Merge "NuPlayer: open audio sink in offload mode when resuming from timeout teardown." into lmp-mr1-dev
diff --git a/include/media/AudioSystem.h b/include/media/AudioSystem.h
index f3b7fbb..d54eca7 100644
--- a/include/media/AudioSystem.h
+++ b/include/media/AudioSystem.h
@@ -373,10 +373,11 @@
     friend class AudioFlingerClient;
     friend class AudioPolicyServiceClient;
 
-    static Mutex gLock;     // protects all members except gAudioPolicyService,
-                            // gAudioPolicyServiceClient, and gAudioPortCallback
-    static Mutex gLockAPS;  // protects gAudioPolicyService and gAudioPolicyServiceClient
-    static Mutex gLockAPC;  // protects gAudioPortCallback
+    static Mutex gLock;      // protects gAudioFlinger and gAudioErrorCallback,
+    static Mutex gLockCache; // protects gOutputs, gPrevInSamplingRate, gPrevInFormat,
+                             // gPrevInChannelMask and gInBuffSize
+    static Mutex gLockAPS;   // protects gAudioPolicyService and gAudioPolicyServiceClient
+    static Mutex gLockAPC;   // protects gAudioPortCallback
     static sp<IAudioFlinger> gAudioFlinger;
     static audio_error_callback gAudioErrorCallback;
 
diff --git a/media/libmedia/AudioSystem.cpp b/media/libmedia/AudioSystem.cpp
index fce4389..0e608f8 100644
--- a/media/libmedia/AudioSystem.cpp
+++ b/media/libmedia/AudioSystem.cpp
@@ -32,6 +32,7 @@
 
 // client singleton for AudioFlinger binder interface
 Mutex AudioSystem::gLock;
+Mutex AudioSystem::gLockCache;
 Mutex AudioSystem::gLockAPS;
 Mutex AudioSystem::gLockAPC;
 sp<IAudioFlinger> AudioSystem::gAudioFlinger;
@@ -250,20 +251,20 @@
 status_t AudioSystem::getSamplingRate(audio_io_handle_t output,
                                       uint32_t* samplingRate)
 {
-    OutputDescriptor *outputDesc;
+    const sp<IAudioFlinger>& af = AudioSystem::get_audio_flinger();
+    if (af == 0) return PERMISSION_DENIED;
 
-    gLock.lock();
-    outputDesc = AudioSystem::gOutputs.valueFor(output);
+    Mutex::Autolock _l(gLockCache);
+
+    OutputDescriptor *outputDesc = AudioSystem::gOutputs.valueFor(output);
     if (outputDesc == NULL) {
         ALOGV("getOutputSamplingRate() no output descriptor for output %d in gOutputs", output);
-        gLock.unlock();
-        const sp<IAudioFlinger>& af = AudioSystem::get_audio_flinger();
-        if (af == 0) return PERMISSION_DENIED;
+        gLockCache.unlock();
         *samplingRate = af->sampleRate(output);
+        gLockCache.lock();
     } else {
         ALOGV("getOutputSamplingRate() reading from output desc");
         *samplingRate = outputDesc->samplingRate;
-        gLock.unlock();
     }
     if (*samplingRate == 0) {
         ALOGE("AudioSystem::getSamplingRate failed for output %d", output);
@@ -294,18 +295,18 @@
 status_t AudioSystem::getFrameCount(audio_io_handle_t output,
                                     size_t* frameCount)
 {
-    OutputDescriptor *outputDesc;
+    const sp<IAudioFlinger>& af = AudioSystem::get_audio_flinger();
+    if (af == 0) return PERMISSION_DENIED;
 
-    gLock.lock();
-    outputDesc = AudioSystem::gOutputs.valueFor(output);
+    Mutex::Autolock _l(gLockCache);
+
+    OutputDescriptor *outputDesc = AudioSystem::gOutputs.valueFor(output);
     if (outputDesc == NULL) {
-        gLock.unlock();
-        const sp<IAudioFlinger>& af = AudioSystem::get_audio_flinger();
-        if (af == 0) return PERMISSION_DENIED;
+        gLockCache.unlock();
         *frameCount = af->frameCount(output);
+        gLockCache.lock();
     } else {
         *frameCount = outputDesc->frameCount;
-        gLock.unlock();
     }
     if (*frameCount == 0) {
         ALOGE("AudioSystem::getFrameCount failed for output %d", output);
@@ -336,18 +337,18 @@
 status_t AudioSystem::getLatency(audio_io_handle_t output,
                                  uint32_t* latency)
 {
-    OutputDescriptor *outputDesc;
+    const sp<IAudioFlinger>& af = AudioSystem::get_audio_flinger();
+    if (af == 0) return PERMISSION_DENIED;
 
