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Delete unused code.

Browse Source 
* Unused audio_coding and video_coding test code.
* Obsolete voice_engine android test app.
* Left-over placeholder files for remoteaudiotrack and
  portallocatorfactory.

In addition, change modules.gyp dependency from rtc_base to
rtc_base_approved.

BUG=
R=henrik.lundin@webrtc.org, henrika@webrtc.org, pbos@webrtc.org, tina.legrand@webrtc.org

Review URL: https://codereview.webrtc.org/2065353002 .

Cr-Commit-Position: refs/heads/master@{#13166}
This commit is contained in:
Niels Möller 2016-06-16 15:51:31 +02:00
parent 2d014be554
commit fc3a8ee47b
22 changed files with 1 additions and 3351 deletions
Show Stats Download Patch File Download Diff File Expand all files Collapse all files

13
webrtc/api/portallocatorfactory.cc
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@ -1,13 +0,0 @@
/*
* Copyright 2011 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.
*/
// TODO(deadbeef): Remove this file once chromium build files no longer
// reference it.
#include "webrtc/api/portallocatorfactory.h"

16
webrtc/api/portallocatorfactory.h
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@ -1,16 +0,0 @@
/*
* Copyright 2011 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.
*/
// TODO(deadbeef): Remove this file once chromium build files no longer
// reference it.
#ifndef WEBRTC_API_PORTALLOCATORFACTORY_H_
#define WEBRTC_API_PORTALLOCATORFACTORY_H_
#endif // WEBRTC_API_PORTALLOCATORFACTORY_H_

11
webrtc/api/remoteaudiotrack.cc
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@ -1,11 +0,0 @@
/*
* Copyright 2015 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.
*/
// TODO(tommi): Delete this file when removed from build files in Chromium.

11
webrtc/api/remoteaudiotrack.h
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@ -1,11 +0,0 @@
/*
* Copyright 2015 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.
*/
// TODO(tommi): Delete this file when removed from build files in Chromium.

72
webrtc/modules/audio_coding/codecs/isac/main/test/debugUtility.h
View File

@ -1,72 +0,0 @@
/*
* Copyright (c) 2011 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_CODING_CODECS_ISAC_MAIN_TEST_DEBUGUTILITY_H_
#define WEBRTC_MODULES_AUDIO_CODING_CODECS_ISAC_MAIN_TEST_DEBUGUTILITY_H_
#include <stdio.h>
#include <string.h>
#include "utility.h"
typedef struct
{
FILE* res0to4FilePtr;
FILE* res4to8FilePtr;
FILE* res8to12FilePtr;
FILE* res8to16FilePtr;
FILE* res0to4DecFilePtr;
FILE* res4to8DecFilePtr;
FILE* res8to12DecFilePtr;
FILE* res8to16DecFilePtr;
FILE* in0to4FilePtr;
FILE* in4to8FilePtr;
FILE* in8to12FilePtr;
FILE* in8to16FilePtr;
FILE* out0to4FilePtr;
FILE* out4to8FilePtr;
FILE* out8to12FilePtr;
FILE* out8to16FilePtr;
FILE* fftFilePtr;
FILE* fftDecFilePtr;
FILE* arrivalTime;
float lastArrivalTime;
int prevPacketLost;
int currPacketLost;
int nextPacketLost;
//double residualSignal4kHZ[240];
int packetLossPercent;
int maxPayloadLB;
int maxPayloadUB;
int lbBytes;
int ubBytes;
}debugStruct;
#define PRINT_ENTROPY_INFO(obj) \
do \
{ \
printf("%10u, %u; ", \
obj->bitstr_obj.streamval, obj->bitstr_obj.stream_index); \
} while(0)
int setupDebugStruct(debugStruct* str);
#endif

