diff --git a/media/libaudioprocessing/AudioMixer.cpp b/media/libaudioprocessing/AudioMixer.cpp
new file mode 100644
index 0000000..a7d9f0f
--- /dev/null
+++ b/media/libaudioprocessing/AudioMixer.cpp
@@ -0,0 +1,2085 @@
+/*
+**
+** 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.
+*/
+
+#define LOG_TAG "AudioMixer"
+//#define LOG_NDEBUG 0
+
+#include <stdint.h>
+#include <string.h>
+#include <stdlib.h>
+#include <math.h>
+#include <sys/types.h>
+
+#include <utils/Errors.h>
+#include <utils/Log.h>
+
+#include <cutils/bitops.h>
+#include <cutils/compiler.h>
+#include <utils/Debug.h>
+
+#include <system/audio.h>
+
+#include <audio_utils/primitives.h>
+#include <audio_utils/format.h>
+#include <media/AudioMixer.h>
+
+#include "AudioMixerOps.h"
+
+// The FCC_2 macro refers to the Fixed Channel Count of 2 for the legacy integer mixer.
+#ifndef FCC_2
+#define FCC_2 2
+#endif
+
+// Look for MONO_HACK for any Mono hack involving legacy mono channel to
+// stereo channel conversion.
+
+/* VERY_VERY_VERBOSE_LOGGING will show exactly which process hook and track hook is
+ * being used. This is a considerable amount of log spam, so don't enable unless you
+ * are verifying the hook based code.
+ */
+//#define VERY_VERY_VERBOSE_LOGGING
+#ifdef VERY_VERY_VERBOSE_LOGGING
+#define ALOGVV ALOGV
+//define ALOGVV printf  // for test-mixer.cpp
+#else
+#define ALOGVV(a...) do { } while (0)
+#endif
+
+#ifndef ARRAY_SIZE
+#define ARRAY_SIZE(x) (sizeof(x)/sizeof((x)[0]))
+#endif
+
+// TODO: Move these macro/inlines to a header file.
+template <typename T>
+static inline
+T max(const T& x, const T& y) {
+    return x > y ? x : y;
+}
+
+// Set kUseNewMixer to true to use the new mixer engine always. Otherwise the
+// original code will be used for stereo sinks, the new mixer for multichannel.
+static const bool kUseNewMixer = true;
+
+// Set kUseFloat to true to allow floating input into the mixer engine.
+// If kUseNewMixer is false, this is ignored or may be overridden internally
+// because of downmix/upmix support.
+static const bool kUseFloat = true;
+
+// Set to default copy buffer size in frames for input processing.
+static const size_t kCopyBufferFrameCount = 256;
+
+namespace android {
+
+// ----------------------------------------------------------------------------
+
+template <typename T>
+T min(const T& a, const T& b)
+{
+    return a < b ? a : b;
+}
+
+// ----------------------------------------------------------------------------
+
+// Ensure mConfiguredNames bitmask is initialized properly on all architectures.
+// The value of 1 << x is undefined in C when x >= 32.
+
+AudioMixer::AudioMixer(size_t frameCount, uint32_t sampleRate, uint32_t maxNumTracks)
+    :   mTrackNames(0), mConfiguredNames((maxNumTracks >= 32 ? 0 : 1 << maxNumTracks) - 1),
+        mSampleRate(sampleRate)
+{
+    ALOG_ASSERT(maxNumTracks <= MAX_NUM_TRACKS, "maxNumTracks %u > MAX_NUM_TRACKS %u",
+            maxNumTracks, MAX_NUM_TRACKS);
+
+    // AudioMixer is not yet capable of more than 32 active track inputs
+    ALOG_ASSERT(32 >= MAX_NUM_TRACKS, "bad MAX_NUM_TRACKS %d", MAX_NUM_TRACKS);
+
+    pthread_once(&sOnceControl, &sInitRoutine);
+
+    mState.enabledTracks= 0;
+    mState.needsChanged = 0;
+    mState.frameCount   = frameCount;
+    mState.hook         = process__nop;
+    mState.outputTemp   = NULL;
+    mState.resampleTemp = NULL;
+    mState.mLog         = &mDummyLog;
+    // mState.reserved
+
+    // FIXME Most of the following initialization is probably redundant since
+    // tracks[i] should only be referenced if (mTrackNames & (1 << i)) != 0
+    // and mTrackNames is initially 0.  However, leave it here until that's verified.
+    track_t* t = mState.tracks;
+    for (unsigned i=0 ; i < MAX_NUM_TRACKS ; i++) {
+        t->resampler = NULL;
+        t->downmixerBufferProvider = NULL;
+        t->mReformatBufferProvider = NULL;
+        t->mTimestretchBufferProvider = NULL;
+        t++;
+    }
+
+}
+
+AudioMixer::~AudioMixer()
+{
+    track_t* t = mState.tracks;
+    for (unsigned i=0 ; i < MAX_NUM_TRACKS ; i++) {
+        delete t->resampler;
+        delete t->downmixerBufferProvider;
+        delete t->mReformatBufferProvider;
+        delete t->mTimestretchBufferProvider;
+        t++;
+    }
+    delete [] mState.outputTemp;
+    delete [] mState.resampleTemp;
+}
+
+void AudioMixer::setLog(NBLog::Writer *log)
+{
+    mState.mLog = log;
+}
+
+static inline audio_format_t selectMixerInFormat(audio_format_t inputFormat __unused) {
+    return kUseFloat && kUseNewMixer ? AUDIO_FORMAT_PCM_FLOAT : AUDIO_FORMAT_PCM_16_BIT;
+}
+
+int AudioMixer::getTrackName(audio_channel_mask_t channelMask,
+        audio_format_t format, int sessionId)
+{
+    if (!isValidPcmTrackFormat(format)) {
+        ALOGE("AudioMixer::getTrackName invalid format (%#x)", format);
+        return -1;
+    }
+    uint32_t names = (~mTrackNames) & mConfiguredNames;
+    if (names != 0) {
+        int n = __builtin_ctz(names);
+        ALOGV("add track (%d)", n);
+        // assume default parameters for the track, except where noted below
+        track_t* t = &mState.tracks[n];
+        t->needs = 0;
+
+        // Integer volume.
+        // Currently integer volume is kept for the legacy integer mixer.
+        // Will be removed when the legacy mixer path is removed.
+        t->volume[0] = UNITY_GAIN_INT;
+        t->volume[1] = UNITY_GAIN_INT;
+        t->prevVolume[0] = UNITY_GAIN_INT << 16;
+        t->prevVolume[1] = UNITY_GAIN_INT << 16;
+        t->volumeInc[0] = 0;
+        t->volumeInc[1] = 0;
+        t->auxLevel = 0;
+        t->auxInc = 0;
+        t->prevAuxLevel = 0;
+
+        // Floating point volume.
+        t->mVolume[0] = UNITY_GAIN_FLOAT;
+        t->mVolume[1] = UNITY_GAIN_FLOAT;
+        t->mPrevVolume[0] = UNITY_GAIN_FLOAT;
+        t->mPrevVolume[1] = UNITY_GAIN_FLOAT;
+        t->mVolumeInc[0] = 0.;
+        t->mVolumeInc[1] = 0.;
+        t->mAuxLevel = 0.;
+        t->mAuxInc = 0.;
+        t->mPrevAuxLevel = 0.;
+
+        // no initialization needed
+        // t->frameCount
+        t->channelCount = audio_channel_count_from_out_mask(channelMask);
+        t->enabled = false;
+        ALOGV_IF(audio_channel_mask_get_bits(channelMask) != AUDIO_CHANNEL_OUT_STEREO,
+                "Non-stereo channel mask: %d\n", channelMask);
+        t->channelMask = channelMask;
+        t->sessionId = sessionId;
+        // setBufferProvider(name, AudioBufferProvider *) is required before enable(name)
+        t->bufferProvider = NULL;
+        t->buffer.raw = NULL;
+        // no initialization needed
+        // t->buffer.frameCount
+        t->hook = NULL;
+        t->in = NULL;
+        t->resampler = NULL;
+        t->sampleRate = mSampleRate;
+        // setParameter(name, TRACK, MAIN_BUFFER, mixBuffer) is required before enable(name)
+        t->mainBuffer = NULL;
+        t->auxBuffer = NULL;
+        t->mInputBufferProvider = NULL;
+        t->mReformatBufferProvider = NULL;
+        t->downmixerBufferProvider = NULL;
+        t->mPostDownmixReformatBufferProvider = NULL;
+        t->mTimestretchBufferProvider = NULL;
+        t->mMixerFormat = AUDIO_FORMAT_PCM_16_BIT;
+        t->mFormat = format;
+        t->mMixerInFormat = selectMixerInFormat(format);
+        t->mDownmixRequiresFormat = AUDIO_FORMAT_INVALID; // no format required
+        t->mMixerChannelMask = audio_channel_mask_from_representation_and_bits(
+                AUDIO_CHANNEL_REPRESENTATION_POSITION, AUDIO_CHANNEL_OUT_STEREO);
+        t->mMixerChannelCount = audio_channel_count_from_out_mask(t->mMixerChannelMask);
+        t->mPlaybackRate = AUDIO_PLAYBACK_RATE_DEFAULT;
+        // Check the downmixing (or upmixing) requirements.
+        status_t status = t->prepareForDownmix();
+        if (status != OK) {
+            ALOGE("AudioMixer::getTrackName invalid channelMask (%#x)", channelMask);
+            return -1;
+        }
+        // prepareForDownmix() may change mDownmixRequiresFormat
+        ALOGVV("mMixerFormat:%#x  mMixerInFormat:%#x\n", t->mMixerFormat, t->mMixerInFormat);
+        t->prepareForReformat();
+        mTrackNames |= 1 << n;
+        return TRACK0 + n;
+    }
+    ALOGE("AudioMixer::getTrackName out of available tracks");
+    return -1;
+}
+
+void AudioMixer::invalidateState(uint32_t mask)
+{
+    if (mask != 0) {
+        mState.needsChanged |= mask;
+        mState.hook = process__validate;
+    }
+ }
+
+// Called when channel masks have changed for a track name
+// TODO: Fix DownmixerBufferProvider not to (possibly) change mixer input format,
+// which will simplify this logic.