-    gLock.lock();
-    outputDesc = AudioSystem::gOutputs.valueFor(output);
+    Mutex::Autolock _l(gLockCache);
+
+    OutputDescriptor *outputDesc = AudioSystem::gOutputs.valueFor(output);
     if (outputDesc == NULL) {
-        gLock.unlock();
-        const sp<IAudioFlinger>& af = AudioSystem::get_audio_flinger();
-        if (af == 0) return PERMISSION_DENIED;
+        gLockCache.unlock();
         *latency = af->latency(output);
+        gLockCache.lock();
     } else {
         *latency = outputDesc->latency;
-        gLock.unlock();
     }
 
     ALOGV("getLatency() output %d, latency %d", output, *latency);
@@ -358,24 +359,24 @@
 status_t AudioSystem::getInputBufferSize(uint32_t sampleRate, audio_format_t format,
         audio_channel_mask_t channelMask, size_t* buffSize)
 {
-    gLock.lock();
+    const sp<IAudioFlinger>& af = AudioSystem::get_audio_flinger();
+    if (af == 0) {
+        return PERMISSION_DENIED;
+    }
+    Mutex::Autolock _l(gLockCache);
     // Do we have a stale gInBufferSize or are we requesting the input buffer size for new values
     size_t inBuffSize = gInBuffSize;
     if ((inBuffSize == 0) || (sampleRate != gPrevInSamplingRate) || (format != gPrevInFormat)
         || (channelMask != gPrevInChannelMask)) {
-        gLock.unlock();
-        const sp<IAudioFlinger>& af = AudioSystem::get_audio_flinger();
-        if (af == 0) {
-            return PERMISSION_DENIED;
-        }
+        gLockCache.unlock();
         inBuffSize = af->getInputBufferSize(sampleRate, format, channelMask);
+        gLockCache.lock();
         if (inBuffSize == 0) {
             ALOGE("AudioSystem::getInputBufferSize failed sampleRate %d format %#x channelMask %x",
                     sampleRate, format, channelMask);
             return BAD_VALUE;
         }
         // A benign race is possible here: we could overwrite a fresher cache entry
-        gLock.lock();
         // save the request params
         gPrevInSamplingRate = sampleRate;
         gPrevInFormat = format;
@@ -383,7 +384,6 @@
 
         gInBuffSize = inBuffSize;
     }
-    gLock.unlock();
     *buffSize = inBuffSize;
 
     return NO_ERROR;
@@ -450,14 +450,21 @@
 
 void AudioSystem::AudioFlingerClient::binderDied(const wp<IBinder>& who __unused)
 {
-    Mutex::Autolock _l(AudioSystem::gLock);
+    audio_error_callback cb = NULL;
+    {
+        Mutex::Autolock _l(AudioSystem::gLock);
+        AudioSystem::gAudioFlinger.clear();
+        cb = gAudioErrorCallback;
+    }
 
-    AudioSystem::gAudioFlinger.clear();
-    // clear output handles and stream to output map caches
-    AudioSystem::gOutputs.clear();
+    {
+        // clear output handles and stream to output map caches
+        Mutex::Autolock _l(gLockCache);
+        AudioSystem::gOutputs.clear();
+    }
 
-    if (gAudioErrorCallback) {
-        gAudioErrorCallback(DEAD_OBJECT);
+    if (cb) {
+        cb(DEAD_OBJECT);
     }
     ALOGW("AudioFlinger server died!");
 }
@@ -470,7 +477,7 @@
 
     if (ioHandle == AUDIO_IO_HANDLE_NONE) return;
 
