am 2ba92c71: do not merge bug 3370834 Cherrypick from master

* commit '2ba92c71b5684dce700cf848bf157153c156df1d':
  do not merge bug 3370834 Cherrypick from master

include/media/AudioSystem.h

@@ -156,6 +156,7 @@ public:
         MODE_NORMAL = 0,
         MODE_RINGTONE,
         MODE_IN_CALL,
+        MODE_IN_COMMUNICATION,
         NUM_MODES  // not a valid entry, denotes end-of-list
     };
 
@@ -466,7 +467,7 @@ public:
     AudioParameter(const String8& keyValuePairs);
     virtual ~AudioParameter();
 
-    // reserved parameter keys for changeing standard parameters with setParameters() function.
+    // reserved parameter keys for changing standard parameters with setParameters() function.
     // Using these keys is mandatory for AudioFlinger to properly monitor audio output/input
     // configuration changes and act accordingly.
     //  keyRouting: to change audio routing, value is an int in AudioSystem::audio_devices
@@ -474,11 +475,14 @@ public:
     //  keyFormat: to change audio format, value is an int in AudioSystem::audio_format
     //  keyChannels: to change audio channel configuration, value is an int in AudioSystem::audio_channels
     //  keyFrameCount: to change audio output frame count, value is an int
+    //  keyInputSource: to change audio input source, value is an int in audio_source
+    //     (defined in media/mediarecorder.h)
     static const char *keyRouting;
     static const char *keySamplingRate;
     static const char *keyFormat;
     static const char *keyChannels;
     static const char *keyFrameCount;
+    static const char *keyInputSource;
 
     String8 toString();
 

include/media/EffectApi.h

@@ -602,9 +602,9 @@ enum audio_device_e {
 
 // Audio mode
 enum audio_mode_e {
-    AUDIO_MODE_NORMAL,      // phone idle
+    AUDIO_MODE_NORMAL,      // device idle
-    AUDIO_MODE_RINGTONE,    // phone ringing
+    AUDIO_MODE_RINGTONE,    // device ringing
-    AUDIO_MODE_IN_CALL      // phone call connected
+    AUDIO_MODE_IN_CALL      // audio call connected (VoIP or telephony)
 };
 
 // Values for "accessMode" field of buffer_config_t:

include/media/mediarecorder.h

@@ -44,7 +44,8 @@ enum audio_source {
     AUDIO_SOURCE_VOICE_CALL = 4,
     AUDIO_SOURCE_CAMCORDER = 5,
     AUDIO_SOURCE_VOICE_RECOGNITION = 6,
-    AUDIO_SOURCE_MAX = AUDIO_SOURCE_VOICE_RECOGNITION,
+    AUDIO_SOURCE_VOICE_COMMUNICATION = 7,
+    AUDIO_SOURCE_MAX = AUDIO_SOURCE_VOICE_COMMUNICATION,
 
     AUDIO_SOURCE_LIST_END  // must be last - used to validate audio source type
 };

media/java/android/media/AudioManager.java

@@ -966,9 +966,14 @@ public class AudioManager {
      */
     public static final int MODE_RINGTONE           = AudioSystem.MODE_RINGTONE;
     /**
-     * In call audio mode. A call is established.
+     * In call audio mode. A telephony call is established.
      */
     public static final int MODE_IN_CALL            = AudioSystem.MODE_IN_CALL;
+    /**
+     * @hide
+     * In communication audio mode. An audio/video chat or VoIP call is established.
+     */
+    public static final int MODE_IN_COMMUNICATION   = AudioSystem.MODE_IN_COMMUNICATION;
 
     /* Routing bits for setRouting/getRouting API */
     /**

media/java/android/media/AudioRecord.java

@@ -252,7 +252,8 @@ public class AudioRecord
         //--------------
         // audio source
         if ( (audioSource < MediaRecorder.AudioSource.DEFAULT) ||
-             (audioSource > MediaRecorder.getAudioSourceMax()) )  {
+                (audioSource > MediaRecorder.AudioSource.VOICE_COMMUNICATION) )  {
+                //(audioSource > MediaRecorder.getAudioSourceMax()) )  {
             throw (new IllegalArgumentException("Invalid audio source."));
         } else {
             mRecordSource = audioSource;

media/java/android/media/AudioService.java

@@ -708,7 +708,7 @@ public class AudioService extends IAudioService.Stub {
             return;
         }
 
