECHO_CANCELLATION_IMPL_H_
#ifndef WEBRTC_MODULES_AUDIO_PROCESSING_ECHO_CANCELLATION_IMPL_H_
#define WEBRTC_MODULES_AUDIO_PROCESSING_ECHO_CANCELLATION_IMPL_H_
#include "audio_processing.h"
#include "processing_component.h"
@ -34,7 +34,6 @@ class EchoCancellationImpl : public EchoCancellation,
// ProcessingComponent implementation.
virtual int Initialize();
virtual int get_version(char* version, int version_len_bytes) const;
private:
// EchoCancellation implementation.
@ -73,4 +72,4 @@ class EchoCancellationImpl : public EchoCancellation,
};
} // namespace webrtc
#endif // WEBRTC_MODULES_AUDIO_PROCESSING_MAIN_SOURCE_ECHO_CANCELLATION_IMPL_H_
#endif // WEBRTC_MODULES_AUDIO_PROCESSING_ECHO_CANCELLATION_IMPL_H_

11
src/modules/audio_processing/echo_control_mobile_impl.cc
View File

@ -1,5 +1,5 @@
/*
* Copyright (c) 2011 The WebRTC project authors. All Rights Reserved.
* 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
@ -248,15 +248,6 @@ int EchoControlMobileImpl::Initialize() {
return ProcessingComponent::Initialize();
}
int EchoControlMobileImpl::get_version(char* version,
int version_len_bytes) const {
if (WebRtcAecm_get_version(version, version_len_bytes) != 0) {
return apm_->kBadParameterError;
}
return apm_->kNoError;
}
void* EchoControlMobileImpl::CreateHandle() const {
Handle* handle = NULL;
if (WebRtcAecm_Create(&handle) != apm_->kNoError) {

9
src/modules/audio_processing/echo_control_mobile_impl.h
View File

@ -1,5 +1,5 @@
/*
* Copyright (c) 2011 The WebRTC project authors. All Rights Reserved.
* 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
@ -8,8 +8,8 @@
* be found in the AUTHORS file in the root of the source tree.
*/
#ifndef WEBRTC_MODULES_AUDIO_PROCESSING_MAIN_SOURCE_ECHO_CONTROL_MOBILE_IMPL_H_
#define WEBRTC_MODULES_AUDIO_PROCESSING_MAIN_SOURCE_ECHO_CONTROL_MOBILE_IMPL_H_
#ifndef WEBRTC_MODULES_AUDIO_PROCESSING_ECHO_CONTROL_MOBILE_IMPL_H_
#define WEBRTC_MODULES_AUDIO_PROCESSING_ECHO_CONTROL_MOBILE_IMPL_H_
#include "audio_processing.h"
#include "processing_component.h"
@ -32,7 +32,6 @@ class EchoControlMobileImpl : public EchoControlMobile,
// ProcessingComponent implementation.
virtual int Initialize();
virtual int get_version(char* version, int version_len_bytes) const;
private:
// EchoControlMobile implementation.
@ -59,4 +58,4 @@ class EchoControlMobileImpl : public EchoControlMobile,
};
} // namespace webrtc
#endif // WEBRTC_MODULES_AUDIO_PROCESSING_MAIN_SOURCE_ECHO_CONTROL_MOBILE_IMPL_H_
#endif // WEBRTC_MODULES_AUDIO_PROCESSING_ECHO_CONTROL_MOBILE_IMPL_H_

10
src/modules/audio_processing/gain_control_impl.cc
View File

@ -1,5 +1,5 @@
/*
* Copyright (c) 2011 The WebRTC project authors. All Rights Reserved.
* 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
@ -333,14 +333,6 @@ int GainControlImpl::Initialize() {
return apm_->kNoError;
}
int GainControlImpl::get_version(char* version, int version_len_bytes) const {
if (WebRtcAgc_Version(version, version_len_bytes) != 0) {
return apm_->kBadParameterError;
}
return apm_->kNoError;
}
void* GainControlImpl::CreateHandle() const {
Handle* handle = NULL;
if (WebRtcAgc_Create(&handle) != apm_->kNoError) {

