Add debug options to test app

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

src/video_engine/main/test/android_test/jni/org_webrtc_videoengineapp_vie_android_java_api.h

@@ -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

src/video_engine/main/test/android_test/jni/vie_android_java_api.cc

@@ -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;
 }

src/video_engine/main/test/android_test/res/layout/aconfig.xml

@@ -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>

src/video_engine/main/test/android_test/res/layout/vconfig.xml

@@ -63,5 +63,11 @@
 		  android:id="@+id/cbNack"
 		  android:text="@string/nack">
 	</CheckBox>
+        <CheckBox android:layout_width="wrap_content"
+                  android:layout_height="wrap_content"
+                  android:id="@+id/cbVideoRTPDump"
+                  android:text="@string/rtpdump">
+        </CheckBox>
+
 	</LinearLayout>
 </LinearLayout>

src/video_engine/main/test/android_test/res/values/strings.xml

@@ -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>

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

@@ -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);
 }

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

@@ -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) {