-        if (mode < AudioSystem.MODE_CURRENT || mode > AudioSystem.MODE_IN_CALL) {
+        if (mode < AudioSystem.MODE_CURRENT || mode >= AudioSystem.NUM_MODES) {
             return;
         }
 
@@ -2259,6 +2259,7 @@ public class AudioService extends IAudioService.Stub {
                 //      add modify the phone app to take advantage of the new API
                 synchronized(mRingingLock) {
                     if (mIsRinging || (getMode() == AudioSystem.MODE_IN_CALL) ||
+                            (getMode() == AudioSystem.MODE_IN_COMMUNICATION) ||
                             (getMode() == AudioSystem.MODE_RINGTONE) ) {
                         return;
                     }

media/java/android/media/AudioSystem.java

@@ -106,7 +106,8 @@ public class AudioSystem
     public static final int MODE_NORMAL             = 0;
     public static final int MODE_RINGTONE           = 1;
     public static final int MODE_IN_CALL            = 2;
-    public static final int NUM_MODES               = 3;
+    public static final int MODE_IN_COMMUNICATION   = 3;
+    public static final int NUM_MODES               = 4;
 
 
     /* Routing bits for setRouting/getRouting API */

media/java/android/media/MediaRecorder.java

@@ -145,6 +145,15 @@ public class MediaRecorder
         /** Microphone audio source tuned for voice recognition if available, behaves like
          *  {@link #DEFAULT} otherwise. */
         public static final int VOICE_RECOGNITION = 6;
+
+        /**
+         * @hide
+         * Microphone audio source tuned for voice communications such as VoIP. It
+         * will for instance take advantage of echo cancellation or automatic gain control
+         * if available. It otherwise behaves like {@link #DEFAULT} if no voice processing
+         * is available.
+         */
+        public static final int VOICE_COMMUNICATION = 7;
     }
 
     /**

media/libmedia/AudioSystem.cpp

@@ -835,6 +835,7 @@ const char *AudioParameter::keySamplingRate = "sampling_rate";
 const char *AudioParameter::keyFormat = "format";
 const char *AudioParameter::keyChannels = "channels";
 const char *AudioParameter::keyFrameCount = "frame_count";
+const char *AudioParameter::keyInputSource = "input_source";
 
 AudioParameter::AudioParameter(const String8& keyValuePairs)
 {

services/audioflinger/AudioFlinger.cpp

@@ -5810,7 +5810,8 @@ uint32_t AudioFlinger::EffectModule::deviceAudioSystemToEffectApi(uint32_t devic
 const uint32_t AudioFlinger::EffectModule::sModeConvTable[] = {
     AUDIO_MODE_NORMAL,   // AudioSystem::MODE_NORMAL
     AUDIO_MODE_RINGTONE, // AudioSystem::MODE_RINGTONE
-    AUDIO_MODE_IN_CALL   // AudioSystem::MODE_IN_CALL
+    AUDIO_MODE_IN_CALL,  // AudioSystem::MODE_IN_CALL
+    AUDIO_MODE_IN_CALL   // AudioSystem::MODE_IN_COMMUNICATION, same conversion as for MODE_IN_CALL
 };
 
 int AudioFlinger::EffectModule::modeAudioSystemToEffectApi(uint32_t mode)

services/audioflinger/AudioHardwareInterface.cpp

@@ -48,14 +48,15 @@ static const char* routingModeStrings[] =
     "CURRENT",
     "NORMAL",
     "RINGTONE",
-    "IN_CALL"
+    "IN_CALL",
+    "IN_COMMUNICATION"
 };
 
 static const char* routeNone = "NONE";
 
 static const char* displayMode(int mode)
 {
-    if ((mode < -2) || (mode > 2))
+    if ((mode < AudioSystem::MODE_INVALID) || (mode >= AudioSystem::NUM_MODES))
         return routingModeStrings[0];
     return routingModeStrings[mode+3];
 }

services/audioflinger/AudioPolicyManagerBase.cpp

@@ -235,7 +235,7 @@ void AudioPolicyManagerBase::setPhoneState(int state)
 