-    Mutex::Autolock _l(AudioSystem::gLock);
+    Mutex::Autolock _l(AudioSystem::gLockCache);
 
     switch (event) {
     case STREAM_CONFIG_CHANGED:
@@ -829,8 +836,11 @@
     // called by restoreTrack_l(), which needs new IAudioFlinger and IAudioPolicyService instances
     ALOGV("clearAudioConfigCache()");
     {
-        Mutex::Autolock _l(gLock);
+        Mutex::Autolock _l(gLockCache);
         gOutputs.clear();
+    }
+    {
+        Mutex::Autolock _l(gLock);
         gAudioFlinger.clear();
     }
     {
diff --git a/media/libmedia/AudioTrack.cpp b/media/libmedia/AudioTrack.cpp
index 2f57b9d..c11050e 100644
--- a/media/libmedia/AudioTrack.cpp
+++ b/media/libmedia/AudioTrack.cpp
@@ -2149,6 +2149,11 @@
             mStreamType = AUDIO_STREAM_ALARM;
             break;
         }
+        audio_mode_t phoneState = AudioSystem::getPhoneState();
+        if (phoneState == AUDIO_MODE_IN_CALL || phoneState == AUDIO_MODE_IN_COMMUNICATION) {
+            mStreamType = AUDIO_STREAM_VOICE_CALL;
+            break;
+        }
     }    /// FALL THROUGH
     case AUDIO_USAGE_MEDIA:
     case AUDIO_USAGE_GAME:
diff --git a/media/libmediaplayerservice/nuplayer/GenericSource.cpp b/media/libmediaplayerservice/nuplayer/GenericSource.cpp
index d446cec..e619926 100644
--- a/media/libmediaplayerservice/nuplayer/GenericSource.cpp
+++ b/media/libmediaplayerservice/nuplayer/GenericSource.cpp
@@ -999,11 +999,12 @@
     return -1;
 }
 
-status_t NuPlayer::GenericSource::selectTrack(size_t trackIndex, bool select) {
+status_t NuPlayer::GenericSource::selectTrack(size_t trackIndex, bool select, int64_t timeUs) {
     ALOGV("%s track: %zu", select ? "select" : "deselect", trackIndex);
     sp<AMessage> msg = new AMessage(kWhatSelectTrack, id());
     msg->setInt32("trackIndex", trackIndex);
     msg->setInt32("select", select);
+    msg->setInt64("timeUs", timeUs);
 
     sp<AMessage> response;
     status_t err = msg->postAndAwaitResponse(&response);
@@ -1016,11 +1017,13 @@
 
 void NuPlayer::GenericSource::onSelectTrack(sp<AMessage> msg) {
     int32_t trackIndex, select;
+    int64_t timeUs;
     CHECK(msg->findInt32("trackIndex", &trackIndex));
     CHECK(msg->findInt32("select", &select));
+    CHECK(msg->findInt64("timeUs", &timeUs));
 
     sp<AMessage> response = new AMessage;
-    status_t err = doSelectTrack(trackIndex, select);
+    status_t err = doSelectTrack(trackIndex, select, timeUs);
     response->setInt32("err", err);
 
     uint32_t replyID;
@@ -1028,7 +1031,7 @@
     response->postReply(replyID);
 }
 
-status_t NuPlayer::GenericSource::doSelectTrack(size_t trackIndex, bool select) {
+status_t NuPlayer::GenericSource::doSelectTrack(size_t trackIndex, bool select, int64_t timeUs) {
     if (trackIndex >= mSources.size()) {
         return BAD_INDEX;
     }
@@ -1081,6 +1084,23 @@
             mFetchTimedTextDataGeneration++;
         }
 