9
src/modules/audio_processing/gain_control_impl.h
View File

@ -1,5 +1,5 @@
/*
* Copyright (c) 2011 The WebRTC project authors. All Rights Reserved.
* 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
@ -8,8 +8,8 @@
* be found in the AUTHORS file in the root of the source tree.
*/
#ifndef WEBRTC_MODULES_AUDIO_PROCESSING_MAIN_SOURCE_GAIN_CONTROL_IMPL_H_
#define WEBRTC_MODULES_AUDIO_PROCESSING_MAIN_SOURCE_GAIN_CONTROL_IMPL_H_
#ifndef WEBRTC_MODULES_AUDIO_PROCESSING_GAIN_CONTROL_IMPL_H_
#define WEBRTC_MODULES_AUDIO_PROCESSING_GAIN_CONTROL_IMPL_H_
#include <vector>
@ -32,7 +32,6 @@ class GainControlImpl : public GainControl,
// ProcessingComponent implementation.
virtual int Initialize();
virtual int get_version(char* version, int version_len_bytes) const;
// GainControl implementation.
virtual bool is_enabled() const;
@ -77,4 +76,4 @@ class GainControlImpl : public GainControl,
};
} // namespace webrtc
#endif // WEBRTC_MODULES_AUDIO_PROCESSING_MAIN_SOURCE_GAIN_CONTROL_IMPL_H_
#endif // WEBRTC_MODULES_AUDIO_PROCESSING_GAIN_CONTROL_IMPL_H_

9
src/modules/audio_processing/high_pass_filter_impl.cc
View File

@ -1,5 +1,5 @@
/*
* Copyright (c) 2011 The WebRTC project authors. All Rights Reserved.
* 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
@ -143,13 +143,6 @@ bool HighPassFilterImpl::is_enabled() const {
return is_component_enabled();
}
int HighPassFilterImpl::get_version(char* version,
int version_len_bytes) const {
// An empty string is used to indicate no version information.
memset(version, 0, version_len_bytes);
return apm_->kNoError;
}
void* HighPassFilterImpl::CreateHandle() const {
return new FilterState;
}

11
src/modules/audio_processing/high_pass_filter_impl.h
View File

@ -1,5 +1,5 @@
/*
* Copyright (c) 2011 The WebRTC project authors. All Rights Reserved.
* 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
@ -8,8 +8,8 @@
* be found in the AUTHORS file in the root of the source tree.
*/
#ifndef WEBRTC_MODULES_AUDIO_PROCESSING_MAIN_SOURCE_HIGH_PASS_FILTER_IMPL_H_
#define WEBRTC_MODULES_AUDIO_PROCESSING_MAIN_SOURCE_HIGH_PASS_FILTER_IMPL_H_
#ifndef WEBRTC_MODULES_AUDIO_PROCESSING_HIGH_PASS_FILTER_IMPL_H_
#define WEBRTC_MODULES_AUDIO_PROCESSING_HIGH_PASS_FILTER_IMPL_H_
#include "audio_processing.h"
#include "processing_component.h"
@ -29,9 +29,6 @@ class HighPassFilterImpl : public HighPassFilter,
// HighPassFilter implementation.
virtual bool is_enabled() const;
// ProcessingComponent implementation.
virtual int get_version(char* version, int version_len_bytes) const;
private:
// HighPassFilter implementation.
virtual int Enable(bool enable);
@ -48,4 +45,4 @@ class HighPassFilterImpl : public HighPassFilter,
};
} // namespace webrtc
#endif // WEBRTC_MODULES_AUDIO_PROCESSING_MAIN_SOURCE_HIGH_PASS_FILTER_IMPL_H_
#endif // WEBRTC_MODULES_AUDIO_PROCESSING_HIGH_PASS_FILTER_IMPL_H_

9
src/modules/audio_processing/level_estimator_impl.cc
View File

@ -1,5 +1,5 @@
/*
* Copyright (c) 2011 The WebRTC project authors. All Rights Reserved.
* 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
@ -130,13 +130,6 @@ int LevelEstimatorImpl::RMS() {
return level->RMS();
}
int LevelEstimatorImpl::get_version(char* version,
int version_len_bytes) const {
// An empty string is used to indicate no version information.
memset(version, 0, version_len_bytes);
return apm_->kNoError;
}
void* LevelEstimatorImpl::CreateHandle() const {
return new Level;
}

