Merge "Issue 2667801: [Audio Effect Framework] AudioFlinger, AudioMixer AudioTrack modifications." into kraken

include/media/AudioRecord.h

@@ -142,7 +142,8 @@ public:
                                     uint32_t flags      = 0,
                                     callback_t cbf = 0,
                                     void* user = 0,
-                                    int notificationFrames = 0);
+                                    int notificationFrames = 0,
+                                    int sessionId = 0);
 
 
     /* Terminates the AudioRecord and unregisters it from AudioFlinger.
@@ -168,7 +169,8 @@ public:
                             callback_t cbf = 0,
                             void* user = 0,
                             int notificationFrames = 0,
-                            bool threadCanCallJava = false);
+                            bool threadCanCallJava = false,
+                            int sessionId = 0);
 
 
     /* Result of constructing the AudioRecord. This must be checked
@@ -270,6 +272,16 @@ public:
      */
             audio_io_handle_t    getInput();
 
+    /* returns the audio session ID associated to this AudioRecord.
+     *
+     * Parameters:
+     *  none.
+     *
+     * Returned value:
+     *  AudioRecord session ID.
+     */
+            int    getSessionId();
+
     /* obtains a buffer of "frameCount" frames. The buffer must be
      * filled entirely. If the track is stopped, obtainBuffer() returns
      * STOPPED instead of NO_ERROR as long as there are buffers availlable,
@@ -356,6 +368,7 @@ private:
     uint32_t                mFlags;
     uint32_t                mChannels;
     audio_io_handle_t       mInput;
+    int                     mSessionId;
 };
 
 }; // namespace android

include/media/AudioSystem.h

@@ -230,6 +230,8 @@ public:
     static status_t getRenderPosition(uint32_t *halFrames, uint32_t *dspFrames, int stream = DEFAULT);
 
     static unsigned int  getInputFramesLost(audio_io_handle_t ioHandle);
+
+    static int newAudioSessionId();
     //
     // AudioPolicyService interface
     //

include/media/AudioTrack.h

@@ -138,7 +138,8 @@ public:
                                     uint32_t flags       = 0,
                                     callback_t cbf       = 0,
                                     void* user           = 0,
-                                    int notificationFrames = 0);
+                                    int notificationFrames = 0,
+                                    int sessionId = 0);
 
     /* Creates an audio track and registers it with AudioFlinger. With this constructor,
      * The PCM data to be rendered by AudioTrack is passed in a shared memory buffer
@@ -157,7 +158,8 @@ public:
                                     uint32_t flags      = 0,
                                     callback_t cbf      = 0,
                                     void* user          = 0,
-                                    int notificationFrames = 0);
+                                    int notificationFrames = 0,
+                                    int sessionId = 0);
 
     /* Terminates the AudioTrack and unregisters it from AudioFlinger.
      * Also destroys all resources assotiated with the AudioTrack.
@@ -182,7 +184,8 @@ public:
                             void* user          = 0,
                             int notificationFrames = 0,
                             const sp<IMemory>& sharedBuffer = 0,
-                            bool threadCanCallJava = false);
+                            bool threadCanCallJava = false,
+                            int sessionId = 0);
 
 
     /* Result of constructing the AudioTrack. This must be checked
@@ -239,10 +242,17 @@ public:
 
 
     /* set volume for this track, mostly used for games' sound effects
+     * left and right volumes. Levels must be <= 1.0.
      */
-            void        setVolume(float left, float right);
+            status_t    setVolume(float left, float right);
             void        getVolume(float* left, float* right);
 
+    /* set the send level for this track. An auxiliary effect should be attached
+     * to the track with attachEffect(). Level must be <= 1.0.
+     */
+            status_t    setSendLevel(float level);
+            void        getSendLevel(float* level);
+
     /* set sample rate for this track, mostly used for games' sound effects
      */
             status_t    setSampleRate(int sampleRate);
@@ -340,6 +350,31 @@ public:
      */
             audio_io_handle_t    getOutput();
 
+    /* returns the unique ID associated to this track.
+     *
+     * Parameters:
+     *  none.
+     *
+     * Returned value:
+     *  AudioTrack ID.
+     */
+            int    getSessionId();
+
+
+    /* Attach track auxiliary output to specified effect. Used effectId = 0
+     * to detach track from effect.
+     *
+     * Parameters:
+     *
+     * effectId:  effectId obtained from AudioEffect::id().
+     *
+     * Returned status (from utils/Errors.h) can be:
+     *  - NO_ERROR: successful operation
+     *  - INVALID_OPERATION: the effect is not an auxiliary effect.
+     *  - BAD_VALUE: The specified effect ID is invalid
+     */
+            status_t    attachAuxEffect(int effectId);
+
     /* obtains a buffer of "frameCount" frames. The buffer must be
      * filled entirely. If the track is stopped, obtainBuffer() returns
      * STOPPED instead of NO_ERROR as long as there are buffers availlable,
@@ -406,6 +441,7 @@ private:
     sp<AudioTrackThread>    mAudioTrackThread;
 
     float                   mVolume[2];
+    float                   mSendLevel;
     uint32_t                mFrameCount;
 
     audio_track_cblk_t*     mCblk;
@@ -431,6 +467,7 @@ private:
     uint32_t                mNewPosition;
     uint32_t                mUpdatePeriod;
     uint32_t                mFlags;
+    int                     mSessionId;
 };
 
 

include/media/EffectApi.h

@@ -114,7 +114,8 @@ typedef struct effect_descriptor_s {
 //  +---------------------------+-----------+-----------------------------------
 //  | Volume management         | 5..6      | 0 none
 //  |                           |           | 1 implements volume control
-//  |                           |           | 2..3 reserved
+//  |                           |           | 2 requires volume indication
+//  |                           |           | 3 reserved
 //  +---------------------------+-----------+-----------------------------------
 //  | Device management         | 7..8      | 0 none
 //  |                           |           | 1 requires device updates
@@ -154,6 +155,7 @@ typedef struct effect_descriptor_s {
 // volume control
 #define EFFECT_FLAG_VOLUME_MASK         0x00000060
 #define EFFECT_FLAG_VOLUME_CTRL         0x00000020
+#define EFFECT_FLAG_VOLUME_IND          0x00000040
 #define EFFECT_FLAG_VOLUME_NONE         0x00000000
 
 // device control
@@ -296,10 +298,12 @@ struct effect_interface_s {
 //  | Set and get volume. Used by    | EFFECT_CMD_SET_VOLUME         | size: n * sizeof(uint32_t)    | size: n * sizeof(uint32_t)
 //  | audio framework to delegate    |                               | data: volume for each channel | data: volume for each channel
 //  | volume control to effect engine|                               | defined in effect_config_t in | defined in effect_config_t in
-//  | The engine must return the     |                               | 8.24 fixed point format       | 8.24 fixed point format
+//  | If volume control flag is set  |                               | 8.24 fixed point format       | 8.24 fixed point format
-//  | volume that should be applied  |                               |                               |
+//  | in the effect descriptor, the  |                               |                               | It is legal to receive a null
-//  | before the effect is processed |                               |                               |
+//  | effect engine must return the  |                               |                               | pointer as pReplyData in which
-//  | The overall volume (the volume |                               |                               |
+//  | volume that should be applied  |                               |                               | case the effect framework has
+//  | before the effect is processed |                               |                               | delegated volume control to
+//  | The overall volume (the volume |                               |                               | another effect.
 //  | actually applied by the effect |                               |                               |
 //  | multiplied by the returned     |                               |                               |
 //  | value) should match the        |                               |                               |
@@ -370,7 +374,7 @@ typedef struct buffer_provider_s {
 // structure that defines both input and output buffer configurations and is
 // passed by the EFFECT_CMD_CONFIGURE command.
 typedef struct buffer_config_s {
-    audio_buffer_t  buffer;     // buffer for use by process() function is not passed explicitly
+    audio_buffer_t  buffer;     // buffer for use by process() function if not passed explicitly
     uint32_t   samplingRate;    // sampling rate
     uint32_t   channels;        // channel mask (see audio_channels_e in AudioCommon.h)
     buffer_provider_t bufferProvider;   // buffer provider
@@ -457,7 +461,7 @@ typedef struct effect_param_s {
 //
 //    Function:       EffectQueryNumberEffects
 //
-//    Description:    Returns the number of different effect exposed by the
+//    Description:    Returns the number of different effects exposed by the
 //          library. Each effect must have a unique effect uuid (see
 //          effect_descriptor_t). This function together with EffectQueryNext()
 //          is used to enumerate all effects present in the library.
@@ -475,7 +479,7 @@ typedef struct effect_param_s {
 //        *pNumEffects:     updated with number of effects in library
 //
 ////////////////////////////////////////////////////////////////////////////////
-typedef int32_t (*effect_QueryNumberEffects_t)(int32_t *pNumEffects);
+typedef int32_t (*effect_QueryNumberEffects_t)(uint32_t *pNumEffects);
 
 ////////////////////////////////////////////////////////////////////////////////
 //
@@ -521,7 +525,7 @@ typedef int32_t (*effect_QueryNextEffect_t)(effect_descriptor_t *pDescriptor);
 //        returned value:    0          successful operation.
 //                          -ENODEV     library failed to initialize
 //                          -EINVAL     invalid pEffectUuid or pInterface
-//                          -ENOENT     No effect with this uuid found
+//                          -ENOENT     no effect with this uuid found
 //        *pInterface:     updated with the effect interface handle.
 //
 ////////////////////////////////////////////////////////////////////////////////

include/media/EffectFactoryApi.h

@@ -34,7 +34,7 @@ extern "C" {
 //
 //    Function:       EffectQueryNumberEffects
 //
-//    Description:    Returns the number of different effect in all loaded libraries.
+//    Description:    Returns the number of different effects in all loaded libraries.
 //          Each effect must have a different effect uuid (see
 //          effect_descriptor_t). This function together with EffectQueryNext()
 //          is used to enumerate all effects present in all loaded libraries.
@@ -52,7 +52,7 @@ extern "C" {
 //        *pNumEffects:     updated with number of effects in factory
 //
 ////////////////////////////////////////////////////////////////////////////////
-int EffectQueryNumberEffects(int *pNumEffects);
+int EffectQueryNumberEffects(uint32_t *pNumEffects);
 
 ////////////////////////////////////////////////////////////////////////////////
 //
@@ -98,7 +98,7 @@ int EffectQueryNext(effect_descriptor_t *pDescriptor);
 //        returned value:    0          successful operation.
 //                          -ENODEV     factory failed to initialize
 //                          -EINVAL     invalid pEffectUuid or pInterface
-//                          -ENOENT     No effect with this uuid found
+//                          -ENOENT     no effect with this uuid found
 //        *pInterface:     updated with the effect interface.
 //
 ////////////////////////////////////////////////////////////////////////////////
@@ -140,7 +140,7 @@ int EffectRelease(effect_interface_t interface);
 //
 //    Output:
 //        returned value:    0          successful operation.
-//                          -ENODEV     Effect factory not initialized or
+//                          -ENODEV     effect factory not initialized or
 //                                      library could not be loaded or
 //                                      library does not implement required functions
 //                          -EINVAL     invalid libPath string or handle
@@ -159,7 +159,7 @@ int EffectLoadLibrary(const char *libPath, int *handle);
 //
 //    Output:
 //        returned value:    0          successful operation.
-//                          -ENODEV     Effect factory not initialized
+//                          -ENODEV     effect factory not initialized
 //                          -ENOENT     invalid handle
 //
 ////////////////////////////////////////////////////////////////////////////////
@@ -184,7 +184,7 @@ int EffectUnloadLibrary(int handle);
 //        returned value:    0          successful operation.
 //                          -ENODEV     factory failed to initialize
 //                          -EINVAL     invalid pEffectUuid or pDescriptor
-//                          -ENOENT     No effect with this uuid found
+//                          -ENOENT     no effect with this uuid found
 //        *pDescriptor:     updated with the effect descriptor.
 //
 ////////////////////////////////////////////////////////////////////////////////

include/media/IAudioFlinger.h

@@ -27,6 +27,9 @@
 #include <media/IAudioTrack.h>
 #include <media/IAudioRecord.h>
 #include <media/IAudioFlingerClient.h>
+#include <media/EffectApi.h>
+#include <media/IEffect.h>
+#include <media/IEffectClient.h>
 #include <utils/String8.h>
 
 namespace android {
@@ -51,6 +54,7 @@ public:
                                 uint32_t flags,
                                 const sp<IMemory>& sharedBuffer,
                                 int output,
+                                int *sessionId,
                                 status_t *status) = 0;
 
     virtual sp<IAudioRecord> openRecord(
@@ -61,6 +65,7 @@ public:
                                 int channelCount,
                                 int frameCount,
                                 uint32_t flags,
+                                int *sessionId,
                                 status_t *status) = 0;
 