+        status_t eosResult; // ignored
+        if (mSubtitleTrack.mSource != NULL
+                && !mSubtitleTrack.mPackets->hasBufferAvailable(&eosResult)) {
+            sp<AMessage> msg = new AMessage(kWhatFetchSubtitleData, id());
+            msg->setInt64("timeUs", timeUs);
+            msg->setInt32("generation", mFetchSubtitleDataGeneration);
+            msg->post();
+        }
+
+        if (mTimedTextTrack.mSource != NULL
+                && !mTimedTextTrack.mPackets->hasBufferAvailable(&eosResult)) {
+            sp<AMessage> msg = new AMessage(kWhatFetchTimedTextData, id());
+            msg->setInt64("timeUs", timeUs);
+            msg->setInt32("generation", mFetchTimedTextDataGeneration);
+            msg->post();
+        }
+
         return OK;
     } else if (!strncasecmp(mime, "audio/", 6) || !strncasecmp(mime, "video/", 6)) {
         bool audio = !strncasecmp(mime, "audio/", 6);
diff --git a/media/libmediaplayerservice/nuplayer/GenericSource.h b/media/libmediaplayerservice/nuplayer/GenericSource.h
index 7a03df0..f2528a9 100644
--- a/media/libmediaplayerservice/nuplayer/GenericSource.h
+++ b/media/libmediaplayerservice/nuplayer/GenericSource.h
@@ -67,7 +67,7 @@
     virtual size_t getTrackCount() const;
     virtual sp<AMessage> getTrackInfo(size_t trackIndex) const;
     virtual ssize_t getSelectedTrack(media_track_type type) const;
-    virtual status_t selectTrack(size_t trackIndex, bool select);
+    virtual status_t selectTrack(size_t trackIndex, bool select, int64_t timeUs);
     virtual status_t seekTo(int64_t seekTimeUs);
 
     virtual status_t setBuffers(bool audio, Vector<MediaBuffer *> &buffers);
@@ -164,7 +164,7 @@
     ssize_t doGetSelectedTrack(media_track_type type) const;
 
     void onSelectTrack(sp<AMessage> msg);
-    status_t doSelectTrack(size_t trackIndex, bool select);
+    status_t doSelectTrack(size_t trackIndex, bool select, int64_t timeUs);
 
     void onSeek(sp<AMessage> msg);
     status_t doSeek(int64_t seekTimeUs);
diff --git a/media/libmediaplayerservice/nuplayer/HTTPLiveSource.cpp b/media/libmediaplayerservice/nuplayer/HTTPLiveSource.cpp
index e335055..a26ef9e 100644
--- a/media/libmediaplayerservice/nuplayer/HTTPLiveSource.cpp
+++ b/media/libmediaplayerservice/nuplayer/HTTPLiveSource.cpp
@@ -147,7 +147,7 @@
     }
 }
 
-status_t NuPlayer::HTTPLiveSource::selectTrack(size_t trackIndex, bool select) {
+status_t NuPlayer::HTTPLiveSource::selectTrack(size_t trackIndex, bool select, int64_t /*timeUs*/) {
     status_t err = mLiveSession->selectTrack(trackIndex, select);
 
     if (err == OK) {
diff --git a/media/libmediaplayerservice/nuplayer/HTTPLiveSource.h b/media/libmediaplayerservice/nuplayer/HTTPLiveSource.h
index 98b826e..bbb8981 100644
--- a/media/libmediaplayerservice/nuplayer/HTTPLiveSource.h
+++ b/media/libmediaplayerservice/nuplayer/HTTPLiveSource.h
@@ -43,7 +43,7 @@
     virtual size_t getTrackCount() const;
     virtual sp<AMessage> getTrackInfo(size_t trackIndex) const;
     virtual ssize_t getSelectedTrack(media_track_type /* type */) const;
-    virtual status_t selectTrack(size_t trackIndex, bool select);
+    virtual status_t selectTrack(size_t trackIndex, bool select, int64_t timeUs);
     virtual status_t seekTo(int64_t seekTimeUs);
 
 protected:
diff --git a/media/libmediaplayerservice/nuplayer/NuPlayer.cpp b/media/libmediaplayerservice/nuplayer/NuPlayer.cpp
index 336a9c5..59b070c 100644
--- a/media/libmediaplayerservice/nuplayer/NuPlayer.cpp
+++ b/media/libmediaplayerservice/nuplayer/NuPlayer.cpp
@@ -453,8 +453,10 @@
 
             size_t trackIndex;
             int32_t select;
+            int64_t timeUs;
             CHECK(msg->findSize("trackIndex", &trackIndex));
             CHECK(msg->findInt32("select", &select));
+            CHECK(msg->findInt64("timeUs", &timeUs));
 
             status_t err = INVALID_OPERATION;
 
@@ -468,7 +470,7 @@
             }
 
             if (trackIndex < inbandTracks) {
-                err = mSource->selectTrack(trackIndex, select);
+                err = mSource->selectTrack(trackIndex, select, timeUs);
 
                 if (!select && err == OK) {
                     int32_t type;
@@ -1633,10 +1635,11 @@
     return err;
 }
 