+bool AudioMixer::setChannelMasks(int name,
+        audio_channel_mask_t trackChannelMask, audio_channel_mask_t mixerChannelMask) {
+    track_t &track = mState.tracks[name];
+
+    if (trackChannelMask == track.channelMask
+            && mixerChannelMask == track.mMixerChannelMask) {
+        return false;  // no need to change
+    }
+    // always recompute for both channel masks even if only one has changed.
+    const uint32_t trackChannelCount = audio_channel_count_from_out_mask(trackChannelMask);
+    const uint32_t mixerChannelCount = audio_channel_count_from_out_mask(mixerChannelMask);
+    const bool mixerChannelCountChanged = track.mMixerChannelCount != mixerChannelCount;
+
+    ALOG_ASSERT((trackChannelCount <= MAX_NUM_CHANNELS_TO_DOWNMIX)
+            && trackChannelCount
+            && mixerChannelCount);
+    track.channelMask = trackChannelMask;
+    track.channelCount = trackChannelCount;
+    track.mMixerChannelMask = mixerChannelMask;
+    track.mMixerChannelCount = mixerChannelCount;
+
+    // channel masks have changed, does this track need a downmixer?
+    // update to try using our desired format (if we aren't already using it)
+    const audio_format_t prevDownmixerFormat = track.mDownmixRequiresFormat;
+    const status_t status = mState.tracks[name].prepareForDownmix();
+    ALOGE_IF(status != OK,
+            "prepareForDownmix error %d, track channel mask %#x, mixer channel mask %#x",
+            status, track.channelMask, track.mMixerChannelMask);
+
+    if (prevDownmixerFormat != track.mDownmixRequiresFormat) {
+        track.prepareForReformat(); // because of downmixer, track format may change!
+    }
+
+    if (track.resampler && mixerChannelCountChanged) {
+        // resampler channels may have changed.
+        const uint32_t resetToSampleRate = track.sampleRate;
+        delete track.resampler;
+        track.resampler = NULL;
+        track.sampleRate = mSampleRate; // without resampler, track rate is device sample rate.
+        // recreate the resampler with updated format, channels, saved sampleRate.
+        track.setResampler(resetToSampleRate /*trackSampleRate*/, mSampleRate /*devSampleRate*/);
+    }
+    return true;
+}
+
+void AudioMixer::track_t::unprepareForDownmix() {
+    ALOGV("AudioMixer::unprepareForDownmix(%p)", this);
+
+    mDownmixRequiresFormat = AUDIO_FORMAT_INVALID;
+    if (downmixerBufferProvider != NULL) {
+        // this track had previously been configured with a downmixer, delete it
+        ALOGV(" deleting old downmixer");
+        delete downmixerBufferProvider;
+        downmixerBufferProvider = NULL;
+        reconfigureBufferProviders();
+    } else {
+        ALOGV(" nothing to do, no downmixer to delete");
+    }
+}
+
+status_t AudioMixer::track_t::prepareForDownmix()
+{
+    ALOGV("AudioMixer::prepareForDownmix(%p) with mask 0x%x",
+            this, channelMask);
+
+    // discard the previous downmixer if there was one
+    unprepareForDownmix();
+    // MONO_HACK Only remix (upmix or downmix) if the track and mixer/device channel masks
+    // are not the same and not handled internally, as mono -> stereo currently is.
+    if (channelMask == mMixerChannelMask
+            || (channelMask == AUDIO_CHANNEL_OUT_MONO
+                    && mMixerChannelMask == AUDIO_CHANNEL_OUT_STEREO)) {
+        return NO_ERROR;
+    }
+    // DownmixerBufferProvider is only used for position masks.
+    if (audio_channel_mask_get_representation(channelMask)
+                == AUDIO_CHANNEL_REPRESENTATION_POSITION
+            && DownmixerBufferProvider::isMultichannelCapable()) {
+        DownmixerBufferProvider* pDbp = new DownmixerBufferProvider(channelMask,
+                mMixerChannelMask,
+                AUDIO_FORMAT_PCM_16_BIT /* TODO: use mMixerInFormat, now only PCM 16 */,
+                sampleRate, sessionId, kCopyBufferFrameCount);
+
+        if (pDbp->isValid()) { // if constructor completed properly
+            mDownmixRequiresFormat = AUDIO_FORMAT_PCM_16_BIT; // PCM 16 bit required for downmix
+            downmixerBufferProvider = pDbp;
+            reconfigureBufferProviders();
+            return NO_ERROR;
+        }
+        delete pDbp;
+    }
+
+    // Effect downmixer does not accept the channel conversion.  Let's use our remixer.
+    RemixBufferProvider* pRbp = new RemixBufferProvider(channelMask,
+            mMixerChannelMask, mMixerInFormat, kCopyBufferFrameCount);
+    // Remix always finds a conversion whereas Downmixer effect above may fail.
+    downmixerBufferProvider = pRbp;
+    reconfigureBufferProviders();
+    return NO_ERROR;
+}
+
+void AudioMixer::track_t::unprepareForReformat() {
+    ALOGV("AudioMixer::unprepareForReformat(%p)", this);
+    bool requiresReconfigure = false;
+    if (mReformatBufferProvider != NULL) {
+        delete mReformatBufferProvider;
+        mReformatBufferProvider = NULL;
+        requiresReconfigure = true;
+    }
+    if (mPostDownmixReformatBufferProvider != NULL) {
+        delete mPostDownmixReformatBufferProvider;
+        mPostDownmixReformatBufferProvider = NULL;
+        requiresReconfigure = true;
+    }
+    if (requiresReconfigure) {
+        reconfigureBufferProviders();
+    }
+}
+
+status_t AudioMixer::track_t::prepareForReformat()
+{
+    ALOGV("AudioMixer::prepareForReformat(%p) with format %#x", this, mFormat);
+    // discard previous reformatters
+    unprepareForReformat();
+    // only configure reformatters as needed
+    const audio_format_t targetFormat = mDownmixRequiresFormat != AUDIO_FORMAT_INVALID
+            ? mDownmixRequiresFormat : mMixerInFormat;
+    bool requiresReconfigure = false;
+    if (mFormat != targetFormat) {
+        mReformatBufferProvider = new ReformatBufferProvider(
+                audio_channel_count_from_out_mask(channelMask),
+                mFormat,
+                targetFormat,
+                kCopyBufferFrameCount);
+        requiresReconfigure = true;
+    }
+    if (targetFormat != mMixerInFormat) {
+        mPostDownmixReformatBufferProvider = new ReformatBufferProvider(
+                audio_channel_count_from_out_mask(mMixerChannelMask),
+                targetFormat,
+                mMixerInFormat,
+                kCopyBufferFrameCount);
+        requiresReconfigure = true;
+    }
+    if (requiresReconfigure) {
+        reconfigureBufferProviders();
+    }
+    return NO_ERROR;
+}
+
+void AudioMixer::track_t::reconfigureBufferProviders()
+{
+    bufferProvider = mInputBufferProvider;
+    if (mReformatBufferProvider) {
+        mReformatBufferProvider->setBufferProvider(bufferProvider);
+        bufferProvider = mReformatBufferProvider;
+    }
+    if (downmixerBufferProvider) {
+        downmixerBufferProvider->setBufferProvider(bufferProvider);
+        bufferProvider = downmixerBufferProvider;
+    }
+    if (mPostDownmixReformatBufferProvider) {
+        mPostDownmixReformatBufferProvider->setBufferProvider(bufferProvider);
+        bufferProvider = mPostDownmixReformatBufferProvider;
+    }
+    if (mTimestretchBufferProvider) {
+        mTimestretchBufferProvider->setBufferProvider(bufferProvider);
+        bufferProvider = mTimestretchBufferProvider;
+    }
+}
+
+void AudioMixer::deleteTrackName(int name)
+{
+    ALOGV("AudioMixer::deleteTrackName(%d)", name);
+    name -= TRACK0;
+    LOG_ALWAYS_FATAL_IF(name < 0 || name >= (int)MAX_NUM_TRACKS, "bad track name %d", name);
+    ALOGV("deleteTrackName(%d)", name);
+    track_t& track(mState.tracks[ name ]);
+    if (track.enabled) {
+        track.enabled = false;
+        invalidateState(1<<name);
+    }
+    // delete the resampler
+    delete track.resampler;
+    track.resampler = NULL;
+    // delete the downmixer
+    mState.tracks[name].unprepareForDownmix();
+    // delete the reformatter
+    mState.tracks[name].unprepareForReformat();
+    // delete the timestretch provider
+    delete track.mTimestretchBufferProvider;
+    track.mTimestretchBufferProvider = NULL;
+    mTrackNames &= ~(1<<name);
+}
+
+void AudioMixer::enable(int name)
+{
+    name -= TRACK0;
+    ALOG_ASSERT(uint32_t(name) < MAX_NUM_TRACKS, "bad track name %d", name);
+    track_t& track = mState.tracks[name];
+
+    if (!track.enabled) {
+        track.enabled = true;
+        ALOGV("enable(%d)", name);
+        invalidateState(1 << name);
+    }
+}
+
+void AudioMixer::disable(int name)
+{
+    name -= TRACK0;
+    ALOG_ASSERT(uint32_t(name) < MAX_NUM_TRACKS, "bad track name %d", name);
+    track_t& track = mState.tracks[name];
+
+    if (track.enabled) {
+        track.enabled = false;
+        ALOGV("disable(%d)", name);
+        invalidateState(1 << name);
+    }
+}
+
+/* Sets the volume ramp variables for the AudioMixer.
+ *
+ * The volume ramp variables are used to transition from the previous
+ * volume to the set volume.  ramp controls the duration of the transition.
+ * Its value is typically one state framecount period, but may also be 0,
+ * meaning "immediate."
