멀티미디어 예제

02_day/test-HAL/atoi.c

@@ -0,0 +1,20 @@
+#include "main.h"
+
+
+int my_atoi(char *buff)
+{
+	int i, sum=0;
+	for(i=0; buff[i]; i++ )
+		sum = sum*10 + buff[i] - '0';
+
+	return sum;
+}
+
+
+int my_add(int a, int b )
+{
+	return a+b;
+}
+
+
+struct _HMI HMI = { my_atoi, my_add, 10 };

02_day/test-HAL/main.c

@@ -0,0 +1,26 @@
+#include <stdio.h>
+#include <dlfcn.h>
+
+int main()
+{
+	int *global;
+	int (*func)(char*);
+	int (*add)(int,int);
+	int data;
+	void *handle = dlopen("libatoi.so", RTLD_LAZY);
+
+	func = dlsym( handle, "my_atoi" );
+	data = func("123");
+	printf("data=%d\n", data );
+
+	add  = dlsym( handle, "my_add" );
+	data = add(1,2);
+	printf("data=%d\n", data );
+
+	*global  = dlsym( handle, "global" );
+	data = add(1,2);
+	printf("data=%d\n", data );
+	dlclose(handle);
+}
+
+

02_day/test-HAL/main.h

@@ -0,0 +1,10 @@
+
+int my_atoi(char *buff);
+int my_add(int a, int b);
+
+struct _HMI
+{
+	int (*atoi)(char *buff);
+	int (*add)(int , int );
+	int global;
+};

02_day/test-ashmem1/Android.mk

@@ -0,0 +1,14 @@
+LOCAL_PATH:= $(call my-dir)
+include $(CLEAR_VARS)
+LOCAL_SRC_FILES:= server.cpp
+LOCAL_MODULE := my_server
+LOCAL_SHARED_LIBRARIES:= libcutils libutils libbinder
+LOCAL_MODULE_TAGS := optional
+include $(BUILD_EXECUTABLE)
+
+include $(CLEAR_VARS)
+LOCAL_SRC_FILES:= client.cpp
+LOCAL_MODULE := my_client
+LOCAL_SHARED_LIBRARIES:= libcutils libutils libbinder
+LOCAL_MODULE_TAGS := optional
+include $(BUILD_EXECUTABLE)

02_day/test-ashmem1/client.cpp

@@ -0,0 +1,17 @@
+#include <binder/IServiceManager.h>
+#include <utils/StrongPointer.h>
+#include <binder/MemoryHeapBase.h>
+#include <binder/IPCThreadState.h>
+#include <stdio.h>
+
+using namespace android;
+
+int main()
+{
+	sp<IServiceManager> sm = defaultServiceManager();
+	sp<IBinder> binder = sm->getService( String16("my.ashmem1") );
+	sp<IMemoryHeap> heap = interface_cast<IMemoryHeap>(binder); 
+	char *p = (char*)heap->getBase();
+	printf("[%s]\n", p );
+	return 0;
+}

02_day/test-ashmem1/server.cpp

@@ -0,0 +1,18 @@
+#include <binder/IServiceManager.h>
+#include <utils/StrongPointer.h>
+#include <binder/MemoryHeapBase.h>
+#include <binder/IPCThreadState.h>
+
+using namespace android;
+
+int main()
+{
+	sp<IServiceManager> sm = defaultServiceManager();
+	sp<MemoryHeapBase>  heap = new MemoryHeapBase(4096);
+	sm->addService( String16("my.ashmem1") , heap );
+
+	char *p = (char*)heap->getBase();
+	sprintf(p, "Hello Client!!\n" );
+	IPCThreadState::self()->joinThreadPool();
+	return 0;
+}

02_day/test-ashmem2/Android.mk

@@ -0,0 +1,14 @@
+LOCAL_PATH:= $(call my-dir)
+include $(CLEAR_VARS)
+LOCAL_SRC_FILES:= server.cpp
+LOCAL_MODULE := my_server
+LOCAL_SHARED_LIBRARIES:= libcutils libutils libbinder
+LOCAL_MODULE_TAGS := optional
+include $(BUILD_EXECUTABLE)
+
+include $(CLEAR_VARS)
+LOCAL_SRC_FILES:= client.cpp
+LOCAL_MODULE := my_client
+LOCAL_SHARED_LIBRARIES:= libcutils libutils libbinder
+LOCAL_MODULE_TAGS := optional
+include $(BUILD_EXECUTABLE)

02_day/test-ashmem2/client.cpp

@@ -0,0 +1,21 @@
+#include <binder/IServiceManager.h>
+#include <utils/StrongPointer.h>
+#include <binder/MemoryHeapBase.h>
+#include <binder/IPCThreadState.h>
+#include <stdio.h>
+
+using namespace android;
+
+int main()
+{
+	sp<IServiceManager> sm = defaultServiceManager();
+	sp<IBinder> binder = sm->getService( String16("my.ashmem1") );
+	sp<IMemory> memory = interface_cast<IMemory>(binder); 
+	ssize_t offset=0;
+	size_t size=0;
+    sp<IMemoryHeap> heap = memory->getMemory(&offset, &size);
+
+	char *p = (char*)heap->getBase();
+	printf("[%s]\n", p+offset );
+	return 0;
+}

02_day/test-ashmem2/server.cpp

@@ -0,0 +1,19 @@
+#include <binder/IServiceManager.h>
+#include <utils/StrongPointer.h>
+#include <binder/MemoryHeapBase.h>
+#include <binder/MemoryBase.h>
+#include <binder/IPCThreadState.h>
+
+using namespace android;
+
+int main()
+{
+	sp<IServiceManager> sm = defaultServiceManager();
+	sp<MemoryHeapBase>  heap = new MemoryHeapBase(4096);
+	sm->addService( String16("my.ashmem1") , new MemoryBase(heap, 100, 100) );
+
+	char *p = (char*)heap->getBase();
+	sprintf(p+100, "Hello Client!!\n" );
+	IPCThreadState::self()->joinThreadPool();
+	return 0;
+}

02_day/test-mixer/Android.mk

@@ -0,0 +1,36 @@
+LOCAL_PATH:= $(call my-dir)
+
+include $(CLEAR_VARS)
+
+LOCAL_SRC_FILES:= \
+	test-mixer.cpp \
+	AudioMixer.cpp.arm \
+
+LOCAL_C_INCLUDES := \
+	bionic \
+	bionic/libstdc++/include \
+	external/stlport/stlport \
+	$(call include-path-for, audio-effects) \
+	$(call include-path-for, audio-utils) \
+	frameworks/av/services/audioflinger
+
+LOCAL_STATIC_LIBRARIES := \
+	libsndfile
+
+LOCAL_SHARED_LIBRARIES := \
+	libstlport \
+	libeffects \
+	libnbaio \
+	libcommon_time_client \
+	libaudioresampler \
+	libaudioutils \
+	libdl \
+	libcutils \
+	libutils \
+	liblog
+
+LOCAL_MODULE:= test-mixer
+
+LOCAL_MODULE_TAGS := optional
+
+include $(BUILD_EXECUTABLE)