-status_t NuPlayer::selectTrack(size_t trackIndex, bool select) {
+status_t NuPlayer::selectTrack(size_t trackIndex, bool select, int64_t timeUs) {
     sp<AMessage> msg = new AMessage(kWhatSelectTrack, id());
     msg->setSize("trackIndex", trackIndex);
     msg->setInt32("select", select);
+    msg->setInt64("timeUs", timeUs);
 
     sp<AMessage> response;
     status_t err = msg->postAndAwaitResponse(&response);
diff --git a/media/libmediaplayerservice/nuplayer/NuPlayer.h b/media/libmediaplayerservice/nuplayer/NuPlayer.h
index 5f6deee..901cfbd 100644
--- a/media/libmediaplayerservice/nuplayer/NuPlayer.h
+++ b/media/libmediaplayerservice/nuplayer/NuPlayer.h
@@ -65,7 +65,7 @@
     status_t setVideoScalingMode(int32_t mode);
     status_t getTrackInfo(Parcel* reply) const;
     status_t getSelectedTrack(int32_t type, Parcel* reply) const;
-    status_t selectTrack(size_t trackIndex, bool select);
+    status_t selectTrack(size_t trackIndex, bool select, int64_t timeUs);
     status_t getCurrentPosition(int64_t *mediaUs);
     void getStats(int64_t *mNumFramesTotal, int64_t *mNumFramesDropped);
 
diff --git a/media/libmediaplayerservice/nuplayer/NuPlayerDriver.cpp b/media/libmediaplayerservice/nuplayer/NuPlayerDriver.cpp
index b42b480..e09567a 100644
--- a/media/libmediaplayerservice/nuplayer/NuPlayerDriver.cpp
+++ b/media/libmediaplayerservice/nuplayer/NuPlayerDriver.cpp
@@ -480,13 +480,16 @@
         case INVOKE_ID_SELECT_TRACK:
         {
             int trackIndex = request.readInt32();
-            return mPlayer->selectTrack(trackIndex, true /* select */);
+            int msec = 0;
+            // getCurrentPosition should always return OK
+            getCurrentPosition(&msec);
+            return mPlayer->selectTrack(trackIndex, true /* select */, msec * 1000ll);
         }
 
         case INVOKE_ID_UNSELECT_TRACK:
         {
             int trackIndex = request.readInt32();
-            return mPlayer->selectTrack(trackIndex, false /* select */);
+            return mPlayer->selectTrack(trackIndex, false /* select */, 0xdeadbeef /* not used */);
         }
 
         case INVOKE_ID_GET_SELECTED_TRACK:
@@ -625,6 +628,16 @@
         case MEDIA_PLAYBACK_COMPLETE:
         {
             if (mState != STATE_RESET_IN_PROGRESS) {
+                if (mAutoLoop) {
+                    audio_stream_type_t streamType = AUDIO_STREAM_MUSIC;
+                    if (mAudioSink != NULL) {
+                        streamType = mAudioSink->getAudioStreamType();
+                    }
+                    if (streamType == AUDIO_STREAM_NOTIFICATION) {
+                        ALOGW("disabling auto-loop for notification");
+                        mAutoLoop = false;
+                    }
+                }
                 if (mLooping || (mAutoLoop
                         && (mAudioSink == NULL || mAudioSink->realtime()))) {
                     mPlayer->seekToAsync(0);
diff --git a/media/libmediaplayerservice/nuplayer/NuPlayerSource.h b/media/libmediaplayerservice/nuplayer/NuPlayerSource.h
index 2f06c31..2b0ac47 100644
--- a/media/libmediaplayerservice/nuplayer/NuPlayerSource.h
+++ b/media/libmediaplayerservice/nuplayer/NuPlayerSource.h
@@ -97,7 +97,7 @@
         return INVALID_OPERATION;
     }
 
