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Add debug options to test app

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Description:
1. add apm debug option
2. add voice/video rtp dump option
3. front facing camera as the default, minor change

BUG=None
TEST=local
Review URL: https://webrtc-codereview.appspot.com/728012

git-svn-id: http://webrtc.googlecode.com/svn/trunk@2650 4adac7df-926f-26a2-2b94-8c16560cd09d
This commit is contained in:
leozwang@webrtc.org 2012-08-21 18:07:14 +00:00
parent bdb03d48ae
commit d6fcf7f0da
7 changed files with 311 additions and 3 deletions
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48
src/video_engine/main/test/android_test/jni/org_webrtc_videoengineapp_vie_android_java_api.h
View File

@ -219,6 +219,22 @@ JNIEXPORT jint JNICALL Java_org_webrtc_videoengineapp_ViEAndroidJavaAPI_EnablePL
JNIEXPORT jint JNICALL Java_org_webrtc_videoengineapp_ViEAndroidJavaAPI_SetCallback
(JNIEnv *, jobject, jint, jobject);
/*
* Class: org_webrtc_videoengineapp_ViEAndroidJavaAPI
* Method: StartIncomingRTPDump
* Signature: (ILjava/lang/String;)I
*/
JNIEXPORT jint JNICALL Java_org_webrtc_videoengineapp_ViEAndroidJavaAPI_StartIncomingRTPDump
(JNIEnv *, jobject, jint, jstring);
/*
* Class: org_webrtc_videoengineapp_ViEAndroidJavaAPI
* Method: StopIncomingRTPDump
* Signature: (I)I
*/
JNIEXPORT jint JNICALL Java_org_webrtc_videoengineapp_ViEAndroidJavaAPI_StopIncomingRTPDump
(JNIEnv *, jobject, jint);
/*
* Class: org_webrtc_videoengineapp_ViEAndroidJavaAPI
* Method: VoE_Create
@ -429,6 +445,38 @@ JNIEXPORT jint JNICALL Java_org_webrtc_videoengineapp_ViEAndroidJavaAPI_VoE_1Set
JNIEXPORT jint JNICALL Java_org_webrtc_videoengineapp_ViEAndroidJavaAPI_VoE_1SetNSStatus
(JNIEnv *, jobject, jboolean);
/*
* Class: org_webrtc_videoengineapp_ViEAndroidJavaAPI
* Method: VoE_StartDebugRecording
* Signature: (Ljava/lang/String)I
*/
JNIEXPORT jint JNICALL Java_org_webrtc_videoengineapp_ViEAndroidJavaAPI_VoE_1StartDebugRecording
(JNIEnv *, jobject, jstring);
/*
* Class: org_webrtc_videoengineapp_ViEAndroidJavaAPI
* Method: VoE_StopDebugRecording
* Signature: ()I
*/
JNIEXPORT jint JNICALL Java_org_webrtc_videoengineapp_ViEAndroidJavaAPI_VoE_1StopDebugRecording
(JNIEnv *, jobject);
/*
* Class: org_webrtc_videoengineapp_ViEAndroidJavaAPI
* Method: VoE_StartIncomingRTPDump
* Signature: (ILjava/lang/String;)I
*/
JNIEXPORT jint JNICALL Java_org_webrtc_videoengineapp_ViEAndroidJavaAPI_VoE_1StartIncomingRTPDump
(JNIEnv *, jobject, jint, jstring);
/*
* Class: org_webrtc_videoengineapp_ViEAndroidJavaAPI
* Method: VoE_StopIncomingRTPDump
* Signature: (I)I
*/
JNIEXPORT jint JNICALL Java_org_webrtc_videoengineapp_ViEAndroidJavaAPI_VoE_1StopIncomingRTPDump
(JNIEnv *, jobject, jint);
#ifdef __cplusplus
}
#endif

165
src/video_engine/main/test/android_test/jni/vie_android_java_api.cc
View File