11
src/modules/audio_processing/level_estimator_impl.h
View File

@ -1,5 +1,5 @@
/*
* Copyright (c) 2011 The WebRTC project authors. All Rights Reserved.
* 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
@ -8,8 +8,8 @@
* be found in the AUTHORS file in the root of the source tree.
*/
#ifndef WEBRTC_MODULES_AUDIO_PROCESSING_MAIN_SOURCE_LEVEL_ESTIMATOR_IMPL_H_
#define WEBRTC_MODULES_AUDIO_PROCESSING_MAIN_SOURCE_LEVEL_ESTIMATOR_IMPL_H_
#ifndef WEBRTC_MODULES_AUDIO_PROCESSING_LEVEL_ESTIMATOR_IMPL_H_
#define WEBRTC_MODULES_AUDIO_PROCESSING_LEVEL_ESTIMATOR_IMPL_H_
#include "audio_processing.h"
#include "processing_component.h"
@ -29,9 +29,6 @@ class LevelEstimatorImpl : public LevelEstimator,
// LevelEstimator implementation.
virtual bool is_enabled() const;
// ProcessingComponent implementation.
virtual int get_version(char* version, int version_len_bytes) const;
private:
// LevelEstimator implementation.
virtual int Enable(bool enable);
@ -49,4 +46,4 @@ class LevelEstimatorImpl : public LevelEstimator,
};
} // namespace webrtc
#endif // WEBRTC_MODULES_AUDIO_PROCESSING_MAIN_SOURCE_LEVEL_ESTIMATOR_IMPL_H_
#endif // WEBRTC_MODULES_AUDIO_PROCESSING_LEVEL_ESTIMATOR_IMPL_H_

16
src/modules/audio_processing/noise_suppression_impl.cc
View File

@ -1,5 +1,5 @@
/*
* Copyright (c) 2011 The WebRTC project authors. All Rights Reserved.
* 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
@ -110,20 +110,6 @@ NoiseSuppression::Level NoiseSuppressionImpl::level() const {
return level_;
}
int NoiseSuppressionImpl::get_version(char* version,
int version_len_bytes) const {
#if defined(WEBRTC_NS_FLOAT)
if (WebRtcNs_get_version(version, version_len_bytes) != 0)
#elif defined(WEBRTC_NS_FIXED)
if (WebRtcNsx_get_version(version, version_len_bytes) != 0)
#endif
{
return apm_->kBadParameterError;
}
return apm_->kNoError;
}
void* NoiseSuppressionImpl::CreateHandle() const {
Handle* handle = NULL;
#if defined(WEBRTC_NS_FLOAT)

11
src/modules/audio_processing/noise_suppression_impl.h
View File

@ -1,5 +1,5 @@
/*
* Copyright (c) 2011 The WebRTC project authors. All Rights Reserved.
* 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
@ -8,8 +8,8 @@
* be found in the AUTHORS file in the root of the source tree.
*/
#ifndef WEBRTC_MODULES_AUDIO_PROCESSING_MAIN_SOURCE_NOISE_SUPPRESSION_IMPL_H_
#define WEBRTC_MODULES_AUDIO_PROCESSING_MAIN_SOURCE_NOISE_SUPPRESSION_IMPL_H_
#ifndef WEBRTC_MODULES_AUDIO_PROCESSING_NOISE_SUPPRESSION_IMPL_H_
#define WEBRTC_MODULES_AUDIO_PROCESSING_NOISE_SUPPRESSION_IMPL_H_
#include "audio_processing.h"
#include "processing_component.h"
@ -29,9 +29,6 @@ class NoiseSuppressionImpl : public NoiseSuppression,
// NoiseSuppression implementation.
virtual bool is_enabled() const;
// ProcessingComponent implementation.
virtual int get_version(char* version, int version_len_bytes) const;
private:
// NoiseSuppression implementation.
virtual int Enable(bool enable);
@ -51,4 +48,4 @@ class NoiseSuppressionImpl : public NoiseSuppression,
};
} // namespace webrtc
#endif // WEBRTC_MODULES_AUDIO_PROCESSING_MAIN_SOURCE_NOISE_SUPPRESSION_IMPL_H_
#endif // WEBRTC_MODULES_AUDIO_PROCESSING_NOISE_SUPPRESSION_IMPL_H_