-    virtual status_t selectTrack(size_t /* trackIndex */, bool /* select */) {
+    virtual status_t selectTrack(size_t /* trackIndex */, bool /* select */, int64_t /* timeUs*/) {
         return INVALID_OPERATION;
     }
 
diff --git a/media/libstagefright/AwesomePlayer.cpp b/media/libstagefright/AwesomePlayer.cpp
index 6a56729..007c090 100644
--- a/media/libstagefright/AwesomePlayer.cpp
+++ b/media/libstagefright/AwesomePlayer.cpp
@@ -878,6 +878,16 @@
         return;
     }
 
+    if (mFlags & AUTO_LOOPING) {
+        audio_stream_type_t streamType = AUDIO_STREAM_MUSIC;
+        if (mAudioSink != NULL) {
+            streamType = mAudioSink->getAudioStreamType();
+        }
+        if (streamType == AUDIO_STREAM_NOTIFICATION) {
+            ALOGW("disabling auto-loop for notification");
+            modifyFlags(AUTO_LOOPING, CLEAR);
+        }
+    }
     if ((mFlags & LOOPING)
             || ((mFlags & AUTO_LOOPING)
                 && (mAudioSink == NULL || mAudioSink->realtime()))) {
diff --git a/media/libstagefright/httplive/PlaylistFetcher.cpp b/media/libstagefright/httplive/PlaylistFetcher.cpp
index e247550..d8eed5b 100644
--- a/media/libstagefright/httplive/PlaylistFetcher.cpp
+++ b/media/libstagefright/httplive/PlaylistFetcher.cpp
@@ -826,6 +826,18 @@
                   "  mStartup=%d, was  looking for %d in %d-%d",
                     mStartup, mSeqNumber, firstSeqNumberInPlaylist,
                     lastSeqNumberInPlaylist);
+            if (mStopParams != NULL) {
+                // we should have kept on fetching until we hit the boundaries in mStopParams,
+                // but since the segments we are supposed to fetch have already rolled off
+                // the playlist, i.e. we have already missed the boat, we inevitably have to
+                // skip.
+                for (size_t i = 0; i < mPacketSources.size(); i++) {
+                    sp<ABuffer> formatChange = mSession->createFormatChangeBuffer();
+                    mPacketSources.valueAt(i)->queueAccessUnit(formatChange);
+                }
+                stopAsync(/* clear = */ false);
+                return;
+            }
             mSeqNumber = lastSeqNumberInPlaylist - 3;
             if (mSeqNumber < firstSeqNumberInPlaylist) {
                 mSeqNumber = firstSeqNumberInPlaylist;
@@ -1266,6 +1278,11 @@
 
             CHECK(accessUnit->meta()->findInt64("timeUs", &timeUs));
             if (mStartTimeUsNotify != NULL && timeUs > mStartTimeUs) {
+                int32_t firstSeqNumberInPlaylist;
+                if (mPlaylist->meta() == NULL || !mPlaylist->meta()->findInt32(
+                            "media-sequence", &firstSeqNumberInPlaylist)) {
+                    firstSeqNumberInPlaylist = 0;
+                }
 
                 int32_t targetDurationSecs;
                 CHECK(mPlaylist->meta()->findInt32("target-duration", &targetDurationSecs));
@@ -1276,6 +1293,8 @@
                 //   mStartTimeUs.
                 // mSegmentStartTimeUs >= 0
                 //   mSegmentStartTimeUs is non-negative when adapting or switching tracks
+                // mSeqNumber > firstSeqNumberInPlaylist
+                //   don't decrement mSeqNumber if it already points to the 1st segment
                 // timeUs - mStartTimeUs > targetDurationUs:
                 //   This and the 2 above conditions should only happen when adapting in a live
                 //   stream; the old fetcher has already fetched to mStartTimeUs; the new fetcher
@@ -1285,6 +1304,7 @@
                 //   stop as early as possible. The definition of being "too far ahead" is
                 //   arbitrary; here we use targetDurationUs as threshold.
                 if (mStartup && mSegmentStartTimeUs >= 0
+                        && mSeqNumber > firstSeqNumberInPlaylist
                         && timeUs - mStartTimeUs > targetDurationUs) {
                     // we just guessed a starting timestamp that is too high when adapting in a
                     // live stream; re-adjust based on the actual timestamp extracted from the
diff --git a/services/audiopolicy/AudioPolicyManager.cpp b/services/audiopolicy/AudioPolicyManager.cpp
index fff7746..20a72b0 100644
--- a/services/audiopolicy/AudioPolicyManager.cpp
+++ b/services/audiopolicy/AudioPolicyManager.cpp
@@ -3983,9 +3983,17 @@
 