@ -21,6 +21,7 @@
#include "voe_audio_processing.h"
#include "voe_volume_control.h"
#include "voe_hardware.h"
#include "voe_rtp_rtcp.h"
#include "vie_base.h"
#include "vie_codec.h"
@ -76,6 +77,14 @@
return -1; \
}
#define VALIDATE_RTP_POINTER \
if (!voeData.rtp) \
{ \
__android_log_write(ANDROID_LOG_ERROR, WEBRTC_LOG_TAG, \
"rtp pointer doesn't exist"); \
return -1; \
}
using namespace webrtc;
//Forward declaration.
@ -94,6 +103,7 @@ typedef struct
VoEAudioProcessing* apm;
VoEVolumeControl* volume;
VoEHardware* hardware;
VoERTP_RTCP* rtp;
JavaVM* jvm;
} VoiceEngineData;
@ -1031,6 +1041,57 @@ JNIEXPORT jint JNICALL Java_org_webrtc_videoengineapp_ViEAndroidJavaAPI_SetCallb
return 0;
}
/*
* Class: org_webrtc_videoengineapp_ViEAndroidJavaAPI
* Method: StartIncomingRTPDump
* Signature: (ILjava/lang/String;)I
*/
JNIEXPORT jint JNICALL Java_org_webrtc_videoengineapp_ViEAndroidJavaAPI_StartIncomingRTPDump(
JNIEnv* env,
jobject,
jint channel,
jstring filename) {
if (NULL == vieData.rtp) {
__android_log_write(ANDROID_LOG_ERROR, WEBRTC_LOG_TAG,
"video RTP_RTCP interface is null");
return -1;
}
const char* file = env->GetStringUTFChars(filename, NULL);
if (!file) {
__android_log_print(ANDROID_LOG_ERROR, WEBRTC_LOG_TAG,
"Video StartRTPDump file name error");
return -1;
}
if (vieData.rtp->StartRTPDump(channel, file, kRtpIncoming) != 0) {
__android_log_print(ANDROID_LOG_ERROR, WEBRTC_LOG_TAG,
"Video StartRTPDump error");
return -1;
}
return 0;
}
/*
* Class: org_webrtc_videoengineapp_ViEAndroidJavaAPI
* Method: StopIncomingRTPDump
* Signature: (I)I
*/
JNIEXPORT jint JNICALL Java_org_webrtc_videoengineapp_ViEAndroidJavaAPI_StopIncomingRTPDump(
JNIEnv *,
jobject,
jint channel) {
if (NULL == vieData.rtp) {
__android_log_write(ANDROID_LOG_ERROR, WEBRTC_LOG_TAG,
"video RTP_RTCP interface is null");
return -1;
}
if (vieData.rtp->StopRTPDump(channel, kRtpIncoming) != 0) {
__android_log_print(ANDROID_LOG_ERROR, WEBRTC_LOG_TAG,
"Video StopRTPDump error");
return -1;
}
return 0;
}
//
// VoiceEngine API wrapper functions
//
@ -1569,6 +1630,91 @@ JNIEXPORT jint JNICALL Java_org_webrtc_videoengineapp_ViEAndroidJavaAPI_VoE_1Set
return 0;
}
/*
* Class: org_webrtc_videoengineapp_ViEAndroidJavaAPI
* Method: VoE_StartDebugRecording
* Signature: (Ljava/lang/String)I
*/
JNIEXPORT jint JNICALL Java_org_webrtc_videoengineapp_ViEAndroidJavaAPI_VoE_1StartDebugRecording(
JNIEnv* env,
jobject,
jstring filename) {
VALIDATE_APM_POINTER;
const char* file = env->GetStringUTFChars(filename, NULL);
if (!file) {
__android_log_print(ANDROID_LOG_ERROR, WEBRTC_LOG_TAG,
"Voice StartDebugRecording file error");
return -1;
}
if (voeData.apm->StartDebugRecording(file) != 0) {
__android_log_print(ANDROID_LOG_ERROR, WEBRTC_LOG_TAG,
"Voice StartDebugRecording error");
return -1;
}
return 0;
}
/*
* Class: org_webrtc_videoengineapp_ViEAndroidJavaAPI
* Method: VoE_StopDebugRecording
* Signature: ()I
*/
JNIEXPORT jint JNICALL Java_org_webrtc_videoengineapp_ViEAndroidJavaAPI_VoE_1StopDebugRecording(
JNIEnv *,
jobject) {
VALIDATE_APM_POINTER;
if (voeData.apm->StopDebugRecording() < 0) {
__android_log_print(ANDROID_LOG_ERROR, WEBRTC_LOG_TAG,
"Voice StopDebugRecording error");
return -1;
}
return 0;
}
/*
* Class: org_webrtc_videoengineapp_ViEAndroidJavaAPI
* Method: VoE_StartIncomingRTPDump
* Signature: (ILjava/lang/String;)I
*/
JNIEXPORT jint JNICALL Java_org_webrtc_videoengineapp_ViEAndroidJavaAPI_VoE_1StartIncomingRTPDump(
JNIEnv* env,
jobject,
jint channel,
jstring filename) {
VALIDATE_RTP_POINTER;
const char* file = env->GetStringUTFChars(filename, NULL);
if (!file) {
__android_log_print(ANDROID_LOG_ERROR, WEBRTC_LOG_TAG,
"Voice StartRTPDump file error");
return -1;
}
if (voeData.rtp->StartRTPDump(channel, file, kRtpIncoming) != 0) {
__android_log_print(ANDROID_LOG_ERROR, WEBRTC_LOG_TAG,
"Voice StartRTPDump error");
return -1;
}
return 0;
}
/*
* Class: org_webrtc_videoengineapp_ViEAndroidJavaAPI
* Method: VoE_StopRTPDump
* Signature: (I)I
*/
JNIEXPORT jint JNICALL Java_org_webrtc_videoengineapp_ViEAndroidJavaAPI_VoE_1StopRTPDump(
JNIEnv *,
jobject,
jint channel) {
VALIDATE_RTP_POINTER;
if (voeData.rtp->StopRTPDump(channel) < 0) {
__android_log_print(ANDROID_LOG_ERROR, WEBRTC_LOG_TAG,
"Voice StopRTPDump error");
return -1;
}
return 0;
}
//
// local function
//
@ -1633,6 +1779,14 @@ bool VE_GetSubApis() {
getOK = false;
}
// RTP
voeData.rtp = VoERTP_RTCP::GetInterface(voeData.ve);
if (!voeData.rtp) {
__android_log_write(ANDROID_LOG_ERROR, WEBRTC_LOG_TAG,
"Get rtp sub-API failed");
getOK = false;
}
return getOK;
}
@ -1724,5 +1878,16 @@ bool VE_ReleaseSubApis() {
}
}
if (voeData.rtp) {
if (0 != voeData.rtp->Release()) {
__android_log_write(ANDROID_LOG_ERROR, WEBRTC_LOG_TAG,
"Release rtp sub-API failed");
releaseOK = false;
}
else {
voeData.rtp = NULL;
}
}
return releaseOK;
}