20
src/modules/audio_processing/ns/include/noise_suppression.h
View File

@ -8,8 +8,8 @@
* be found in the AUTHORS file in the root of the source tree.
*/
#ifndef WEBRTC_MODULES_AUDIO_PROCESSING_NS_MAIN_INTERFACE_NOISE_SUPPRESSION_H_
#define WEBRTC_MODULES_AUDIO_PROCESSING_NS_MAIN_INTERFACE_NOISE_SUPPRESSION_H_
#ifndef WEBRTC_MODULES_AUDIO_PROCESSING_NS_INCLUDE_NOISE_SUPPRESSION_H_
#define WEBRTC_MODULES_AUDIO_PROCESSING_NS_INCLUDE_NOISE_SUPPRESSION_H_
#include "typedefs.h"
@ -19,20 +19,6 @@ typedef struct NsHandleT NsHandle;
extern "C" {
#endif
/*
* This function returns the version number of the code.
*
* Input:
* - version : Pointer to a character array where the version
* info is stored.
* - length : Length of version.
*
* Return value : 0 - Ok
* -1 - Error (probably length is not sufficient)
*/
int WebRtcNs_get_version(char* version, short length);
/*
* This function creates an instance to the noise reduction structure
*
@ -121,4 +107,4 @@ int WebRtcNs_Process(NsHandle* NS_inst,
}
#endif
#endif // WEBRTC_MODULES_AUDIO_PROCESSING_NS_MAIN_INTERFACE_NOISE_SUPPRESSION_H_
#endif // WEBRTC_MODULES_AUDIO_PROCESSING_NS_INCLUDE_NOISE_SUPPRESSION_H_

20
src/modules/audio_processing/ns/include/noise_suppression_x.h
View File

@ -8,8 +8,8 @@
* be found in the AUTHORS file in the root of the source tree.
*/
#ifndef WEBRTC_MODULES_AUDIO_PROCESSING_NS_MAIN_INTERFACE_NOISE_SUPPRESSION_X_H_
#define WEBRTC_MODULES_AUDIO_PROCESSING_NS_MAIN_INTERFACE_NOISE_SUPPRESSION_X_H_
#ifndef WEBRTC_MODULES_AUDIO_PROCESSING_NS_INCLUDE_NOISE_SUPPRESSION_X_H_
#define WEBRTC_MODULES_AUDIO_PROCESSING_NS_INCLUDE_NOISE_SUPPRESSION_X_H_
#include "typedefs.h"
@ -19,20 +19,6 @@ typedef struct NsxHandleT NsxHandle;
extern "C" {
#endif
/*
* This function returns the version number of the code.
*
* Input:
* - version : Pointer to a character array where the version
* info is stored.
* - length : Length of version.
*
* Return value : 0 - Ok
* -1 - Error (probably length is not sufficient)
*/
int WebRtcNsx_get_version(char* version, short length);
/*
* This function creates an instance to the noise reduction structure
*
@ -120,4 +106,4 @@ int WebRtcNsx_Process(NsxHandle* nsxInst,
}
#endif
#endif // WEBRTC_MODULES_AUDIO_PROCESSING_NS_MAIN_INTERFACE_NOISE_SUPPRESSION_X_H_
#endif // WEBRTC_MODULES_AUDIO_PROCESSING_NS_INCLUDE_NOISE_SUPPRESSION_X_H_

19
src/modules/audio_processing/ns/noise_suppression.c
View File

@ -1,5 +1,5 @@
/*
* Copyright (c) 2011 The WebRTC project authors. All Rights Reserved.
* 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
@ -15,23 +15,6 @@
#include "ns_core.h"
#include "defines.h"
int WebRtcNs_get_version(char* versionStr, short length) {
const char version[] = "NS 2.2.0";
const short versionLen = (short)strlen(version) + 1; // +1: null-termination
if (versionStr == NULL) {
return -1;
}
if (versionLen > length) {
return -1;
}
strncpy(versionStr, version, versionLen);
return 0;
}
int WebRtcNs_Create(NsHandle** NS_inst) {
*NS_inst = (NsHandle*) malloc(sizeof(NSinst_t));
if (*NS_inst != NULL) {