     /* query the audio hardware state. This state never changes,
@@ -134,6 +139,28 @@ public:
     virtual status_t getRenderPosition(uint32_t *halFrames, uint32_t *dspFrames, int output) = 0;
 
     virtual unsigned int  getInputFramesLost(int ioHandle) = 0;
+
+    virtual int newAudioSessionId() = 0;
+
+    virtual status_t loadEffectLibrary(const char *libPath, int *handle) = 0;
+
+    virtual status_t unloadEffectLibrary(int handle) = 0;
+
+    virtual status_t queryNumberEffects(uint32_t *numEffects) = 0;
+
+    virtual status_t queryNextEffect(effect_descriptor_t *pDescriptor) = 0;
+
+    virtual status_t getEffectDescriptor(effect_uuid_t *pEffectUUID, effect_descriptor_t *pDescriptor) = 0;
+
+    virtual sp<IEffect> createEffect(pid_t pid,
+                                    effect_descriptor_t *pDesc,
+                                    const sp<IEffectClient>& client,
+                                    int32_t priority,
+                                    int output,
+                                    int sessionId,
+                                    status_t *status,
+                                    int *id,
+                                    int *enabled) = 0;
 };
 
 

include/media/IAudioTrack.h

@@ -62,6 +62,11 @@ public:
      */
     virtual void        pause() = 0;
 
+    /* Attach track auxiliary output to specified effect. Use effectId = 0
+     * to detach track from effect.
+     */
+    virtual status_t    attachAuxEffect(int effectId) = 0;
+
     /* get this tracks control block */
     virtual sp<IMemory> getCblk() const = 0;    
 };

include/private/media/AudioEffectShared.h

@@ -0,0 +1,51 @@
+/*
+ * Copyright (C) 2010 The Android Open Source Project
+ *
+ * Licensed under the Apache License, Version 2.0 (the "License");
+ * you may not use this file except in compliance with the License.
+ * You may obtain a copy of the License at
+ *
+ *      http://www.apache.org/licenses/LICENSE-2.0
+ *
+ * Unless required by applicable law or agreed to in writing, software
+ * distributed under the License is distributed on an "AS IS" BASIS,
+ * WITHOUT WARRANTIES OR CONDITIONS OF ANY KIND, either express or implied.
+ * See the License for the specific language governing permissions and
+ * limitations under the License.
+ */
+
+#ifndef ANDROID_EFFECTCBASESHARED_H
+#define ANDROID_EFFECTCBASESHARED_H
+
+#include <stdint.h>
+#include <sys/types.h>
+
+#include <utils/threads.h>
+
+namespace android {
+
+// ----------------------------------------------------------------------------
+
+// Size of buffer used to exchange parameters between application and mediaserver processes.
+#define EFFECT_PARAM_BUFFER_SIZE 1024
+
+
+// Shared memory area used to exchange parameters between application and mediaserver
+// process.
+struct effect_param_cblk_t
+{
+                Mutex       lock;
+    volatile    uint32_t    clientIndex;    // Current read/write index for application
+    volatile    uint32_t    serverIndex;    // Current read/write index for mediaserver
+                uint8_t*    buffer;         // start of parameter buffer
+
+                effect_param_cblk_t()
+                    : lock(Mutex::SHARED), clientIndex(0), serverIndex(0) {}
+};
+
+
+// ----------------------------------------------------------------------------
+
+}; // namespace android
+
+#endif // ANDROID_EFFECTCBASESHARED_H

include/private/media/AudioTrackShared.h

@@ -78,7 +78,8 @@ struct audio_track_cblk_t
                 uint16_t    bufferTimeoutMs; // Maximum cumulated timeout before restarting audioflinger
                 uint16_t    waitTimeMs;      // Cumulated wait time
 
-                uint32_t    reserved;
+                uint16_t    sendLevel;
+                uint16_t    reserved;
                 // Cache line boundary (32 bytes)
                             audio_track_cblk_t();
                 uint32_t    stepUser(uint32_t frameCount);

libs/audioflinger/Android.mk

@@ -87,7 +87,8 @@ LOCAL_SHARED_LIBRARIES := \
     libutils \
     libbinder \
     libmedia \
-    libhardware_legacy
+    libhardware_legacy \
+    libeffects
 
 ifeq ($(strip $(BOARD_USES_GENERIC_AUDIO)),true)
   LOCAL_STATIC_LIBRARIES += libaudiointerface libaudiopolicybase

libs/audioflinger/AudioFlinger.h

@@ -42,6 +42,7 @@
 namespace android {
 
 class audio_track_cblk_t;
+class effect_param_cblk_t;
 class AudioMixer;
 class AudioBuffer;
 class AudioResampler;
@@ -75,6 +76,7 @@ public:
                                 uint32_t flags,
                                 const sp<IMemory>& sharedBuffer,
                                 int output,
+                                int *sessionId,
                                 status_t *status);
 
     virtual     uint32_t    sampleRate(int output) const;
@@ -139,6 +141,28 @@ public:
 
     virtual status_t getRenderPosition(uint32_t *halFrames, uint32_t *dspFrames, int output);
 
+    virtual int newAudioSessionId();
+
+    virtual status_t loadEffectLibrary(const char *libPath, int *handle);
+
+    virtual status_t unloadEffectLibrary(int handle);
+
+    virtual status_t queryNumberEffects(uint32_t *numEffects);
+
+    virtual status_t queryNextEffect(effect_descriptor_t *descriptor);
+
+    virtual status_t getEffectDescriptor(effect_uuid_t *pUuid, effect_descriptor_t *descriptor);
+
+    virtual sp<IEffect> createEffect(pid_t pid,
+                        effect_descriptor_t *pDesc,
+                        const sp<IEffectClient>& effectClient,
+                        int32_t priority,
+                        int output,
+                        int sessionId,
+                        status_t *status,
+                        int *id,
+                        int *enabled);
+
     enum hardware_call_state {
         AUDIO_HW_IDLE = 0,
         AUDIO_HW_INIT,
@@ -167,6 +191,7 @@ public:
                                 int channelCount,
                                 int frameCount,
                                 uint32_t flags,
+                                int *sessionId,
                                 status_t *status);
 
     virtual     status_t    onTransact(
@@ -233,6 +258,9 @@ private:
     class DuplicatingThread;
     class Track;
     class RecordTrack;
+    class EffectModule;
+    class EffectHandle;
+    class EffectChain;
 
     class ThreadBase : public Thread {
     public:
@@ -268,13 +296,15 @@ private:
                                         int channelCount,
                                         int frameCount,
                                         uint32_t flags,
-                                        const sp<IMemory>& sharedBuffer);
+                                        const sp<IMemory>& sharedBuffer,
+                                        int sessionId);
                                 ~TrackBase();
 
             virtual status_t    start() = 0;
             virtual void        stop() = 0;
                     sp<IMemory> getCblk() const;
                     audio_track_cblk_t* cblk() const { return mCblk; }
+                    int         sessionId() { return mSessionId; }
 
         protected:
             friend class ThreadBase;
@@ -323,6 +353,7 @@ private:
             int                 mClientTid;
             uint8_t             mFormat;
             uint32_t            mFlags;
+            int                 mSessionId;
         };
 
         class ConfigEvent {
@@ -405,7 +436,8 @@ private:
                                         int format,
                                         int channelCount,
                                         int frameCount,
-                                        const sp<IMemory>& sharedBuffer);
+                                        const sp<IMemory>& sharedBuffer,
+                                        int sessionId);
                                 ~Track();
 
                     void        dump(char* buffer, size_t size);
@@ -424,6 +456,12 @@ private:
                     int type() const {
                         return mStreamType;
                     }
+                    status_t    attachAuxEffect(int EffectId);
+                    void        setAuxBuffer(int EffectId, int32_t *buffer);
+                    int32_t     *auxBuffer() { return mAuxBuffer; }
+                    void        setMainBuffer(int16_t *buffer) { mMainBuffer = buffer; }
+                    int16_t     *mainBuffer() { return mMainBuffer; }
+                    int         auxEffectId() { return mAuxEffectId; }
 
 
         protected:
@@ -464,6 +502,9 @@ private:
             bool                mResetDone;
             int                 mStreamType;
             int                 mName;
+            int16_t             *mMainBuffer;
+            int32_t             *mAuxBuffer;
+            int                 mAuxEffectId;
         };  // end of Track
 
 
@@ -505,7 +546,7 @@ private:
             DuplicatingThread*          mSourceThread;
         };  // end of OutputTrack
 
-        PlaybackThread (const sp<AudioFlinger>& audioFlinger, AudioStreamOut* output, int id);
+        PlaybackThread (const sp<AudioFlinger>& audioFlinger, AudioStreamOut* output, int id, uint32_t device);
         virtual             ~PlaybackThread();
 
         virtual     status_t    dump(int fd, const Vector<String16>& args);
@@ -538,6 +579,7 @@ private:
                                     int channelCount,
                                     int frameCount,
                                     const sp<IMemory>& sharedBuffer,
+                                    int sessionId,
                                     status_t *status);
 
                     AudioStreamOut* getOutput() { return mOutput; }
@@ -549,6 +591,29 @@ private:
         virtual     String8     getParameters(const String8& keys);
         virtual     void        audioConfigChanged_l(int event, int param = 0);
         virtual     status_t    getRenderPosition(uint32_t *halFrames, uint32_t *dspFrames);
+                    int16_t     *mixBuffer() { return mMixBuffer; };
+
+                    sp<EffectHandle> createEffect_l(
+                                        const sp<AudioFlinger::Client>& client,
+                                        const sp<IEffectClient>& effectClient,
+                                        int32_t priority,
+                                        int sessionId,
+                                        effect_descriptor_t *desc,
+                                        int *enabled,
+                                        status_t *status);
+
+                    bool hasAudioSession(int sessionId);
+                    sp<EffectChain> getEffectChain(int sessionId);
+                    sp<EffectChain> getEffectChain_l(int sessionId);
+                    status_t addEffectChain_l(const sp<EffectChain>& chain);
+                    size_t removeEffectChain_l(const sp<EffectChain>& chain);
+                    void lockEffectChains_l();
+                    void unlockEffectChains();
+
+                    sp<AudioFlinger::EffectModule> getEffect_l(int sessionId, int effectId);
+                    void detachAuxEffect_l(int effectId);
+                    status_t attachAuxEffect(const sp<AudioFlinger::PlaybackThread::Track> track, int EffectId);
+                    status_t attachAuxEffect_l(const sp<AudioFlinger::PlaybackThread::Track> track, int EffectId);
 
         struct  stream_type_t {
             stream_type_t()
@@ -591,8 +656,11 @@ private:
 
         void        readOutputParameters();
 
+        uint32_t    device() { return mDevice; }
+
         virtual status_t    dumpInternals(int fd, const Vector<String16>& args);
         status_t    dumpTracks(int fd, const Vector<String16>& args);
+        status_t    dumpEffectChains(int fd, const Vector<String16>& args);
 
         SortedVector< sp<Track> >       mTracks;
         // mStreamTypes[] uses 1 additionnal stream type internally for the OutputTrack used by DuplicatingThread
@@ -603,11 +671,13 @@ private:
         int                             mNumWrites;
         int                             mNumDelayedWrites;
         bool                            mInWrite;
+        Vector< sp<EffectChain> >       mEffectChains;
+        uint32_t                        mDevice;
     };
 
     class MixerThread : public PlaybackThread {
     public:
-        MixerThread (const sp<AudioFlinger>& audioFlinger, AudioStreamOut* output, int id);
+        MixerThread (const sp<AudioFlinger>& audioFlinger, AudioStreamOut* output, int id, uint32_t device);
         virtual             ~MixerThread();
 
         // Thread virtuals
@@ -630,7 +700,7 @@ private:
     class DirectOutputThread : public PlaybackThread {
     public:
 
-        DirectOutputThread (const sp<AudioFlinger>& audioFlinger, AudioStreamOut* output, int id);
+        DirectOutputThread (const sp<AudioFlinger>& audioFlinger, AudioStreamOut* output, int id, uint32_t device);
         ~DirectOutputThread();
 
         // Thread virtuals
@@ -645,8 +715,12 @@ private:
         virtual     uint32_t    idleSleepTimeUs();
 
     private:
-        float mLeftVolume;
+        void applyVolume(uint16_t leftVol, uint16_t rightVol, bool ramp);
-        float mRightVolume;
+
+        float mLeftVolFloat;
+        float mRightVolFloat;
+        uint16_t mLeftVolShort;
+        uint16_t mRightVolShort;
     };
 
     class DuplicatingThread : public MixerThread {
@@ -676,6 +750,8 @@ private:
               float streamVolumeInternal(int stream) const { return mStreamTypes[stream].volume; }
               void audioConfigChanged_l(int event, int ioHandle, void *param2);
 
+              int  nextUniqueId();
+
     friend class AudioBuffer;
 
     class TrackHandle : public android::BnAudioTrack {
@@ -689,6 +765,7 @@ private:
         virtual void        pause();
         virtual void        setVolume(float left, float right);
         virtual sp<IMemory> getCblk() const;
+        virtual status_t    attachAuxEffect(int effectId);
         virtual status_t onTransact(
             uint32_t code, const Parcel& data, Parcel* reply, uint32_t flags);
     private:
@@ -717,7 +794,8 @@ private:
                                         int format,
                                         int channelCount,
                                         int frameCount,
-                                        uint32_t flags);
+                                        uint32_t flags,
+                                        int sessionId);
                                 ~RecordTrack();
 
             virtual status_t    start();
@@ -792,6 +870,215 @@ private:
         sp<RecordThread::RecordTrack> mRecordTrack;
     };
 