     // usage to strategy mapping
     switch (attr->usage) {
+    case AUDIO_USAGE_ASSISTANCE_ACCESSIBILITY:
+        if (isStreamActive(AUDIO_STREAM_RING) || isStreamActive(AUDIO_STREAM_ALARM)) {
+            return (uint32_t) STRATEGY_SONIFICATION;
+        }
+        if (isInCall()) {
+            return (uint32_t) STRATEGY_PHONE;
+        }
+        // FALL THROUGH
+
     case AUDIO_USAGE_MEDIA:
     case AUDIO_USAGE_GAME:
-    case AUDIO_USAGE_ASSISTANCE_ACCESSIBILITY:
     case AUDIO_USAGE_ASSISTANCE_NAVIGATION_GUIDANCE:
     case AUDIO_USAGE_ASSISTANCE_SONIFICATION:
         return (uint32_t) STRATEGY_MEDIA;
@@ -4185,7 +4193,7 @@
             // when not in a phone call, phone strategy should route STREAM_VOICE_CALL to A2DP
             if (!isInCall() &&
                     (mForceUse[AUDIO_POLICY_FORCE_FOR_MEDIA] != AUDIO_POLICY_FORCE_NO_BT_A2DP) &&
-                    (getA2dpOutput() != 0) && !mA2dpSuspended) {
+                    (getA2dpOutput() != 0)) {
                 device = availableOutputDeviceTypes & AUDIO_DEVICE_OUT_BLUETOOTH_A2DP;
                 if (device) break;
                 device = availableOutputDeviceTypes & AUDIO_DEVICE_OUT_BLUETOOTH_A2DP_HEADPHONES;
@@ -4220,7 +4228,7 @@
             // A2DP speaker when forcing to speaker output
             if (!isInCall() &&
                     (mForceUse[AUDIO_POLICY_FORCE_FOR_MEDIA] != AUDIO_POLICY_FORCE_NO_BT_A2DP) &&
-                    (getA2dpOutput() != 0) && !mA2dpSuspended) {
+                    (getA2dpOutput() != 0)) {
                 device = availableOutputDeviceTypes & AUDIO_DEVICE_OUT_BLUETOOTH_A2DP_SPEAKER;
                 if (device) break;
             }
@@ -4282,7 +4290,7 @@
         }
         if ((device2 == AUDIO_DEVICE_NONE) &&
                 (mForceUse[AUDIO_POLICY_FORCE_FOR_MEDIA] != AUDIO_POLICY_FORCE_NO_BT_A2DP) &&
-                (getA2dpOutput() != 0) && !mA2dpSuspended) {
+                (getA2dpOutput() != 0)) {
             device2 = availableOutputDeviceTypes & AUDIO_DEVICE_OUT_BLUETOOTH_A2DP;
             if (device2 == AUDIO_DEVICE_NONE) {
                 device2 = availableOutputDeviceTypes & AUDIO_DEVICE_OUT_BLUETOOTH_A2DP_HEADPHONES;
diff --git a/services/camera/libcameraservice/api1/Camera2Client.cpp b/services/camera/libcameraservice/api1/Camera2Client.cpp
index f3a88a1..6e7824e 100644
--- a/services/camera/libcameraservice/api1/Camera2Client.cpp
+++ b/services/camera/libcameraservice/api1/Camera2Client.cpp
@@ -929,13 +929,6 @@
                         "stop preview: %s (%d)",
                         __FUNCTION__, mCameraId, strerror(-res), res);
             }
-            {
-                // Ideally we should recover the override after recording stopped, but
-                // right now recording stream will live until here, so we are forced to
-                // recover here. TODO: find a better way to handle that (b/17495165)
-                SharedParameters::Lock l(mParameters);
-                l.mParameters.recoverOverriddenJpegSize();
-            }
             // no break
         case Parameters::WAITING_FOR_PREVIEW_WINDOW: {
             SharedParameters::Lock l(mParameters);
@@ -1206,6 +1199,28 @@
 
     mCameraService->playSound(CameraService::SOUND_RECORDING);
 