02_day/test-mixer/AudioMixer.h

@@ -0,0 +1,470 @@
+/*
+**
+** Copyright 2007, 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_AUDIO_MIXER_H
+#define ANDROID_AUDIO_MIXER_H
+
+#include <stdint.h>
+#include <sys/types.h>
+
+#include <utils/threads.h>
+
+#include <media/AudioBufferProvider.h>
+#include "AudioResampler.h"
+
+#include <hardware/audio_effect.h>
+#include <system/audio.h>
+#include <media/nbaio/NBLog.h>
+
+// FIXME This is actually unity gain, which might not be max in future, expressed in U.12
+#define MAX_GAIN_INT AudioMixer::UNITY_GAIN_INT
+
+namespace android {
+
+// ----------------------------------------------------------------------------
+
+class AudioMixer
+{
+public:
+                            AudioMixer(size_t frameCount, uint32_t sampleRate,
+                                       uint32_t maxNumTracks = MAX_NUM_TRACKS);
+
+    /*virtual*/             ~AudioMixer();  // non-virtual saves a v-table, restore if sub-classed
+
+
+    // This mixer has a hard-coded upper limit of 32 active track inputs.
+    // Adding support for > 32 tracks would require more than simply changing this value.
+    static const uint32_t MAX_NUM_TRACKS = 32;
+    // maximum number of channels supported by the mixer
+
+    // This mixer has a hard-coded upper limit of 8 channels for output.
+    static const uint32_t MAX_NUM_CHANNELS = 8;
+    static const uint32_t MAX_NUM_VOLUMES = 2; // stereo volume only
+    // maximum number of channels supported for the content
+    static const uint32_t MAX_NUM_CHANNELS_TO_DOWNMIX = AUDIO_CHANNEL_COUNT_MAX;
+
+    static const uint16_t UNITY_GAIN_INT = 0x1000;
+    static const float    UNITY_GAIN_FLOAT = 1.0f;
+
+    enum { // names
+
+        // track names (MAX_NUM_TRACKS units)
+        TRACK0          = 0x1000,
+
+        // 0x2000 is unused
+
+        // setParameter targets
+        TRACK           = 0x3000,
+        RESAMPLE        = 0x3001,
+        RAMP_VOLUME     = 0x3002, // ramp to new volume
+        VOLUME          = 0x3003, // don't ramp
+
+        // set Parameter names
+        // for target TRACK
+        CHANNEL_MASK    = 0x4000,
+        FORMAT          = 0x4001,
+        MAIN_BUFFER     = 0x4002,
+        AUX_BUFFER      = 0x4003,
+        DOWNMIX_TYPE    = 0X4004,
+        MIXER_FORMAT    = 0x4005, // AUDIO_FORMAT_PCM_(FLOAT|16_BIT)
+        MIXER_CHANNEL_MASK = 0x4006, // Channel mask for mixer output
+        // for target RESAMPLE
+        SAMPLE_RATE     = 0x4100, // Configure sample rate conversion on this track name;
+                                  // parameter 'value' is the new sample rate in Hz.
+                                  // Only creates a sample rate converter the first time that
+                                  // the track sample rate is different from the mix sample rate.
+                                  // If the new sample rate is the same as the mix sample rate,
+                                  // and a sample rate converter already exists,
+                                  // then the sample rate converter remains present but is a no-op.
+        RESET           = 0x4101, // Reset sample rate converter without changing sample rate.
+                                  // This clears out the resampler's input buffer.
+        REMOVE          = 0x4102, // Remove the sample rate converter on this track name;
+                                  // the track is restored to the mix sample rate.
+        // for target RAMP_VOLUME and VOLUME (8 channels max)
+        // FIXME use float for these 3 to improve the dynamic range
+        VOLUME0         = 0x4200,
+        VOLUME1         = 0x4201,
+        AUXLEVEL        = 0x4210,
+    };
+
+
+    // For all APIs with "name": TRACK0 <= name < TRACK0 + MAX_NUM_TRACKS
+
+    // Allocate a track name.  Returns new track name if successful, -1 on failure.
+    // The failure could be because of an invalid channelMask or format, or that
+    // the track capacity of the mixer is exceeded.
+    int         getTrackName(audio_channel_mask_t channelMask,
+                             audio_format_t format, int sessionId);
+
+    // Free an allocated track by name
+    void        deleteTrackName(int name);
+
+    // Enable or disable an allocated track by name
+    void        enable(int name);
+    void        disable(int name);
+
+    void        setParameter(int name, int target, int param, void *value);
+
+    void        setBufferProvider(int name, AudioBufferProvider* bufferProvider);
+    void        process(int64_t pts);
+
+    uint32_t    trackNames() const { return mTrackNames; }
+
+    size_t      getUnreleasedFrames(int name) const;
+
+    static inline bool isValidPcmTrackFormat(audio_format_t format) {
+        return format == AUDIO_FORMAT_PCM_16_BIT ||
+                format == AUDIO_FORMAT_PCM_24_BIT_PACKED ||
+                format == AUDIO_FORMAT_PCM_32_BIT ||
+                format == AUDIO_FORMAT_PCM_FLOAT;
+    }
+
+private:
+
+    enum {
+        // FIXME this representation permits up to 8 channels
+        NEEDS_CHANNEL_COUNT__MASK   = 0x00000007,
+    };
+
+    enum {
+        NEEDS_CHANNEL_1             = 0x00000000,   // mono
+        NEEDS_CHANNEL_2             = 0x00000001,   // stereo
+
+        // sample format is not explicitly specified, and is assumed to be AUDIO_FORMAT_PCM_16_BIT
+
+        NEEDS_MUTE                  = 0x00000100,
+        NEEDS_RESAMPLE              = 0x00001000,
+        NEEDS_AUX                   = 0x00010000,
+    };
+
+    struct state_t;
+    struct track_t;
+    class CopyBufferProvider;
+
+    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 {
+        uint32_t    needs;
+
+        // TODO: Eventually remove legacy integer volume settings
+        union {
+        int16_t     volume[MAX_NUM_VOLUMES]; // U4.12 fixed point (top bit should be zero)
+        int32_t     volumeRL;
+        };
+
+        int32_t     prevVolume[MAX_NUM_VOLUMES];
+
+        // 16-byte boundary
+
+        int32_t     volumeInc[MAX_NUM_VOLUMES];
+        int32_t     auxInc;
+        int32_t     prevAuxLevel;
+
+        // 16-byte boundary
+
+        int16_t     auxLevel;       // 0 <= auxLevel <= MAX_GAIN_INT, but signed for mul performance
+        uint16_t    frameCount;
+
+        uint8_t     channelCount;   // 1 or 2, redundant with (needs & NEEDS_CHANNEL_COUNT__MASK)
+        uint8_t     unused_padding; // formerly format, was always 16
+        uint16_t    enabled;        // actually bool
+        audio_channel_mask_t channelMask;
+
+        // actual buffer provider used by the track hooks, see DownmixerBufferProvider below
+        //  for how the Track buffer provider is wrapped by another one when dowmixing is required
+        AudioBufferProvider*                bufferProvider;
+
+        // 16-byte boundary
+
+        mutable AudioBufferProvider::Buffer buffer; // 8 bytes
+
+        hook_t      hook;
+        const void* in;             // current location in buffer
+
+        // 16-byte boundary
+
+        AudioResampler*     resampler;
+        uint32_t            sampleRate;
+        int32_t*           mainBuffer;
+        int32_t*           auxBuffer;
+
+        // 16-byte boundary
+        AudioBufferProvider*     mInputBufferProvider;    // externally provided buffer provider.
+        CopyBufferProvider*      mReformatBufferProvider; // provider wrapper for reformatting.
+        CopyBufferProvider*      downmixerBufferProvider; // wrapper for channel conversion.
+
+        int32_t     sessionId;
+
+        // 16-byte boundary
+        audio_format_t mMixerFormat;     // output mix format: AUDIO_FORMAT_PCM_(FLOAT|16_BIT)
+        audio_format_t mFormat;          // input track format
+        audio_format_t mMixerInFormat;   // mix internal format AUDIO_FORMAT_PCM_(FLOAT|16_BIT)
+                                         // each track must be converted to this format.
+
+        float          mVolume[MAX_NUM_VOLUMES];     // floating point set volume
+        float          mPrevVolume[MAX_NUM_VOLUMES]; // floating point previous volume
+        float          mVolumeInc[MAX_NUM_VOLUMES];  // floating point volume increment
+
+        float          mAuxLevel;                     // floating point set aux level
+        float          mPrevAuxLevel;                 // floating point prev aux level
+        float          mAuxInc;                       // floating point aux increment
+
+        // 16-byte boundary
+        audio_channel_mask_t mMixerChannelMask;
+        uint32_t             mMixerChannelCount;
+
+        bool        needsRamp() { return (volumeInc[0] | volumeInc[1] | auxInc) != 0; }
+        bool        setResampler(uint32_t trackSampleRate, uint32_t devSampleRate);
+        bool        doesResample() const { return resampler != NULL; }
+        void        resetResampler() { if (resampler != NULL) resampler->reset(); }
+        void        adjustVolumeRamp(bool aux, bool useFloat = false);
+        size_t      getUnreleasedFrames() const { return resampler != NULL ?
+                                                    resampler->getUnreleasedFrames() : 0; };
+    };
+
+    typedef void (*process_hook_t)(state_t* state, int64_t pts);
+
+    // pad to 32-bytes to fill cache line
+    struct state_t {
+        uint32_t        enabledTracks;
+        uint32_t        needsChanged;
+        size_t          frameCount;
+        process_hook_t  hook;   // one of process__*, never NULL
+        int32_t         *outputTemp;
+        int32_t         *resampleTemp;
+        NBLog::Writer*  mLog;
+        int32_t         reserved[1];