+ *
+ * FIXME: 1) Volume ramp is enabled only if there is a nonzero integer increment
+ * even if there is a nonzero floating point increment (in that case, the volume
+ * change is immediate).  This restriction should be changed when the legacy mixer
+ * is removed (see #2).
+ * FIXME: 2) Integer volume variables are used for Legacy mixing and should be removed
+ * when no longer needed.
+ *
+ * @param newVolume set volume target in floating point [0.0, 1.0].
+ * @param ramp number of frames to increment over. if ramp is 0, the volume
+ * should be set immediately.  Currently ramp should not exceed 65535 (frames).
+ * @param pIntSetVolume pointer to the U4.12 integer target volume, set on return.
+ * @param pIntPrevVolume pointer to the U4.28 integer previous volume, set on return.
+ * @param pIntVolumeInc pointer to the U4.28 increment per output audio frame, set on return.
+ * @param pSetVolume pointer to the float target volume, set on return.
+ * @param pPrevVolume pointer to the float previous volume, set on return.
+ * @param pVolumeInc pointer to the float increment per output audio frame, set on return.
+ * @return true if the volume has changed, false if volume is same.
+ */
+static inline bool setVolumeRampVariables(float newVolume, int32_t ramp,
+        int16_t *pIntSetVolume, int32_t *pIntPrevVolume, int32_t *pIntVolumeInc,
+        float *pSetVolume, float *pPrevVolume, float *pVolumeInc) {
+    // check floating point volume to see if it is identical to the previously
+    // set volume.
+    // We do not use a tolerance here (and reject changes too small)
+    // as it may be confusing to use a different value than the one set.
+    // If the resulting volume is too small to ramp, it is a direct set of the volume.
+    if (newVolume == *pSetVolume) {
+        return false;
+    }
+    if (newVolume < 0) {
+        newVolume = 0; // should not have negative volumes
+    } else {
+        switch (fpclassify(newVolume)) {
+        case FP_SUBNORMAL:
+        case FP_NAN:
+            newVolume = 0;
+            break;
+        case FP_ZERO:
+            break; // zero volume is fine
+        case FP_INFINITE:
+            // Infinite volume could be handled consistently since
+            // floating point math saturates at infinities,
+            // but we limit volume to unity gain float.
+            // ramp = 0; break;
+            //
+            newVolume = AudioMixer::UNITY_GAIN_FLOAT;
+            break;
+        case FP_NORMAL:
+        default:
+            // Floating point does not have problems with overflow wrap
+            // that integer has.  However, we limit the volume to
+            // unity gain here.
+            // TODO: Revisit the volume limitation and perhaps parameterize.
+            if (newVolume > AudioMixer::UNITY_GAIN_FLOAT) {
+                newVolume = AudioMixer::UNITY_GAIN_FLOAT;
+            }
+            break;
+        }
+    }
+
+    // set floating point volume ramp
+    if (ramp != 0) {
+        // when the ramp completes, *pPrevVolume is set to *pSetVolume, so there
+        // is no computational mismatch; hence equality is checked here.
+        ALOGD_IF(*pPrevVolume != *pSetVolume, "previous float ramp hasn't finished,"
+                " prev:%f  set_to:%f", *pPrevVolume, *pSetVolume);
+        const float inc = (newVolume - *pPrevVolume) / ramp; // could be inf, nan, subnormal
+        const float maxv = max(newVolume, *pPrevVolume); // could be inf, cannot be nan, subnormal
+
+        if (isnormal(inc) // inc must be a normal number (no subnormals, infinite, nan)
+                && maxv + inc != maxv) { // inc must make forward progress
+            *pVolumeInc = inc;
+            // ramp is set now.
+            // Note: if newVolume is 0, then near the end of the ramp,
+            // it may be possible that the ramped volume may be subnormal or
+            // temporarily negative by a small amount or subnormal due to floating
+            // point inaccuracies.
+        } else {
+            ramp = 0; // ramp not allowed
+        }
+    }
+
+    // compute and check integer volume, no need to check negative values
+    // The integer volume is limited to "unity_gain" to avoid wrapping and other
+    // audio artifacts, so it never reaches the range limit of U4.28.
+    // We safely use signed 16 and 32 bit integers here.
+    const float scaledVolume = newVolume * AudioMixer::UNITY_GAIN_INT; // not neg, subnormal, nan
+    const int32_t intVolume = (scaledVolume >= (float)AudioMixer::UNITY_GAIN_INT) ?
+            AudioMixer::UNITY_GAIN_INT : (int32_t)scaledVolume;
+
+    // set integer volume ramp
+    if (ramp != 0) {
+        // integer volume is U4.12 (to use 16 bit multiplies), but ramping uses U4.28.
+        // when the ramp completes, *pIntPrevVolume is set to *pIntSetVolume << 16, so there
+        // is no computational mismatch; hence equality is checked here.
+        ALOGD_IF(*pIntPrevVolume != *pIntSetVolume << 16, "previous int ramp hasn't finished,"
+                " prev:%d  set_to:%d", *pIntPrevVolume, *pIntSetVolume << 16);
+        const int32_t inc = ((intVolume << 16) - *pIntPrevVolume) / ramp;
+
+        if (inc != 0) { // inc must make forward progress
+            *pIntVolumeInc = inc;
+        } else {
+            ramp = 0; // ramp not allowed
+        }
+    }
+
+    // if no ramp, or ramp not allowed, then clear float and integer increments
+    if (ramp == 0) {
+        *pVolumeInc = 0;
+        *pPrevVolume = newVolume;
+        *pIntVolumeInc = 0;
+        *pIntPrevVolume = intVolume << 16;
+    }
+    *pSetVolume = newVolume;
+    *pIntSetVolume = intVolume;
+    return true;
+}
+
+void AudioMixer::setParameter(int name, int target, int param, void *value)
+{
+    name -= TRACK0;
+    ALOG_ASSERT(uint32_t(name) < MAX_NUM_TRACKS, "bad track name %d", name);
+    track_t& track = mState.tracks[name];
+
+    int valueInt = static_cast<int>(reinterpret_cast<uintptr_t>(value));
+    int32_t *valueBuf = reinterpret_cast<int32_t*>(value);
+
+    switch (target) {
+
+    case TRACK:
+        switch (param) {
+        case CHANNEL_MASK: {
+            const audio_channel_mask_t trackChannelMask =
+                static_cast<audio_channel_mask_t>(valueInt);
+            if (setChannelMasks(name, trackChannelMask, track.mMixerChannelMask)) {
+                ALOGV("setParameter(TRACK, CHANNEL_MASK, %x)", trackChannelMask);
+                invalidateState(1 << name);
+            }
+            } break;
+        case MAIN_BUFFER:
+            if (track.mainBuffer != valueBuf) {
+                track.mainBuffer = valueBuf;
+                ALOGV("setParameter(TRACK, MAIN_BUFFER, %p)", valueBuf);
+                invalidateState(1 << name);
+            }
+            break;
+        case AUX_BUFFER:
+            if (track.auxBuffer != valueBuf) {
+                track.auxBuffer = valueBuf;
+                ALOGV("setParameter(TRACK, AUX_BUFFER, %p)", valueBuf);
+                invalidateState(1 << name);
+            }
+            break;
+        case FORMAT: {
+            audio_format_t format = static_cast<audio_format_t>(valueInt);
+            if (track.mFormat != format) {
+                ALOG_ASSERT(audio_is_linear_pcm(format), "Invalid format %#x", format);
+                track.mFormat = format;
+                ALOGV("setParameter(TRACK, FORMAT, %#x)", format);
+                track.prepareForReformat();
+                invalidateState(1 << name);
+            }
+            } break;
+        // FIXME do we want to support setting the downmix type from AudioFlinger?
+        //         for a specific track? or per mixer?