+    // Remove recording stream to prevent it from slowing down takePicture later
+    if (!l.mParameters.recordingHint && l.mParameters.isJpegSizeOverridden()) {
+        res = stopStream();
+        if (res != OK) {
+            ALOGE("%s: Camera %d: Can't stop streaming: %s (%d)",
+                    __FUNCTION__, mCameraId, strerror(-res), res);
+        }
+        res = mDevice->waitUntilDrained();
+        if (res != OK) {
+            ALOGE("%s: Camera %d: Waiting to stop streaming failed: %s (%d)",
+                    __FUNCTION__, mCameraId, strerror(-res), res);
+        }
+        // Clean up recording stream
+        res = mStreamingProcessor->deleteRecordingStream();
+        if (res != OK) {
+            ALOGE("%s: Camera %d: Unable to delete recording stream before "
+                    "stop preview: %s (%d)",
+                    __FUNCTION__, mCameraId, strerror(-res), res);
+        }
+        l.mParameters.recoverOverriddenJpegSize();
+    }
+
     res = startPreviewL(l.mParameters, true);
     if (res != OK) {
         ALOGE("%s: Camera %d: Unable to return to preview",
@@ -1388,6 +1403,34 @@
                     return res;
                 }
                 l.mParameters.state = Parameters::STILL_CAPTURE;
+
+                // Remove recording stream to prevent video snapshot jpeg logic kicking in
+                if (l.mParameters.isJpegSizeOverridden() &&
+                        mStreamingProcessor->getRecordingStreamId() != NO_STREAM) {
+                    res = mStreamingProcessor->togglePauseStream(/*pause*/true);
+                    if (res != OK) {
+                        ALOGE("%s: Camera %d: Can't pause streaming: %s (%d)",
+                                __FUNCTION__, mCameraId, strerror(-res), res);
+                    }
+                    res = mDevice->waitUntilDrained();
+                    if (res != OK) {
+                        ALOGE("%s: Camera %d: Waiting to stop streaming failed: %s (%d)",
+                                __FUNCTION__, mCameraId, strerror(-res), res);
+                    }
+                    // Clean up recording stream
+                    res = mStreamingProcessor->deleteRecordingStream();
+                    if (res != OK) {
+                        ALOGE("%s: Camera %d: Unable to delete recording stream before "
+                                "stop preview: %s (%d)",
+                                __FUNCTION__, mCameraId, strerror(-res), res);
+                    }
+                    res = mStreamingProcessor->togglePauseStream(/*pause*/false);
+                    if (res != OK) {
+                        ALOGE("%s: Camera %d: Can't unpause streaming: %s (%d)",
+                                __FUNCTION__, mCameraId, strerror(-res), res);
+                    }
+                    l.mParameters.recoverOverriddenJpegSize();
+                }
                 break;
             case Parameters::RECORD:
                 // Good to go for video snapshot
diff --git a/services/camera/libcameraservice/api1/client2/Parameters.cpp b/services/camera/libcameraservice/api1/client2/Parameters.cpp
index 74bbb9d..234247b 100644
--- a/services/camera/libcameraservice/api1/client2/Parameters.cpp
+++ b/services/camera/libcameraservice/api1/client2/Parameters.cpp
@@ -2203,6 +2203,10 @@
     return OK;
 }
 
+bool Parameters::isJpegSizeOverridden() {
+    return pictureSizeOverriden;
+}
+
 const char* Parameters::getStateName(State state) {
 #define CASE_ENUM_TO_CHAR(x) case x: return(#x); break;
     switch(state) {
diff --git a/services/camera/libcameraservice/api1/client2/Parameters.h b/services/camera/libcameraservice/api1/client2/Parameters.h
index 389cb92..7e5be84 100644
--- a/services/camera/libcameraservice/api1/client2/Parameters.h
+++ b/services/camera/libcameraservice/api1/client2/Parameters.h
@@ -266,6 +266,8 @@
     status_t overrideJpegSizeByVideoSize();
     // Recover overridden jpeg size.  Called during stopRecording.
     status_t recoverOverriddenJpegSize();
+    // if video snapshot size is currently overridden
+    bool isJpegSizeOverridden();
 
     // Calculate the crop region rectangle based on current stream sizes
     struct CropRegion {