19
src/modules/audio_processing/ns/noise_suppression_x.c
View File

@ -1,5 +1,5 @@
/*
* Copyright (c) 2011 The WebRTC project authors. All Rights Reserved.
* 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
@ -15,23 +15,6 @@
#include "nsx_core.h"
#include "nsx_defines.h"
int WebRtcNsx_get_version(char* versionStr, short length) {
const char version[] = "NS\t3.1.0";
const short versionLen = (short)strlen(version) + 1; // +1: null-termination
if (versionStr == NULL) {
return -1;
}
if (versionLen > length) {
return -1;
}
strncpy(versionStr, version, versionLen);
return 0;
}
int WebRtcNsx_Create(NsxHandle** nsxInst) {
*nsxInst = (NsxHandle*)malloc(sizeof(NsxInst_t));
if (*nsxInst != NULL) {

9
src/modules/audio_processing/processing_component.h
View File

@ -1,5 +1,5 @@
/*
* Copyright (c) 2011 The WebRTC project authors. All Rights Reserved.
* 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
@ -8,8 +8,8 @@
* be found in the AUTHORS file in the root of the source tree.
*/
#ifndef WEBRTC_MODULES_AUDIO_PROCESSING_MAIN_SOURCE_PROCESSING_COMPONENT_H_
#define WEBRTC_MODULES_AUDIO_PROCESSING_MAIN_SOURCE_PROCESSING_COMPONENT_H_
#ifndef WEBRTC_MODULES_AUDIO_PROCESSING_PROCESSING_COMPONENT_H_
#define WEBRTC_MODULES_AUDIO_PROCESSING_PROCESSING_COMPONENT_H_
#include <vector>
@ -25,7 +25,6 @@ class ProcessingComponent {
virtual int Initialize();
virtual int Destroy();
virtual int get_version(char* version, int version_len_bytes) const = 0;
bool is_component_enabled() const;
@ -51,4 +50,4 @@ class ProcessingComponent {
};
} // namespace webrtc
#endif // WEBRTC_MODULES_AUDIO_PROCESSING_MAIN_SOURCE_PROCESSING_COMPONENT_H__
#endif // WEBRTC_MODULES_AUDIO_PROCESSING_PROCESSING_COMPONENT_H__

9
src/modules/audio_processing/voice_detection_impl.cc
View File

@ -157,15 +157,6 @@ int VoiceDetectionImpl::Initialize() {
return apm_->kNoError;
}
int VoiceDetectionImpl::get_version(char* version,
int version_len_bytes) const {
if (WebRtcVad_get_version(version, version_len_bytes) != 0) {
return apm_->kBadParameterError;
}
return apm_->kNoError;
}
void* VoiceDetectionImpl::CreateHandle() const {
Handle* handle = NULL;
if (WebRtcVad_Create(&handle) != apm_->kNoError) {

9
src/modules/audio_processing/voice_detection_impl.h
View File

@ -1,5 +1,5 @@
/*
* Copyright (c) 2011 The WebRTC project authors. All Rights Reserved.
* 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
@ -8,8 +8,8 @@
* be found in the AUTHORS file in the root of the source tree.
*/
#ifndef WEBRTC_MODULES_AUDIO_PROCESSING_MAIN_SOURCE_VOICE_DETECTION_IMPL_H_
#define WEBRTC_MODULES_AUDIO_PROCESSING_MAIN_SOURCE_VOICE_DETECTION_IMPL_H_
#ifndef WEBRTC_MODULES_AUDIO_PROCESSING_VOICE_DETECTION_IMPL_H_
#define WEBRTC_MODULES_AUDIO_PROCESSING_VOICE_DETECTION_IMPL_H_
#include "audio_processing.h"
#include "processing_component.h"
@ -31,7 +31,6 @@ class VoiceDetectionImpl : public VoiceDetection,
// ProcessingComponent implementation.
virtual int Initialize();
virtual int get_version(char* version, int version_len_bytes) const;
private:
// VoiceDetection implementation.
@ -60,4 +59,4 @@ class VoiceDetectionImpl : public VoiceDetection,
};
} // namespace webrtc
#endif // WEBRTC_MODULES_AUDIO_PROCESSING_MAIN_SOURCE_VOICE_DETECTION_IMPL_H_
#endif // WEBRTC_MODULES_AUDIO_PROCESSING_VOICE_DETECTION_IMPL_H_