+        /* case DOWNMIX_TYPE:
+            break          */
+        case MIXER_FORMAT: {
+            audio_format_t format = static_cast<audio_format_t>(valueInt);
+            if (track.mMixerFormat != format) {
+                track.mMixerFormat = format;
+                ALOGV("setParameter(TRACK, MIXER_FORMAT, %#x)", format);
+            }
+            } break;
+        case MIXER_CHANNEL_MASK: {
+            const audio_channel_mask_t mixerChannelMask =
+                    static_cast<audio_channel_mask_t>(valueInt);
+            if (setChannelMasks(name, track.channelMask, mixerChannelMask)) {
+                ALOGV("setParameter(TRACK, MIXER_CHANNEL_MASK, %#x)", mixerChannelMask);
+                invalidateState(1 << name);
+            }
+            } break;
+        default:
+            LOG_ALWAYS_FATAL("setParameter track: bad param %d", param);
+        }
+        break;
+
+    case RESAMPLE:
+        switch (param) {
+        case SAMPLE_RATE:
+            ALOG_ASSERT(valueInt > 0, "bad sample rate %d", valueInt);
+            if (track.setResampler(uint32_t(valueInt), mSampleRate)) {
+                ALOGV("setParameter(RESAMPLE, SAMPLE_RATE, %u)",
+                        uint32_t(valueInt));
+                invalidateState(1 << name);
+            }
+            break;
+        case RESET:
+            track.resetResampler();
+            invalidateState(1 << name);
+            break;
+        case REMOVE:
+            delete track.resampler;
+            track.resampler = NULL;
+            track.sampleRate = mSampleRate;
+            invalidateState(1 << name);
+            break;
+        default:
+            LOG_ALWAYS_FATAL("setParameter resample: bad param %d", param);
+        }
+        break;
+
+    case RAMP_VOLUME:
+    case VOLUME:
+        switch (param) {
+        case AUXLEVEL:
+            if (setVolumeRampVariables(*reinterpret_cast<float*>(value),
+                    target == RAMP_VOLUME ? mState.frameCount : 0,
+                    &track.auxLevel, &track.prevAuxLevel, &track.auxInc,
+                    &track.mAuxLevel, &track.mPrevAuxLevel, &track.mAuxInc)) {
+                ALOGV("setParameter(%s, AUXLEVEL: %04x)",
+                        target == VOLUME ? "VOLUME" : "RAMP_VOLUME", track.auxLevel);
+                invalidateState(1 << name);
+            }
+            break;
+        default:
+            if ((unsigned)param >= VOLUME0 && (unsigned)param < VOLUME0 + MAX_NUM_VOLUMES) {
+                if (setVolumeRampVariables(*reinterpret_cast<float*>(value),
+                        target == RAMP_VOLUME ? mState.frameCount : 0,
+                        &track.volume[param - VOLUME0], &track.prevVolume[param - VOLUME0],
+                        &track.volumeInc[param - VOLUME0],
+                        &track.mVolume[param - VOLUME0], &track.mPrevVolume[param - VOLUME0],
+                        &track.mVolumeInc[param - VOLUME0])) {
+                    ALOGV("setParameter(%s, VOLUME%d: %04x)",
+                            target == VOLUME ? "VOLUME" : "RAMP_VOLUME", param - VOLUME0,
+                                    track.volume[param - VOLUME0]);
+                    invalidateState(1 << name);
+                }
+            } else {
+                LOG_ALWAYS_FATAL("setParameter volume: bad param %d", param);
+            }
+        }
+        break;
+        case TIMESTRETCH:
+            switch (param) {
+            case PLAYBACK_RATE: {
+                const AudioPlaybackRate *playbackRate =
+                        reinterpret_cast<AudioPlaybackRate*>(value);
+                ALOGW_IF(!isAudioPlaybackRateValid(*playbackRate),
+                        "bad parameters speed %f, pitch %f",playbackRate->mSpeed,
+                        playbackRate->mPitch);
+                if (track.setPlaybackRate(*playbackRate)) {
+                    ALOGV("setParameter(TIMESTRETCH, PLAYBACK_RATE, STRETCH_MODE, FALLBACK_MODE "
+                            "%f %f %d %d",
+                            playbackRate->mSpeed,
+                            playbackRate->mPitch,
+                            playbackRate->mStretchMode,
+                            playbackRate->mFallbackMode);
+                    // invalidateState(1 << name);
+                }
+            } break;
+            default:
+                LOG_ALWAYS_FATAL("setParameter timestretch: bad param %d", param);
+            }
+            break;
+
+    default:
+        LOG_ALWAYS_FATAL("setParameter: bad target %d", target);
+    }
+}
+
+bool AudioMixer::track_t::setResampler(uint32_t trackSampleRate, uint32_t devSampleRate)
+{
+    if (trackSampleRate != devSampleRate || resampler != NULL) {
+        if (sampleRate != trackSampleRate) {
+            sampleRate = trackSampleRate;
+            if (resampler == NULL) {
+                ALOGV("Creating resampler from track %d Hz to device %d Hz",
+                        trackSampleRate, devSampleRate);
+                AudioResampler::src_quality quality;
+                // force lowest quality level resampler if use case isn't music or video
+                // FIXME this is flawed for dynamic sample rates, as we choose the resampler
+                // quality level based on the initial ratio, but that could change later.
+                // Should have a way to distinguish tracks with static ratios vs. dynamic ratios.
+                if (isMusicRate(trackSampleRate)) {
+                    quality = AudioResampler::DEFAULT_QUALITY;
+                } else {
+                    quality = AudioResampler::DYN_LOW_QUALITY;
+                }
+
+                // TODO: Remove MONO_HACK. Resampler sees #channels after the downmixer
+                // but if none exists, it is the channel count (1 for mono).
+                const int resamplerChannelCount = downmixerBufferProvider != NULL
+                        ? mMixerChannelCount : channelCount;
+                ALOGVV("Creating resampler:"
+                        " format(%#x) channels(%d) devSampleRate(%u) quality(%d)\n",
+                        mMixerInFormat, resamplerChannelCount, devSampleRate, quality);
+                resampler = AudioResampler::create(
+                        mMixerInFormat,
+                        resamplerChannelCount,
+                        devSampleRate, quality);
+            }
+            return true;
+        }
+    }
+    return false;
+}
+
+bool AudioMixer::track_t::setPlaybackRate(const AudioPlaybackRate &playbackRate)
+{
+    if ((mTimestretchBufferProvider == NULL &&
+            fabs(playbackRate.mSpeed - mPlaybackRate.mSpeed) < AUDIO_TIMESTRETCH_SPEED_MIN_DELTA &&
+            fabs(playbackRate.mPitch - mPlaybackRate.mPitch) < AUDIO_TIMESTRETCH_PITCH_MIN_DELTA) ||
+            isAudioPlaybackRateEqual(playbackRate, mPlaybackRate)) {
+        return false;
+    }
+    mPlaybackRate = playbackRate;
+    if (mTimestretchBufferProvider == NULL) {
+        // TODO: Remove MONO_HACK. Resampler sees #channels after the downmixer
+        // but if none exists, it is the channel count (1 for mono).
+        const int timestretchChannelCount = downmixerBufferProvider != NULL
+                ? mMixerChannelCount : channelCount;
+        mTimestretchBufferProvider = new TimestretchBufferProvider(timestretchChannelCount,
+                mMixerInFormat, sampleRate, playbackRate);
+        reconfigureBufferProviders();
+    } else {
+        reinterpret_cast<TimestretchBufferProvider*>(mTimestretchBufferProvider)
+                ->setPlaybackRate(playbackRate);
+    }
+    return true;
+}
+
+/* Checks to see if the volume ramp has completed and clears the increment
+ * variables appropriately.
+ *
+ * FIXME: There is code to handle int/float ramp variable switchover should it not
+ * complete within a mixer buffer processing call, but it is preferred to avoid switchover
+ * due to precision issues.  The switchover code is included for legacy code purposes
+ * and can be removed once the integer volume is removed.
+ *
+ * It is not sufficient to clear only the volumeInc integer variable because
+ * if one channel requires ramping, all channels are ramped.
+ *
+ * There is a bit of duplicated code here, but it keeps backward compatibility.
+ */
+inline void AudioMixer::track_t::adjustVolumeRamp(bool aux, bool useFloat)
+{
+    if (useFloat) {
+        for (uint32_t i = 0; i < MAX_NUM_VOLUMES; i++) {
+            if ((mVolumeInc[i] > 0 && mPrevVolume[i] + mVolumeInc[i] >= mVolume[i]) ||
+                     (mVolumeInc[i] < 0 && mPrevVolume[i] + mVolumeInc[i] <= mVolume[i])) {
+                volumeInc[i] = 0;
+                prevVolume[i] = volume[i] << 16;
+                mVolumeInc[i] = 0.;
+                mPrevVolume[i] = mVolume[i];
+            } else {
+                //ALOGV("ramp: %f %f %f", mVolume[i], mPrevVolume[i], mVolumeInc[i]);
+                prevVolume[i] = u4_28_from_float(mPrevVolume[i]);
+            }
+        }
+    } else {
+        for (uint32_t i = 0; i < MAX_NUM_VOLUMES; i++) {
+            if (((volumeInc[i]>0) && (((prevVolume[i]+volumeInc[i])>>16) >= volume[i])) ||
+                    ((volumeInc[i]<0) && (((prevVolume[i]+volumeInc[i])>>16) <= volume[i]))) {
+                volumeInc[i] = 0;
+                prevVolume[i] = volume[i] << 16;
+                mVolumeInc[i] = 0.;
+                mPrevVolume[i] = mVolume[i];
+            } else {
+                //ALOGV("ramp: %d %d %d", volume[i] << 16, prevVolume[i], volumeInc[i]);
+                mPrevVolume[i]  = float_from_u4_28(prevVolume[i]);
+            }
+        }
+    }
+    /* TODO: aux is always integer regardless of output buffer type */
+    if (aux) {
+        if (((auxInc>0) && (((prevAuxLevel+auxInc)>>16) >= auxLevel)) ||
+                ((auxInc<0) && (((prevAuxLevel+auxInc)>>16) <= auxLevel))) {
+            auxInc = 0;
+            prevAuxLevel = auxLevel << 16;
+            mAuxInc = 0.;
+            mPrevAuxLevel = mAuxLevel;
+        } else {
+            //ALOGV("aux ramp: %d %d %d", auxLevel << 16, prevAuxLevel, auxInc);
+        }
+    }
+}
+
+size_t AudioMixer::getUnreleasedFrames(int name) const
+{
+    name -= TRACK0;
+    if (uint32_t(name) < MAX_NUM_TRACKS) {
+        return mState.tracks[name].getUnreleasedFrames();
+    }
+    return 0;
+}
+
+void AudioMixer::setBufferProvider(int name, AudioBufferProvider* bufferProvider)
+{
+    name -= TRACK0;
+    ALOG_ASSERT(uint32_t(name) < MAX_NUM_TRACKS, "bad track name %d", name);
+
+    if (mState.tracks[name].mInputBufferProvider == bufferProvider) {
+        return; // don't reset any buffer providers if identical.
+    }
+    if (mState.tracks[name].mReformatBufferProvider != NULL) {
+        mState.tracks[name].mReformatBufferProvider->reset();
+    } else if (mState.tracks[name].downmixerBufferProvider != NULL) {
+        mState.tracks[name].downmixerBufferProvider->reset();
+    } else if (mState.tracks[name].mPostDownmixReformatBufferProvider != NULL) {
+        mState.tracks[name].mPostDownmixReformatBufferProvider->reset();
+    } else if (mState.tracks[name].mTimestretchBufferProvider != NULL) {
+        mState.tracks[name].mTimestretchBufferProvider->reset();
+    }
+
+    mState.tracks[name].mInputBufferProvider = bufferProvider;
+    mState.tracks[name].reconfigureBufferProviders();
+}
+
+
+void AudioMixer::process()
+{
+    mState.hook(&mState);
+}
+
+
+void AudioMixer::process__validate(state_t* state)
+{
+    ALOGW_IF(!state->needsChanged,
+        "in process__validate() but nothing's invalid");
+
+    uint32_t changed = state->needsChanged;
+    state->needsChanged = 0; // clear the validation flag
+
+    // recompute which tracks are enabled / disabled
+    uint32_t enabled = 0;
+    uint32_t disabled = 0;
+    while (changed) {
+        const int i = 31 - __builtin_clz(changed);
+        const uint32_t mask = 1<<i;
+        changed &= ~mask;
+        track_t& t = state->tracks[i];
+        (t.enabled ? enabled : disabled) |= mask;
+    }
+    state->enabledTracks &= ~disabled;
+    state->enabledTracks |=  enabled;
+
+    // compute everything we need...