111
webrtc/modules/audio_coding/codecs/isac/main/test/simpleKenny.c
View File

@ -29,7 +29,6 @@
#include "isac.h"
#include "utility.h"
#include "webrtc/base/format_macros.h"
//#include "commonDefs.h"
/* max number of samples per frame (= 60 ms frame) */
#define MAX_FRAMESAMPLES_SWB 1920
@ -41,13 +40,6 @@
#define FS_SWB 32000
#define FS_WB 16000
//#define CHANGE_OUTPUT_NAME
#ifdef HAVE_DEBUG_INFO
#include "debugUtility.h"
debugStruct debugInfo;
#endif
unsigned long framecnt = 0;
int main(int argc, char* argv[]) {
@ -69,7 +61,6 @@ int main(int argc, char* argv[]) {
int16_t codingMode = 1;
int16_t shortdata[FRAMESAMPLES_SWB_10ms];
int16_t decoded[MAX_FRAMESAMPLES_SWB];
// uint16_t streamdata[1000];
int16_t speechType[1];
int16_t payloadLimit;
int32_t rateLimit;
@ -257,18 +248,6 @@ int main(int argc, char* argv[]) {
}
WebRtcIsac_DecoderInit(ISAC_main_inst);
// {
// int32_t b1, b2;
// FILE* fileID = fopen("GetBNTest.txt", "w");
// b2 = 32100;
// while (b2 <= 52000) {
// WebRtcIsac_Control(ISAC_main_inst, b2, frameSize);
// WebRtcIsac_GetUplinkBw(ISAC_main_inst, &b1);
// fprintf(fileID, "%5d %5d\n", b2, b1);
// b2 += 10;
// }
// }
if (codingMode == 1) {
if (WebRtcIsac_Control(ISAC_main_inst, bottleneck, frameSize) < 0) {
printf("cannot set bottleneck\n");
@ -293,14 +272,6 @@ int main(int argc, char* argv[]) {
}
}
//=====================================
//#ifdef HAVE_DEBUG_INFO
// if(setupDebugStruct(&debugInfo) < 0)
// {
// exit(1);
// }
//#endif
while (endfile == 0) {
fprintf(stderr, " \rframe = %7li", framecnt);
@ -445,13 +416,6 @@ int main(int argc, char* argv[]) {
printf("Rate Limit.................. %d bits/sec \n", rateLimit);
#ifdef WIN32
#ifdef HAVE_DEBUG_INFO
rateLB =
((double)debugInfo.lbBytes * 8. * (sampFreqKHz)) / (double)totalsmpls;
rateUB =
((double)debugInfo.ubBytes * 8. * (sampFreqKHz)) / (double)totalsmpls;
#endif
fprintf(bitrateFile, "%d %10u %d %6.3f %6.3f %6.3f\n",
sampFreqKHz, framecnt, bottleneck, rateLB, rateUB, rate);
fclose(bitrateFile);
@ -465,18 +429,6 @@ int main(int argc, char* argv[]) {
printf("Measured packet-loss........ %0.1f%% \n",
100.0f * (float)lostPacketCntr / (float)packetCntr);
// #ifdef HAVE_DEBUG_INFO
// printf("Measured lower-band bit-rate %0.3f kbps (%.0f%%)\n",
// rateLB, (double)(rateLB) * 100. /(double)(rate));
// printf("Measured upper-band bit-rate %0.3f kbps (%.0f%%)\n",
// rateUB, (double)(rateUB) * 100. /(double)(rate));
//
// printf("Maximum payload lower-band.. %d bytes (%0.3f kbps)\n",
// debugInfo.maxPayloadLB, debugInfo.maxPayloadLB * 8.0 / 0.03);
// printf("Maximum payload upper-band.. %d bytes (%0.3f kbps)\n",
// debugInfo.maxPayloadUB, debugInfo.maxPayloadUB * 8.0 / 0.03);
// #endif
printf("\n");
/* Runtime statistics */
@ -511,68 +463,5 @@ int main(int argc, char* argv[]) {
WebRtcIsac_Free(ISAC_main_inst);
#ifdef CHANGE_OUTPUT_NAME
{
char* p;
char myExt[50];
char bitRateStr[10];
char newOutName[500];
strcpy(newOutName, outname);
myExt[0] = '\0';
p = strchr(newOutName, '.');
if (p != NULL) {
strcpy(myExt, p);
*p = '_';
p++;
*p = '\0';
} else {
strcat(newOutName, "_");
}
sprintf(bitRateStr, "%0.0fkbps", rate);
strcat(newOutName, bitRateStr);
strcat(newOutName, myExt);
rename(outname, newOutName);
}
#endif
exit(0);
}
#ifdef HAVE_DEBUG_INFO
int setupDebugStruct(debugStruct* str) {
str->prevPacketLost = 0;
str->currPacketLost = 0;
OPEN_FILE_WB(str->res0to4FilePtr, "Res0to4.dat");
OPEN_FILE_WB(str->res4to8FilePtr, "Res4to8.dat");
OPEN_FILE_WB(str->res8to12FilePtr, "Res8to12.dat");
OPEN_FILE_WB(str->res8to16FilePtr, "Res8to16.dat");
OPEN_FILE_WB(str->res0to4DecFilePtr, "Res0to4Dec.dat");
OPEN_FILE_WB(str->res4to8DecFilePtr, "Res4to8Dec.dat");
OPEN_FILE_WB(str->res8to12DecFilePtr, "Res8to12Dec.dat");
OPEN_FILE_WB(str->res8to16DecFilePtr, "Res8to16Dec.dat");
OPEN_FILE_WB(str->in0to4FilePtr, "in0to4.dat");
OPEN_FILE_WB(str->in4to8FilePtr, "in4to8.dat");
OPEN_FILE_WB(str->in8to12FilePtr, "in8to12.dat");
OPEN_FILE_WB(str->in8to16FilePtr, "in8to16.dat");
OPEN_FILE_WB(str->out0to4FilePtr, "out0to4.dat");
OPEN_FILE_WB(str->out4to8FilePtr, "out4to8.dat");
OPEN_FILE_WB(str->out8to12FilePtr, "out8to12.dat");
OPEN_FILE_WB(str->out8to16FilePtr, "out8to16.dat");
OPEN_FILE_WB(str->fftFilePtr, "riFFT.dat");
OPEN_FILE_WB(str->fftDecFilePtr, "riFFTDec.dat");
OPEN_FILE_WB(str->arrivalTime, NULL /*"ArivalTime.dat"*/);
str->lastArrivalTime = 0;
str->maxPayloadLB = 0;
str->maxPayloadUB = 0;
str->lbBytes = 0;
str->ubBytes = 0;
return 0;
};
#endif