+        // FIXME allocate dynamically to save some memory when maxNumTracks < MAX_NUM_TRACKS
+        track_t         tracks[MAX_NUM_TRACKS] __attribute__((aligned(32)));
+    };
+
+    // Base AudioBufferProvider class used for DownMixerBufferProvider, RemixBufferProvider,
+    // and ReformatBufferProvider.
+    // It handles a private buffer for use in converting format or channel masks from the
+    // input data to a form acceptable by the mixer.
+    // TODO: Make a ResamplerBufferProvider when integers are entirely removed from the
+    // processing pipeline.
+    class CopyBufferProvider : public AudioBufferProvider {
+    public:
+        // Use a private buffer of bufferFrameCount frames (each frame is outputFrameSize bytes).
+        // If bufferFrameCount is 0, no private buffer is created and in-place modification of
+        // the upstream buffer provider's buffers is performed by copyFrames().
+        CopyBufferProvider(size_t inputFrameSize, size_t outputFrameSize,
+                size_t bufferFrameCount);
+        virtual ~CopyBufferProvider();
+
+        // Overrides AudioBufferProvider methods
+        virtual status_t getNextBuffer(Buffer* buffer, int64_t pts);
+        virtual void releaseBuffer(Buffer* buffer);
+
+        // Other public methods
+
+        // call this to release the buffer to the upstream provider.
+        // treat it as an audio discontinuity for future samples.
+        virtual void reset();
+
+        // this function should be supplied by the derived class.  It converts
+        // #frames in the *src pointer to the *dst pointer.  It is public because
+        // some providers will allow this to work on arbitrary buffers outside
+        // of the internal buffers.
+        virtual void copyFrames(void *dst, const void *src, size_t frames) = 0;
+
+        // set the upstream buffer provider. Consider calling "reset" before this function.
+        void setBufferProvider(AudioBufferProvider *p) {
+            mTrackBufferProvider = p;
+        }
+
+    protected:
+        AudioBufferProvider* mTrackBufferProvider;
+        const size_t         mInputFrameSize;
+        const size_t         mOutputFrameSize;
+    private:
+        AudioBufferProvider::Buffer mBuffer;
+        const size_t         mLocalBufferFrameCount;
+        void*                mLocalBufferData;
+        size_t               mConsumed;
+    };
+
+    // DownmixerBufferProvider wraps a track AudioBufferProvider to provide
+    // position dependent downmixing by an Audio Effect.
+    class DownmixerBufferProvider : public CopyBufferProvider {
+    public:
+        DownmixerBufferProvider(audio_channel_mask_t inputChannelMask,
+                audio_channel_mask_t outputChannelMask, audio_format_t format,
+                uint32_t sampleRate, int32_t sessionId, size_t bufferFrameCount);
+        virtual ~DownmixerBufferProvider();
+        virtual void copyFrames(void *dst, const void *src, size_t frames);
+        bool isValid() const { return mDownmixHandle != NULL; }
+
+        static status_t init();
+        static bool isMultichannelCapable() { return sIsMultichannelCapable; }
+
+    protected:
+        effect_handle_t    mDownmixHandle;
+        effect_config_t    mDownmixConfig;
+
+        // effect descriptor for the downmixer used by the mixer
+        static effect_descriptor_t sDwnmFxDesc;
+        // indicates whether a downmix effect has been found and is usable by this mixer
+        static bool                sIsMultichannelCapable;
+        // FIXME: should we allow effects outside of the framework?
+        // We need to here. A special ioId that must be <= -2 so it does not map to a session.
+        static const int32_t SESSION_ID_INVALID_AND_IGNORED = -2;
+    };
+
+    // RemixBufferProvider wraps a track AudioBufferProvider to perform an
+    // upmix or downmix to the proper channel count and mask.
+    class RemixBufferProvider : public CopyBufferProvider {
+    public:
+        RemixBufferProvider(audio_channel_mask_t inputChannelMask,
+                audio_channel_mask_t outputChannelMask, audio_format_t format,
+                size_t bufferFrameCount);
+        virtual void copyFrames(void *dst, const void *src, size_t frames);
+
+    protected:
+        const audio_format_t mFormat;
+        const size_t         mSampleSize;
+        const size_t         mInputChannels;
+        const size_t         mOutputChannels;
+        int8_t               mIdxAry[sizeof(uint32_t)*8]; // 32 bits => channel indices
+    };
+
+    // ReformatBufferProvider wraps a track AudioBufferProvider to convert the input data
+    // to an acceptable mixer input format type.
+    class ReformatBufferProvider : public CopyBufferProvider {
+    public:
+        ReformatBufferProvider(int32_t channels,
+                audio_format_t inputFormat, audio_format_t outputFormat,
+                size_t bufferFrameCount);
+        virtual void copyFrames(void *dst, const void *src, size_t frames);
+
+    protected:
+        const int32_t        mChannels;
+        const audio_format_t mInputFormat;
+        const audio_format_t mOutputFormat;
+    };
+
+    // bitmask of allocated track names, where bit 0 corresponds to TRACK0 etc.
+    uint32_t        mTrackNames;
+
+    // bitmask of configured track names; ~0 if maxNumTracks == MAX_NUM_TRACKS,
+    // but will have fewer bits set if maxNumTracks < MAX_NUM_TRACKS
+    const uint32_t  mConfiguredNames;
+
+    const uint32_t  mSampleRate;
+
+    NBLog::Writer   mDummyLog;
+public:
+    void            setLog(NBLog::Writer* log);
+private:
+    state_t         mState __attribute__((aligned(32)));
+
+    // Call after changing either the enabled status of a track, or parameters of an enabled track.
+    // OK to call more often than that, but unnecessary.
+    void invalidateState(uint32_t mask);
+
+    bool setChannelMasks(int name,
+            audio_channel_mask_t trackChannelMask, audio_channel_mask_t mixerChannelMask);
+
+    // TODO: remove unused trackName/trackNum from functions below.
+    static status_t initTrackDownmix(track_t* pTrack, int trackName);
+    static status_t prepareTrackForDownmix(track_t* pTrack, int trackNum);
+    static void unprepareTrackForDownmix(track_t* pTrack, int trackName);
+    static status_t prepareTrackForReformat(track_t* pTrack, int trackNum);
+    static void unprepareTrackForReformat(track_t* pTrack, int trackName);
+    static void reconfigureBufferProviders(track_t* pTrack);
+
+    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, int32_t* aux);
+    static void track__16BitsStereo(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,
+            int32_t* aux);
+    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, int64_t pts);
+    static void process__nop(state_t* state, int64_t pts);
+    static void process__genericNoResampling(state_t* state, int64_t pts);
+    static void process__genericResampling(state_t* state, int64_t pts);
+    static void process__OneTrack16BitsStereoNoResampling(state_t* state,
+                                                          int64_t pts);
+
+    static int64_t calculateOutputPTS(const track_t& t, int64_t basePTS,
+                                      int outputFrameIndex);
+
+    static uint64_t         sLocalTimeFreq;
+    static pthread_once_t   sOnceControl;
+    static void             sInitRoutine();
+
+    /* multi-format volume mixing function (calls template functions
+     * in AudioMixerOps.h).  The template parameters are as follows:
+     *
+     *   MIXTYPE     (see AudioMixerOps.h MIXTYPE_* enumeration)
+     *   USEFLOATVOL (set to true if float volume is used)
+     *   ADJUSTVOL   (set to true if volume ramp parameters needs adjustment afterwards)
+     *   TO: int32_t (Q4.27) or float
+     *   TI: int32_t (Q4.27) or int16_t (Q0.15) or float
+     *   TA: int32_t (Q4.27)
+     */
+    template <int MIXTYPE, bool USEFLOATVOL, bool ADJUSTVOL,
+        typename TO, typename TI, typename TA>
+    static void volumeMix(TO *out, size_t outFrames,
+            const TI *in, TA *aux, bool ramp, AudioMixer::track_t *t);
+
+    // multi-format process hooks
+    template <int MIXTYPE, typename TO, typename TI, typename TA>
+    static void process_NoResampleOneTrack(state_t* state, int64_t pts);
+
+    // multi-format track hooks
+    template <int MIXTYPE, typename TO, typename TI, typename TA>
+    static void track__Resample(track_t* t, TO* out, size_t frameCount,
+            TO* temp __unused, TA* aux);
+    template <int MIXTYPE, typename TO, typename TI, typename TA>
+    static void track__NoResample(track_t* t, TO* out, size_t frameCount,
+            TO* temp __unused, TA* aux);
+
+    static void convertMixerFormat(void *out, audio_format_t mixerOutFormat,
+            void *in, audio_format_t mixerInFormat, size_t sampleCount);
+
+    // hook types
+    enum {
+        PROCESSTYPE_NORESAMPLEONETRACK,
+    };
+    enum {
+        TRACKTYPE_NOP,
+        TRACKTYPE_RESAMPLE,
+        TRACKTYPE_NORESAMPLE,
+        TRACKTYPE_NORESAMPLEMONO,
+    };
+
+    // functions for determining the proper process and track hooks.
+    static process_hook_t getProcessHook(int processType, uint32_t channelCount,
+            audio_format_t mixerInFormat, audio_format_t mixerOutFormat);
+    static hook_t getTrackHook(int trackType, uint32_t channelCount,
+            audio_format_t mixerInFormat, audio_format_t mixerOutFormat);
+};
+
+// ----------------------------------------------------------------------------
+}; // namespace android
+
+#endif // ANDROID_AUDIO_MIXER_H