     // if leaving call state, handle special case of active streams
     // pertaining to sonification strategy see handleIncallSonification()
-    if (mPhoneState == AudioSystem::MODE_IN_CALL) {
+    if (isInCall()) {
         LOGV("setPhoneState() in call state management: new state is %d", state);
         for (int stream = 0; stream < AudioSystem::NUM_STREAM_TYPES; stream++) {
             handleIncallSonification(stream, false, true);
@@ -248,16 +248,21 @@ void AudioPolicyManagerBase::setPhoneState(int state)
     bool force = false;
 
     // are we entering or starting a call
-    if ((oldState != AudioSystem::MODE_IN_CALL) && (state == AudioSystem::MODE_IN_CALL)) {
+    if (!isStateInCall(oldState) && isStateInCall(state)) {
         LOGV("  Entering call in setPhoneState()");
         // force routing command to audio hardware when starting a call
         // even if no device change is needed
         force = true;
-    } else if ((oldState == AudioSystem::MODE_IN_CALL) && (state != AudioSystem::MODE_IN_CALL)) {
+    } else if (isStateInCall(oldState) && !isStateInCall(state)) {
         LOGV("  Exiting call in setPhoneState()");
         // force routing command to audio hardware when exiting a call
         // even if no device change is needed
         force = true;
+    } else if (isStateInCall(state) && (state != oldState)) {
+        LOGV("  Switching between telephony and VoIP in setPhoneState()");
+        // force routing command to audio hardware when switching between telephony and VoIP
+        // even if no device change is needed
+        force = true;
     }
 
     // check for device and output changes triggered by new phone state
@@ -272,7 +277,7 @@ void AudioPolicyManagerBase::setPhoneState(int state)
 
     // force routing command to audio hardware when ending call
     // even if no device change is needed
-    if (oldState == AudioSystem::MODE_IN_CALL && newDevice == 0) {
+    if (isStateInCall(oldState) && newDevice == 0) {
         newDevice = hwOutputDesc->device();
     }
 
@@ -280,7 +285,7 @@ void AudioPolicyManagerBase::setPhoneState(int state)
     // immediately and delay the route change to avoid sending the ring tone
     // tail into the earpiece or headset.
     int delayMs = 0;
-    if (state == AudioSystem::MODE_IN_CALL && oldState == AudioSystem::MODE_RINGTONE) {
+    if (isStateInCall(state) && oldState == AudioSystem::MODE_RINGTONE) {
         // delay the device change command by twice the output latency to have some margin
         // and be sure that audio buffers not yet affected by the mute are out when
         // we actually apply the route change
@@ -293,7 +298,7 @@ void AudioPolicyManagerBase::setPhoneState(int state)
 
     // if entering in call state, handle special case of active streams
     // pertaining to sonification strategy see handleIncallSonification()
-    if (state == AudioSystem::MODE_IN_CALL) {
+    if (isStateInCall(state)) {
         LOGV("setPhoneState() in call state management: new state is %d", state);
         // unmute the ringing tone after a sufficient delay if it was muted before
         // setting output device above
@@ -564,7 +569,7 @@ status_t AudioPolicyManagerBase::startOutput(audio_io_handle_t output,
     setOutputDevice(output, getNewDevice(output));
 
     // handle special case for sonification while in call
-    if (mPhoneState == AudioSystem::MODE_IN_CALL) {
+    if (isInCall()) {
         handleIncallSonification(stream, true, false);
     }
 
@@ -589,7 +594,7 @@ status_t AudioPolicyManagerBase::stopOutput(audio_io_handle_t output,
     routing_strategy strategy = getStrategy((AudioSystem::stream_type)stream);
 
     // handle special case for sonification while in call
-    if (mPhoneState == AudioSystem::MODE_IN_CALL) {
+    if (isInCall()) {
         handleIncallSonification(stream, false, false);
     }
 
@@ -738,10 +743,8 @@ status_t AudioPolicyManagerBase::startInput(audio_io_handle_t input)
     AudioParameter param = AudioParameter();
     param.addInt(String8(AudioParameter::keyRouting), (int)inputDesc->mDevice);
 