58
webrtc/modules/audio_coding/test/TimedTrace.cc
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@ -1,58 +0,0 @@
/*
* Copyright (c) 2011 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.
*/
#include "TimedTrace.h"
#include <math.h>
double TimedTrace::_timeEllapsedSec = 0;
FILE* TimedTrace::_timedTraceFile = NULL;
TimedTrace::TimedTrace() {
}
TimedTrace::~TimedTrace() {
if (_timedTraceFile != NULL) {
fclose(_timedTraceFile);
}
_timedTraceFile = NULL;
}
int16_t TimedTrace::SetUp(char* fileName) {
if (_timedTraceFile == NULL) {
_timedTraceFile = fopen(fileName, "w");
}
if (_timedTraceFile == NULL) {
return -1;
}
return 0;
}
void TimedTrace::SetTimeEllapsed(double timeEllapsedSec) {
_timeEllapsedSec = timeEllapsedSec;
}
double TimedTrace::TimeEllapsed() {
return _timeEllapsedSec;
}
void TimedTrace::Tick10Msec() {
_timeEllapsedSec += 0.010;
}
void TimedTrace::TimedLogg(char* message) {
unsigned int minutes = (uint32_t) floor(_timeEllapsedSec / 60.0);
double seconds = _timeEllapsedSec - minutes * 60;
//char myFormat[100] = "%8.2f, %3u:%05.2f: %s\n";
if (_timedTraceFile != NULL) {
fprintf(_timedTraceFile, "%8.2f, %3u:%05.2f: %s\n", _timeEllapsedSec,
minutes, seconds, message);
}
}