+    //--- Audio Effect Management
+
+    // EffectModule and EffectChain classes both have their own mutex to protect
+    // state changes or resource modifications. Always respect the following order
+    // if multiple mutexes must be acquired to avoid cross deadlock:
+    // AudioFlinger -> ThreadBase -> EffectChain -> EffectModule
+
+    // The EffectModule class is a wrapper object controlling the effect engine implementation
+    // in the effect library. It prevents concurrent calls to process() and command() functions
+    // from different client threads. It keeps a list of EffectHandle objects corresponding
+    // to all client applications using this effect and notifies applications of effect state,
+    // control or parameter changes. It manages the activation state machine to send appropriate
+    // reset, enable, disable commands to effect engine and provide volume
+    // ramping when effects are activated/deactivated.
+    // When controlling an auxiliary effect, the EffectModule also provides an input buffer used by
+    // the attached track(s) to accumulate their auxiliary channel.
+    class EffectModule: public RefBase {
+    public:
+        EffectModule(const wp<ThreadBase>& wThread,
+                        const wp<AudioFlinger::EffectChain>& chain,
+                        effect_descriptor_t *desc,
+                        int id,
+                        int sessionId);
+        ~EffectModule();
+
+        enum effect_state {
+            IDLE,
+            RESET,
+            STARTING,
+            ACTIVE,
+            STOPPING,
+            STOPPED
+        };
+
+        int         id() { return mId; }
+        void process();
+        status_t command(int cmdCode, int cmdSize, void *pCmdData, int *replySize, void *pReplyData);
+
+        void reset();
+        status_t configure();
+        status_t init();
+        uint32_t state() {
+            return mState;
+        }
+        uint32_t status() {
+            return mStatus;
+        }
+        status_t    setEnabled(bool enabled);
+        bool isEnabled();
+
+        void        setInBuffer(int16_t *buffer) { mConfig.inputCfg.buffer.s16 = buffer; }
+        int16_t     *inBuffer() { return mConfig.inputCfg.buffer.s16; }
+        void        setOutBuffer(int16_t *buffer) { mConfig.outputCfg.buffer.s16 = buffer; }
+        int16_t     *outBuffer() { return mConfig.outputCfg.buffer.s16; }
+
+        status_t addHandle(sp<EffectHandle>& handle);
+        void disconnect(const wp<EffectHandle>& handle);
+        size_t removeHandle (const wp<EffectHandle>& handle);
+
+        effect_descriptor_t& desc() { return mDescriptor; }
+
+        status_t         setDevice(uint32_t device);
+        status_t         setVolume(uint32_t *left, uint32_t *right, bool controller);
+
+        status_t         dump(int fd, const Vector<String16>& args);
+
+    protected:
+
+        EffectModule(const EffectModule&);
+        EffectModule& operator = (const EffectModule&);
+
+        status_t start();
+        status_t stop();
+
+        Mutex               mLock;      // mutex for process, commands and handles list protection
+        wp<ThreadBase>      mThread;    // parent thread
+        wp<EffectChain>     mChain;     // parent effect chain
+        int                 mId;        // this instance unique ID
+        int                 mSessionId; // audio session ID
+        effect_descriptor_t mDescriptor;// effect descriptor received from effect engine
+        effect_config_t     mConfig;    // input and output audio configuration
+        effect_interface_t  mEffectInterface; // Effect module C API
+        status_t mStatus;               // initialization status
+        uint32_t mState;                // current activation state (effect_state)
+        Vector< wp<EffectHandle> > mHandles;    // list of client handles
+    };
+
+    // The EffectHandle class implements the IEffect interface. It provides resources
+    // to receive parameter updates, keeps track of effect control
+    // ownership and state and has a pointer to the EffectModule object it is controlling.
+    // There is one EffectHandle object for each application controlling (or using)
+    // an effect module.
+    // The EffectHandle is obtained by calling AudioFlinger::createEffect().
+    class EffectHandle: public android::BnEffect {
+    public:
+
+        EffectHandle(const sp<EffectModule>& effect,
+                const sp<AudioFlinger::Client>& client,
+                const sp<IEffectClient>& effectClient,
+                int32_t priority);
+        virtual ~EffectHandle();
+
+        // IEffect
+        virtual status_t enable();
+        virtual status_t disable();
+        virtual status_t command(int cmdCode, int cmdSize, void *pCmdData, int *replySize, void *pReplyData);
+        virtual void disconnect();
+        virtual sp<IMemory> getCblk() const;
+        virtual status_t onTransact(uint32_t code, const Parcel& data,
+                Parcel* reply, uint32_t flags);
+
+
+        // Give or take control of effect module
+        void setControl(bool hasControl, bool signal);
+        void commandExecuted(int cmdCode, int cmdSize, void *pCmdData, int replySize, void *pReplyData);
+        void setEnabled(bool enabled);
+
+        // Getters
+        int id() { return mEffect->id(); }
+        int priority() { return mPriority; }
+        bool hasControl() { return mHasControl; }
+        sp<EffectModule> effect() { return mEffect; }
+
+        void dump(char* buffer, size_t size);
+
+    protected:
+
+        EffectHandle(const EffectHandle&);
+        EffectHandle& operator =(const EffectHandle&);
+
+        sp<EffectModule> mEffect;           // pointer to controlled EffectModule
+        sp<IEffectClient> mEffectClient;    // callback interface for client notifications
+        sp<Client>          mClient;        // client for shared memory allocation
+        sp<IMemory>         mCblkMemory;    // shared memory for control block
+        effect_param_cblk_t* mCblk;         // control block for deferred parameter setting via shared memory
+        uint8_t*            mBuffer;        // pointer to parameter area in shared memory
+        int mPriority;                      // client application priority to control the effect
+        bool mHasControl;                   // true if this handle is controlling the effect
+    };
+
+    // the EffectChain class represents a group of effects associated to one audio session.
+    // There can be any number of EffectChain objects per output mixer thread (PlaybackThread).
+    // The EffecChain with session ID 0 contains global effects applied to the output mix.
+    // Effects in this chain can be insert or auxiliary. Effects in other chains (attached to tracks)
+    // are insert only. The EffectChain maintains an ordered list of effect module, the order corresponding
+    // in the effect process order. When attached to a track (session ID != 0), it also provide it's own
+    // input buffer used by the track as accumulation buffer.
+    class EffectChain: public RefBase {
+    public:
+        EffectChain(const wp<ThreadBase>& wThread, int sessionId);
+        ~EffectChain();
+
+        void process_l();
+
+        void lock() {
+            mLock.lock();
+        }
+        void unlock() {
+            mLock.unlock();
+        }
+
+        status_t addEffect(sp<EffectModule>& handle);
+        size_t removeEffect(const sp<EffectModule>& handle);
+
+        int sessionId() {
+            return mSessionId;
+        }
+        sp<EffectModule> getEffectFromDesc(effect_descriptor_t *descriptor);
+        sp<EffectModule> getEffectFromId(int id);
+        sp<EffectModule> getVolumeController();
+        bool setVolume(uint32_t *left, uint32_t *right);
+        void setDevice(uint32_t device);
+
+        void setInBuffer(int16_t *buffer, bool ownsBuffer = false) {
+            mInBuffer = buffer;
+            mOwnInBuffer = ownsBuffer;
+        }
+        int16_t *inBuffer() {
+            return mInBuffer;
+        }
+        void setOutBuffer(int16_t *buffer) {
+            mOutBuffer = buffer;
+        }
+        int16_t *outBuffer() {
+            return mOutBuffer;
+        }
+
+        void startTrack() {mActiveTrackCnt++;}
+        void stopTrack() {mActiveTrackCnt--;}
+        int activeTracks() { return mActiveTrackCnt;}
+
+        status_t dump(int fd, const Vector<String16>& args);
+
+    protected:
+
+        EffectChain(const EffectChain&);
+        EffectChain& operator =(const EffectChain&);
+
+        wp<ThreadBase> mThread;     // parent mixer thread
+        Mutex mLock;                // mutex protecting effect list
+        Vector<sp<EffectModule> > mEffects; // list of effect modules
+        int mSessionId;             // audio session ID
+        int16_t *mInBuffer;         // chain input buffer
+        int16_t *mOutBuffer;        // chain output buffer
+        int mVolumeCtrlIdx;         // index of insert effect having control over volume
+        int mActiveTrackCnt;        // number of active tracks connected
+        bool mOwnInBuffer;          // true if the chain owns its input buffer
+    };
+
     friend class RecordThread;
     friend class PlaybackThread;
 
@@ -813,7 +1100,7 @@ private:
                 DefaultKeyedVector< int, sp<RecordThread> >    mRecordThreads;
 
                 DefaultKeyedVector< pid_t, sp<NotificationClient> >    mNotificationClients;
-                int                                 mNextThreadId;
+                volatile int32_t                    mNextUniqueId;
 #ifdef LVMX
                 int mLifeVibesClientPid;
 #endif

libs/audioflinger/AudioMixer.cpp

@@ -56,6 +56,8 @@ AudioMixer::AudioMixer(size_t frameCount, uint32_t sampleRate)
         t->volume[1] = UNITY_GAIN;
         t->volumeInc[0] = 0;
         t->volumeInc[1] = 0;
+        t->auxLevel = 0;
+        t->auxInc = 0;
         t->channelCount = 2;
         t->enabled = 0;
         t->format = 16;
@@ -65,6 +67,8 @@ AudioMixer::AudioMixer(size_t frameCount, uint32_t sampleRate)
         t->resampler = 0;
         t->sampleRate = mSampleRate;
         t->in = 0;
+        t->mainBuffer = NULL;
+        t->auxBuffer = NULL;
         t++;
     }
 }
@@ -169,28 +173,48 @@ status_t AudioMixer::setActiveTrack(int track)
     return NO_ERROR;
 }
 
-status_t AudioMixer::setParameter(int target, int name, int value)
+status_t AudioMixer::setParameter(int target, int name, void *value)
 {
+    int valueInt = (int)value;
+    int32_t *valueBuf = (int32_t *)value;
+
     switch (target) {
     case TRACK:
         if (name == CHANNEL_COUNT) {
-            if ((uint32_t(value) <= MAX_NUM_CHANNELS) && (value)) {
+            if ((uint32_t(valueInt) <= MAX_NUM_CHANNELS) && (valueInt)) {
-                if (mState.tracks[ mActiveTrack ].channelCount != value) {
+                if (mState.tracks[ mActiveTrack ].channelCount != valueInt) {
-                    mState.tracks[ mActiveTrack ].channelCount = value;
+                    mState.tracks[ mActiveTrack ].channelCount = valueInt;
-                    LOGV("setParameter(TRACK, CHANNEL_COUNT, %d)", value);
+                    LOGV("setParameter(TRACK, CHANNEL_COUNT, %d)", valueInt);
                     invalidateState(1<<mActiveTrack);
                 }
                 return NO_ERROR;
             }
         }
+        if (name == MAIN_BUFFER) {
+            if (mState.tracks[ mActiveTrack ].mainBuffer != valueBuf) {
+                mState.tracks[ mActiveTrack ].mainBuffer = valueBuf;
+                LOGV("setParameter(TRACK, MAIN_BUFFER, %p)", valueBuf);
+                invalidateState(1<<mActiveTrack);
+            }
+            return NO_ERROR;
+        }
+        if (name == AUX_BUFFER) {
+            if (mState.tracks[ mActiveTrack ].auxBuffer != valueBuf) {
+                mState.tracks[ mActiveTrack ].auxBuffer = valueBuf;
+                LOGV("setParameter(TRACK, AUX_BUFFER, %p)", valueBuf);
+                invalidateState(1<<mActiveTrack);
+            }
+            return NO_ERROR;
+        }
+
         break;
     case RESAMPLE:
         if (name == SAMPLE_RATE) {
-            if (value > 0) {
+            if (valueInt > 0) {
                 track_t& track = mState.tracks[ mActiveTrack ];
-                if (track.setResampler(uint32_t(value), mSampleRate)) {
+                if (track.setResampler(uint32_t(valueInt), mSampleRate)) {
                     LOGV("setParameter(RESAMPLE, SAMPLE_RATE, %u)",
-                            uint32_t(value));
+                            uint32_t(valueInt));
                     invalidateState(1<<mActiveTrack);
                 }
                 return NO_ERROR;
@@ -201,18 +225,39 @@ status_t AudioMixer::setParameter(int target, int name, int value)
     case VOLUME:
         if ((uint32_t(name-VOLUME0) < MAX_NUM_CHANNELS)) {
             track_t& track = mState.tracks[ mActiveTrack ];
-            if (track.volume[name-VOLUME0] != value) {
+            if (track.volume[name-VOLUME0] != valueInt) {
+                LOGV("setParameter(VOLUME, VOLUME0/1: %04x)", valueInt);
                 track.prevVolume[name-VOLUME0] = track.volume[name-VOLUME0] << 16;
-                track.volume[name-VOLUME0] = value;
+                track.volume[name-VOLUME0] = valueInt;
                 if (target == VOLUME) {
-                    track.prevVolume[name-VOLUME0] = value << 16;
+                    track.prevVolume[name-VOLUME0] = valueInt << 16;
                     track.volumeInc[name-VOLUME0] = 0;
                 } else {
-                    int32_t d = (value<<16) - track.prevVolume[name-VOLUME0];
+                    int32_t d = (valueInt<<16) - track.prevVolume[name-VOLUME0];
                     int32_t volInc = d / int32_t(mState.frameCount);
                     track.volumeInc[name-VOLUME0] = volInc;
                     if (volInc == 0) {
-                        track.prevVolume[name-VOLUME0] = value << 16;
+                        track.prevVolume[name-VOLUME0] = valueInt << 16;
+                    }
+                }
+                invalidateState(1<<mActiveTrack);
+            }
+            return NO_ERROR;
+        } else if (name == AUXLEVEL) {
+            track_t& track = mState.tracks[ mActiveTrack ];
+            if (track.auxLevel != valueInt) {
+                LOGV("setParameter(VOLUME, AUXLEVEL: %04x)", valueInt);
+                track.prevAuxLevel = track.auxLevel << 16;
+                track.auxLevel = valueInt;
+                if (target == VOLUME) {
+                    track.prevAuxLevel = valueInt << 16;
+                    track.auxInc = 0;
+                } else {
+                    int32_t d = (valueInt<<16) - track.prevAuxLevel;
+                    int32_t volInc = d / int32_t(mState.frameCount);
+                    track.auxInc = volInc;
+                    if (volInc == 0) {
+                        track.prevAuxLevel = valueInt << 16;
                     }
                 }
                 invalidateState(1<<mActiveTrack);
@@ -245,7 +290,7 @@ bool AudioMixer::track_t::doesResample() const
 }
 