-    // use Voice Recognition mode or not for this input based on input source
+    param.addInt(String8(AudioParameter::keyInputSource), (int)inputDesc->mInputSource);
-    int vr_enabled = inputDesc->mInputSource == AUDIO_SOURCE_VOICE_RECOGNITION ? 1 : 0;
+    LOGV("AudioPolicyManager::startInput() input source = %d", inputDesc->mInputSource);
-    param.addInt(String8("vr_mode"), vr_enabled);
-    LOGV("AudioPolicyManager::startInput(%d), setting vr_mode to %d", inputDesc->mInputSource, vr_enabled);
 
     mpClientInterface->setParameters(input, param.toString());
 
@@ -1494,7 +1497,7 @@ uint32_t AudioPolicyManagerBase::getNewDevice(audio_io_handle_t output, bool fro
     //      use device for strategy media
     // 4: the strategy DTMF is active on the hardware output:
     //      use device for strategy DTMF
-    if (mPhoneState == AudioSystem::MODE_IN_CALL ||
+    if (isInCall() ||
         outputDesc->isUsedByStrategy(STRATEGY_PHONE)) {
         device = getDeviceForStrategy(STRATEGY_PHONE, fromCache);
     } else if (outputDesc->isUsedByStrategy(STRATEGY_SONIFICATION)) {
@@ -1549,7 +1552,7 @@ uint32_t AudioPolicyManagerBase::getDeviceForStrategy(routing_strategy strategy,
 
     switch (strategy) {
     case STRATEGY_DTMF:
-        if (mPhoneState != AudioSystem::MODE_IN_CALL) {
+        if (!isInCall()) {
             // when off call, DTMF strategy follows the same rules as MEDIA strategy
             device = getDeviceForStrategy(STRATEGY_MEDIA, false);
             break;
@@ -1562,7 +1565,7 @@ uint32_t AudioPolicyManagerBase::getDeviceForStrategy(routing_strategy strategy,
         // of priority
         switch (mForceUse[AudioSystem::FOR_COMMUNICATION]) {
         case AudioSystem::FORCE_BT_SCO:
-            if (mPhoneState != AudioSystem::MODE_IN_CALL || strategy != STRATEGY_DTMF) {
+            if (!isInCall() || strategy != STRATEGY_DTMF) {
                 device = mAvailableOutputDevices & AudioSystem::DEVICE_OUT_BLUETOOTH_SCO_CARKIT;
                 if (device) break;
             }
@@ -1580,7 +1583,7 @@ uint32_t AudioPolicyManagerBase::getDeviceForStrategy(routing_strategy strategy,
             if (device) break;
 #ifdef WITH_A2DP
             // when not in a phone call, phone strategy should route STREAM_VOICE_CALL to A2DP
-            if (mPhoneState != AudioSystem::MODE_IN_CALL) {
+            if (!isInCall()) {
                 device = mAvailableOutputDevices & AudioSystem::DEVICE_OUT_BLUETOOTH_A2DP;
                 if (device) break;
                 device = mAvailableOutputDevices & AudioSystem::DEVICE_OUT_BLUETOOTH_A2DP_HEADPHONES;
@@ -1594,14 +1597,14 @@ uint32_t AudioPolicyManagerBase::getDeviceForStrategy(routing_strategy strategy,
             break;
 
         case AudioSystem::FORCE_SPEAKER:
-            if (mPhoneState != AudioSystem::MODE_IN_CALL || strategy != STRATEGY_DTMF) {
+            if (!isInCall() || strategy != STRATEGY_DTMF) {
                 device = mAvailableOutputDevices & AudioSystem::DEVICE_OUT_BLUETOOTH_SCO_CARKIT;
                 if (device) break;
             }
 #ifdef WITH_A2DP
             // when not in a phone call, phone strategy should route STREAM_VOICE_CALL to
             // A2DP speaker when forcing to speaker output
-            if (mPhoneState != AudioSystem::MODE_IN_CALL) {
+            if (!isInCall()) {
                 device = mAvailableOutputDevices & AudioSystem::DEVICE_OUT_BLUETOOTH_A2DP_SPEAKER;
                 if (device) break;
             }
@@ -1618,7 +1621,7 @@ uint32_t AudioPolicyManagerBase::getDeviceForStrategy(routing_strategy strategy,
 