36
webrtc/modules/audio_coding/test/TimedTrace.h
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@ -1,36 +0,0 @@
/*
* Copyright (c) 2011 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_CODING_TEST_TIMEDTRACE_H_
#define WEBRTC_MODULES_AUDIO_CODING_TEST_TIMEDTRACE_H_
#include "webrtc/typedefs.h"
#include <stdio.h>
#include <stdlib.h>
class TimedTrace {
public:
TimedTrace();
~TimedTrace();
void SetTimeEllapsed(double myTime);
double TimeEllapsed();
void Tick10Msec();
int16_t SetUp(char* fileName);
void TimedLogg(char* message);
private:
static double _timeEllapsedSec;
static FILE* _timedTraceFile;
};
#endif // WEBRTC_MODULES_AUDIO_CODING_TEST_TIMEDTRACE_H_

3
webrtc/modules/modules.gyp
View File

@ -89,7 +89,6 @@
'audio_coding/test/TestStereo.cc',
'audio_coding/test/TestVADDTX.cc',
'audio_coding/test/Tester.cc',
'audio_coding/test/TimedTrace.cc',
'audio_coding/test/TwoWayCommunication.cc',
'audio_coding/test/iSACTest.cc',
'audio_coding/test/opus_test.cc',
@ -151,7 +150,7 @@
'<(DEPTH)/testing/gmock.gyp:gmock',
'<(DEPTH)/testing/gtest.gyp:gtest',
'<(DEPTH)/third_party/gflags/gflags.gyp:gflags',
'<(webrtc_root)/base/base.gyp:rtc_base',
'<(webrtc_root)/base/base.gyp:rtc_base_approved',
'<(webrtc_root)/common.gyp:webrtc_common',
'<(webrtc_root)/common_audio/common_audio.gyp:common_audio',
'<(webrtc_root)/common_video/common_video.gyp:common_video',

1
webrtc/modules/video_coding/test/receiver_tests.h
View File

@ -19,7 +19,6 @@
#include "webrtc/modules/rtp_rtcp/include/rtp_rtcp.h"
#include "webrtc/modules/video_coding/include/video_coding.h"
#include "webrtc/modules/video_coding/test/test_util.h"
#include "webrtc/modules/video_coding/test/video_source.h"
#include "webrtc/typedefs.h"
class RtpDataCallback : public webrtc::NullRtpData {

85
webrtc/modules/video_coding/test/video_source.h
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@ -1,85 +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_VIDEO_CODING_TEST_VIDEO_SOURCE_H_
#define WEBRTC_MODULES_VIDEO_CODING_TEST_VIDEO_SOURCE_H_
#include <string>
#include "webrtc/common_video/libyuv/include/webrtc_libyuv.h"
#include "webrtc/typedefs.h"
enum VideoSize {
kUndefined,
kSQCIF, // 128*96 = 12 288
kQQVGA, // 160*120 = 19 200
kQCIF, // 176*144 = 25 344
kCGA, // 320*200 = 64 000
kQVGA, // 320*240 = 76 800
kSIF, // 352*240 = 84 480
kWQVGA, // 400*240 = 96 000
kCIF, // 352*288 = 101 376
kW288p, // 512*288 = 147 456 (WCIF)
k448p, // 576*448 = 281 088
kVGA, // 640*480 = 307 200
k432p, // 720*432 = 311 040
kW432p, // 768*432 = 331 776
k4SIF, // 704*480 = 337 920
kW448p, // 768*448 = 344 064
kNTSC, // 720*480 = 345 600
kFW448p, // 800*448 = 358 400
kWVGA, // 800*480 = 384 000
k4CIF, // 704*576 = 405 504
kSVGA, // 800*600 = 480 000
kW544p, // 960*544 = 522 240
kW576p, // 1024*576 = 589 824 (W4CIF)
kHD, // 960*720 = 691 200
kXGA, // 1024*768 = 786 432
kWHD, // 1280*720 = 921 600
kFullHD, // 1440*1080 = 1 555 200
kWFullHD, // 1920*1080 = 2 073 600
kNumberOfVideoSizes
};
class VideoSource {
public:
VideoSource();
VideoSource(std::string fileName,
VideoSize size,
float frameRate,
webrtc::VideoType type = webrtc::kI420);
VideoSource(std::string fileName,
uint16_t width,
uint16_t height,
float frameRate = 30,
webrtc::VideoType type = webrtc::kI420);
std::string GetFileName() const { return _fileName; }
uint16_t GetWidth() const { return _width; }
uint16_t GetHeight() const { return _height; }
webrtc::VideoType GetType() const { return _type; }
float GetFrameRate() const { return _frameRate; }
int GetWidthHeight(VideoSize size);
// Returns the filename with the path (including the leading slash) removed.
std::string GetName() const;
size_t GetFrameLength() const;
private:
std::string _fileName;
uint16_t _width;
uint16_t _height;
webrtc::VideoType _type;
float _frameRate;
};
#endif // WEBRTC_MODULES_VIDEO_CODING_TEST_VIDEO_SOURCE_H_