+    int countActiveTracks = 0;
+    // TODO: fix all16BitsStereNoResample logic to
+    // either properly handle muted tracks (it should ignore them)
+    // or remove altogether as an obsolete optimization.
+    bool all16BitsStereoNoResample = true;
+    bool resampling = false;
+    bool volumeRamp = false;
+    uint32_t en = state->enabledTracks;
+    while (en) {
+        const int i = 31 - __builtin_clz(en);
+        en &= ~(1<<i);
+
+        countActiveTracks++;
+        track_t& t = state->tracks[i];
+        uint32_t n = 0;
+        // FIXME can overflow (mask is only 3 bits)
+        n |= NEEDS_CHANNEL_1 + t.channelCount - 1;
+        if (t.doesResample()) {
+            n |= NEEDS_RESAMPLE;
+        }
+        if (t.auxLevel != 0 && t.auxBuffer != NULL) {
+            n |= NEEDS_AUX;
+        }
+
+        if (t.volumeInc[0]|t.volumeInc[1]) {
+            volumeRamp = true;
+        } else if (!t.doesResample() && t.volumeRL == 0) {
+            n |= NEEDS_MUTE;
+        }
+        t.needs = n;
+
+        if (n & NEEDS_MUTE) {
+            t.hook = track__nop;
+        } else {
+            if (n & NEEDS_AUX) {
+                all16BitsStereoNoResample = false;
+            }
+            if (n & NEEDS_RESAMPLE) {
+                all16BitsStereoNoResample = false;
+                resampling = true;
+                t.hook = getTrackHook(TRACKTYPE_RESAMPLE, t.mMixerChannelCount,
+                        t.mMixerInFormat, t.mMixerFormat);
+                ALOGV_IF((n & NEEDS_CHANNEL_COUNT__MASK) > NEEDS_CHANNEL_2,
+                        "Track %d needs downmix + resample", i);
+            } else {
+                if ((n & NEEDS_CHANNEL_COUNT__MASK) == NEEDS_CHANNEL_1){
+                    t.hook = getTrackHook(
+                            (t.mMixerChannelMask == AUDIO_CHANNEL_OUT_STEREO  // TODO: MONO_HACK
+                                    && t.channelMask == AUDIO_CHANNEL_OUT_MONO)
+                                ? TRACKTYPE_NORESAMPLEMONO : TRACKTYPE_NORESAMPLE,
+                            t.mMixerChannelCount,
+                            t.mMixerInFormat, t.mMixerFormat);
+                    all16BitsStereoNoResample = false;
+                }
+                if ((n & NEEDS_CHANNEL_COUNT__MASK) >= NEEDS_CHANNEL_2){
+                    t.hook = getTrackHook(TRACKTYPE_NORESAMPLE, t.mMixerChannelCount,
+                            t.mMixerInFormat, t.mMixerFormat);
+                    ALOGV_IF((n & NEEDS_CHANNEL_COUNT__MASK) > NEEDS_CHANNEL_2,
+                            "Track %d needs downmix", i);
+                }
+            }
+        }
+    }
+
+    // select the processing hooks
+    state->hook = process__nop;
+    if (countActiveTracks > 0) {
+        if (resampling) {
+            if (!state->outputTemp) {
+                state->outputTemp = new int32_t[MAX_NUM_CHANNELS * state->frameCount];
+            }
+            if (!state->resampleTemp) {
+                state->resampleTemp = new int32_t[MAX_NUM_CHANNELS * state->frameCount];
+            }
+            state->hook = process__genericResampling;
+        } else {
+            if (state->outputTemp) {
+                delete [] state->outputTemp;
+                state->outputTemp = NULL;
+            }
+            if (state->resampleTemp) {
+                delete [] state->resampleTemp;
+                state->resampleTemp = NULL;
+            }
+            state->hook = process__genericNoResampling;
+            if (all16BitsStereoNoResample && !volumeRamp) {
+                if (countActiveTracks == 1) {
+                    const int i = 31 - __builtin_clz(state->enabledTracks);
+                    track_t& t = state->tracks[i];
+                    if ((t.needs & NEEDS_MUTE) == 0) {
+                        // The check prevents a muted track from acquiring a process hook.
+                        //
+                        // This is dangerous if the track is MONO as that requires
+                        // special case handling due to implicit channel duplication.
+                        // Stereo or Multichannel should actually be fine here.
+                        state->hook = getProcessHook(PROCESSTYPE_NORESAMPLEONETRACK,
+                                t.mMixerChannelCount, t.mMixerInFormat, t.mMixerFormat);
+                    }
+                }
+            }
+        }
+    }
+
+    ALOGV("mixer configuration change: %d activeTracks (%08x) "
+        "all16BitsStereoNoResample=%d, resampling=%d, volumeRamp=%d",
+        countActiveTracks, state->enabledTracks,
+        all16BitsStereoNoResample, resampling, volumeRamp);
+
+   state->hook(state);
+
+    // Now that the volume ramp has been done, set optimal state and
+    // track hooks for subsequent mixer process
+    if (countActiveTracks > 0) {
+        bool allMuted = true;
+        uint32_t en = state->enabledTracks;
+        while (en) {
+            const int i = 31 - __builtin_clz(en);
+            en &= ~(1<<i);
+            track_t& t = state->tracks[i];
+            if (!t.doesResample() && t.volumeRL == 0) {
+                t.needs |= NEEDS_MUTE;
+                t.hook = track__nop;
+            } else {
+                allMuted = false;
+            }
+        }
+        if (allMuted) {
+            state->hook = process__nop;
+        } else if (all16BitsStereoNoResample) {
+            if (countActiveTracks == 1) {
+                const int i = 31 - __builtin_clz(state->enabledTracks);
+                track_t& t = state->tracks[i];
+                // Muted single tracks handled by allMuted above.
+                state->hook = getProcessHook(PROCESSTYPE_NORESAMPLEONETRACK,
+                        t.mMixerChannelCount, t.mMixerInFormat, t.mMixerFormat);
+            }
+        }
+    }
+}
+
+
+void AudioMixer::track__genericResample(track_t* t, int32_t* out, size_t outFrameCount,
+        int32_t* temp, int32_t* aux)
+{
+    ALOGVV("track__genericResample\n");
+    t->resampler->setSampleRate(t->sampleRate);
+
+    // ramp gain - resample to temp buffer and scale/mix in 2nd step
+    if (aux != NULL) {
+        // always resample with unity gain when sending to auxiliary buffer to be able
+        // to apply send level after resampling
+        t->resampler->setVolume(UNITY_GAIN_FLOAT, UNITY_GAIN_FLOAT);
+        memset(temp, 0, outFrameCount * t->mMixerChannelCount * sizeof(int32_t));
+        t->resampler->resample(temp, outFrameCount, t->bufferProvider);
+        if (CC_UNLIKELY(t->volumeInc[0]|t->volumeInc[1]|t->auxInc)) {
+            volumeRampStereo(t, out, outFrameCount, temp, aux);
+        } else {
+            volumeStereo(t, out, outFrameCount, temp, aux);
+        }
+    } else {
+        if (CC_UNLIKELY(t->volumeInc[0]|t->volumeInc[1])) {
+            t->resampler->setVolume(UNITY_GAIN_FLOAT, UNITY_GAIN_FLOAT);
+            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
+        else {
+            t->resampler->setVolume(t->mVolume[0], t->mVolume[1]);
+            t->resampler->resample(out, outFrameCount, t->bufferProvider);
+        }
+    }
+}
+
+void AudioMixer::track__nop(track_t* t __unused, int32_t* out __unused,
+        size_t outFrameCount __unused, int32_t* temp __unused, int32_t* aux __unused)
+{
+}
+
+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];
+    const int32_t vlInc = t->volumeInc[0];
+    const int32_t vrInc = t->volumeInc[1];
+
+    //ALOGD("[0] %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);
+
+    // ramp volume
+    if (CC_UNLIKELY(aux != NULL)) {
+        int32_t va = t->prevAuxLevel;
+        const int32_t vaInc = t->auxInc;
+        int32_t l;
+        int32_t r;
+
+        do {
+            l = (*temp++ >> 12);
+            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(aux != NULL);
+}
+
+void AudioMixer::volumeStereo(track_t* t, int32_t* out, size_t frameCount, int32_t* temp,
+        int32_t* aux)
+{
+    const int16_t vl = t->volume[0];
+    const int16_t vr = t->volume[1];
+
+    if (CC_UNLIKELY(aux != NULL)) {
+        const int16_t va = t->auxLevel;
+        do {
+            int16_t l = (int16_t)(*temp++ >> 12);
+            int16_t r = (int16_t)(*temp++ >> 12);
+            out[0] = mulAdd(l, vl, out[0]);
+            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 __unused, int32_t* aux)
+{
+    ALOGVV("track__16BitsStereo\n");
+    const int16_t *in = static_cast<const int16_t *>(t->in);
+
+    if (CC_UNLIKELY(aux != NULL)) {
+        int32_t l;
+        int32_t r;
+        // ramp gain
+        if (CC_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;
+            // ALOGD("[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<const uint32_t *>(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 (CC_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];
+
+            // ALOGD("[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<const uint32_t *>(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 __unused, int32_t* aux)
+{
+    ALOGVV("track__16BitsMono\n");
+    const int16_t *in = static_cast<int16_t const *>(t->in);
+
+    if (CC_UNLIKELY(aux != NULL)) {
+        // ramp gain
+        if (CC_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;
+
+            // ALOGD("[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;
+                *aux++ += (va >> 16) * l;
+                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 int16_t vl = t->volume[0];
+            const int16_t vr = t->volume[1];
+            const int16_t va = (int16_t)t->auxLevel;
+            do {
+                int16_t l = *in++;