         // If incall, just select the STRATEGY_PHONE device: The rest of the behavior is handled by
         // handleIncallSonification().
-        if (mPhoneState == AudioSystem::MODE_IN_CALL) {
+        if (isInCall()) {
             device = getDeviceForStrategy(STRATEGY_PHONE, false);
             break;
         }
@@ -1739,6 +1742,7 @@ uint32_t AudioPolicyManagerBase::getDeviceForInputSource(int inputSource)
     case AUDIO_SOURCE_DEFAULT:
     case AUDIO_SOURCE_MIC:
     case AUDIO_SOURCE_VOICE_RECOGNITION:
+    case AUDIO_SOURCE_VOICE_COMMUNICATION:
         if (mForceUse[AudioSystem::FOR_RECORD] == AudioSystem::FORCE_BT_SCO &&
             mAvailableInputDevices & AudioSystem::DEVICE_IN_BLUETOOTH_SCO_HEADSET) {
             device = AudioSystem::DEVICE_IN_BLUETOOTH_SCO_HEADSET;
@@ -1960,6 +1964,16 @@ void AudioPolicyManagerBase::handleIncallSonification(int stream, bool starting,
     }
 }
 
+bool AudioPolicyManagerBase::isInCall()
+{
+    return isStateInCall(mPhoneState);
+}
+
+bool AudioPolicyManagerBase::isStateInCall(int state) {
+    return ((state == AudioSystem::MODE_IN_CALL) ||
+            (state == AudioSystem::MODE_IN_COMMUNICATION));
+}
+
 bool AudioPolicyManagerBase::needsDirectOuput(AudioSystem::stream_type stream,
                                     uint32_t samplingRate,
                                     uint32_t format,

telephony/java/com/android/internal/telephony/CallManager.java

@@ -380,12 +380,17 @@ public final class CallManager {
                 break;
             case OFFHOOK:
                 Phone fgPhone = getFgPhone();
-                // Enable IN_CALL mode while foreground call is in DIALING,
+                // While foreground call is in DIALING,
-                // ALERTING, ACTIVE and DISCONNECTING state and not from sipPhone
+                // ALERTING, ACTIVE and DISCONNECTING state
                 if (getActiveFgCallState() != Call.State.IDLE
-                        && getActiveFgCallState() != Call.State.DISCONNECTED
+                        && getActiveFgCallState() != Call.State.DISCONNECTED) {
-                        && !(fgPhone instanceof SipPhone)) {
+                    if (fgPhone instanceof SipPhone) {
-                    mode = AudioManager.MODE_IN_CALL;
+                        // enable IN_COMMUNICATION audio mode for sipPhone
+                        mode = AudioManager.MODE_IN_COMMUNICATION;
+                    } else {
+                        // enable IN_CALL audio mode for telephony
+                        mode = AudioManager.MODE_IN_CALL;
+                    }
                 }
                 break;
         }

voip/jni/rtp/AudioGroup.cpp

@@ -483,7 +483,7 @@ public:
         ON_HOLD = 0,
         MUTED = 1,
         NORMAL = 2,
-        EC_ENABLED = 3,
+        ECHO_SUPPRESSION = 3,
         LAST_MODE = 3,
     };
 
@@ -619,6 +619,10 @@ bool AudioGroup::setMode(int mode)
     if (mode < 0 || mode > LAST_MODE) {
         return false;
     }
+    if (mode == ECHO_SUPPRESSION && AudioSystem::getParameters(
+        0, String8("ec_supported")) == "ec_supported=yes") {
+        mode = NORMAL;
+    }
     if (mMode == mode) {
         return true;
     }
@@ -775,8 +779,8 @@ bool AudioGroup::DeviceThread::threadLoop()
     AudioTrack track;
     AudioRecord record;
     if (track.set(AudioSystem::VOICE_CALL, sampleRate, AudioSystem::PCM_16_BIT,
-        AudioSystem::CHANNEL_OUT_MONO, output) != NO_ERROR ||
+        AudioSystem::CHANNEL_OUT_MONO, output) != NO_ERROR || record.set(
-        record.set(AUDIO_SOURCE_MIC, sampleRate, AudioSystem::PCM_16_BIT,
+        AUDIO_SOURCE_VOICE_COMMUNICATION, sampleRate, AudioSystem::PCM_16_BIT,
         AudioSystem::CHANNEL_IN_MONO, input) != NO_ERROR) {
         LOGE("cannot initialize audio device");
         return false;