02_day/test-mixer/test-mixer.cpp

@@ -0,0 +1,306 @@
+/*
+ * Copyright (C) 2014 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.
+ */
+
+#include <stdio.h>
+#include <inttypes.h>
+#include <math.h>
+#include <vector>
+#include <audio_utils/primitives.h>
+#include <audio_utils/sndfile.h>
+#include <media/AudioBufferProvider.h>
+#include "AudioMixer.h"
+#include "test_utils.h"
+
+/* Testing is typically through creation of an output WAV file from several
+ * source inputs, to be later analyzed by an audio program such as Audacity.
+ *
+ * Sine or chirp functions are typically more useful as input to the mixer
+ * as they show up as straight lines on a spectrogram if successfully mixed.
+ *
+ * A sample shell script is provided: mixer_to_wave_tests.sh
+ */
+
+using namespace android;
+
+static void usage(const char* name) {
+    fprintf(stderr, "Usage: %s [-f] [-m] [-c channels]"
+                    " [-s sample-rate] [-o <output-file>] [-a <aux-buffer-file>] [-P csv]"
+                    " (<input-file> | <command>)+\n", name);
+    fprintf(stderr, "    -f    enable floating point input track\n");
+    fprintf(stderr, "    -m    enable floating point mixer output\n");
+    fprintf(stderr, "    -c    number of mixer output channels\n");
+    fprintf(stderr, "    -s    mixer sample-rate\n");
+    fprintf(stderr, "    -o    <output-file> WAV file, pcm16 (or float if -m specified)\n");
+    fprintf(stderr, "    -a    <aux-buffer-file>\n");
+    fprintf(stderr, "    -P    # frames provided per call to resample() in CSV format\n");
+    fprintf(stderr, "    <input-file> is a WAV file\n");
+    fprintf(stderr, "    <command> can be 'sine:<channels>,<frequency>,<samplerate>'\n");
+    fprintf(stderr, "                     'chirp:<channels>,<samplerate>'\n");
+}
+
+static int writeFile(const char *filename, const void *buffer,
+        uint32_t sampleRate, uint32_t channels, size_t frames, bool isBufferFloat) {
+    if (filename == NULL) {
+        return 0; // ok to pass in NULL filename
+    }
+    // write output to file.
+    SF_INFO info;
+    info.frames = 0;
+    info.samplerate = sampleRate;
+    info.channels = channels;
+    info.format = SF_FORMAT_WAV | (isBufferFloat ? SF_FORMAT_FLOAT : SF_FORMAT_PCM_16);
+    printf("saving file:%s  channels:%u  samplerate:%u  frames:%zu\n",
+            filename, info.channels, info.samplerate, frames);
+    SNDFILE *sf = sf_open(filename, SFM_WRITE, &info);
+    if (sf == NULL) {
+        perror(filename);
+        return EXIT_FAILURE;
+    }
+    if (isBufferFloat) {
+        (void) sf_writef_float(sf, (float*)buffer, frames);
+    } else {
+        (void) sf_writef_short(sf, (short*)buffer, frames);
+    }
+    sf_close(sf);
+    return EXIT_SUCCESS;
+}
+
+int main(int argc, char* argv[]) {
+    const char* const progname = argv[0];
+    bool useInputFloat = false;
+    bool useMixerFloat = false;
+    bool useRamp = true;
+    uint32_t outputSampleRate = 48000;
+    uint32_t outputChannels = 2; // stereo for now
+    std::vector<int> Pvalues;
+    const char* outputFilename = NULL;
+    const char* auxFilename = NULL;
+    std::vector<int32_t> Names;
+    std::vector<SignalProvider> Providers;
+
+    for (int ch; (ch = getopt(argc, argv, "fmc:s:o:a:P:")) != -1;) {
+        switch (ch) {
+        case 'f':
+            useInputFloat = true;
+            break;
+        case 'm':
+            useMixerFloat = true;
+            break;
+        case 'c':
+            outputChannels = atoi(optarg);
+            break;
+        case 's':
+            outputSampleRate = atoi(optarg);
+            break;
+        case 'o':
+            outputFilename = optarg;
+            break;
+        case 'a':
+            auxFilename = optarg;
+            break;
+        case 'P':
+            if (parseCSV(optarg, Pvalues) < 0) {
+                fprintf(stderr, "incorrect syntax for -P option\n");
+                return EXIT_FAILURE;
+            }
+            break;
+        case '?':
+        default:
+            usage(progname);
+            return EXIT_FAILURE;
+        }
+    }
+    argc -= optind;
+    argv += optind;
+
+    if (argc == 0) {
+        usage(progname);
+        return EXIT_FAILURE;
+    }
+    if ((unsigned)argc > AudioMixer::MAX_NUM_TRACKS) {
+        fprintf(stderr, "too many tracks: %d > %u", argc, AudioMixer::MAX_NUM_TRACKS);
+        return EXIT_FAILURE;
+    }
+
+    size_t outputFrames = 0;
+
+    // create providers for each track
+    Providers.resize(argc);
+    for (int i = 0; i < argc; ++i) {
+        static const char chirp[] = "chirp:";
+        static const char sine[] = "sine:";
+        static const double kSeconds = 10;
+
+        if (!strncmp(argv[i], chirp, strlen(chirp))) {
+            std::vector<int> v;
+
+            parseCSV(argv[i] + strlen(chirp), v);
+            if (v.size() == 2) {
+                printf("creating chirp(%d %d)\n", v[0], v[1]);
+                if (useInputFloat) {
+                    Providers[i].setChirp<float>(v[0], 0, v[1]/2, v[1], kSeconds);
+                } else {
+                    Providers[i].setChirp<int16_t>(v[0], 0, v[1]/2, v[1], kSeconds);
+                }
+                Providers[i].setIncr(Pvalues);
+            } else {
+                fprintf(stderr, "malformed input '%s'\n", argv[i]);
+            }
+        } else if (!strncmp(argv[i], sine, strlen(sine))) {
+            std::vector<int> v;
+
+            parseCSV(argv[i] + strlen(sine), v);
+            if (v.size() == 3) {
+                printf("creating sine(%d %d %d)\n", v[0], v[1], v[2]);
+                if (useInputFloat) {
+                    Providers[i].setSine<float>(v[0], v[1], v[2], kSeconds);
+                } else {
+                    Providers[i].setSine<int16_t>(v[0], v[1], v[2], kSeconds);
+                }
+                Providers[i].setIncr(Pvalues);
+            } else {
+                fprintf(stderr, "malformed input '%s'\n", argv[i]);
+            }
+        } else {
+            printf("creating filename(%s)\n", argv[i]);
+            if (useInputFloat) {
+                Providers[i].setFile<float>(argv[i]);
+            } else {
+                Providers[i].setFile<short>(argv[i]);
+            }
+            Providers[i].setIncr(Pvalues);
+        }
+        // calculate the number of output frames
+        size_t nframes = (int64_t) Providers[i].getNumFrames() * outputSampleRate
+                / Providers[i].getSampleRate();
+        if (i == 0 || outputFrames > nframes) { // choose minimum for outputFrames
+            outputFrames = nframes;
+        }
+    }
+
+    // create the output buffer.
+    const size_t outputFrameSize = outputChannels
+            * (useMixerFloat ? sizeof(float) : sizeof(int16_t));
+    const size_t outputSize = outputFrames * outputFrameSize;
+    const audio_channel_mask_t outputChannelMask =
+            audio_channel_out_mask_from_count(outputChannels);
+    void *outputAddr = NULL;
+    (void) posix_memalign(&outputAddr, 32, outputSize);
+    memset(outputAddr, 0, outputSize);
+
+    // create the aux buffer, if needed.
+    const size_t auxFrameSize = sizeof(int32_t); // Q4.27 always
+    const size_t auxSize = outputFrames * auxFrameSize;
+    void *auxAddr = NULL;
+    if (auxFilename) {
+        (void) posix_memalign(&auxAddr, 32, auxSize);
+        memset(auxAddr, 0, auxSize);
+    }
+
+    // create the mixer.
+    const size_t mixerFrameCount = 320; // typical numbers may range from 240 or 960
+    AudioMixer *mixer = new AudioMixer(mixerFrameCount, outputSampleRate);
+    audio_format_t inputFormat = useInputFloat
+            ? AUDIO_FORMAT_PCM_FLOAT : AUDIO_FORMAT_PCM_16_BIT;
+    audio_format_t mixerFormat = useMixerFloat
+            ? AUDIO_FORMAT_PCM_FLOAT : AUDIO_FORMAT_PCM_16_BIT;
+    float f = AudioMixer::UNITY_GAIN_FLOAT / Providers.size(); // normalize volume by # tracks
+    static float f0; // zero
+
+    // set up the tracks.
+    for (size_t i = 0; i < Providers.size(); ++i) {
+        //printf("track %d out of %d\n", i, Providers.size());
+        uint32_t channelMask = audio_channel_out_mask_from_count(Providers[i].getNumChannels());
+        int32_t name = mixer->getTrackName(channelMask,
+                inputFormat, AUDIO_SESSION_OUTPUT_MIX);
+        ALOG_ASSERT(name >= 0);
+        Names.push_back(name);
+        mixer->setBufferProvider(name, &Providers[i]);
+        mixer->setParameter(name, AudioMixer::TRACK, AudioMixer::MAIN_BUFFER,
+                (void *)outputAddr);
+        mixer->setParameter(
+                name,
+                AudioMixer::TRACK,
+                AudioMixer::MIXER_FORMAT,
+                (void *)(uintptr_t)mixerFormat);
+        mixer->setParameter(
+                name,
+                AudioMixer::TRACK,
+                AudioMixer::FORMAT,
+                (void *)(uintptr_t)inputFormat);
+        mixer->setParameter(
+                name,
+                AudioMixer::TRACK,
+                AudioMixer::MIXER_CHANNEL_MASK,
+                (void *)(uintptr_t)outputChannelMask);
+        mixer->setParameter(
+                name,
+                AudioMixer::TRACK,
+                AudioMixer::CHANNEL_MASK,
+                (void *)(uintptr_t)channelMask);
+        mixer->setParameter(
+                name,
+                AudioMixer::RESAMPLE,
+                AudioMixer::SAMPLE_RATE,
+                (void *)(uintptr_t)Providers[i].getSampleRate());
+        if (useRamp) {
+            mixer->setParameter(name, AudioMixer::VOLUME, AudioMixer::VOLUME0, &f0);
+            mixer->setParameter(name, AudioMixer::VOLUME, AudioMixer::VOLUME1, &f0);
+            mixer->setParameter(name, AudioMixer::RAMP_VOLUME, AudioMixer::VOLUME0, &f);
+            mixer->setParameter(name, AudioMixer::RAMP_VOLUME, AudioMixer::VOLUME1, &f);
+        } else {
+            mixer->setParameter(name, AudioMixer::VOLUME, AudioMixer::VOLUME0, &f);
+            mixer->setParameter(name, AudioMixer::VOLUME, AudioMixer::VOLUME1, &f);
+        }
+        if (auxFilename) {
+            mixer->setParameter(name, AudioMixer::TRACK, AudioMixer::AUX_BUFFER,
+                    (void *) auxAddr);
+            mixer->setParameter(name, AudioMixer::VOLUME, AudioMixer::AUXLEVEL, &f0);
+            mixer->setParameter(name, AudioMixer::RAMP_VOLUME, AudioMixer::AUXLEVEL, &f);
+        }
+        mixer->enable(name);
+    }
+
+    // pump the mixer to process data.
+    size_t i;
+    for (i = 0; i < outputFrames - mixerFrameCount; i += mixerFrameCount) {
+        for (size_t j = 0; j < Names.size(); ++j) {
+            mixer->setParameter(Names[j], AudioMixer::TRACK, AudioMixer::MAIN_BUFFER,
+                    (char *) outputAddr + i * outputFrameSize);
+            if (auxFilename) {
+                mixer->setParameter(Names[j], AudioMixer::TRACK, AudioMixer::AUX_BUFFER,
+                        (char *) auxAddr + i * auxFrameSize);
+            }
+        }
+        mixer->process(AudioBufferProvider::kInvalidPTS);
+    }
+    outputFrames = i; // reset output frames to the data actually produced.
+
+    // write to files
+    writeFile(outputFilename, outputAddr,
+            outputSampleRate, outputChannels, outputFrames, useMixerFloat);
+    if (auxFilename) {
+        // Aux buffer is always in q4_27 format for now.
+        // memcpy_to_i16_from_q4_27(), but with stereo frame count (not sample count)
+        ditherAndClamp((int32_t*)auxAddr, (int32_t*)auxAddr, outputFrames >> 1);
+        writeFile(auxFilename, auxAddr, outputSampleRate, 1, outputFrames, false);
+    }
+
+    delete mixer;
+    free(outputAddr);
+    free(auxAddr);
+    return EXIT_SUCCESS;
+}