+                out[0] = mulAdd(l, vl, out[0]);
+                out[1] = mulAdd(l, vr, out[1]);
+                out += 2;
+                aux[0] = mulAdd(l, va, aux[0]);
+                aux++;
+            } while (--frameCount);
+        }
+    } else {
+        // ramp gain
+        if (CC_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];
+
+            // ALOGD("[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;
+}
+
+// no-op case
+void AudioMixer::process__nop(state_t* state)
+{
+    ALOGVV("process__nop\n");
+    uint32_t e0 = state->enabledTracks;
+    while (e0) {
+        // process by group of tracks with same output buffer to
+        // avoid multiple memset() on same buffer
+        uint32_t e1 = e0, e2 = e0;
+        int i = 31 - __builtin_clz(e1);
+        {
+            track_t& t1 = state->tracks[i];
+            e2 &= ~(1<<i);
+            while (e2) {
+                i = 31 - __builtin_clz(e2);
+                e2 &= ~(1<<i);
+                track_t& t2 = state->tracks[i];
+                if (CC_UNLIKELY(t2.mainBuffer != t1.mainBuffer)) {
+                    e1 &= ~(1<<i);
+                }
+            }
+            e0 &= ~(e1);
+
+            memset(t1.mainBuffer, 0, state->frameCount * t1.mMixerChannelCount
+                    * audio_bytes_per_sample(t1.mMixerFormat));
+        }
+
+        while (e1) {
+            i = 31 - __builtin_clz(e1);
+            e1 &= ~(1<<i);
+            {
+                track_t& t3 = state->tracks[i];
+                size_t outFrames = state->frameCount;
+                while (outFrames) {
+                    t3.buffer.frameCount = outFrames;
+                    t3.bufferProvider->getNextBuffer(&t3.buffer);
+                    if (t3.buffer.raw == NULL) break;
+                    outFrames -= t3.buffer.frameCount;
+                    t3.bufferProvider->releaseBuffer(&t3.buffer);
+                }
+            }
+        }
+    }
+}
+
+// generic code without resampling
+void AudioMixer::process__genericNoResampling(state_t* state)
+{
+    ALOGVV("process__genericNoResampling\n");
+    int32_t outTemp[BLOCKSIZE * MAX_NUM_CHANNELS] __attribute__((aligned(32)));
+
+    // acquire each track's buffer
+    uint32_t enabledTracks = state->enabledTracks;
+    uint32_t e0 = enabledTracks;
+    while (e0) {
+        const int i = 31 - __builtin_clz(e0);
+        e0 &= ~(1<<i);
+        track_t& t = state->tracks[i];
+        t.buffer.frameCount = state->frameCount;
+        t.bufferProvider->getNextBuffer(&t.buffer);
+        t.frameCount = t.buffer.frameCount;
+        t.in = t.buffer.raw;
+    }
+
+    e0 = enabledTracks;
+    while (e0) {
+        // process by group of tracks with same output buffer to
+        // optimize cache use
+        uint32_t e1 = e0, e2 = e0;
+        int j = 31 - __builtin_clz(e1);
+        track_t& t1 = state->tracks[j];
+        e2 &= ~(1<<j);
+        while (e2) {
+            j = 31 - __builtin_clz(e2);
+            e2 &= ~(1<<j);
+            track_t& t2 = state->tracks[j];
+            if (CC_UNLIKELY(t2.mainBuffer != t1.mainBuffer)) {
+                e1 &= ~(1<<j);
+            }
+        }
+        e0 &= ~(e1);
+        // this assumes output 16 bits stereo, no resampling
+        int32_t *out = t1.mainBuffer;
+        size_t numFrames = 0;
+        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 (CC_UNLIKELY(t.needs & NEEDS_AUX)) {
+                    aux = t.auxBuffer + numFrames;
+                }
+                while (outFrames) {
+                    // t.in == NULL can happen if the track was flushed just after having
+                    // been enabled for mixing.
+                   if (t.in == NULL) {
+                        enabledTracks &= ~(1<<i);
+                        e1 &= ~(1<<i);
+                        break;
+                    }
+                    size_t inFrames = (t.frameCount > outFrames)?outFrames:t.frameCount;
+                    if (inFrames > 0) {
+                        t.hook(&t, outTemp + (BLOCKSIZE - outFrames) * t.mMixerChannelCount,
+                                inFrames, state->resampleTemp, aux);
+                        t.frameCount -= inFrames;
+                        outFrames -= inFrames;
+                        if (CC_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;
+                    }
+                }
+            }
+
+            convertMixerFormat(out, t1.mMixerFormat, outTemp, t1.mMixerInFormat,
+                    BLOCKSIZE * t1.mMixerChannelCount);
+            // TODO: fix ugly casting due to choice of out pointer type
+            out = reinterpret_cast<int32_t*>((uint8_t*)out
+                    + BLOCKSIZE * t1.mMixerChannelCount
+                        * audio_bytes_per_sample(t1.mMixerFormat));
+            numFrames += BLOCKSIZE;
+        } while (numFrames < state->frameCount);
+    }
+
+    // release each track's buffer
+    e0 = enabledTracks;
+    while (e0) {
+        const int i = 31 - __builtin_clz(e0);
+        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)
+{
+    ALOGVV("process__genericResampling\n");
+    // this const just means that local variable outTemp doesn't change
+    int32_t* const outTemp = state->outputTemp;
+    size_t numFrames = state->frameCount;
+
+    uint32_t e0 = state->enabledTracks;
+    while (e0) {
+        // process by group of tracks with same output buffer
+        // to optimize cache use
+        uint32_t e1 = e0, e2 = e0;
+        int j = 31 - __builtin_clz(e1);
+        track_t& t1 = state->tracks[j];
+        e2 &= ~(1<<j);
+        while (e2) {
+            j = 31 - __builtin_clz(e2);
+            e2 &= ~(1<<j);
+            track_t& t2 = state->tracks[j];
+            if (CC_UNLIKELY(t2.mainBuffer != t1.mainBuffer)) {
+                e1 &= ~(1<<j);
+            }
+        }
+        e0 &= ~(e1);
+        int32_t *out = t1.mainBuffer;
+        memset(outTemp, 0, sizeof(*outTemp) * t1.mMixerChannelCount * state->frameCount);
+        while (e1) {
+            const int i = 31 - __builtin_clz(e1);
+            e1 &= ~(1<<i);
+            track_t& t = state->tracks[i];
+            int32_t *aux = NULL;
+            if (CC_UNLIKELY(t.needs & NEEDS_AUX)) {
+                aux = t.auxBuffer;
+            }
+
+            // 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) {
+                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 (CC_UNLIKELY(aux != NULL)) {
+                        aux += outFrames;
+                    }
+                    t.hook(&t, outTemp + outFrames * t.mMixerChannelCount, t.buffer.frameCount,
+                            state->resampleTemp, aux);
+                    outFrames += t.buffer.frameCount;
+                    t.bufferProvider->releaseBuffer(&t.buffer);
+                }
+            }
+        }
+        convertMixerFormat(out, t1.mMixerFormat,
+                outTemp, t1.mMixerInFormat, numFrames * t1.mMixerChannelCount);
+    }
+}
+
+// one track, 16 bits stereo without resampling is the most common case
+void AudioMixer::process__OneTrack16BitsStereoNoResampling(state_t* state)
+{
+    ALOGVV("process__OneTrack16BitsStereoNoResampling\n");
+    // This method is only called when state->enabledTracks has exactly
+    // one bit set.  The asserts below would verify this, but are commented out
+    // since the whole point of this method is to optimize performance.
+    //ALOG_ASSERT(0 != state->enabledTracks, "no tracks enabled");
+    const int i = 31 - __builtin_clz(state->enabledTracks);
+    //ALOG_ASSERT((1 << i) == state->enabledTracks, "more than 1 track enabled");
+    const track_t& t = state->tracks[i];
+
+    AudioBufferProvider::Buffer& b(t.buffer);
+
+    int32_t* out = t.mainBuffer;
+    float *fout = reinterpret_cast<float*>(out);
+    size_t numFrames = state->frameCount;
+
+    const int16_t vl = t.volume[0];
+    const int16_t vr = t.volume[1];
+    const uint32_t vrl = t.volumeRL;
+    while (numFrames) {
+        b.frameCount = numFrames;
+        t.bufferProvider->getNextBuffer(&b);
+        const int16_t *in = b.i16;
+
+        // in == NULL can happen if the track was flushed just after having
+        // been enabled for mixing.
+        if (in == NULL || (((uintptr_t)in) & 3)) {
+            if ( AUDIO_FORMAT_PCM_FLOAT == t.mMixerFormat ) {
+                 memset((char*)fout, 0, numFrames
+                         * t.mMixerChannelCount * audio_bytes_per_sample(t.mMixerFormat));
+            } else {
+                 memset((char*)out, 0, numFrames
+                         * t.mMixerChannelCount * audio_bytes_per_sample(t.mMixerFormat));
+            }
+            ALOGE_IF((((uintptr_t)in) & 3),
+                    "process__OneTrack16BitsStereoNoResampling: misaligned buffer"
+                    " %p track %d, channels %d, needs %08x, volume %08x vfl %f vfr %f",
+                    in, i, t.channelCount, t.needs, vrl, t.mVolume[0], t.mVolume[1]);
+            return;
+        }
+        size_t outFrames = b.frameCount;
+
+        switch (t.mMixerFormat) {
+        case AUDIO_FORMAT_PCM_FLOAT:
+            do {
+                uint32_t rl = *reinterpret_cast<const uint32_t *>(in);
+                in += 2;
+                int32_t l = mulRL(1, rl, vrl);
+                int32_t r = mulRL(0, rl, vrl);
+                *fout++ = float_from_q4_27(l);
+                *fout++ = float_from_q4_27(r);
+                // Note: In case of later int16_t sink output,
+                // conversion and clamping is done by memcpy_to_i16_from_float().