7
webrtc/voice_engine/test/android/android_test/.classpath
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@ -1,7 +0,0 @@
<?xml version="1.0" encoding="UTF-8"?>
<classpath>
<classpathentry kind="con" path="com.android.ide.eclipse.adt.ANDROID_FRAMEWORK"/>
<classpathentry kind="src" path="src"/>
<classpathentry kind="src" path="gen"/>
<classpathentry kind="output" path="bin"/>
</classpath>

33
webrtc/voice_engine/test/android/android_test/.project
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@ -1,33 +0,0 @@
<?xml version="1.0" encoding="UTF-8"?>
<projectDescription>
<name>AndroidTest</name>
<comment></comment>
<projects>
</projects>
<buildSpec>
<buildCommand>
<name>com.android.ide.eclipse.adt.ResourceManagerBuilder</name>
<arguments>
</arguments>
</buildCommand>
<buildCommand>
<name>com.android.ide.eclipse.adt.PreCompilerBuilder</name>
<arguments>
</arguments>
</buildCommand>
<buildCommand>
<name>org.eclipse.jdt.core.javabuilder</name>
<arguments>
</arguments>
</buildCommand>
<buildCommand>
<name>com.android.ide.eclipse.adt.ApkBuilder</name>
<arguments>
</arguments>
</buildCommand>
</buildSpec>
<natures>
<nature>com.android.ide.eclipse.adt.AndroidNature</nature>
<nature>org.eclipse.jdt.core.javanature</nature>
</natures>
</projectDescription>

32
webrtc/voice_engine/test/android/android_test/AndroidManifest.xml
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@ -1,32 +0,0 @@
<?xml version="1.0" encoding="utf-8"?>
<!-- Copyright (c) 2011 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. -->
<manifest xmlns:android="http://schemas.android.com/apk/res/android"
android:versionCode="1"
android:versionName="1.0" package="org.webrtc.voiceengine.test">
<application android:icon="@drawable/icon"
android:label="@string/app_name"
android:debuggable="true">
<activity android:name=".AndroidTest"
android:label="@string/app_name"
android:screenOrientation="portrait">
<intent-filter>
<action android:name="android.intent.action.MAIN" />
<category android:name="android.intent.category.LAUNCHER" />
</intent-filter>
</activity>
</application>
<uses-sdk android:minSdkVersion="3" />
<uses-permission android:name="android.permission.MODIFY_AUDIO_SETTINGS" />
<uses-permission android:name="android.permission.RECORD_AUDIO" />
<uses-permission android:name="android.permission.INTERNET" />
</manifest>

11
webrtc/voice_engine/test/android/android_test/default.properties
View File

@ -1,11 +0,0 @@
# This file is automatically generated by Android Tools.
# Do not modify this file -- YOUR CHANGES WILL BE ERASED!
#
# This file must be checked in Version Control Systems.
#
# To customize properties used by the Ant build system use,
# "build.properties", and override values to adapt the script to your
# project structure.
# Project target, OpenSL ES requires API level 9
target=android-9

30
webrtc/voice_engine/test/android/android_test/gen/org/webrtc/voiceengine/test/R.java
View File