11
src/video_engine/main/test/android_test/res/layout/aconfig.xml
View File

@ -71,6 +71,17 @@
android:id="@+id/cbSpeaker"
android:text="@string/speaker">
</CheckBox>
<CheckBox android:layout_width="wrap_content"
android:layout_height="wrap_content"
android:id="@+id/cbDebugRecording"
android:text="@string/debugrecording">
</CheckBox>
<CheckBox android:layout_width="wrap_content"
android:layout_height="wrap_content"
android:id="@+id/cbVoiceRTPDump"
android:text="@string/rtpdump">
</CheckBox>
</LinearLayout>
</LinearLayout>

8
src/video_engine/main/test/android_test/res/layout/vconfig.xml
View File

@ -62,6 +62,12 @@
android:layout_height="wrap_content"
android:id="@+id/cbNack"
android:text="@string/nack">
</CheckBox>
</CheckBox>
<CheckBox android:layout_width="wrap_content"
android:layout_height="wrap_content"
android:id="@+id/cbVideoRTPDump"
android:text="@string/rtpdump">
</CheckBox>
</LinearLayout>
</LinearLayout>

2
src/video_engine/main/test/android_test/res/values/strings.xml
View File

@ -33,6 +33,8 @@
<string name="stopCall">StopCall</string>
<string name="exit">Exit</string>
<string name="speaker">Speaker</string>
<string name="debugrecording">APMRecord</string>
<string name="rtpdump">rtpdump</string>
<string name="surfaceview">SurfaceView</string>
<string name="opengl">OpenGL</string>
</resources>

9
src/video_engine/main/test/android_test/src/org/webrtc/videoengineapp/ViEAndroidJavaAPI.java
View File

@ -81,6 +81,9 @@ public class ViEAndroidJavaAPI {
// Enable stats callback
public native int SetCallback(int channel, IViEAndroidCallback callback);
public native int StartIncomingRTPDump(int channel, String file);
public native int StopIncomingRTPDump(int channel);
// Voice Engine API
// Create and Delete functions
public native boolean VoE_Create();
@ -132,8 +135,12 @@ public class ViEAndroidJavaAPI {
public native String[] VoE_GetCodecs();
public native int VoE_SetSendCodec(int channel, int index);
//VE funtions
//VoiceEngine funtions
public native int VoE_SetECStatus(boolean enable);
public native int VoE_SetAGCStatus(boolean enable);
public native int VoE_SetNSStatus(boolean enable);
public native int VoE_StartDebugRecording(String file);
public native int VoE_StopDebugRecording();
public native int VoE_StartIncomingRTPDump(int channel, String file);
public native int VoE_StopIncomingRTPDump(int channel);
}

71
src/video_engine/main/test/android_test/src/org/webrtc/videoengineapp/WebRTCDemo.java
View File