 inline
-void AudioMixer::track_t::adjustVolumeRamp()
+void AudioMixer::track_t::adjustVolumeRamp(bool aux)
 {
     for (int i=0 ; i<2 ; i++) {
         if (((volumeInc[i]>0) && (((prevVolume[i]+volumeInc[i])>>16) >= volume[i])) ||
@@ -254,6 +299,13 @@ void AudioMixer::track_t::adjustVolumeRamp()
             prevVolume[i] = volume[i]<<16;
         }
     }
+    if (aux) {
+        if (((auxInc>0) && (((prevAuxLevel+auxInc)>>16) >= auxLevel)) ||
+            ((auxInc<0) && (((prevAuxLevel+auxInc)>>16) <= auxLevel))) {
+            auxInc = 0;
+            prevAuxLevel = auxLevel<<16;
+        }
+    }
 }
 
 
@@ -265,13 +317,13 @@ status_t AudioMixer::setBufferProvider(AudioBufferProvider* buffer)
 
 
 
-void AudioMixer::process(void* output)
+void AudioMixer::process()
 {
-    mState.hook(&mState, output);
+    mState.hook(&mState);
 }
 
 
-void AudioMixer::process__validate(state_t* state, void* output)
+void AudioMixer::process__validate(state_t* state)
 {
     LOGW_IF(!state->needsChanged,
         "in process__validate() but nothing's invalid");
@@ -308,6 +360,9 @@ void AudioMixer::process__validate(state_t* state, void* output)
         n |= NEEDS_CHANNEL_1 + t.channelCount - 1;
         n |= NEEDS_FORMAT_16;
         n |= t.doesResample() ? NEEDS_RESAMPLE_ENABLED : NEEDS_RESAMPLE_DISABLED;
+        if (t.auxLevel != 0 && t.auxBuffer != NULL) {
+            n |= NEEDS_AUX_ENABLED;
+        }
 
         if (t.volumeInc[0]|t.volumeInc[1]) {
             volumeRamp = 1;
@@ -319,6 +374,9 @@ void AudioMixer::process__validate(state_t* state, void* output)
         if ((n & NEEDS_MUTE__MASK) == NEEDS_MUTE_ENABLED) {
             t.hook = track__nop;
         } else {
+            if ((n & NEEDS_AUX__MASK) == NEEDS_AUX_ENABLED) {
+                all16BitsStereoNoResample = 0;
+            }
             if ((n & NEEDS_RESAMPLE__MASK) == NEEDS_RESAMPLE_ENABLED) {
                 all16BitsStereoNoResample = 0;
                 resampling = 1;
@@ -369,7 +427,7 @@ void AudioMixer::process__validate(state_t* state, void* output)
         countActiveTracks, state->enabledTracks,
         all16BitsStereoNoResample, resampling, volumeRamp);
 
-   state->hook(state, output);
+   state->hook(state);
 
    // Now that the volume ramp has been done, set optimal state and
    // track hooks for subsequent mixer process
@@ -390,7 +448,7 @@ void AudioMixer::process__validate(state_t* state, void* output)
        }
        if (allMuted) {
            state->hook = process__nop;
-       } else if (!resampling && all16BitsStereoNoResample) {
+       } else if (all16BitsStereoNoResample) {
            if (countActiveTracks == 1) {
               state->hook = process__OneTrack16BitsStereoNoResampling;
            }
@@ -481,30 +539,44 @@ int32_t mulRL(int left, uint32_t inRL, uint32_t vRL)
 }
 
 
-void AudioMixer::track__genericResample(track_t* t, int32_t* out, size_t outFrameCount, int32_t* temp)
+void AudioMixer::track__genericResample(track_t* t, int32_t* out, size_t outFrameCount, int32_t* temp, int32_t* aux)
 {
     t->resampler->setSampleRate(t->sampleRate);
 
     // ramp gain - resample to temp buffer and scale/mix in 2nd step
-    if UNLIKELY(t->volumeInc[0]|t->volumeInc[1]) {
+    if (aux != NULL) {
+        // always resample with unity gain when sending to auxiliary buffer to be able
+        // to apply send level after resampling
+        // TODO: modify each resampler to support aux channel?
         t->resampler->setVolume(UNITY_GAIN, UNITY_GAIN);
         memset(temp, 0, outFrameCount * MAX_NUM_CHANNELS * sizeof(int32_t));
         t->resampler->resample(temp, outFrameCount, t->bufferProvider);
-        volumeRampStereo(t, out, outFrameCount, temp);
+        if UNLIKELY(t->volumeInc[0]|t->volumeInc[1]|t->auxInc) {
-    }
+            volumeRampStereo(t, out, outFrameCount, temp, aux);
+        } else {
+            volumeStereo(t, out, outFrameCount, temp, aux);
+        }
+    } else {
+        if UNLIKELY(t->volumeInc[0]|t->volumeInc[1]) {
+            t->resampler->setVolume(UNITY_GAIN, UNITY_GAIN);
+            memset(temp, 0, outFrameCount * MAX_NUM_CHANNELS * sizeof(int32_t));
+            t->resampler->resample(temp, outFrameCount, t->bufferProvider);
+            volumeRampStereo(t, out, outFrameCount, temp, aux);
+        }
 
-    // constant gain
+        // constant gain
-    else {
+        else {
-        t->resampler->setVolume(t->volume[0], t->volume[1]);
+            t->resampler->setVolume(t->volume[0], t->volume[1]);
-        t->resampler->resample(out, outFrameCount, t->bufferProvider);
+            t->resampler->resample(out, outFrameCount, t->bufferProvider);
+        }
     }
 }
 
-void AudioMixer::track__nop(track_t* t, int32_t* out, size_t outFrameCount, int32_t* temp)
+void AudioMixer::track__nop(track_t* t, int32_t* out, size_t outFrameCount, int32_t* temp, int32_t* aux)
 {
 }
 
-void AudioMixer::volumeRampStereo(track_t* t, int32_t* out, size_t frameCount, int32_t* temp)
+void AudioMixer::volumeRampStereo(track_t* t, int32_t* out, size_t frameCount, int32_t* temp, int32_t* aux)
 {
     int32_t vl = t->prevVolume[0];
     int32_t vr = t->prevVolume[1];
@@ -516,96 +588,236 @@ void AudioMixer::volumeRampStereo(track_t* t, int32_t* out, size_t frameCount, i
     //       (vl + vlInc*frameCount)/65536.0f, frameCount);
 
     // ramp volume
-    do {
+    if UNLIKELY(aux != NULL) {
-        *out++ += (vl >> 16) * (*temp++ >> 12);
+        int32_t va = t->prevAuxLevel;
-        *out++ += (vr >> 16) * (*temp++ >> 12);
+        const int32_t vaInc = t->auxInc;
-        vl += vlInc;
+        int32_t l;
-        vr += vrInc;
+        int32_t r;
-    } while (--frameCount);
-
-    t->prevVolume[0] = vl;
-    t->prevVolume[1] = vr;
-    t->adjustVolumeRamp();
-}
-
-void AudioMixer::track__16BitsStereo(track_t* t, int32_t* out, size_t frameCount, int32_t* temp)
-{
-    int16_t const *in = static_cast<int16_t const *>(t->in);
-
-    // ramp gain
-    if UNLIKELY(t->volumeInc[0]|t->volumeInc[1]) {
-        int32_t vl = t->prevVolume[0];
-        int32_t vr = t->prevVolume[1];
-        const int32_t vlInc = t->volumeInc[0];
-        const int32_t vrInc = t->volumeInc[1];
-
-        // LOGD("[1] %p: inc=%f, v0=%f, v1=%d, final=%f, count=%d",
-        //        t, vlInc/65536.0f, vl/65536.0f, t->volume[0],
-        //        (vl + vlInc*frameCount)/65536.0f, frameCount);
 
         do {
-            *out++ += (vl >> 16) * (int32_t) *in++;
+            l = (*temp++ >> 12);
-            *out++ += (vr >> 16) * (int32_t) *in++;
+            r = (*temp++ >> 12);
+            *out++ += (vl >> 16) * l;
+            *out++ += (vr >> 16) * r;
+            *aux++ += (va >> 17) * (l + r);
+            vl += vlInc;
+            vr += vrInc;
+            va += vaInc;
+        } while (--frameCount);
+        t->prevAuxLevel = va;
+    } else {
+        do {
+            *out++ += (vl >> 16) * (*temp++ >> 12);
+            *out++ += (vr >> 16) * (*temp++ >> 12);
             vl += vlInc;
             vr += vrInc;
         } while (--frameCount);
-       
-        t->prevVolume[0] = vl;
-        t->prevVolume[1] = vr;
-        t->adjustVolumeRamp();
     }
+    t->prevVolume[0] = vl;
+    t->prevVolume[1] = vr;
+    t->adjustVolumeRamp((aux != NULL));
+}
 