@ -1,30 +0,0 @@
/* AUTO-GENERATED FILE. DO NOT MODIFY.
*
* This class was automatically generated by the
* aapt tool from the resource data it found. It
* should not be modified by hand.
*/
package org.webrtc.voiceengine.test;
public final class R {
public static final class attr {
}
public static final class drawable {
public static final int icon=0x7f020000;
}
public static final class id {
public static final int Button01=0x7f050002;
public static final int Button02=0x7f050005;
public static final int EditText01=0x7f050001;
public static final int Spinner01=0x7f050003;
public static final int Spinner02=0x7f050004;
public static final int TextView01=0x7f050000;
}
public static final class layout {
public static final int main=0x7f030000;
}
public static final class string {
public static final int app_name=0x7f040000;
}
}

1331
webrtc/voice_engine/test/android/android_test/jni/android_test.cc
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File diff suppressed because it is too large Load Diff

245
webrtc/voice_engine/test/android/android_test/jni/org_webrtc_voiceengine_test_AndroidTest.h
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@ -1,245 +0,0 @@
/* DO NOT EDIT THIS FILE - it is machine generated */
#include <jni.h>
/* Header for class org_webrtc_voiceengine_test_AndroidTest */
#ifndef _Included_org_webrtc_voiceengine_test_AndroidTest
#define _Included_org_webrtc_voiceengine_test_AndroidTest
#ifdef __cplusplus
extern "C" {
#endif
/*
* Class: org_webrtc_voiceengine_test_AndroidTest
* Method: NativeInit
* Signature: ()Z
*/
JNIEXPORT jboolean JNICALL Java_org_webrtc_voiceengine_test_AndroidTest_NativeInit
(JNIEnv *, jclass);
/*
* Class: org_webrtc_voiceengine_test_AndroidTest
* Method: RunAutoTest
* Signature: (II)I
*/
JNIEXPORT jint JNICALL Java_org_webrtc_voiceengine_test_AndroidTest_RunAutoTest
(JNIEnv *, jobject, jint, jint);
/*
* Class: org_webrtc_voiceengine_test_AndroidTest
* Method: Create
* Signature: ()Z
*/
JNIEXPORT jboolean JNICALL Java_org_webrtc_voiceengine_test_AndroidTest_Create
(JNIEnv *, jobject);
/*
* Class: org_webrtc_voiceengine_test_AndroidTest
* Method: Delete
* Signature: ()Z
*/
JNIEXPORT jboolean JNICALL Java_org_webrtc_voiceengine_test_AndroidTest_Delete
(JNIEnv *, jobject);
/*
* Class: org_webrtc_voiceengine_test_AndroidTest
* Method: Init
* Signature: (IIIZZ)I
*/
JNIEXPORT jint JNICALL Java_org_webrtc_voiceengine_test_AndroidTest_Init
(JNIEnv *, jobject, jboolean, jboolean);
/*
* Class: org_webrtc_voiceengine_test_AndroidTest
* Method: Terminate
* Signature: ()I
*/
JNIEXPORT jint JNICALL Java_org_webrtc_voiceengine_test_AndroidTest_Terminate
(JNIEnv *, jobject);
/*
* Class: org_webrtc_voiceengine_test_AndroidTest
* Method: CreateChannel
* Signature: ()I
*/
JNIEXPORT jint JNICALL Java_org_webrtc_voiceengine_test_AndroidTest_CreateChannel
(JNIEnv *, jobject);
/*
* Class: org_webrtc_voiceengine_test_AndroidTest
* Method: DeleteChannel
* Signature: (I)I
*/
JNIEXPORT jint JNICALL Java_org_webrtc_voiceengine_test_AndroidTest_DeleteChannel
(JNIEnv *, jobject, jint);
/*
* Class: org_webrtc_voiceengine_test_AndroidTest
* Method: SetLocalReceiver
* Signature: (II)I
*/
JNIEXPORT jint JNICALL Java_org_webrtc_voiceengine_test_AndroidTest_SetLocalReceiver
(JNIEnv *, jobject, jint, jint);
/*
* Class: org_webrtc_voiceengine_test_AndroidTest
* Method: SetSendDestination
* Signature: (IILjava/lang/String;)I
*/
JNIEXPORT jint JNICALL Java_org_webrtc_voiceengine_test_AndroidTest_SetSendDestination
(JNIEnv *, jobject, jint, jint, jstring);
/*
* Class: org_webrtc_voiceengine_test_AndroidTest
* Method: StartListen
* Signature: (I)I
*/
JNIEXPORT jint JNICALL Java_org_webrtc_voiceengine_test_AndroidTest_StartListen
(JNIEnv *, jobject, jint);
/*
* Class: org_webrtc_voiceengine_test_AndroidTest
* Method: StartPlayout
* Signature: (I)I
*/