+            } while (--outFrames);
+            break;
+        case AUDIO_FORMAT_PCM_16_BIT:
+            if (CC_UNLIKELY(uint32_t(vl) > UNITY_GAIN_INT || uint32_t(vr) > UNITY_GAIN_INT)) {
+                // volume is boosted, so we might need to clamp even though
+                // we process only one track.
+                do {
+                    uint32_t rl = *reinterpret_cast<const uint32_t *>(in);
+                    in += 2;
+                    int32_t l = mulRL(1, rl, vrl) >> 12;
+                    int32_t r = mulRL(0, rl, vrl) >> 12;
+                    // clamping...
+                    l = clamp16(l);
+                    r = clamp16(r);
+                    *out++ = (r<<16) | (l & 0xFFFF);
+                } while (--outFrames);
+            } else {
+                do {
+                    uint32_t rl = *reinterpret_cast<const uint32_t *>(in);
+                    in += 2;
+                    int32_t l = mulRL(1, rl, vrl) >> 12;
+                    int32_t r = mulRL(0, rl, vrl) >> 12;
+                    *out++ = (r<<16) | (l & 0xFFFF);
+                } while (--outFrames);
+            }
+            break;
+        default:
+            LOG_ALWAYS_FATAL("bad mixer format: %d", t.mMixerFormat);
+        }
+        numFrames -= b.frameCount;
+        t.bufferProvider->releaseBuffer(&b);
+    }
+}
+
+/*static*/ pthread_once_t AudioMixer::sOnceControl = PTHREAD_ONCE_INIT;
+
+/*static*/ void AudioMixer::sInitRoutine()
+{
+    DownmixerBufferProvider::init(); // for the downmixer
+}
+
+/* TODO: consider whether this level of optimization is necessary.
+ * Perhaps just stick with a single for loop.
+ */
+
+// Needs to derive a compile time constant (constexpr).  Could be targeted to go
+// to a MONOVOL mixtype based on MAX_NUM_VOLUMES, but that's an unnecessary complication.
+#define MIXTYPE_MONOVOL(mixtype) ((mixtype) == MIXTYPE_MULTI ? MIXTYPE_MULTI_MONOVOL : \
+        (mixtype) == MIXTYPE_MULTI_SAVEONLY ? MIXTYPE_MULTI_SAVEONLY_MONOVOL : (mixtype))
+
+/* MIXTYPE     (see AudioMixerOps.h MIXTYPE_* enumeration)
+ * 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,
+        typename TO, typename TI, typename TV, typename TA, typename TAV>
+static void volumeRampMulti(uint32_t channels, TO* out, size_t frameCount,
+        const TI* in, TA* aux, TV *vol, const TV *volinc, TAV *vola, TAV volainc)
+{
+    switch (channels) {
+    case 1:
+        volumeRampMulti<MIXTYPE, 1>(out, frameCount, in, aux, vol, volinc, vola, volainc);
+        break;
+    case 2:
+        volumeRampMulti<MIXTYPE, 2>(out, frameCount, in, aux, vol, volinc, vola, volainc);
+        break;
+    case 3:
+        volumeRampMulti<MIXTYPE_MONOVOL(MIXTYPE), 3>(out,
+                frameCount, in, aux, vol, volinc, vola, volainc);
+        break;
+    case 4:
+        volumeRampMulti<MIXTYPE_MONOVOL(MIXTYPE), 4>(out,
+                frameCount, in, aux, vol, volinc, vola, volainc);
+        break;
+    case 5:
+        volumeRampMulti<MIXTYPE_MONOVOL(MIXTYPE), 5>(out,
+                frameCount, in, aux, vol, volinc, vola, volainc);
+        break;
+    case 6:
+        volumeRampMulti<MIXTYPE_MONOVOL(MIXTYPE), 6>(out,
+                frameCount, in, aux, vol, volinc, vola, volainc);
+        break;
+    case 7:
+        volumeRampMulti<MIXTYPE_MONOVOL(MIXTYPE), 7>(out,
+                frameCount, in, aux, vol, volinc, vola, volainc);
+        break;
+    case 8:
+        volumeRampMulti<MIXTYPE_MONOVOL(MIXTYPE), 8>(out,
+                frameCount, in, aux, vol, volinc, vola, volainc);
+        break;
+    }
+}
+
+/* MIXTYPE     (see AudioMixerOps.h MIXTYPE_* enumeration)
+ * 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,
+        typename TO, typename TI, typename TV, typename TA, typename TAV>
+static void volumeMulti(uint32_t channels, TO* out, size_t frameCount,
+        const TI* in, TA* aux, const TV *vol, TAV vola)
+{
+    switch (channels) {
+    case 1:
+        volumeMulti<MIXTYPE, 1>(out, frameCount, in, aux, vol, vola);
+        break;
+    case 2:
+        volumeMulti<MIXTYPE, 2>(out, frameCount, in, aux, vol, vola);
+        break;
+    case 3:
+        volumeMulti<MIXTYPE_MONOVOL(MIXTYPE), 3>(out, frameCount, in, aux, vol, vola);
+        break;
+    case 4:
+        volumeMulti<MIXTYPE_MONOVOL(MIXTYPE), 4>(out, frameCount, in, aux, vol, vola);
+        break;
+    case 5:
+        volumeMulti<MIXTYPE_MONOVOL(MIXTYPE), 5>(out, frameCount, in, aux, vol, vola);
+        break;
+    case 6:
+        volumeMulti<MIXTYPE_MONOVOL(MIXTYPE), 6>(out, frameCount, in, aux, vol, vola);
+        break;
+    case 7:
+        volumeMulti<MIXTYPE_MONOVOL(MIXTYPE), 7>(out, frameCount, in, aux, vol, vola);
+        break;
+    case 8:
+        volumeMulti<MIXTYPE_MONOVOL(MIXTYPE), 8>(out, frameCount, in, aux, vol, vola);
+        break;
+    }
+}
+
+/* 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>
+void AudioMixer::volumeMix(TO *out, size_t outFrames,
+        const TI *in, TA *aux, bool ramp, AudioMixer::track_t *t)
+{
+    if (USEFLOATVOL) {
+        if (ramp) {
+            volumeRampMulti<MIXTYPE>(t->mMixerChannelCount, out, outFrames, in, aux,
+                    t->mPrevVolume, t->mVolumeInc, &t->prevAuxLevel, t->auxInc);
+            if (ADJUSTVOL) {
+                t->adjustVolumeRamp(aux != NULL, true);
+            }
+        } else {
+            volumeMulti<MIXTYPE>(t->mMixerChannelCount, out, outFrames, in, aux,
+                    t->mVolume, t->auxLevel);
+        }
+    } else {
+        if (ramp) {
+            volumeRampMulti<MIXTYPE>(t->mMixerChannelCount, out, outFrames, in, aux,
+                    t->prevVolume, t->volumeInc, &t->prevAuxLevel, t->auxInc);
+            if (ADJUSTVOL) {
+                t->adjustVolumeRamp(aux != NULL);
+            }
+        } else {
+            volumeMulti<MIXTYPE>(t->mMixerChannelCount, out, outFrames, in, aux,
+                    t->volume, t->auxLevel);
+        }
+    }
+}
+
+/* This process hook is called when there is a single track without
+ * aux buffer, volume ramp, or resampling.
+ * TODO: Update the hook selection: this can properly handle aux and ramp.
+ *
+ * MIXTYPE     (see AudioMixerOps.h MIXTYPE_* enumeration)
+ * 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, typename TO, typename TI, typename TA>
+void AudioMixer::process_NoResampleOneTrack(state_t* state)
+{
+    ALOGVV("process_NoResampleOneTrack\n");
+    // CLZ is faster than CTZ on ARM, though really not sure if true after 31 - clz.
+    const int i = 31 - __builtin_clz(state->enabledTracks);
+    ALOG_ASSERT((1 << i) == state->enabledTracks, "more than 1 track enabled");
+    track_t *t = &state->tracks[i];
+    const uint32_t channels = t->mMixerChannelCount;
+    TO* out = reinterpret_cast<TO*>(t->mainBuffer);
+    TA* aux = reinterpret_cast<TA*>(t->auxBuffer);
+    const bool ramp = t->needsRamp();
+
+    for (size_t numFrames = state->frameCount; numFrames; ) {
+        AudioBufferProvider::Buffer& b(t->buffer);
+        // get input buffer
+        b.frameCount = numFrames;
+        t->bufferProvider->getNextBuffer(&b);
+        const TI *in = reinterpret_cast<TI*>(b.raw);
+
+        // in == NULL can happen if the track was flushed just after having
+        // been enabled for mixing.
+        if (in == NULL || (((uintptr_t)in) & 3)) {
+            memset(out, 0, numFrames
+                    * channels * audio_bytes_per_sample(t->mMixerFormat));
+            ALOGE_IF((((uintptr_t)in) & 3), "process_NoResampleOneTrack: bus error: "
+                    "buffer %p track %p, channels %d, needs %#x",
+                    in, t, t->channelCount, t->needs);
+            return;
+        }
+
+        const size_t outFrames = b.frameCount;
+        volumeMix<MIXTYPE, is_same<TI, float>::value, false> (
+                out, outFrames, in, aux, ramp, t);
+
+        out += outFrames * channels;
+        if (aux != NULL) {
+            aux += channels;
+        }
+        numFrames -= b.frameCount;
+
+        // release buffer
+        t->bufferProvider->releaseBuffer(&b);
+    }
+    if (ramp) {
+        t->adjustVolumeRamp(aux != NULL, is_same<TI, float>::value);
+    }
+}
+
+/* This track hook is called to do resampling then mixing,
+ * pulling from the track's upstream AudioBufferProvider.