-    // constant gain
+void AudioMixer::volumeStereo(track_t* t, int32_t* out, size_t frameCount, int32_t* temp, int32_t* aux)
-    else {
+{
-        const uint32_t vrl = t->volumeRL;
+    const int16_t vl = t->volume[0];
+    const int16_t vr = t->volume[1];
+
+    if UNLIKELY(aux != NULL) {
+        const int16_t va = (int16_t)t->auxLevel;
         do {
-            uint32_t rl = *reinterpret_cast<uint32_t const *>(in);
+            int16_t l = (int16_t)(*temp++ >> 12);
-            in += 2;
+            int16_t r = (int16_t)(*temp++ >> 12);
-            out[0] = mulAddRL(1, rl, vrl, out[0]);
+            out[0] = mulAdd(l, vl, out[0]);
-            out[1] = mulAddRL(0, rl, vrl, out[1]);
+            int16_t a = (int16_t)(((int32_t)l + r) >> 1);
+            out[1] = mulAdd(r, vr, out[1]);
+            out += 2;
+            aux[0] = mulAdd(a, va, aux[0]);
+            aux++;
+        } while (--frameCount);
+    } else {
+        do {
+            int16_t l = (int16_t)(*temp++ >> 12);
+            int16_t r = (int16_t)(*temp++ >> 12);
+            out[0] = mulAdd(l, vl, out[0]);
+            out[1] = mulAdd(r, vr, out[1]);
             out += 2;
         } while (--frameCount);
     }
+}
+
+void AudioMixer::track__16BitsStereo(track_t* t, int32_t* out, size_t frameCount, int32_t* temp, int32_t* aux)
+{
+    int16_t const *in = static_cast<int16_t const *>(t->in);
+
+    if UNLIKELY(aux != NULL) {
+        int32_t l;
+        int32_t r;
+        // ramp gain
+        if UNLIKELY(t->volumeInc[0]|t->volumeInc[1]|t->auxInc) {
+            int32_t vl = t->prevVolume[0];
+            int32_t vr = t->prevVolume[1];
+            int32_t va = t->prevAuxLevel;
+            const int32_t vlInc = t->volumeInc[0];
+            const int32_t vrInc = t->volumeInc[1];
+            const int32_t vaInc = t->auxInc;
+            // LOGD("[1] %p: inc=%f, v0=%f, v1=%d, final=%f, count=%d",
+            //        t, vlInc/65536.0f, vl/65536.0f, t->volume[0],
+            //        (vl + vlInc*frameCount)/65536.0f, frameCount);
+
+            do {
+                l = (int32_t)*in++;
+                r = (int32_t)*in++;
+                *out++ += (vl >> 16) * l;
+                *out++ += (vr >> 16) * r;
+                *aux++ += (va >> 17) * (l + r);
+                vl += vlInc;
+                vr += vrInc;
+                va += vaInc;
+            } while (--frameCount);
+
+            t->prevVolume[0] = vl;
+            t->prevVolume[1] = vr;
+            t->prevAuxLevel = va;
+            t->adjustVolumeRamp(true);
+        }
+
+        // constant gain
+        else {
+            const uint32_t vrl = t->volumeRL;
+            const int16_t va = (int16_t)t->auxLevel;
+            do {
+                uint32_t rl = *reinterpret_cast<uint32_t const *>(in);
+                int16_t a = (int16_t)(((int32_t)in[0] + in[1]) >> 1);
+                in += 2;
+                out[0] = mulAddRL(1, rl, vrl, out[0]);
+                out[1] = mulAddRL(0, rl, vrl, out[1]);
+                out += 2;
+                aux[0] = mulAdd(a, va, aux[0]);
+                aux++;
+            } while (--frameCount);
+        }
+    } else {
+        // ramp gain
+        if UNLIKELY(t->volumeInc[0]|t->volumeInc[1]) {
+            int32_t vl = t->prevVolume[0];
+            int32_t vr = t->prevVolume[1];
+            const int32_t vlInc = t->volumeInc[0];
+            const int32_t vrInc = t->volumeInc[1];
+
+            // LOGD("[1] %p: inc=%f, v0=%f, v1=%d, final=%f, count=%d",
+            //        t, vlInc/65536.0f, vl/65536.0f, t->volume[0],
+            //        (vl + vlInc*frameCount)/65536.0f, frameCount);
+
+            do {
+                *out++ += (vl >> 16) * (int32_t) *in++;
+                *out++ += (vr >> 16) * (int32_t) *in++;
+                vl += vlInc;
+                vr += vrInc;
+            } while (--frameCount);
+
+            t->prevVolume[0] = vl;
+            t->prevVolume[1] = vr;
+            t->adjustVolumeRamp(false);
+        }
+
+        // constant gain
+        else {
+            const uint32_t vrl = t->volumeRL;
+            do {
+                uint32_t rl = *reinterpret_cast<uint32_t const *>(in);
+                in += 2;
+                out[0] = mulAddRL(1, rl, vrl, out[0]);
+                out[1] = mulAddRL(0, rl, vrl, out[1]);
+                out += 2;
+            } while (--frameCount);
+        }
+    }
     t->in = in;
 }
 
-void AudioMixer::track__16BitsMono(track_t* t, int32_t* out, size_t frameCount, int32_t* temp)
+void AudioMixer::track__16BitsMono(track_t* t, int32_t* out, size_t frameCount, int32_t* temp, int32_t* aux)
 {
     int16_t const *in = static_cast<int16_t const *>(t->in);
 
-    // ramp gain
+    if UNLIKELY(aux != NULL) {
-    if UNLIKELY(t->volumeInc[0]|t->volumeInc[1]) {
+        // ramp gain
-        int32_t vl = t->prevVolume[0];
+        if UNLIKELY(t->volumeInc[0]|t->volumeInc[1]|t->auxInc) {
-        int32_t vr = t->prevVolume[1];
+            int32_t vl = t->prevVolume[0];
-        const int32_t vlInc = t->volumeInc[0];
+            int32_t vr = t->prevVolume[1];
-        const int32_t vrInc = t->volumeInc[1];
+            int32_t va = t->prevAuxLevel;
+            const int32_t vlInc = t->volumeInc[0];
+            const int32_t vrInc = t->volumeInc[1];
+            const int32_t vaInc = t->auxInc;
 
-        // LOGD("[2] %p: inc=%f, v0=%f, v1=%d, final=%f, count=%d",
+            // LOGD("[2] %p: inc=%f, v0=%f, v1=%d, final=%f, count=%d",
-        //         t, vlInc/65536.0f, vl/65536.0f, t->volume[0],
+            //         t, vlInc/65536.0f, vl/65536.0f, t->volume[0],
-        //         (vl + vlInc*frameCount)/65536.0f, frameCount);
+            //         (vl + vlInc*frameCount)/65536.0f, frameCount);
 
-        do {
+            do {
-            int32_t l = *in++;
+                int32_t l = *in++;
-            *out++ += (vl >> 16) * l;
+                *out++ += (vl >> 16) * l;
-            *out++ += (vr >> 16) * l;
+                *out++ += (vr >> 16) * l;
-            vl += vlInc;
+                *aux++ += (va >> 16) * l;
-            vr += vrInc;
+                vl += vlInc;
-        } while (--frameCount);
+                vr += vrInc;
+                va += vaInc;
+            } while (--frameCount);
 
-        t->prevVolume[0] = vl;
+            t->prevVolume[0] = vl;
-        t->prevVolume[1] = vr;
+            t->prevVolume[1] = vr;
-        t->adjustVolumeRamp();
+            t->prevAuxLevel = va;
-    }
+            t->adjustVolumeRamp(true);
-    // constant gain
+        }
-    else {
+        // constant gain
-        const int16_t vl = t->volume[0];
+        else {
-        const int16_t vr = t->volume[1];
+            const int16_t vl = t->volume[0];
-        do {
+            const int16_t vr = t->volume[1];
-            int16_t l = *in++;
+            const int16_t va = (int16_t)t->auxLevel;
-            out[0] = mulAdd(l, vl, out[0]);
+            do {
-            out[1] = mulAdd(l, vr, out[1]);
+                int16_t l = *in++;
-            out += 2;
+                out[0] = mulAdd(l, vl, out[0]);
-        } while (--frameCount);
+                out[1] = mulAdd(l, vr, out[1]);
+                out += 2;
+                aux[0] = mulAdd(l, va, aux[0]);
+                aux++;
+            } while (--frameCount);
+        }
+    } else {
+        // ramp gain
+        if UNLIKELY(t->volumeInc[0]|t->volumeInc[1]) {
+            int32_t vl = t->prevVolume[0];
+            int32_t vr = t->prevVolume[1];
+            const int32_t vlInc = t->volumeInc[0];
+            const int32_t vrInc = t->volumeInc[1];
+
+            // LOGD("[2] %p: inc=%f, v0=%f, v1=%d, final=%f, count=%d",
+            //         t, vlInc/65536.0f, vl/65536.0f, t->volume[0],
+            //         (vl + vlInc*frameCount)/65536.0f, frameCount);
+
+            do {
+                int32_t l = *in++;
+                *out++ += (vl >> 16) * l;
+                *out++ += (vr >> 16) * l;
+                vl += vlInc;
+                vr += vrInc;
+            } while (--frameCount);
+
+            t->prevVolume[0] = vl;
+            t->prevVolume[1] = vr;
+            t->adjustVolumeRamp(false);
+        }
+        // constant gain
+        else {
+            const int16_t vl = t->volume[0];
+            const int16_t vr = t->volume[1];
+            do {
+                int16_t l = *in++;
+                out[0] = mulAdd(l, vl, out[0]);
+                out[1] = mulAdd(l, vr, out[1]);
+                out += 2;
+            } while (--frameCount);
+        }
     }
     t->in = in;
 }
@@ -624,37 +836,56 @@ void AudioMixer::ditherAndClamp(int32_t* out, int32_t const *sums, size_t c)
 }
 
 // no-op case
-void AudioMixer::process__nop(state_t* state, void* output)
+void AudioMixer::process__nop(state_t* state)
 {
-    // this assumes output 16 bits stereo, no resampling
+    uint32_t e0 = state->enabledTracks;
-    memset(output, 0, state->frameCount*4);
+    size_t bufSize = state->frameCount * sizeof(int16_t) * MAX_NUM_CHANNELS;
-    uint32_t en = state->enabledTracks;
+    while (e0) {
-    while (en) {
+        // process by group of tracks with same output buffer to
-        const int i = 31 - __builtin_clz(en);
+        // avoid multiple memset() on same buffer
-        en &= ~(1<<i);
+        uint32_t e1 = e0, e2 = e0;
-        track_t& t = state->tracks[i];
+        int i = 31 - __builtin_clz(e1);
-        size_t outFrames = state->frameCount;
+        track_t& t1 = state->tracks[i];
-        while (outFrames) {
+        e2 &= ~(1<<i);
-            t.buffer.frameCount = outFrames;
+        while (e2) {
-            t.bufferProvider->getNextBuffer(&t.buffer);
+            i = 31 - __builtin_clz(e2);
-            if (!t.buffer.raw) break;
+            e2 &= ~(1<<i);
-            outFrames -= t.buffer.frameCount;
+            track_t& t2 = state->tracks[i];
-            t.bufferProvider->releaseBuffer(&t.buffer);
+            if UNLIKELY(t2.mainBuffer != t1.mainBuffer) {
+                e1 &= ~(1<<i);
+            }
+        }
+        e0 &= ~(e1);
+
+        memset(t1.mainBuffer, 0, bufSize);
+
+        while (e1) {
+            i = 31 - __builtin_clz(e1);
+            e1 &= ~(1<<i);
+            t1 = state->tracks[i];
+            size_t outFrames = state->frameCount;
+            while (outFrames) {
+                t1.buffer.frameCount = outFrames;
+                t1.bufferProvider->getNextBuffer(&t1.buffer);
+                if (!t1.buffer.raw) break;
+                outFrames -= t1.buffer.frameCount;
+                t1.bufferProvider->releaseBuffer(&t1.buffer);
+            }
         }
     }
 }
 
 // generic code without resampling
-void AudioMixer::process__genericNoResampling(state_t* state, void* output)
+void AudioMixer::process__genericNoResampling(state_t* state)
 {
     int32_t outTemp[BLOCKSIZE * MAX_NUM_CHANNELS] __attribute__((aligned(32)));
 
     // acquire each track's buffer
     uint32_t enabledTracks = state->enabledTracks;
-    uint32_t en = enabledTracks;
+    uint32_t e0 = enabledTracks;
-    while (en) {
+    while (e0) {
-        const int i = 31 - __builtin_clz(en);
+        const int i = 31 - __builtin_clz(e0);
-        en &= ~(1<<i);
+        e0 &= ~(1<<i);
         track_t& t = state->tracks[i];
         t.buffer.frameCount = state->frameCount;
         t.bufferProvider->getNextBuffer(&t.buffer);
@@ -666,108 +897,154 @@ void AudioMixer::process__genericNoResampling(state_t* state, void* output)
             enabledTracks &= ~(1<<i);
     }
 
-    // this assumes output 16 bits stereo, no resampling
+    e0 = enabledTracks;
-    int32_t* out = static_cast<int32_t*>(output);
+    while (e0) {
-    size_t numFrames = state->frameCount;
+        // process by group of tracks with same output buffer to
-    do {
+        // optimize cache use
-        memset(outTemp, 0, sizeof(outTemp));
+        uint32_t e1 = e0, e2 = e0;
-
+        int j = 31 - __builtin_clz(e1);
-        en = enabledTracks;
+        track_t& t1 = state->tracks[j];
-        while (en) {
+        e2 &= ~(1<<j);
-            const int i = 31 - __builtin_clz(en);
+        while (e2) {
-            en &= ~(1<<i);
+            j = 31 - __builtin_clz(e2);
-            track_t& t = state->tracks[i];
+            e2 &= ~(1<<j);
-            size_t outFrames = BLOCKSIZE;
+            track_t& t2 = state->tracks[j];
-           
+            if UNLIKELY(t2.mainBuffer != t1.mainBuffer) {
-            while (outFrames) {
+                e1 &= ~(1<<j);
-                size_t inFrames = (t.frameCount > outFrames)?outFrames:t.frameCount;
-                if (inFrames) {
-                    (t.hook)(&t, outTemp + (BLOCKSIZE-outFrames)*MAX_NUM_CHANNELS, inFrames, state->resampleTemp);
-                    t.frameCount -= inFrames;
-                    outFrames -= inFrames;
-                }
-                if (t.frameCount == 0 && outFrames) {
-                    t.bufferProvider->releaseBuffer(&t.buffer);
-                    t.buffer.frameCount = numFrames - (BLOCKSIZE - outFrames);
-                    t.bufferProvider->getNextBuffer(&t.buffer);
-                    t.in = t.buffer.raw;
-                    if (t.in == NULL) {
-                        enabledTracks &= ~(1<<i);
-                        break;
-                    }
-                    t.frameCount = t.buffer.frameCount;
-                 }
             }
         }
-
+        e0 &= ~(e1);
-        ditherAndClamp(out, outTemp, BLOCKSIZE);
+        // this assumes output 16 bits stereo, no resampling
-        out += BLOCKSIZE;
+        int32_t *out = t1.mainBuffer;
-        numFrames -= BLOCKSIZE;
+        size_t numFrames = 0;
-    } while (numFrames);
+        do {
-
+            memset(outTemp, 0, sizeof(outTemp));
+            e2 = e1;
+            while (e2) {
+                const int i = 31 - __builtin_clz(e2);
+                e2 &= ~(1<<i);
+                track_t& t = state->tracks[i];
+                size_t outFrames = BLOCKSIZE;
+                int32_t *aux = NULL;
+                if UNLIKELY((t.needs & NEEDS_AUX__MASK) == NEEDS_AUX_ENABLED) {
+                    aux = t.auxBuffer + numFrames;
+                }
+                while (outFrames) {
+                    size_t inFrames = (t.frameCount > outFrames)?outFrames:t.frameCount;
+                    if (inFrames) {
+                        (t.hook)(&t, outTemp + (BLOCKSIZE-outFrames)*MAX_NUM_CHANNELS, inFrames, state->resampleTemp, aux);
+                        t.frameCount -= inFrames;
+                        outFrames -= inFrames;
+                        if UNLIKELY(aux != NULL) {
+                            aux += inFrames;
+                        }
+                    }
+                    if (t.frameCount == 0 && outFrames) {
+                        t.bufferProvider->releaseBuffer(&t.buffer);
+                        t.buffer.frameCount = (state->frameCount - numFrames) - (BLOCKSIZE - outFrames);
+                        t.bufferProvider->getNextBuffer(&t.buffer);
+                        t.in = t.buffer.raw;
+                        if (t.in == NULL) {
+                            enabledTracks &= ~(1<<i);
+                            e1 &= ~(1<<i);
+                            break;
+                        }
+                        t.frameCount = t.buffer.frameCount;
+                    }
+                }
+            }
+            ditherAndClamp(out, outTemp, BLOCKSIZE);
+            out += BLOCKSIZE;
+            numFrames += BLOCKSIZE;
+        } while (numFrames < state->frameCount);
+    }
 