JNIEXPORT jint JNICALL Java_org_webrtc_voiceengine_test_AndroidTest_StartPlayout
(JNIEnv *, jobject, jint);
/*
* Class: org_webrtc_voiceengine_test_AndroidTest
* Method: StartSend
* Signature: (I)I
*/
JNIEXPORT jint JNICALL Java_org_webrtc_voiceengine_test_AndroidTest_StartSend
(JNIEnv *, jobject, jint);
/*
* Class: org_webrtc_voiceengine_test_AndroidTest
* Method: StopListen
* Signature: (I)I
*/
JNIEXPORT jint JNICALL Java_org_webrtc_voiceengine_test_AndroidTest_StopListen
(JNIEnv *, jobject, jint);
/*
* Class: org_webrtc_voiceengine_test_AndroidTest
* Method: StopPlayout
* Signature: (I)I
*/
JNIEXPORT jint JNICALL Java_org_webrtc_voiceengine_test_AndroidTest_StopPlayout
(JNIEnv *, jobject, jint);
/*
* Class: org_webrtc_voiceengine_test_AndroidTest
* Method: StopSend
* Signature: (I)I
*/
JNIEXPORT jint JNICALL Java_org_webrtc_voiceengine_test_AndroidTest_StopSend
(JNIEnv *, jobject, jint);
/*
* Class: org_webrtc_voiceengine_test_AndroidTest
* Method: StartPlayingFileLocally
* Signature: (ILjava/lang/String;Z)I
*/
JNIEXPORT jint JNICALL Java_org_webrtc_voiceengine_test_AndroidTest_StartPlayingFileLocally
(JNIEnv *, jobject, jint, jstring, jboolean);
/*
* Class: org_webrtc_voiceengine_test_AndroidTest
* Method: StopPlayingFileLocally
* Signature: (I)I
*/
JNIEXPORT jint JNICALL Java_org_webrtc_voiceengine_test_AndroidTest_StopPlayingFileLocally
(JNIEnv *, jobject, jint);
/*
* Class: org_webrtc_voiceengine_test_AndroidTest
* Method: StartRecordingPlayout
* Signature: (ILjava/lang/String;Z)I
*/
JNIEXPORT jint JNICALL Java_org_webrtc_voiceengine_test_AndroidTest_StartRecordingPlayout
(JNIEnv *, jobject, jint, jstring, jboolean);
/*
* Class: org_webrtc_voiceengine_test_AndroidTest
* Method: StopRecordingPlayout
* Signature: (I)I
*/
JNIEXPORT jint JNICALL Java_org_webrtc_voiceengine_test_AndroidTest_StopRecordingPlayout
(JNIEnv *, jobject, jint);
/*
* Class: org_webrtc_voiceengine_test_AndroidTest
* Method: StartPlayingFileAsMicrophone
* Signature: (ILjava/lang/String;Z)I
*/
JNIEXPORT jint JNICALL Java_org_webrtc_voiceengine_test_AndroidTest_StartPlayingFileAsMicrophone
(JNIEnv *, jobject, jint, jstring, jboolean);
/*
* Class: org_webrtc_voiceengine_test_AndroidTest
* Method: StopPlayingFileAsMicrophone
* Signature: (I)I
*/
JNIEXPORT jint JNICALL Java_org_webrtc_voiceengine_test_AndroidTest_StopPlayingFileAsMicrophone
(JNIEnv *, jobject, jint);
/*
* Class: org_webrtc_voiceengine_test_AndroidTest
* Method: NumOfCodecs
* Signature: ()I
*/
JNIEXPORT jint JNICALL Java_org_webrtc_voiceengine_test_AndroidTest_NumOfCodecs
(JNIEnv *, jobject);
/*
* Class: org_webrtc_voiceengine_test_AndroidTest
* Method: SetSendCodec
* Signature: (II)I
*/
JNIEXPORT jint JNICALL Java_org_webrtc_voiceengine_test_AndroidTest_SetSendCodec
(JNIEnv *, jobject, jint, jint);
/*
* Class: org_webrtc_voiceengine_test_AndroidTest
* Method: SetVADStatus
* Signature: (IZI)I
*/
JNIEXPORT jint JNICALL Java_org_webrtc_voiceengine_test_AndroidTest_SetVADStatus
(JNIEnv *, jobject, jint, jboolean, jint);
/*
* Class: org_webrtc_voiceengine_test_AndroidTest
* Method: SetNSStatus
* Signature: (ZI)I
*/
JNIEXPORT jint JNICALL Java_org_webrtc_voiceengine_test_AndroidTest_SetNSStatus
(JNIEnv *, jobject, jboolean, jint);
/*
* Class: org_webrtc_voiceengine_test_AndroidTest
* Method: SetAGCStatus
* Signature: (ZI)I
*/
JNIEXPORT jint JNICALL Java_org_webrtc_voiceengine_test_AndroidTest_SetAGCStatus
(JNIEnv *, jobject, jboolean, jint);
/*
* Class: org_webrtc_voiceengine_test_AndroidTest
* Method: SetECStatus
* Signature: (ZI)I
*/
JNIEXPORT jint JNICALL Java_org_webrtc_voiceengine_test_AndroidTest_SetECStatus
(JNIEnv *, jobject, jboolean, jint);
/*
* Class: org_webrtc_voiceengine_test_AndroidTest
* Method: SetSpeakerVolume
* Signature: (I)I
*/
JNIEXPORT jint JNICALL Java_org_webrtc_voiceengine_test_AndroidTest_SetSpeakerVolume
(JNIEnv *, jobject, jint);
#ifdef __cplusplus
}
#endif
#endif