02_day/test-mixer/test_utils.h

@@ -0,0 +1,307 @@
+/*
+ * Copyright (C) 2014 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_AUDIO_TEST_UTILS_H
+#define ANDROID_AUDIO_TEST_UTILS_H
+
+#include <audio_utils/sndfile.h>
+
+#ifndef ARRAY_SIZE
+#define ARRAY_SIZE(a) (sizeof(a) / sizeof((a)[0]))
+#endif
+
+template<typename T, typename U>
+struct is_same
+{
+    static const bool value = false;
+};
+
+template<typename T>
+struct is_same<T, T>  // partial specialization
+{
+    static const bool value = true;
+};
+
+template<typename T>
+static inline T convertValue(double val)
+{
+    if (is_same<T, int16_t>::value) {
+        return floor(val * 32767.0 + 0.5);
+    } else if (is_same<T, int32_t>::value) {
+        return floor(val * (1UL<<31) + 0.5);
+    }
+    return val; // assume float or double
+}
+
+// Convert a list of integers in CSV format to a Vector of those values.
+// Returns the number of elements in the list, or -1 on error.
+static inline int parseCSV(const char *string, std::vector<int>& values)
+{
+    // pass 1: count the number of values and do syntax check
+    size_t numValues = 0;
+    bool hadDigit = false;
+    for (const char *p = string; ; ) {
+        switch (*p++) {
+        case '0': case '1': case '2': case '3': case '4':
+        case '5': case '6': case '7': case '8': case '9':
+            hadDigit = true;
+            break;
+        case '\0':
+            if (hadDigit) {
+                // pass 2: allocate and initialize vector of values
+                values.resize(++numValues);
+                values[0] = atoi(p = string);
+                for (size_t i = 1; i < numValues; ) {
+                    if (*p++ == ',') {
+                        values[i++] = atoi(p);
+                    }
+                }
+                return numValues;
+            }
+            // fall through
+        case ',':
+            if (hadDigit) {
+                hadDigit = false;
+                numValues++;
+                break;
+            }
+            // fall through
+        default:
+            return -1;
+        }
+    }
+}
+
+/* Creates a type-independent audio buffer provider from
+ * a buffer base address, size, framesize, and input increment array.
+ *
+ * No allocation or deallocation of the provided buffer is done.
+ */
+class TestProvider : public android::AudioBufferProvider {
+public:
+    TestProvider(void* addr, size_t frames, size_t frameSize,
+            const std::vector<int>& inputIncr)
+    : mAddr(addr),
+      mNumFrames(frames),
+      mFrameSize(frameSize),
+      mNextFrame(0), mUnrel(0), mInputIncr(inputIncr), mNextIdx(0)
+    {
+    }
+
+    TestProvider()
+    : mAddr(NULL), mNumFrames(0), mFrameSize(0),
+      mNextFrame(0), mUnrel(0), mNextIdx(0)
+    {
+    }
+
+    void setIncr(const std::vector<int>& inputIncr) {
+        mInputIncr = inputIncr;
+        mNextIdx = 0;
+    }
+
+    virtual android::status_t getNextBuffer(Buffer* buffer, int64_t pts __unused = kInvalidPTS)
+    {
+        size_t requestedFrames = buffer->frameCount;
+        if (requestedFrames > mNumFrames - mNextFrame) {
+            buffer->frameCount = mNumFrames - mNextFrame;
+        }
+        if (!mInputIncr.empty()) {
+            size_t provided = mInputIncr[mNextIdx++];
+            ALOGV("getNextBuffer() mValue[%zu]=%zu not %zu",
+                    mNextIdx-1, provided, buffer->frameCount);
+            if (provided < buffer->frameCount) {
+                buffer->frameCount = provided;
+            }
+            if (mNextIdx >= mInputIncr.size()) {
+                mNextIdx = 0;
+            }
+        }
+        ALOGV("getNextBuffer() requested %zu frames out of %zu frames available"
+                " and returned %zu frames",
+                requestedFrames, mNumFrames - mNextFrame, buffer->frameCount);
+        mUnrel = buffer->frameCount;
+        if (buffer->frameCount > 0) {
+            buffer->raw = (char *)mAddr + mFrameSize * mNextFrame;
+            return android::NO_ERROR;
+        } else {
+            buffer->raw = NULL;
+            return android::NOT_ENOUGH_DATA;
+        }
+    }
+
+    virtual void releaseBuffer(Buffer* buffer)
+    {
+        if (buffer->frameCount > mUnrel) {
+            ALOGE("releaseBuffer() released %zu frames but only %zu available "
+                    "to release", buffer->frameCount, mUnrel);
+            mNextFrame += mUnrel;
+            mUnrel = 0;
+        } else {
+
+            ALOGV("releaseBuffer() released %zu frames out of %zu frames available "
+                    "to release", buffer->frameCount, mUnrel);
+            mNextFrame += buffer->frameCount;
+            mUnrel -= buffer->frameCount;
+        }
+        buffer->frameCount = 0;
+        buffer->raw = NULL;
+    }
+
+    void reset()
+    {
+        mNextFrame = 0;
+    }
+
+    size_t getNumFrames()
+    {
+        return mNumFrames;
+    }
+
+
+protected:
+    void* mAddr;   // base address
+    size_t mNumFrames;   // total frames
+    int mFrameSize;      // frame size (# channels * bytes per sample)
+    size_t mNextFrame;   // index of next frame to provide
+    size_t mUnrel;       // number of frames not yet released
+    std::vector<int> mInputIncr; // number of frames provided per call
+    size_t mNextIdx;     // index of next entry in mInputIncr to use
+};
+
+/* Creates a buffer filled with a sine wave.
+ */
+template<typename T>
+static void createSine(void *vbuffer, size_t frames,
+        size_t channels, double sampleRate, double freq)
+{
+    double tscale = 1. / sampleRate;
+    T* buffer = reinterpret_cast<T*>(vbuffer);
+    for (size_t i = 0; i < frames; ++i) {
+        double t = i * tscale;
+        double y = sin(2. * M_PI * freq * t);
+        T yt = convertValue<T>(y);
+
+        for (size_t j = 0; j < channels; ++j) {
+            buffer[i*channels + j] = yt / T(j + 1);
+        }
+    }
+}
+
+/* Creates a buffer filled with a chirp signal (a sine wave sweep).
+ *
+ * When creating the Chirp, note that the frequency is the true sinusoidal
+ * frequency not the sampling rate.
+ *
+ * http://en.wikipedia.org/wiki/Chirp
+ */
+template<typename T>
+static void createChirp(void *vbuffer, size_t frames,
+        size_t channels, double sampleRate,  double minfreq, double maxfreq)
+{
+    double tscale = 1. / sampleRate;
+    T *buffer = reinterpret_cast<T*>(vbuffer);
+    // note the chirp constant k has a divide-by-two.
+    double k = (maxfreq - minfreq) / (2. * tscale * frames);
+    for (size_t i = 0; i < frames; ++i) {
+        double t = i * tscale;
+        double y = sin(2. * M_PI * (k * t + minfreq) * t);
+        T yt = convertValue<T>(y);
+
+        for (size_t j = 0; j < channels; ++j) {
+            buffer[i*channels + j] = yt / T(j + 1);
+        }
+    }
+}
+
+/* This derived class creates a buffer provider of datatype T,
+ * consisting of an input signal, e.g. from createChirp().
+ * The number of frames can be obtained from the base class
+ * TestProvider::getNumFrames().
+ */
+
+class SignalProvider : public TestProvider {
+public:
+    SignalProvider()
+    : mSampleRate(0),
+      mChannels(0)
+    {
+    }
+
+    virtual ~SignalProvider()
+    {
+        free(mAddr);
+        mAddr = NULL;
+    }
+
+    template <typename T>
+    void setChirp(size_t channels, double minfreq, double maxfreq, double sampleRate, double time)
+    {
+        createBufferByFrames<T>(channels, sampleRate, sampleRate*time);
+        createChirp<T>(mAddr, mNumFrames, mChannels, mSampleRate, minfreq, maxfreq);
+    }
+
+    template <typename T>
+    void setSine(size_t channels,
+            double freq, double sampleRate, double time)
+    {
+        createBufferByFrames<T>(channels, sampleRate, sampleRate*time);
+        createSine<T>(mAddr, mNumFrames,  mChannels, mSampleRate, freq);
+    }
+
+    template <typename T>
+    void setFile(const char *file_in)
+    {
+        SF_INFO info;
+        info.format = 0;
+        SNDFILE *sf = sf_open(file_in, SFM_READ, &info);
+        if (sf == NULL) {
+            perror(file_in);
+            return;
+        }
+        createBufferByFrames<T>(info.channels, info.samplerate, info.frames);
+        if (is_same<T, float>::value) {
+            (void) sf_readf_float(sf, (float *) mAddr, mNumFrames);
+        } else if (is_same<T, short>::value) {
+            (void) sf_readf_short(sf, (short *) mAddr, mNumFrames);
+        }
+        sf_close(sf);
+    }
+
+    template <typename T>
+    void createBufferByFrames(size_t channels, uint32_t sampleRate, size_t frames)
+    {
+        mNumFrames = frames;
+        mChannels = channels;
+        mFrameSize = mChannels * sizeof(T);
+        free(mAddr);
+        mAddr = malloc(mFrameSize * mNumFrames);
+        mSampleRate = sampleRate;
+    }
+
+    uint32_t getSampleRate() const {
+        return mSampleRate;
+    }
+
+    uint32_t getNumChannels() const {
+        return mChannels;
+    }
+
+protected:
+    uint32_t mSampleRate;
+    uint32_t mChannels;
+};
+
+#endif // ANDROID_AUDIO_TEST_UTILS_H