     // release each track's buffer
-    en = enabledTracks;
+    e0 = enabledTracks;
-    while (en) {
+    while (e0) {
-        const int i = 31 - __builtin_clz(en);
+        const int i = 31 - __builtin_clz(e0);
-        en &= ~(1<<i);
+        e0 &= ~(1<<i);
         track_t& t = state->tracks[i];
         t.bufferProvider->releaseBuffer(&t.buffer);
     }
 }
 
-// generic code with resampling
+
-void AudioMixer::process__genericResampling(state_t* state, void* output)
+  // generic code with resampling
+void AudioMixer::process__genericResampling(state_t* state)
 {
     int32_t* const outTemp = state->outputTemp;
     const size_t size = sizeof(int32_t) * MAX_NUM_CHANNELS * state->frameCount;
     memset(outTemp, 0, size);
 
-    int32_t* out = static_cast<int32_t*>(output);
     size_t numFrames = state->frameCount;
 
-    uint32_t en = state->enabledTracks;
+    uint32_t e0 = state->enabledTracks;
-    while (en) {
+    while (e0) {
-        const int i = 31 - __builtin_clz(en);
+        // process by group of tracks with same output buffer
-        en &= ~(1<<i);
+        // to optimize cache use
-        track_t& t = state->tracks[i];
+        uint32_t e1 = e0, e2 = e0;
-
+        int j = 31 - __builtin_clz(e1);
-        // this is a little goofy, on the resampling case we don't
+        track_t& t1 = state->tracks[j];
-        // acquire/release the buffers because it's done by
+        e2 &= ~(1<<j);
-        // the resampler.
+        while (e2) {
-        if ((t.needs & NEEDS_RESAMPLE__MASK) == NEEDS_RESAMPLE_ENABLED) {
+            j = 31 - __builtin_clz(e2);
-            (t.hook)(&t, outTemp, numFrames, state->resampleTemp);
+            e2 &= ~(1<<j);
-        } else {
+            track_t& t2 = state->tracks[j];
-
+            if UNLIKELY(t2.mainBuffer != t1.mainBuffer) {
-            size_t outFrames = numFrames;
+                e1 &= ~(1<<j);
-           
-            while (outFrames) {
-                t.buffer.frameCount = outFrames;
-                t.bufferProvider->getNextBuffer(&t.buffer);
-                t.in = t.buffer.raw;
-                // t.in == NULL can happen if the track was flushed just after having
-                // been enabled for mixing.
-                if (t.in == NULL) break;
-
-                (t.hook)(&t, outTemp + (numFrames-outFrames)*MAX_NUM_CHANNELS, t.buffer.frameCount, state->resampleTemp);
-                outFrames -= t.buffer.frameCount;
-                t.bufferProvider->releaseBuffer(&t.buffer);
             }
         }
-    }
+        e0 &= ~(e1);
+        int32_t *out = t1.mainBuffer;
+        while (e1) {
+            const int i = 31 - __builtin_clz(e1);
+            e1 &= ~(1<<i);
+            track_t& t = state->tracks[i];
+            int32_t *aux = NULL;
+            if UNLIKELY((t.needs & NEEDS_AUX__MASK) == NEEDS_AUX_ENABLED) {
+                aux = t.auxBuffer;
+            }
 
-    ditherAndClamp(out, outTemp, numFrames);
+            // this is a little goofy, on the resampling case we don't
+            // acquire/release the buffers because it's done by
+            // the resampler.
+            if ((t.needs & NEEDS_RESAMPLE__MASK) == NEEDS_RESAMPLE_ENABLED) {
+                (t.hook)(&t, outTemp, numFrames, state->resampleTemp, aux);
+            } else {
+
+                size_t outFrames = 0;
+
+                while (outFrames < numFrames) {
+                    t.buffer.frameCount = numFrames - outFrames;
+                    t.bufferProvider->getNextBuffer(&t.buffer);
+                    t.in = t.buffer.raw;
+                    // t.in == NULL can happen if the track was flushed just after having
+                    // been enabled for mixing.
+                    if (t.in == NULL) break;
+
+                    if UNLIKELY(aux != NULL) {
+                        aux += outFrames;
+                    }
+                    (t.hook)(&t, outTemp + outFrames*MAX_NUM_CHANNELS, t.buffer.frameCount, state->resampleTemp, aux);
+                    outFrames += t.buffer.frameCount;
+                    t.bufferProvider->releaseBuffer(&t.buffer);
+                }
+            }
+        }
+        ditherAndClamp(out, outTemp, numFrames);
+    }
 }
 
 // one track, 16 bits stereo without resampling is the most common case
-void AudioMixer::process__OneTrack16BitsStereoNoResampling(state_t* state, void* output)
+void AudioMixer::process__OneTrack16BitsStereoNoResampling(state_t* state)
 {
     const int i = 31 - __builtin_clz(state->enabledTracks);
     const track_t& t = state->tracks[i];
 
     AudioBufferProvider::Buffer& b(t.buffer);
 
-    int32_t* out = static_cast<int32_t*>(output);
+    int32_t* out = t.mainBuffer;
     size_t numFrames = state->frameCount;
 
     const int16_t vl = t.volume[0];
@@ -816,7 +1093,9 @@ void AudioMixer::process__OneTrack16BitsStereoNoResampling(state_t* state, void*
 }
 
 // 2 tracks is also a common case
-void AudioMixer::process__TwoTracks16BitsStereoNoResampling(state_t* state, void* output)
+// NEVER used in current implementation of process__validate()
+// only use if the 2 tracks have the same output buffer
+void AudioMixer::process__TwoTracks16BitsStereoNoResampling(state_t* state)
 {
     int i;
     uint32_t en = state->enabledTracks;
@@ -840,7 +1119,8 @@ void AudioMixer::process__TwoTracks16BitsStereoNoResampling(state_t* state, void
     const int16_t vr1 = t1.volume[1];
     size_t frameCount1 = 0;
 
-    int32_t* out = static_cast<int32_t*>(output);
+    //FIXME: only works if two tracks use same buffer
+    int32_t* out = t0.mainBuffer;
     size_t numFrames = state->frameCount;
     int16_t const *buff = NULL;
 

libs/audioflinger/AudioMixer.h

@@ -63,11 +63,14 @@ public:
         // for target TRACK
         CHANNEL_COUNT   = 0x4000,
         FORMAT          = 0x4001,
+        MAIN_BUFFER     = 0x4002,
+        AUX_BUFFER      = 0x4003,
         // for TARGET RESAMPLE
         SAMPLE_RATE     = 0x4100,
         // for TARGET VOLUME (8 channels max)
         VOLUME0         = 0x4200,
         VOLUME1         = 0x4201,
+        AUXLEVEL        = 0x4210,
     };
 
 
@@ -78,10 +81,10 @@ public:
     status_t    disable(int name);
 
     status_t    setActiveTrack(int track);
-    status_t    setParameter(int target, int name, int value);
+    status_t    setParameter(int target, int name, void *value);
 
     status_t    setBufferProvider(AudioBufferProvider* bufferProvider);
-    void        process(void* output);
+    void        process();
 
     uint32_t    trackNames() const { return mTrackNames; }
 
@@ -94,6 +97,7 @@ private:
         NEEDS_FORMAT__MASK          = 0x000000F0,
         NEEDS_MUTE__MASK            = 0x00000100,
         NEEDS_RESAMPLE__MASK        = 0x00001000,
+        NEEDS_AUX__MASK             = 0x00010000,
     };
 
     enum {
@@ -107,6 +111,9 @@ private:
 
         NEEDS_RESAMPLE_DISABLED     = 0x00000000,
         NEEDS_RESAMPLE_ENABLED      = 0x00001000,
+
+        NEEDS_AUX_DISABLED     = 0x00000000,
+        NEEDS_AUX_ENABLED      = 0x00010000,
     };
 
     static inline int32_t applyVolume(int32_t in, int32_t v) {
@@ -115,9 +122,10 @@ private:
 
 
     struct state_t;
+    struct track_t;
 
-    typedef void (*mix_t)(state_t* state, void* output);
+    typedef void (*mix_t)(state_t* state);
-
+    typedef void (*hook_t)(track_t* t, int32_t* output, size_t numOutFrames, int32_t* temp, int32_t* aux);
     static const int BLOCKSIZE = 16; // 4 cache lines
 
     struct track_t {
@@ -131,6 +139,9 @@ private:
         int32_t     prevVolume[2];
 
         int32_t     volumeInc[2];
+        int32_t     auxLevel;
+        int32_t     auxInc;
+        int32_t     prevAuxLevel;
 
         uint16_t    frameCount;
 
@@ -142,15 +153,17 @@ private:
         AudioBufferProvider*                bufferProvider;
         mutable AudioBufferProvider::Buffer buffer;
 
-        void (*hook)(track_t* t, int32_t* output, size_t numOutFrames, int32_t* temp);
+        hook_t      hook;
         void const* in;             // current location in buffer
 
         AudioResampler*     resampler;
         uint32_t            sampleRate;
+        int32_t*           mainBuffer;
+        int32_t*           auxBuffer;
 
         bool        setResampler(uint32_t sampleRate, uint32_t devSampleRate);
         bool        doesResample() const;
-        void        adjustVolumeRamp();
+        void        adjustVolumeRamp(bool aux);
     };
 
     // pad to 32-bytes to fill cache line
@@ -173,18 +186,19 @@ private:
 
     void invalidateState(uint32_t mask);
 
-    static void track__genericResample(track_t* t, int32_t* out, size_t numFrames, int32_t* temp);
+    static void track__genericResample(track_t* t, int32_t* out, size_t numFrames, int32_t* temp, int32_t* aux);
-    static void track__nop(track_t* t, int32_t* out, size_t numFrames, int32_t* temp);
+    static void track__nop(track_t* t, int32_t* out, size_t numFrames, int32_t* temp, int32_t* aux);
-    static void volumeRampStereo(track_t* t, int32_t* out, size_t frameCount, int32_t* temp);
+    static void track__16BitsStereo(track_t* t, int32_t* out, size_t numFrames, int32_t* temp, int32_t* aux);
-    static void track__16BitsStereo(track_t* t, int32_t* out, size_t numFrames, int32_t* temp);
+    static void track__16BitsMono(track_t* t, int32_t* out, size_t numFrames, int32_t* temp, int32_t* aux);
-    static void track__16BitsMono(track_t* t, int32_t* out, size_t numFrames, int32_t* temp);
+    static void volumeRampStereo(track_t* t, int32_t* out, size_t frameCount, int32_t* temp, int32_t* aux);
+    static void volumeStereo(track_t* t, int32_t* out, size_t frameCount, int32_t* temp, int32_t* aux);
 
-    static void process__validate(state_t* state, void* output);
+    static void process__validate(state_t* state);
-    static void process__nop(state_t* state, void* output);
+    static void process__nop(state_t* state);
-    static void process__genericNoResampling(state_t* state, void* output);
+    static void process__genericNoResampling(state_t* state);
-    static void process__genericResampling(state_t* state, void* output);
+    static void process__genericResampling(state_t* state);
-    static void process__OneTrack16BitsStereoNoResampling(state_t* state, void* output);
+    static void process__OneTrack16BitsStereoNoResampling(state_t* state);
-    static void process__TwoTracks16BitsStereoNoResampling(state_t* state, void* output);
+    static void process__TwoTracks16BitsStereoNoResampling(state_t* state);
 };
 