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webrtc/voice_engine/test/android/android_test/res/layout/main.xml
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<?xml version="1.0" encoding="utf-8"?>
<!-- Copyright (c) 2011 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. -->
<LinearLayout xmlns:android="http://schemas.android.com/apk/res/android"
android:orientation="vertical"
android:layout_width="fill_parent"
android:layout_height="fill_parent">
<TextView android:text="@+id/TextView01"
android:id="@+id/TextView01"
android:layout_width="wrap_content"
android:layout_height="wrap_content">
</TextView>
<EditText android:text="@+id/EditText01"
android:id="@+id/EditText01"
android:layout_width="wrap_content"
android:layout_height="wrap_content">
</EditText>
<Button android:text="@+id/Button01"
android:id="@+id/Button01"
android:layout_width="wrap_content"
android:layout_height="wrap_content">
</Button>
<Spinner android:id="@+id/Spinner01"
android:layout_width="wrap_content"
android:layout_height="wrap_content">
</Spinner>
<Spinner android:id="@+id/Spinner02"
android:layout_width="wrap_content"
android:layout_height="wrap_content">
</Spinner>
<Button android:text="@+id/Button02"
android:id="@+id/Button02"
android:layout_width="wrap_content"
android:layout_height="wrap_content">
</Button>
</LinearLayout>

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webrtc/voice_engine/test/android/android_test/res/values/strings.xml
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<?xml version="1.0" encoding="utf-8"?>
<!-- Copyright (c) 2011 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. -->
<resources>
<string name="app_name">WebRtc VoE</string>
</resources>

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webrtc/voice_engine/test/android/android_test/src/org/webrtc/voiceengine/test/AndroidTest.java
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