02_day/test-resample/Android.mk

@@ -0,0 +1,27 @@
+LOCAL_PATH:= $(call my-dir)
+
+
+include $(CLEAR_VARS)
+
+LOCAL_SRC_FILES:=               \
+    test-resample.cpp           \
+
+LOCAL_C_INCLUDES := \
+    $(call include-path-for, audio-utils)
+
+LOCAL_STATIC_LIBRARIES := \
+    libsndfile
+
+LOCAL_SHARED_LIBRARIES := \
+    libaudioresampler \
+    libaudioutils \
+    libdl \
+    libcutils \
+    libutils \
+    liblog
+
+LOCAL_MODULE:= test-resample
+
+LOCAL_MODULE_TAGS := optional
+
+include $(BUILD_EXECUTABLE)

02_day/test-resample/AudioResampler.h

@@ -0,0 +1,174 @@
+/*
+ * Copyright (C) 2007 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_AUDIO_RESAMPLER_H
+#define ANDROID_AUDIO_RESAMPLER_H
+
+#include <stdint.h>
+#include <sys/types.h>
+#include <cutils/compiler.h>
+
+#include <media/AudioBufferProvider.h>
+#include <system/audio.h>
+
+namespace android {
+// ----------------------------------------------------------------------------
+
+class ANDROID_API AudioResampler {
+public:
+    // Determines quality of SRC.
+    //  LOW_QUALITY: linear interpolator (1st order)
+    //  MED_QUALITY: cubic interpolator (3rd order)
+    //  HIGH_QUALITY: fixed multi-tap FIR (e.g. 48KHz->44.1KHz)
+    // NOTE: high quality SRC will only be supported for
+    // certain fixed rate conversions. Sample rate cannot be
+    // changed dynamically.
+    enum src_quality {
+        DEFAULT_QUALITY=0,
+        LOW_QUALITY=1,
+        MED_QUALITY=2,
+        HIGH_QUALITY=3,
+        VERY_HIGH_QUALITY=4,
+        DYN_LOW_QUALITY=5,
+        DYN_MED_QUALITY=6,
+        DYN_HIGH_QUALITY=7,
+    };
+
+    static const float UNITY_GAIN_FLOAT = 1.0f;
+
+    static AudioResampler* create(audio_format_t format, int inChannelCount,
+            int32_t sampleRate, src_quality quality=DEFAULT_QUALITY);
+
+    virtual ~AudioResampler();
+
+    virtual void init() = 0;
+    virtual void setSampleRate(int32_t inSampleRate);
+    virtual void setVolume(float left, float right);
+    virtual void setLocalTimeFreq(uint64_t freq);
+
+    // set the PTS of the next buffer output by the resampler
+    virtual void setPTS(int64_t pts);
+
+    // Resample int16_t samples from provider and accumulate into 'out'.
+    // A mono provider delivers a sequence of samples.
+    // A stereo provider delivers a sequence of interleaved pairs of samples.
+    // Multi-channel providers are not supported.
+    // In either case, 'out' holds interleaved pairs of fixed-point Q4.27.
+    // That is, for a mono provider, there is an implicit up-channeling.
+    // Since this method accumulates, the caller is responsible for clearing 'out' initially.
+    // FIXME assumes provider is always successful; it should return the actual frame count.
+    virtual void resample(int32_t* out, size_t outFrameCount,
+            AudioBufferProvider* provider) = 0;
+
+    virtual void reset();
+    virtual size_t getUnreleasedFrames() const { return mInputIndex; }
+
+    // called from destructor, so must not be virtual
+    src_quality getQuality() const { return mQuality; }
+
+protected:
+    // number of bits for phase fraction - 30 bits allows nearly 2x downsampling
+    static const int kNumPhaseBits = 30;
+
+    // phase mask for fraction
+    static const uint32_t kPhaseMask = (1LU<<kNumPhaseBits)-1;
+
+    // multiplier to calculate fixed point phase increment
+    static const double kPhaseMultiplier;
+
+    AudioResampler(int inChannelCount, int32_t sampleRate, src_quality quality);
+
+    // prevent copying
+    AudioResampler(const AudioResampler&);
+    AudioResampler& operator=(const AudioResampler&);
+
+    int64_t calculateOutputPTS(int outputFrameIndex);
+
+    const int32_t mChannelCount;
+    const int32_t mSampleRate;
+    int32_t mInSampleRate;
+    AudioBufferProvider::Buffer mBuffer;
+    union {
+        int16_t mVolume[2];
+        uint32_t mVolumeRL;
+    };
+    int16_t mTargetVolume[2];
+    size_t mInputIndex;
+    int32_t mPhaseIncrement;
+    uint32_t mPhaseFraction;
+    uint64_t mLocalTimeFreq;
+    int64_t mPTS;
+
+    // returns the inFrameCount required to generate outFrameCount frames.
+    //
+    // Placed here to be a consistent for all resamplers.
+    //
+    // Right now, we use the upper bound without regards to the current state of the
+    // input buffer using integer arithmetic, as follows:
+    //
+    // (static_cast<uint64_t>(outFrameCount)*mInSampleRate + (mSampleRate - 1))/mSampleRate;
+    //
+    // The double precision equivalent (float may not be precise enough):
+    // ceil(static_cast<double>(outFrameCount) * mInSampleRate / mSampleRate);
+    //
+    // this relies on the fact that the mPhaseIncrement is rounded down from
+    // #phases * mInSampleRate/mSampleRate and the fact that Sum(Floor(x)) <= Floor(Sum(x)).
+    // http://www.proofwiki.org/wiki/Sum_of_Floors_Not_Greater_Than_Floor_of_Sums
+    //
+    // (so long as double precision is computed accurately enough to be considered
+    // greater than or equal to the Floor(x) value in int32_t arithmetic; thus this
+    // will not necessarily hold for floats).
+    //
+    // TODO:
+    // Greater accuracy and a tight bound is obtained by:
+    // 1) subtract and adjust for the current state of the AudioBufferProvider buffer.
+    // 2) using the exact integer formula where (ignoring 64b casting)
+    //  inFrameCount = (mPhaseIncrement * (outFrameCount - 1) + mPhaseFraction) / phaseWrapLimit;
+    //  phaseWrapLimit is the wraparound (1 << kNumPhaseBits), if not specified explicitly.
+    //
+    inline size_t getInFrameCountRequired(size_t outFrameCount) {
+        return (static_cast<uint64_t>(outFrameCount)*mInSampleRate
+                + (mSampleRate - 1))/mSampleRate;
+    }
+
+    inline float clampFloatVol(float volume) {
+        if (volume > UNITY_GAIN_FLOAT) {
+            return UNITY_GAIN_FLOAT;
+        } else if (volume >= 0.) {
+            return volume;
+        }
+        return 0.;  // NaN or negative volume maps to 0.
+    }
+
+private:
+    const src_quality mQuality;
+
+    // Return 'true' if the quality level is supported without explicit request
+    static bool qualityIsSupported(src_quality quality);
+
+    // For pthread_once()
+    static void init_routine();
+
+    // Return the estimated CPU load for specific resampler in MHz.
+    // The absolute number is irrelevant, it's the relative values that matter.
+    static uint32_t qualityMHz(src_quality quality);
+};
+
+// ----------------------------------------------------------------------------
+}
+; // namespace android
+
+#endif // ANDROID_AUDIO_RESAMPLER_H