 // ----------------------------------------------------------------------------

media/libeffects/EffectEqualizer.cpp

@@ -114,7 +114,7 @@ int Equalizer_setParameter(AudioEqualizer * pEqualizer, int32_t *pParam, void *p
 //--- Effect Library Interface Implementation
 //
 
-extern "C" int EffectQueryNumberEffects(int *pNumEffects) {
+extern "C" int EffectQueryNumberEffects(uint32_t *pNumEffects) {
     *pNumEffects = 1;
     gEffectIndex = 0;
     return 0;

media/libeffects/EffectReverb.c

@@ -18,7 +18,8 @@
 //
 #define LOG_NDEBUG 0
 #include <cutils/log.h>
-
+#include <stdlib.h>
+#include <string.h>
 #include <stdbool.h>
 #include "EffectReverb.h"
 #include "EffectsMath.h"
@@ -86,7 +87,7 @@ static const effect_descriptor_t * const gDescriptors[] = {
 
 /*--- Effect Library Interface Implementation ---*/
 
-int EffectQueryNumberEffects(int *pNumEffects) {
+int EffectQueryNumberEffects(uint32_t *pNumEffects) {
     *pNumEffects = sizeof(gDescriptors) / sizeof(const effect_descriptor_t *)
             - 1;
     gEffectIndex = 0;

media/libeffects/EffectReverb.h

@@ -292,7 +292,7 @@ typedef struct reverb_module_s {
  * Effect API
  *------------------------------------
 */
-int EffectQueryNumberEffects(int *pNumEffects);
+int EffectQueryNumberEffects(uint32_t *pNumEffects);
 int EffectQueryNext(effect_descriptor_t *pDescriptor);
 int EffectCreate(effect_uuid_t *effectUID, effect_interface_t *pInterface);
 int EffectRelease(effect_interface_t interface);

media/libeffects/EffectsFactory.c

@@ -39,7 +39,7 @@ static int gInitDone; // true is global initialization has been preformed
 static int init();
 static int loadLibrary(const char *libPath, int *handle);
 static int unloadLibrary(int handle);
-static int numEffectModules();
+static uint32_t numEffectModules();
 static int findEffect(effect_uuid_t *uuid, lib_entry_t **lib, effect_descriptor_t **desc);
 static void dumpEffectDescriptor(effect_descriptor_t *desc, char *str, size_t len);
 
@@ -96,7 +96,7 @@ const struct effect_interface_s gInterface = {
 //      Effect Factory Interface functions
 /////////////////////////////////////////////////
 
-int EffectQueryNumberEffects(int *pNumEffects)
+int EffectQueryNumberEffects(uint32_t *pNumEffects)
 {
     int ret = init();
     if (ret < 0) {
@@ -353,8 +353,8 @@ int loadLibrary(const char *libPath, int *handle)
     effect_QueryNextEffect_t queryFx;
     effect_CreateEffect_t createFx;
     effect_ReleaseEffect_t releaseFx;
-    int numFx;
+    uint32_t numFx;
-    int fx;
+    uint32_t fx;
     int ret;
     list_elem_t *e, *descHead = NULL;
     lib_entry_t *l;
@@ -525,9 +525,9 @@ int unloadLibrary(int handle)
 
 
 
-int numEffectModules() {
+uint32_t numEffectModules() {
     list_elem_t *e = gLibraryList;
-    int cnt = 0;
+    uint32_t cnt = 0;
 
     // Reset pointers for EffectQueryNext()
     gCurLib = e;

media/libmedia/AudioRecord.cpp

@@ -45,7 +45,7 @@ namespace android {
 // ---------------------------------------------------------------------------
 
 AudioRecord::AudioRecord()
-    : mStatus(NO_INIT)
+    : mStatus(NO_INIT), mSessionId(0)
 {
 }
 
@@ -58,11 +58,12 @@ AudioRecord::AudioRecord(
         uint32_t flags,
         callback_t cbf,
         void* user,
-        int notificationFrames)
+        int notificationFrames,
-    : mStatus(NO_INIT)
+        int sessionId)
+    : mStatus(NO_INIT), mSessionId(0)
 {
     mStatus = set(inputSource, sampleRate, format, channels,
-            frameCount, flags, cbf, user, notificationFrames);
+            frameCount, flags, cbf, user, notificationFrames, sessionId);
 }
 
 AudioRecord::~AudioRecord()
@@ -91,7 +92,8 @@ status_t AudioRecord::set(
         callback_t cbf,
         void* user,
         int notificationFrames,
-        bool threadCanCallJava)
+        bool threadCanCallJava,
+        int sessionId)
 {
 
     LOGV("set(): sampleRate %d, channels %d, frameCount %d",sampleRate, channels, frameCount);
@@ -119,6 +121,7 @@ status_t AudioRecord::set(
     if (!AudioSystem::isInputChannel(channels)) {
         return BAD_VALUE;
     }
+
     int channelCount = AudioSystem::popCount(channels);
 
     audio_io_handle_t input = AudioSystem::getInput(inputSource,
@@ -164,6 +167,8 @@ status_t AudioRecord::set(
         notificationFrames = frameCount/2;
     }
 
+    mSessionId = sessionId;
+
     // create the IAudioRecord
     status_t status = openRecord(sampleRate, format, channelCount,
                                  frameCount, flags, input);
@@ -414,6 +419,7 @@ status_t AudioRecord::openRecord(
                                                        channelCount,
                                                        frameCount,
                                                        ((uint16_t)flags) << 16,
+                                                       &mSessionId,
                                                        &status);
     if (record == 0) {
         LOGE("AudioFlinger could not create record track, status: %d", status);
@@ -532,6 +538,11 @@ audio_io_handle_t AudioRecord::getInput()
     return mInput;
 }
 
+int AudioRecord::getSessionId()
+{
+    return mSessionId;
+}
+
 // -------------------------------------------------------------------------
 
 ssize_t AudioRecord::read(void* buffer, size_t userSize)

media/libmedia/AudioSystem.cpp

@@ -364,6 +364,12 @@ unsigned int AudioSystem::getInputFramesLost(audio_io_handle_t ioHandle) {
     return result;
 }
 
+int AudioSystem::newAudioSessionId() {
+    const sp<IAudioFlinger>& af = AudioSystem::get_audio_flinger();
+    if (af == 0) return 0;
+    return af->newAudioSessionId();
+}
+
 // ---------------------------------------------------------------------------
 
 void AudioSystem::AudioFlingerClient::binderDied(const wp<IBinder>& who) {

media/libmedia/AudioTrack.cpp

@@ -58,7 +58,8 @@ AudioTrack::AudioTrack(
         uint32_t flags,
         callback_t cbf,
         void* user,
-        int notificationFrames)
+        int notificationFrames,
+        int sessionId)
     : mStatus(NO_INIT)
 {
     mStatus = set(streamType, sampleRate, format, channels,
@@ -74,7 +75,8 @@ AudioTrack::AudioTrack(
         uint32_t flags,
         callback_t cbf,
         void* user,
-        int notificationFrames)
+        int notificationFrames,
+        int sessionId)
     : mStatus(NO_INIT)
 {
     mStatus = set(streamType, sampleRate, format, channels,
@@ -110,7 +112,8 @@ status_t AudioTrack::set(
         void* user,
         int notificationFrames,
         const sp<IMemory>& sharedBuffer,
-        bool threadCanCallJava)
+        bool threadCanCallJava,
+        int sessionId)
 {
 
     LOGV_IF(sharedBuffer != 0, "sharedBuffer: %p, size: %d", sharedBuffer->pointer(), sharedBuffer->size());
@@ -171,8 +174,11 @@ status_t AudioTrack::set(
 
     mVolume[LEFT] = 1.0f;
     mVolume[RIGHT] = 1.0f;
+    mSendLevel = 0;
     mFrameCount = frameCount;
     mNotificationFramesReq = notificationFrames;
+    mSessionId = sessionId;
+
     // create the IAudioTrack
     status_t status = createTrack(streamType, sampleRate, format, channelCount,
                                   frameCount, flags, sharedBuffer, output, true);
@@ -396,19 +402,49 @@ bool AudioTrack::muted() const
     return mMuted;
 }
 
-void AudioTrack::setVolume(float left, float right)
+status_t AudioTrack::setVolume(float left, float right)
 {
+    if (left > 1.0f || right > 1.0f) {
+        return BAD_VALUE;
+    }
+
     mVolume[LEFT] = left;
     mVolume[RIGHT] = right;
 
     // write must be atomic
-    mCblk->volumeLR = (int32_t(int16_t(left * 0x1000)) << 16) | int16_t(right * 0x1000);
+    mCblk->volumeLR = (uint32_t(uint16_t(right * 0x1000)) << 16) | uint16_t(left * 0x1000);
+
+    return NO_ERROR;
 }
 
 void AudioTrack::getVolume(float* left, float* right)
 {
-    *left  = mVolume[LEFT];
+    if (left != NULL) {
-    *right = mVolume[RIGHT];
+        *left  = mVolume[LEFT];
+    }
+    if (right != NULL) {
+        *right = mVolume[RIGHT];
+    }
+}
+
+status_t AudioTrack::setSendLevel(float level)
+{
+    if (level > 1.0f) {
+        return BAD_VALUE;
+    }
+
+    mSendLevel = level;
+
+    mCblk->sendLevel = uint16_t(level * 0x1000);
+
+    return NO_ERROR;
+}
+
+void AudioTrack::getSendLevel(float* level)
+{
+    if (level != NULL) {
+        *level  = mSendLevel;
+    }
 }
 
 status_t AudioTrack::setSampleRate(int rate)
@@ -563,6 +599,16 @@ audio_io_handle_t AudioTrack::getOutput()
             mCblk->sampleRate, mFormat, mChannels, (AudioSystem::output_flags)mFlags);
 }
 
+int AudioTrack::getSessionId()
+{
+    return mSessionId;
+}
+
+status_t AudioTrack::attachAuxEffect(int effectId)
+{
+    return mAudioTrack->attachAuxEffect(effectId);
+}
+
 // -------------------------------------------------------------------------
 
 status_t AudioTrack::createTrack(
@@ -647,6 +693,7 @@ status_t AudioTrack::createTrack(
                                                       ((uint16_t)flags) << 16,
                                                       sharedBuffer,
                                                       output,
+                                                      &mSessionId,
                                                       &status);
 
     if (track == 0) {
@@ -672,7 +719,8 @@ status_t AudioTrack::createTrack(
         mCblk->stepUser(mCblk->frameCount);
     }
 
-    mCblk->volumeLR = (int32_t(int16_t(mVolume[LEFT] * 0x1000)) << 16) | int16_t(mVolume[RIGHT] * 0x1000);
+    mCblk->volumeLR = (uint32_t(uint16_t(mVolume[RIGHT] * 0x1000)) << 16) | uint16_t(mVolume[LEFT] * 0x1000);
+    mCblk->sendLevel = uint16_t(mSendLevel * 0x1000);
     mCblk->bufferTimeoutMs = MAX_STARTUP_TIMEOUT_MS;
     mCblk->waitTimeMs = 0;
     mRemainingFrames = mNotificationFramesAct;
@@ -1016,7 +1064,7 @@ audio_track_cblk_t::audio_track_cblk_t()
     : lock(Mutex::SHARED), cv(Condition::SHARED), user(0), server(0),
     userBase(0), serverBase(0), buffers(0), frameCount(0),
     loopStart(UINT_MAX), loopEnd(UINT_MAX), loopCount(0), volumeLR(0),
-    flags(0)
+    flags(0), sendLevel(0)
 {
 }
 

media/libmedia/IAudioFlinger.cpp

@@ -62,7 +62,14 @@ enum {
     SET_STREAM_OUTPUT,
     SET_VOICE_VOLUME,
     GET_RENDER_POSITION,
-    GET_INPUT_FRAMES_LOST
+    GET_INPUT_FRAMES_LOST,
+    NEW_AUDIO_SESSION_ID,
+    LOAD_EFFECT_LIBRARY,
+    UNLOAD_EFFECT_LIBRARY,
+    QUERY_NUM_EFFECTS,
+    QUERY_NEXT_EFFECT,
+    GET_EFFECT_DESCRIPTOR,
+    CREATE_EFFECT
 };
 