@ -10,6 +10,7 @@
package org.webrtc.videoengineapp;
import java.io.File;
import java.net.InetAddress;
import java.net.NetworkInterface;
import java.net.SocketException;
@ -34,6 +35,7 @@ import android.graphics.Rect;
import android.hardware.SensorManager;
import android.media.AudioManager;
import android.os.Bundle;
import android.os.Environment;
import android.os.PowerManager;
import android.os.PowerManager.WakeLock;
@ -136,6 +138,7 @@ public class WebRTCDemo extends TabActivity implements IViEAndroidCallback,
private int destinationPortVideo = 11111;
private CheckBox cbEnableNack;
private boolean enableNack = false;
private CheckBox cbEnableVideoRTPDump;
// Audio settings
private Spinner spVoiceCodecType;
@ -152,6 +155,8 @@ public class WebRTCDemo extends TabActivity implements IViEAndroidCallback,
private boolean enableAECM = false;
private CheckBox cbEnableNS;
private boolean enableNS = false;
private CheckBox cbEnableDebugAPM;
private CheckBox cbEnableVoiceRTPDump;
// Stats variables
private int frameRateI;
@ -160,9 +165,13 @@ public class WebRTCDemo extends TabActivity implements IViEAndroidCallback,
private int frameRateO;
private int bitRateO;
// Variable for storing variables
private String webrtcName = "/webrtc";
private String webrtcDebugDir = null;
private WakeLock wakeLock;
private boolean usingFrontCamera = false;
private boolean usingFrontCamera = true;
private String[] mVideoCodecsStrings = null;
private String[] mVideoCodecsSizeStrings = { "176x144", "320x240",
@ -261,6 +270,18 @@ public class WebRTCDemo extends TabActivity implements IViEAndroidCallback,
};
orientationListener.enable ();
// Create a folder named webrtc in /scard for debugging
webrtcDebugDir = Environment.getExternalStorageDirectory().toString() +
webrtcName;
File webrtcDir = new File(webrtcDebugDir);
if (!webrtcDir.exists() && webrtcDir.mkdir() == false) {
Log.v(TAG, "Failed to create " + webrtcDebugDir);
}
else if (!webrtcDir.isDirectory()) {
Log.v(TAG, webrtcDebugDir + " exists but not a folder");
webrtcDebugDir = null;
}
StartMain();
return;
}
@ -494,6 +515,15 @@ public class WebRTCDemo extends TabActivity implements IViEAndroidCallback,
cbEnableNS = (CheckBox) findViewById(R.id.cbNoiseSuppression);
cbEnableNS.setChecked(enableNS);
cbEnableDebugAPM = (CheckBox) findViewById(R.id.cbDebugRecording);
cbEnableDebugAPM.setChecked(false); // Disable APM debugging by default
cbEnableVideoRTPDump = (CheckBox) findViewById(R.id.cbVideoRTPDump);
cbEnableVideoRTPDump.setChecked(false); // Disable Video RTP Dump
cbEnableVoiceRTPDump = (CheckBox) findViewById(R.id.cbVoiceRTPDump);
cbEnableVoiceRTPDump.setChecked(false); // Disable Voice RTP Dump
etRemoteIp.setOnClickListener(this);
cbLoopback.setOnClickListener(this);
cbStats.setOnClickListener(this);
@ -502,6 +532,9 @@ public class WebRTCDemo extends TabActivity implements IViEAndroidCallback,
cbEnableAECM.setOnClickListener(this);
cbEnableAGC.setOnClickListener(this);
cbEnableNS.setOnClickListener(this);
cbEnableDebugAPM.setOnClickListener(this);
cbEnableVideoRTPDump.setOnClickListener(this);
cbEnableVoiceRTPDump.setOnClickListener(this);
if (loopbackMode) {
remoteIp = LOOPBACK_IP;
@ -807,6 +840,42 @@ public class WebRTCDemo extends TabActivity implements IViEAndroidCallback,
RouteAudio(enableSpeaker);
}
break;
case R.id.cbDebugRecording:
if(voERunning && webrtcDebugDir != null) {
if (cbEnableDebugAPM.isChecked() ) {
ViEAndroidAPI.VoE_StartDebugRecording(
webrtcDebugDir + String.format("/apm_%d.dat",
System.currentTimeMillis()));
}
else {
ViEAndroidAPI.VoE_StopDebugRecording();
}
}
break;
case R.id.cbVoiceRTPDump:
if(voERunning && webrtcDebugDir != null) {
if (cbEnableVoiceRTPDump.isChecked() ) {
ViEAndroidAPI.VoE_StartIncomingRTPDump(channel,
webrtcDebugDir + String.format("/voe_%d.rtp",
System.currentTimeMillis()));
}
else {
ViEAndroidAPI.VoE_StopIncomingRTPDump(channel);
}
}
break;
case R.id.cbVideoRTPDump:
if(viERunning && webrtcDebugDir != null) {
if (cbEnableVideoRTPDump.isChecked() ) {
ViEAndroidAPI.StartIncomingRTPDump(channel,
webrtcDebugDir + String.format("/vie_%d.rtp",
System.currentTimeMillis()));
}
else {
ViEAndroidAPI.StopIncomingRTPDump(channel);
}
}
break;
case R.id.cbAutoGainControl:
enableAGC=cbEnableAGC.isChecked();
if(voERunning) {