02_day/test-resample/test-resample.cpp

@@ -0,0 +1,509 @@
+/*
+ * Copyright (C) 2012 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.
+ */
+
+#include <unistd.h>
+#include <stdio.h>
+#include <stdlib.h>
+#include <fcntl.h>
+#include <string.h>
+#include <sys/mman.h>
+#include <sys/stat.h>
+#include <errno.h>
+#include <inttypes.h>
+#include <time.h>
+#include <math.h>
+#include <audio_utils/primitives.h>
+#include <audio_utils/sndfile.h>
+#include <utils/Vector.h>
+#include <media/AudioBufferProvider.h>
+#include "AudioResampler.h"
+
+using namespace android;
+
+static bool gVerbose = false;
+
+static int usage(const char* name) {
+    fprintf(stderr,"Usage: %s [-p] [-f] [-F] [-v] [-c channels]"
+                   " [-q {dq|lq|mq|hq|vhq|dlq|dmq|dhq}]"
+                   " [-i input-sample-rate] [-o output-sample-rate]"
+                   " [-O csv] [-P csv] [<input-file>]"
+                   " <output-file>\n", name);
+    fprintf(stderr,"    -p    enable profiling\n");
+    fprintf(stderr,"    -f    enable filter profiling\n");
+    fprintf(stderr,"    -F    enable floating point -q {dlq|dmq|dhq} only");
+    fprintf(stderr,"    -v    verbose : log buffer provider calls\n");
+    fprintf(stderr,"    -c    # channels (1-2 for lq|mq|hq; 1-8 for dlq|dmq|dhq)\n");
+    fprintf(stderr,"    -q    resampler quality\n");
+    fprintf(stderr,"              dq  : default quality\n");
+    fprintf(stderr,"              lq  : low quality\n");
+    fprintf(stderr,"              mq  : medium quality\n");
+    fprintf(stderr,"              hq  : high quality\n");
+    fprintf(stderr,"              vhq : very high quality\n");
+    fprintf(stderr,"              dlq : dynamic low quality\n");
+    fprintf(stderr,"              dmq : dynamic medium quality\n");
+    fprintf(stderr,"              dhq : dynamic high quality\n");
+    fprintf(stderr,"    -i    input file sample rate (ignored if input file is specified)\n");
+    fprintf(stderr,"    -o    output file sample rate\n");
+    fprintf(stderr,"    -O    # frames output per call to resample() in CSV format\n");
+    fprintf(stderr,"    -P    # frames provided per call to resample() in CSV format\n");
+    return -1;
+}
+
+// Convert a list of integers in CSV format to a Vector of those values.
+// Returns the number of elements in the list, or -1 on error.
+int parseCSV(const char *string, Vector<int>& values)
+{
+    // pass 1: count the number of values and do syntax check
+    size_t numValues = 0;
+    bool hadDigit = false;
+    for (const char *p = string; ; ) {
+        switch (*p++) {
+        case '0': case '1': case '2': case '3': case '4':
+        case '5': case '6': case '7': case '8': case '9':
+            hadDigit = true;
+            break;
+        case '\0':
+            if (hadDigit) {
+                // pass 2: allocate and initialize vector of values
+                values.resize(++numValues);
+                values.editItemAt(0) = atoi(p = optarg);
+                for (size_t i = 1; i < numValues; ) {
+                    if (*p++ == ',') {
+                        values.editItemAt(i++) = atoi(p);
+                    }
+                }
+                return numValues;
+            }
+            // fall through
+        case ',':
+            if (hadDigit) {
+                hadDigit = false;
+                numValues++;
+                break;
+            }
+            // fall through
+        default:
+            return -1;
+        }
+    }
+}
+
+int main(int argc, char* argv[]) {
+    const char* const progname = argv[0];
+    bool profileResample = false;
+    bool profileFilter = false;
+    bool useFloat = false;
+    int channels = 1;
+    int input_freq = 0;
+    int output_freq = 0;
+    AudioResampler::src_quality quality = AudioResampler::DEFAULT_QUALITY;
+    Vector<int> Ovalues;
+    Vector<int> Pvalues;
+
+    int ch;
+    while ((ch = getopt(argc, argv, "pfFvc:q:i:o:O:P:")) != -1) {
+        switch (ch) {
+        case 'p':
+            profileResample = true;
+            break;
+        case 'f':
+            profileFilter = true;
+            break;
+        case 'F':
+            useFloat = true;
+            break;
+        case 'v':
+            gVerbose = true;
+            break;
+        case 'c':
+            channels = atoi(optarg);
+            break;
+        case 'q':
+            if (!strcmp(optarg, "dq"))
+                quality = AudioResampler::DEFAULT_QUALITY;
+            else if (!strcmp(optarg, "lq"))
+                quality = AudioResampler::LOW_QUALITY;
+            else if (!strcmp(optarg, "mq"))
+                quality = AudioResampler::MED_QUALITY;
+            else if (!strcmp(optarg, "hq"))
+                quality = AudioResampler::HIGH_QUALITY;
+            else if (!strcmp(optarg, "vhq"))
+                quality = AudioResampler::VERY_HIGH_QUALITY;
+            else if (!strcmp(optarg, "dlq"))
+                quality = AudioResampler::DYN_LOW_QUALITY;
+            else if (!strcmp(optarg, "dmq"))
+                quality = AudioResampler::DYN_MED_QUALITY;
+            else if (!strcmp(optarg, "dhq"))
+                quality = AudioResampler::DYN_HIGH_QUALITY;
+            else {
+                usage(progname);
+                return -1;
+            }
+            break;
+        case 'i':
+            input_freq = atoi(optarg);
+            break;
+        case 'o':
+            output_freq = atoi(optarg);
+            break;
+        case 'O':
+            if (parseCSV(optarg, Ovalues) < 0) {
+                fprintf(stderr, "incorrect syntax for -O option\n");
+                return -1;
+            }
+            break;
+        case 'P':
+            if (parseCSV(optarg, Pvalues) < 0) {
+                fprintf(stderr, "incorrect syntax for -P option\n");
+                return -1;
+            }
+            break;
+        case '?':
+        default:
+            usage(progname);
+            return -1;
+        }
+    }
+
+    if (channels < 1
+            || channels > (quality < AudioResampler::DYN_LOW_QUALITY ? 2 : 8)) {
+        fprintf(stderr, "invalid number of audio channels %d\n", channels);
+        return -1;
+    }
+    if (useFloat && quality < AudioResampler::DYN_LOW_QUALITY) {
+        fprintf(stderr, "float processing is only possible for dynamic resamplers\n");
+        return -1;
+    }
+
+    argc -= optind;
+    argv += optind;
+
+    const char* file_in = NULL;
+    const char* file_out = NULL;
+    if (argc == 1) {
+        file_out = argv[0];
+    } else if (argc == 2) {
+        file_in = argv[0];
+        file_out = argv[1];
+    } else {
+        usage(progname);
+        return -1;
+    }
+
+    // ----------------------------------------------------------
+
+    size_t input_size;
+    void* input_vaddr;
+    if (argc == 2) {
+        SF_INFO info;
+        info.format = 0;
+        SNDFILE *sf = sf_open(file_in, SFM_READ, &info);
+        if (sf == NULL) {
+            perror(file_in);
+            return EXIT_FAILURE;
+        }
+        input_size = info.frames * info.channels * sizeof(short);
+        input_vaddr = malloc(input_size);
+        (void) sf_readf_short(sf, (short *) input_vaddr, info.frames);
+        sf_close(sf);
+        channels = info.channels;
+        input_freq = info.samplerate;
+    } else {
+        // data for testing is exactly (input sampling rate/1000)/2 seconds
+        // so 44.1khz input is 22.05 seconds
+        double k = 1000; // Hz / s
+        double time = (input_freq / 2) / k;
+        size_t input_frames = size_t(input_freq * time);
+        input_size = channels * sizeof(int16_t) * input_frames;
+        input_vaddr = malloc(input_size);
+        int16_t* in = (int16_t*)input_vaddr;
+        for (size_t i=0 ; i<input_frames ; i++) {
+            double t = double(i) / input_freq;
+            double y = sin(M_PI * k * t * t);
+            int16_t yi = floor(y * 32767.0 + 0.5);
+            for (int j = 0; j < channels; j++) {
+                in[i*channels + j] = yi / (1 + j);
+            }
+        }
+    }
+    size_t input_framesize = channels * sizeof(int16_t);
+    size_t input_frames = input_size / input_framesize;
+
+    // For float processing, convert input int16_t to float array
+    if (useFloat) {
+        void *new_vaddr;
+
+        input_framesize = channels * sizeof(float);
+        input_size = input_frames * input_framesize;
+        new_vaddr = malloc(input_size);
+        memcpy_to_float_from_i16(reinterpret_cast<float*>(new_vaddr),
+                reinterpret_cast<int16_t*>(input_vaddr), input_frames * channels);
+        free(input_vaddr);
+        input_vaddr = new_vaddr;
+    }
+
+    // ----------------------------------------------------------
+
+    class Provider: public AudioBufferProvider {
+        const void*     mAddr;      // base address
+        const size_t    mNumFrames; // total frames
+        const size_t    mFrameSize; // size of each frame in bytes
+        size_t          mNextFrame; // index of next frame to provide
+        size_t          mUnrel;     // number of frames not yet released
+        const Vector<int> mPvalues; // number of frames provided per call
+        size_t          mNextPidx;  // index of next entry in mPvalues to use
+    public:
+        Provider(const void* addr, size_t frames, size_t frameSize, const Vector<int>& Pvalues)
+          : mAddr(addr),
+            mNumFrames(frames),
+            mFrameSize(frameSize),
+            mNextFrame(0), mUnrel(0), mPvalues(Pvalues), mNextPidx(0) {
+        }
+        virtual status_t getNextBuffer(Buffer* buffer,
+                int64_t pts = kInvalidPTS) {
+            (void)pts; // suppress warning
+            size_t requestedFrames = buffer->frameCount;
+            if (requestedFrames > mNumFrames - mNextFrame) {
+                buffer->frameCount = mNumFrames - mNextFrame;
+            }
+            if (!mPvalues.isEmpty()) {
+                size_t provided = mPvalues[mNextPidx++];