 class BpAudioFlinger : public BpInterface<IAudioFlinger>
@@ -83,6 +90,7 @@ public:
                                 uint32_t flags,
                                 const sp<IMemory>& sharedBuffer,
                                 int output,
+                                int *sessionId,
                                 status_t *status)
     {
         Parcel data, reply;
@@ -97,10 +105,19 @@ public:
         data.writeInt32(flags);
         data.writeStrongBinder(sharedBuffer->asBinder());
         data.writeInt32(output);
+        int lSessionId = 0;
+        if (sessionId != NULL) {
+            lSessionId = *sessionId;
+        }
+        data.writeInt32(lSessionId);
         status_t lStatus = remote()->transact(CREATE_TRACK, data, &reply);
         if (lStatus != NO_ERROR) {
             LOGE("createTrack error: %s", strerror(-lStatus));
         } else {
+            lSessionId = reply.readInt32();
+            if (sessionId != NULL) {
+                *sessionId = lSessionId;
+            }
             lStatus = reply.readInt32();
             track = interface_cast<IAudioTrack>(reply.readStrongBinder());
         }
@@ -118,6 +135,7 @@ public:
                                 int channelCount,
                                 int frameCount,
                                 uint32_t flags,
+                                int *sessionId,
                                 status_t *status)
     {
         Parcel data, reply;
@@ -130,10 +148,19 @@ public:
         data.writeInt32(channelCount);
         data.writeInt32(frameCount);
         data.writeInt32(flags);
+        int lSessionId = 0;
+        if (sessionId != NULL) {
+            lSessionId = *sessionId;
+        }
+        data.writeInt32(lSessionId);
         status_t lStatus = remote()->transact(OPEN_RECORD, data, &reply);
         if (lStatus != NO_ERROR) {
             LOGE("openRecord error: %s", strerror(-lStatus));
         } else {
+            lSessionId = reply.readInt32();
+            if (sessionId != NULL) {
+                *sessionId = lSessionId;
+            }
             lStatus = reply.readInt32();
             record = interface_cast<IAudioRecord>(reply.readStrongBinder());
         }
@@ -497,6 +524,157 @@ public:
         remote()->transact(GET_INPUT_FRAMES_LOST, data, &reply);
         return reply.readInt32();
     }
+
+    virtual int newAudioSessionId()
+    {
+        Parcel data, reply;
+        data.writeInterfaceToken(IAudioFlinger::getInterfaceDescriptor());
+        status_t status = remote()->transact(NEW_AUDIO_SESSION_ID, data, &reply);
+        int id = 0;
+        if (status == NO_ERROR) {
+            id = reply.readInt32();
+        }
+        return id;
+    }
+
+    virtual status_t loadEffectLibrary(const char *libPath, int *handle)
+    {
+        if (libPath == NULL || handle == NULL) {
+            return BAD_VALUE;
+        }
+        *handle = 0;
+        Parcel data, reply;
+        data.writeInterfaceToken(IAudioFlinger::getInterfaceDescriptor());
+        data.writeCString(libPath);
+        status_t status = remote()->transact(LOAD_EFFECT_LIBRARY, data, &reply);
+        if (status == NO_ERROR) {
+            status = reply.readInt32();
+            if (status == NO_ERROR) {
+                *handle = reply.readInt32();
+            }
+        }
+        return status;
+    }
+
+    virtual status_t unloadEffectLibrary(int handle)
+    {
+        Parcel data, reply;
+        data.writeInterfaceToken(IAudioFlinger::getInterfaceDescriptor());
+        data.writeInt32(handle);
+        status_t status = remote()->transact(UNLOAD_EFFECT_LIBRARY, data, &reply);
+        if (status == NO_ERROR) {
+            status = reply.readInt32();
+        }
+        return status;
+    }
+
+    virtual status_t queryNumberEffects(uint32_t *numEffects)
+    {
+        Parcel data, reply;
+        data.writeInterfaceToken(IAudioFlinger::getInterfaceDescriptor());
+        status_t status = remote()->transact(QUERY_NUM_EFFECTS, data, &reply);
+        if (status != NO_ERROR) {
+            return status;
+        }
+        status = reply.readInt32();
+        if (status != NO_ERROR) {
+            return status;
+        }
+        if (numEffects) {
+            *numEffects = (uint32_t)reply.readInt32();
+        }
+        return NO_ERROR;
+    }
+
+    virtual status_t queryNextEffect(effect_descriptor_t *pDescriptor)
+    {
+        if (pDescriptor == NULL) {
+            return BAD_VALUE;
+        }
+        Parcel data, reply;
+        data.writeInterfaceToken(IAudioFlinger::getInterfaceDescriptor());
+        status_t status = remote()->transact(QUERY_NEXT_EFFECT, data, &reply);
+        if (status != NO_ERROR) {
+            return status;
+        }
+        status = reply.readInt32();
+        if (status != NO_ERROR) {
+            return status;
+        }
+        reply.read(pDescriptor, sizeof(effect_descriptor_t));
+        return NO_ERROR;
+    }
+
+    virtual status_t getEffectDescriptor(effect_uuid_t *pUuid, effect_descriptor_t *pDescriptor)
+    {
+        if (pUuid == NULL || pDescriptor == NULL) {
+            return BAD_VALUE;
+        }
+        Parcel data, reply;
+        data.writeInterfaceToken(IAudioFlinger::getInterfaceDescriptor());
+        data.write(pUuid, sizeof(effect_uuid_t));
+        status_t status = remote()->transact(GET_EFFECT_DESCRIPTOR, data, &reply);
+        if (status != NO_ERROR) {
+            return status;
+        }
+        status = reply.readInt32();
+        if (status != NO_ERROR) {
+            return status;
+        }
+        reply.read(pDescriptor, sizeof(effect_descriptor_t));
+        return NO_ERROR;
+    }
+
+    virtual sp<IEffect> createEffect(pid_t pid,
+                                    effect_descriptor_t *pDesc,
+                                    const sp<IEffectClient>& client,
+                                    int32_t priority,
+                                    int output,
+                                    int sessionId,
+                                    status_t *status,
+                                    int *id,
+                                    int *enabled)
+    {
+        Parcel data, reply;
+        sp<IEffect> effect;
+
+        if (pDesc == NULL) {
+             return effect;
+             if (status) {
+                 *status = BAD_VALUE;
+             }
+         }
+
+        data.writeInterfaceToken(IAudioFlinger::getInterfaceDescriptor());
+        data.writeInt32(pid);
+        data.write(pDesc, sizeof(effect_descriptor_t));
+        data.writeStrongBinder(client->asBinder());
+        data.writeInt32(priority);
+        data.writeInt32(output);
+        data.writeInt32(sessionId);
+
+        status_t lStatus = remote()->transact(CREATE_EFFECT, data, &reply);
+        if (lStatus != NO_ERROR) {
+            LOGE("createEffect error: %s", strerror(-lStatus));
+        } else {
+            lStatus = reply.readInt32();
+            int tmp = reply.readInt32();
+            if (id) {
+                *id = tmp;
+            }
+            tmp = reply.readInt32();
+            if (enabled) {
+                *enabled = tmp;
+            }
+            effect = interface_cast<IEffect>(reply.readStrongBinder());
+            reply.read(pDesc, sizeof(effect_descriptor_t));
+        }
+        if (status) {
+            *status = lStatus;
+        }
+
+        return effect;
+    }
 };
 
 IMPLEMENT_META_INTERFACE(AudioFlinger, "android.media.IAudioFlinger");
@@ -518,10 +696,12 @@ status_t BnAudioFlinger::onTransact(
             uint32_t flags = data.readInt32();
             sp<IMemory> buffer = interface_cast<IMemory>(data.readStrongBinder());
             int output = data.readInt32();
+            int sessionId = data.readInt32();
             status_t status;
             sp<IAudioTrack> track = createTrack(pid,
                     streamType, sampleRate, format,
-                    channelCount, bufferCount, flags, buffer, output, &status);
+                    channelCount, bufferCount, flags, buffer, output, &sessionId, &status);
+            reply->writeInt32(sessionId);
             reply->writeInt32(status);
             reply->writeStrongBinder(track->asBinder());
             return NO_ERROR;
@@ -535,9 +715,11 @@ status_t BnAudioFlinger::onTransact(
             int channelCount = data.readInt32();
             size_t bufferCount = data.readInt32();
             uint32_t flags = data.readInt32();
+            int sessionId = data.readInt32();
             status_t status;
             sp<IAudioRecord> record = openRecord(pid, input,
-                    sampleRate, format, channelCount, bufferCount, flags, &status);
+                    sampleRate, format, channelCount, bufferCount, flags, &sessionId, &status);
+            reply->writeInt32(sessionId);
             reply->writeInt32(status);
             reply->writeStrongBinder(record->asBinder());
             return NO_ERROR;
@@ -768,7 +950,79 @@ status_t BnAudioFlinger::onTransact(
             reply->writeInt32(getInputFramesLost(ioHandle));
             return NO_ERROR;
         } break;
+        case NEW_AUDIO_SESSION_ID: {
+            CHECK_INTERFACE(IAudioFlinger, data, reply);
+            reply->writeInt32(newAudioSessionId());
+            return NO_ERROR;
+        } break;
+        case LOAD_EFFECT_LIBRARY: {
+            CHECK_INTERFACE(IAudioFlinger, data, reply);
+            int handle;
+            status_t status = loadEffectLibrary(data.readCString(), &handle);
+            reply->writeInt32(status);
+            if (status == NO_ERROR) {
+                reply->writeInt32(handle);
+            }
+            return NO_ERROR;
+        }
+        case UNLOAD_EFFECT_LIBRARY: {
+            CHECK_INTERFACE(IAudioFlinger, data, reply);
+            reply->writeInt32(unloadEffectLibrary(data.readInt32()));
+            return NO_ERROR;
+        }
+        case QUERY_NUM_EFFECTS: {
+            CHECK_INTERFACE(IAudioFlinger, data, reply);
+            uint32_t numEffects;
+            status_t status = queryNumberEffects(&numEffects);
+            reply->writeInt32(status);
+            if (status == NO_ERROR) {
+                reply->writeInt32((int32_t)numEffects);
+            }
+            return NO_ERROR;
+        }
+        case QUERY_NEXT_EFFECT: {
+            CHECK_INTERFACE(IAudioFlinger, data, reply);
+            effect_descriptor_t desc;
+            status_t status = queryNextEffect(&desc);
+            reply->writeInt32(status);
+            if (status == NO_ERROR) {
+                reply->write(&desc, sizeof(effect_descriptor_t));
+            }
+            return NO_ERROR;
+        }
+        case GET_EFFECT_DESCRIPTOR: {
+            CHECK_INTERFACE(IAudioFlinger, data, reply);
+            effect_uuid_t uuid;
+            data.read(&uuid, sizeof(effect_uuid_t));
+            effect_descriptor_t desc;
+            status_t status = getEffectDescriptor(&uuid, &desc);
+            reply->writeInt32(status);
+            if (status == NO_ERROR) {
+                reply->write(&desc, sizeof(effect_descriptor_t));
+            }
+            return NO_ERROR;
+        }
+        case CREATE_EFFECT: {
+            CHECK_INTERFACE(IAudioFlinger, data, reply);
+            pid_t pid = data.readInt32();
+            effect_descriptor_t desc;
+            data.read(&desc, sizeof(effect_descriptor_t));
+            sp<IEffectClient> client = interface_cast<IEffectClient>(data.readStrongBinder());
+            int32_t priority = data.readInt32();
+            int output = data.readInt32();
+            int sessionId = data.readInt32();
+            status_t status;
+            int id;
+            int enabled;
 
+            sp<IEffect> effect = createEffect(pid, &desc, client, priority, output, sessionId, &status, &id, &enabled);
+            reply->writeInt32(status);
+            reply->writeInt32(id);
+            reply->writeInt32(enabled);
+            reply->writeStrongBinder(effect->asBinder());
+            reply->write(&desc, sizeof(effect_descriptor_t));
+            return NO_ERROR;
+        } break;
         default:
             return BBinder::onTransact(code, data, reply, flags);
     }

media/libmedia/IAudioTrack.cpp

@@ -34,7 +34,8 @@ enum {
     STOP,
     FLUSH,
     MUTE,
-    PAUSE
+    PAUSE,
+    ATTACH_AUX_EFFECT
 };
 
 class BpAudioTrack : public BpInterface<IAudioTrack>
@@ -98,6 +99,20 @@ public:
         }
         return cblk;
     }
+
+    virtual status_t attachAuxEffect(int effectId)
+    {
+        Parcel data, reply;
+        data.writeInterfaceToken(IAudioTrack::getInterfaceDescriptor());
+        data.writeInt32(effectId);
+        status_t status = remote()->transact(ATTACH_AUX_EFFECT, data, &reply);
+        if (status == NO_ERROR) {
+            status = reply.readInt32();
+        } else {
+            LOGW("attachAuxEffect() error: %s", strerror(-status));
+        }
+        return status;
+    }
 };
 
 IMPLEMENT_META_INTERFACE(AudioTrack, "android.media.IAudioTrack");
@@ -138,6 +153,11 @@ status_t BnAudioTrack::onTransact(
             pause();
             return NO_ERROR;
         }
+        case ATTACH_AUX_EFFECT: {
+            CHECK_INTERFACE(IAudioTrack, data, reply);
+            reply->writeInt32(attachAuxEffect(data.readInt32()));
+            return NO_ERROR;
+        } break;
         default:
             return BBinder::onTransact(code, data, reply, flags);
     }