+ *
+ * MIXTYPE     (see AudioMixerOps.h MIXTYPE_* enumeration)
+ * 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, typename TO, typename TI, typename TA>
+void AudioMixer::track__Resample(track_t* t, TO* out, size_t outFrameCount, TO* temp, TA* aux)
+{
+    ALOGVV("track__Resample\n");
+    t->resampler->setSampleRate(t->sampleRate);
+    const bool ramp = t->needsRamp();
+    if (ramp || aux != NULL) {
+        // if ramp:        resample with unity gain to temp buffer and scale/mix in 2nd step.
+        // if aux != NULL: resample with unity gain to temp buffer then apply send level.
+
+        t->resampler->setVolume(UNITY_GAIN_FLOAT, UNITY_GAIN_FLOAT);
+        memset(temp, 0, outFrameCount * t->mMixerChannelCount * sizeof(TO));
+        t->resampler->resample((int32_t*)temp, outFrameCount, t->bufferProvider);
+
+        volumeMix<MIXTYPE, is_same<TI, float>::value, true>(
+                out, outFrameCount, temp, aux, ramp, t);
+
+    } else { // constant volume gain
+        t->resampler->setVolume(t->mVolume[0], t->mVolume[1]);
+        t->resampler->resample((int32_t*)out, outFrameCount, t->bufferProvider);
+    }
+}
+
+/* This track hook is called to mix a track, when no resampling is required.
+ * The input buffer should be present in t->in.
+ *
+ * MIXTYPE     (see AudioMixerOps.h MIXTYPE_* enumeration)
+ * 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, typename TO, typename TI, typename TA>
+void AudioMixer::track__NoResample(track_t* t, TO* out, size_t frameCount,
+        TO* temp __unused, TA* aux)
+{
+    ALOGVV("track__NoResample\n");
+    const TI *in = static_cast<const TI *>(t->in);
+
+    volumeMix<MIXTYPE, is_same<TI, float>::value, true>(
+            out, frameCount, in, aux, t->needsRamp(), t);
+
+    // MIXTYPE_MONOEXPAND reads a single input channel and expands to NCHAN output channels.
+    // MIXTYPE_MULTI reads NCHAN input channels and places to NCHAN output channels.
+    in += (MIXTYPE == MIXTYPE_MONOEXPAND) ? frameCount : frameCount * t->mMixerChannelCount;
+    t->in = in;
+}
+
+/* The Mixer engine generates either int32_t (Q4_27) or float data.
+ * We use this function to convert the engine buffers
+ * to the desired mixer output format, either int16_t (Q.15) or float.
+ */
+void AudioMixer::convertMixerFormat(void *out, audio_format_t mixerOutFormat,
+        void *in, audio_format_t mixerInFormat, size_t sampleCount)
+{
+    switch (mixerInFormat) {
+    case AUDIO_FORMAT_PCM_FLOAT:
+        switch (mixerOutFormat) {
+        case AUDIO_FORMAT_PCM_FLOAT:
+            memcpy(out, in, sampleCount * sizeof(float)); // MEMCPY. TODO optimize out
+            break;
+        case AUDIO_FORMAT_PCM_16_BIT:
+            memcpy_to_i16_from_float((int16_t*)out, (float*)in, sampleCount);
+            break;
+        default:
+            LOG_ALWAYS_FATAL("bad mixerOutFormat: %#x", mixerOutFormat);
+            break;
+        }
+        break;
+    case AUDIO_FORMAT_PCM_16_BIT:
+        switch (mixerOutFormat) {
+        case AUDIO_FORMAT_PCM_FLOAT:
+            memcpy_to_float_from_q4_27((float*)out, (int32_t*)in, sampleCount);
+            break;
+        case AUDIO_FORMAT_PCM_16_BIT:
+            // two int16_t are produced per iteration
+            ditherAndClamp((int32_t*)out, (int32_t*)in, sampleCount >> 1);
+            break;
+        default:
+            LOG_ALWAYS_FATAL("bad mixerOutFormat: %#x", mixerOutFormat);
+            break;
+        }
+        break;
+    default:
+        LOG_ALWAYS_FATAL("bad mixerInFormat: %#x", mixerInFormat);
+        break;
+    }
+}
+
+/* Returns the proper track hook to use for mixing the track into the output buffer.
+ */
+AudioMixer::hook_t AudioMixer::getTrackHook(int trackType, uint32_t channelCount,
+        audio_format_t mixerInFormat, audio_format_t mixerOutFormat __unused)
+{
+    if (!kUseNewMixer && channelCount == FCC_2 && mixerInFormat == AUDIO_FORMAT_PCM_16_BIT) {
+        switch (trackType) {
+        case TRACKTYPE_NOP:
+            return track__nop;
+        case TRACKTYPE_RESAMPLE:
+            return track__genericResample;
+        case TRACKTYPE_NORESAMPLEMONO:
+            return track__16BitsMono;
+        case TRACKTYPE_NORESAMPLE:
+            return track__16BitsStereo;
+        default:
+            LOG_ALWAYS_FATAL("bad trackType: %d", trackType);
+            break;
+        }
+    }
+    LOG_ALWAYS_FATAL_IF(channelCount > MAX_NUM_CHANNELS);
+    switch (trackType) {
+    case TRACKTYPE_NOP:
+        return track__nop;
+    case TRACKTYPE_RESAMPLE:
+        switch (mixerInFormat) {
+        case AUDIO_FORMAT_PCM_FLOAT:
+            return (AudioMixer::hook_t)
+                    track__Resample<MIXTYPE_MULTI, float /*TO*/, float /*TI*/, int32_t /*TA*/>;
+        case AUDIO_FORMAT_PCM_16_BIT:
+            return (AudioMixer::hook_t)\
+                    track__Resample<MIXTYPE_MULTI, int32_t, int16_t, int32_t>;
+        default:
+            LOG_ALWAYS_FATAL("bad mixerInFormat: %#x", mixerInFormat);
+            break;
+        }
+        break;
+    case TRACKTYPE_NORESAMPLEMONO:
+        switch (mixerInFormat) {
+        case AUDIO_FORMAT_PCM_FLOAT:
+            return (AudioMixer::hook_t)
+                    track__NoResample<MIXTYPE_MONOEXPAND, float, float, int32_t>;
+        case AUDIO_FORMAT_PCM_16_BIT:
+            return (AudioMixer::hook_t)
+                    track__NoResample<MIXTYPE_MONOEXPAND, int32_t, int16_t, int32_t>;
+        default:
+            LOG_ALWAYS_FATAL("bad mixerInFormat: %#x", mixerInFormat);
+            break;
+        }
+        break;
+    case TRACKTYPE_NORESAMPLE:
+        switch (mixerInFormat) {
+        case AUDIO_FORMAT_PCM_FLOAT:
+            return (AudioMixer::hook_t)
+                    track__NoResample<MIXTYPE_MULTI, float, float, int32_t>;
+        case AUDIO_FORMAT_PCM_16_BIT:
+            return (AudioMixer::hook_t)
+                    track__NoResample<MIXTYPE_MULTI, int32_t, int16_t, int32_t>;
+        default:
+            LOG_ALWAYS_FATAL("bad mixerInFormat: %#x", mixerInFormat);
+            break;
+        }
+        break;
+    default:
+        LOG_ALWAYS_FATAL("bad trackType: %d", trackType);
+        break;
+    }
+    return NULL;
+}
+
+/* Returns the proper process hook for mixing tracks. Currently works only for
+ * PROCESSTYPE_NORESAMPLEONETRACK, a mix involving one track, no resampling.
+ *
+ * TODO: Due to the special mixing considerations of duplicating to
+ * a stereo output track, the input track cannot be MONO.  This should be
+ * prevented by the caller.
+ */
+AudioMixer::process_hook_t AudioMixer::getProcessHook(int processType, uint32_t channelCount,
+        audio_format_t mixerInFormat, audio_format_t mixerOutFormat)
+{
+    if (processType != PROCESSTYPE_NORESAMPLEONETRACK) { // Only NORESAMPLEONETRACK
+        LOG_ALWAYS_FATAL("bad processType: %d", processType);
+        return NULL;
+    }
+    if (!kUseNewMixer && channelCount == FCC_2 && mixerInFormat == AUDIO_FORMAT_PCM_16_BIT) {
+        return process__OneTrack16BitsStereoNoResampling;
+    }
+    LOG_ALWAYS_FATAL_IF(channelCount > MAX_NUM_CHANNELS);
+    switch (mixerInFormat) {
+    case AUDIO_FORMAT_PCM_FLOAT:
+        switch (mixerOutFormat) {
+        case AUDIO_FORMAT_PCM_FLOAT:
+            return process_NoResampleOneTrack<MIXTYPE_MULTI_SAVEONLY,
+                    float /*TO*/, float /*TI*/, int32_t /*TA*/>;
+        case AUDIO_FORMAT_PCM_16_BIT:
+            return process_NoResampleOneTrack<MIXTYPE_MULTI_SAVEONLY,
+                    int16_t, float, int32_t>;
+        default:
+            LOG_ALWAYS_FATAL("bad mixerOutFormat: %#x", mixerOutFormat);
+            break;
+        }
+        break;
+    case AUDIO_FORMAT_PCM_16_BIT:
+        switch (mixerOutFormat) {
+        case AUDIO_FORMAT_PCM_FLOAT:
+            return process_NoResampleOneTrack<MIXTYPE_MULTI_SAVEONLY,
+                    float, int16_t, int32_t>;
+        case AUDIO_FORMAT_PCM_16_BIT:
+            return process_NoResampleOneTrack<MIXTYPE_MULTI_SAVEONLY,
+                    int16_t, int16_t, int32_t>;
+        default:
+            LOG_ALWAYS_FATAL("bad mixerOutFormat: %#x", mixerOutFormat);
+            break;
+        }
+        break;
+    default:
+        LOG_ALWAYS_FATAL("bad mixerInFormat: %#x", mixerInFormat);
+        break;
+    }
+    return NULL;
+}
+
+// ----------------------------------------------------------------------------
+} // namespace android