+                printf("mPvalue[%zu]=%zu not %zu\n", mNextPidx-1, provided, buffer->frameCount);
+                if (provided < buffer->frameCount) {
+                    buffer->frameCount = provided;
+                }
+                if (mNextPidx >= mPvalues.size()) {
+                    mNextPidx = 0;
+                }
+            }
+            if (gVerbose) {
+                printf("getNextBuffer() requested %zu frames out of %zu frames available,"
+                        " and returned %zu frames\n",
+                        requestedFrames, (size_t) (mNumFrames - mNextFrame), buffer->frameCount);
+            }
+            mUnrel = buffer->frameCount;
+            if (buffer->frameCount > 0) {
+                buffer->raw = (char *)mAddr + mFrameSize * mNextFrame;
+                return NO_ERROR;
+            } else {
+                buffer->raw = NULL;
+                return NOT_ENOUGH_DATA;
+            }
+        }
+        virtual void releaseBuffer(Buffer* buffer) {
+            if (buffer->frameCount > mUnrel) {
+                fprintf(stderr, "ERROR releaseBuffer() released %zu frames but only %zu available "
+                        "to release\n", buffer->frameCount, mUnrel);
+                mNextFrame += mUnrel;
+                mUnrel = 0;
+            } else {
+                if (gVerbose) {
+                    printf("releaseBuffer() released %zu frames out of %zu frames available "
+                            "to release\n", buffer->frameCount, mUnrel);
+                }
+                mNextFrame += buffer->frameCount;
+                mUnrel -= buffer->frameCount;
+            }
+            buffer->frameCount = 0;
+            buffer->raw = NULL;
+        }
+        void reset() {
+            mNextFrame = 0;
+        }
+    } provider(input_vaddr, input_frames, input_framesize, Pvalues);
+
+    if (gVerbose) {
+        printf("%zu input frames\n", input_frames);
+    }
+
+    audio_format_t format = useFloat ? AUDIO_FORMAT_PCM_FLOAT : AUDIO_FORMAT_PCM_16_BIT;
+    int output_channels = channels > 2 ? channels : 2; // output is at least stereo samples
+    size_t output_framesize = output_channels * (useFloat ? sizeof(float) : sizeof(int32_t));
+    size_t output_frames = ((int64_t) input_frames * output_freq) / input_freq;
+    size_t output_size = output_frames * output_framesize;
+
+    if (profileFilter) {
+        // Check how fast sample rate changes are that require filter changes.
+        // The delta sample rate changes must indicate a downsampling ratio,
+        // and must be larger than 10% changes.
+        //
+        // On fast devices, filters should be generated between 0.1ms - 1ms.
+        // (single threaded).
+        AudioResampler* resampler = AudioResampler::create(format, channels,
+                8000, quality);
+        int looplimit = 100;
+        timespec start, end;
+        clock_gettime(CLOCK_MONOTONIC, &start);
+        for (int i = 0; i < looplimit; ++i) {
+            resampler->setSampleRate(9000);
+            resampler->setSampleRate(12000);
+            resampler->setSampleRate(20000);
+            resampler->setSampleRate(30000);
+        }
+        clock_gettime(CLOCK_MONOTONIC, &end);
+        int64_t start_ns = start.tv_sec * 1000000000LL + start.tv_nsec;
+        int64_t end_ns = end.tv_sec * 1000000000LL + end.tv_nsec;
+        int64_t time = end_ns - start_ns;
+        printf("%.2f sample rate changes with filter calculation/sec\n",
+                looplimit * 4 / (time / 1e9));
+
+        // Check how fast sample rate changes are without filter changes.
+        // This should be very fast, probably 0.1us - 1us per sample rate
+        // change.
+        resampler->setSampleRate(1000);
+        looplimit = 1000;
+        clock_gettime(CLOCK_MONOTONIC, &start);
+        for (int i = 0; i < looplimit; ++i) {
+            resampler->setSampleRate(1000+i);
+        }
+        clock_gettime(CLOCK_MONOTONIC, &end);
+        start_ns = start.tv_sec * 1000000000LL + start.tv_nsec;
+        end_ns = end.tv_sec * 1000000000LL + end.tv_nsec;
+        time = end_ns - start_ns;
+        printf("%.2f sample rate changes without filter calculation/sec\n",
+                looplimit / (time / 1e9));
+        resampler->reset();
+        delete resampler;
+    }
+
+    void* output_vaddr = malloc(output_size);
+    AudioResampler* resampler = AudioResampler::create(format, channels,
+            output_freq, quality);
+
+    resampler->setSampleRate(input_freq);
+    resampler->setVolume(AudioResampler::UNITY_GAIN_FLOAT, AudioResampler::UNITY_GAIN_FLOAT);
+
+    if (profileResample) {
+        /*
+         * For profiling on mobile devices, upon experimentation
+         * it is better to run a few trials with a shorter loop limit,
+         * and take the minimum time.
+         *
+         * Long tests can cause CPU temperature to build up and thermal throttling
+         * to reduce CPU frequency.
+         *
+         * For frequency checks (index=0, or 1, etc.):
+         * "cat /sys/devices/system/cpu/cpu${index}/cpufreq/scaling_*_freq"
+         *
+         * For temperature checks (index=0, or 1, etc.):
+         * "cat /sys/class/thermal/thermal_zone${index}/temp"
+         *
+         * Another way to avoid thermal throttling is to fix the CPU frequency
+         * at a lower level which prevents excessive temperatures.
+         */
+        const int trials = 4;
+        const int looplimit = 4;
+        timespec start, end;
+        int64_t time = 0;
+
+        for (int n = 0; n < trials; ++n) {
+            clock_gettime(CLOCK_MONOTONIC, &start);
+            for (int i = 0; i < looplimit; ++i) {
+                resampler->resample((int*) output_vaddr, output_frames, &provider);
+                provider.reset(); //  during benchmarking reset only the provider
+            }
+            clock_gettime(CLOCK_MONOTONIC, &end);
+            int64_t start_ns = start.tv_sec * 1000000000LL + start.tv_nsec;
+            int64_t end_ns = end.tv_sec * 1000000000LL + end.tv_nsec;
+            int64_t diff_ns = end_ns - start_ns;
+            if (n == 0 || diff_ns < time) {
+                time = diff_ns;   // save the best out of our trials.
+            }
+        }
+        // Mfrms/s is "Millions of output frames per second".
+        printf("quality: %d  channels: %d  msec: %" PRId64 "  Mfrms/s: %.2lf\n",
+                quality, channels, time/1000000, output_frames * looplimit / (time / 1e9) / 1e6);
+        resampler->reset();
+    }
+
+    memset(output_vaddr, 0, output_size);
+    if (gVerbose) {
+        printf("resample() %zu output frames\n", output_frames);
+    }
+    if (Ovalues.isEmpty()) {
+        Ovalues.push(output_frames);
+    }
+    for (size_t i = 0, j = 0; i < output_frames; ) {
+        size_t thisFrames = Ovalues[j++];
+        if (j >= Ovalues.size()) {
+            j = 0;
+        }
+        if (thisFrames == 0 || thisFrames > output_frames - i) {
+            thisFrames = output_frames - i;
+        }
+        resampler->resample((int*) output_vaddr + output_channels*i, thisFrames, &provider);
+        i += thisFrames;
+    }
+    if (gVerbose) {
+        printf("resample() complete\n");
+    }
+    resampler->reset();
+    if (gVerbose) {
+        printf("reset() complete\n");
+    }
+    delete resampler;
+    resampler = NULL;
+
+    // For float processing, convert output format from float to Q4.27,
+    // which is then converted to int16_t for final storage.
+    if (useFloat) {
+        memcpy_to_q4_27_from_float(reinterpret_cast<int32_t*>(output_vaddr),
+                reinterpret_cast<float*>(output_vaddr), output_frames * output_channels);
+    }
+
+    // mono takes left channel only (out of stereo output pair)
+    // stereo and multichannel preserve all channels.
+    int32_t* out = (int32_t*) output_vaddr;
+    int16_t* convert = (int16_t*) malloc(output_frames * channels * sizeof(int16_t));
+
+    const int volumeShift = 12; // shift requirement for Q4.27 to Q.15
+    // round to half towards zero and saturate at int16 (non-dithered)
+    const int roundVal = (1<<(volumeShift-1)) - 1; // volumePrecision > 0
+
+    for (size_t i = 0; i < output_frames; i++) {
+        for (int j = 0; j < channels; j++) {
+            int32_t s = out[i * output_channels + j] + roundVal; // add offset here
+            if (s < 0) {
+                s = (s + 1) >> volumeShift; // round to 0
+                if (s < -32768) {
+                    s = -32768;
+                }
+            } else {
+                s = s >> volumeShift;
+                if (s > 32767) {
+                    s = 32767;
+                }
+            }
+            convert[i * channels + j] = int16_t(s);
+        }
+    }
+
+    // write output to disk
+    SF_INFO info;
+    info.frames = 0;
+    info.samplerate = output_freq;
+    info.channels = channels;
+    info.format = SF_FORMAT_WAV | SF_FORMAT_PCM_16;
+    SNDFILE *sf = sf_open(file_out, SFM_WRITE, &info);
+    if (sf == NULL) {
+        perror(file_out);
+        return EXIT_FAILURE;
+    }
+    (void) sf_writef_short(sf, convert, output_frames);
+    sf_close(sf);
+
+    return EXIT_SUCCESS;
+}

02_day/test-thread/Android.mk

@@ -0,0 +1,8 @@
+LOCAL_PATH:= $(call my-dir)
+include $(CLEAR_VARS)
+LOCAL_SRC_FILES:= thread.cpp
+LOCAL_MODULE := my_thread
+LOCAL_SHARED_LIBRARIES:= libcutils libutils libbinder
+LOCAL_MODULE_TAGS := optional
+include $(BUILD_EXECUTABLE)
+

02_day/test-thread/thread.cpp

@@ -0,0 +1,27 @@
+#include <binder/IServiceManager.h>
+#include <utils/StrongPointer.h>
+#include <utils/Thread.h>
+#include <binder/MemoryHeapBase.h>
+#include <binder/IPCThreadState.h>
+
+using namespace android;
+
+
+class MyThread : public Thread 
+{
+	public : 
+		bool threadLoop()
+		{
+			printf("MyThread::threadLoop()\n"); 
+			sleep(1);
+			return true;
+		}
+};
+
+int main()
+{
+	sp<Thread> thread = new MyThread;
+	thread->run();
+	getchar();
+	return 0;
+}