Squashed commit of the following:

commit 9128d6ffec43731d723f9b394f243d940f4c7e41
Author: Glenn Kasten <gkasten@google.com>
Date:   Tue May 13 10:38:42 2014 -0700

    Use of fast capture by normal capture

    Will only configure fast capture path if the input buffer size is less than
    10 ms and the input sample rate is same as the primary output sample rate.

    Change-Id: I4a7cdc6069d750845412c626d27e83f72a1ab397

commit 2e5e0806a5abe7499848358ef5fde5c26405000d
Author: Glenn Kasten <gkasten@google.com>
Date:   Mon Jun 2 08:29:22 2014 -0700

    Add mPrimaryOutputSampleRate

    Change-Id: I46b527fc3f2b5a5720a74b4f0b9a8f2e0d570b09

commit baf1d73467923996d1b1f2a9237260cc5697e050
Author: Andy Hung <hunga@google.com>
Date:   Fri May 30 10:42:03 2014 -0700

    Change parameter type for volume to float in AudioMixer

    Change-Id: I4da1505ce852505f86f8e5b87f60e8edceeb30e0

commit 40fe20fa9760cd03c69778c2021cf7a490d75ece
Author: Andy Hung <hunga@google.com>
Date:   Fri May 30 10:35:47 2014 -0700

    Rename UNITY_GAIN to UNITY_GAIN_INT in AudioMixer

    Change-Id: Ic040311305026f0b4c4280a5b3bef7a447ac1da3

commit 37c9a2b49f876abc5ff537a9ec036d7f0a423775
Author: Andy Hung <hunga@google.com>
Date:   Thu May 29 21:33:13 2014 -0700

    Refactor setVolumeRampVariables in AudioMixer

    Change-Id: I8fcf3101bcea292de7c65433fa578f1c9cdd0974

commit 397070eca31f121d5d3993de1bfea99aaea5d4f3
Author: Andy Hung <hunga@google.com>
Date:   Thu May 29 18:52:38 2014 -0700

    Fix floating point output from mixer

    A buffer pointer was being erroneously reset to buffer start,
    potentially causing an audio glitch. The floating point
    output mode is not enabled at this time, but will be in
    the future.

    Change-Id: If8b6414d232f064f3a2e2c5a6da889a91b27fb24

commit 2e61aa5b33b2247bbc5d4eaa0b519df9accd4bbc
Author: Andy Hung <hunga@google.com>
Date:   Fri May 23 21:22:17 2014 -0700

    Add multiple format capability to FastMixer

    Floating point data from MixerThread into FastMixer.
    Multiple output format capability from FastMixer to Sink.

    Change-Id: I0da17810ee71381a39a006c46faec71108d22c26

commit b9ea653c702a785bbd23a66c5e588d40b4192c4e
Author: Andy Hung <hunga@google.com>
Date:   Thu May 29 15:53:09 2014 -0700

    Avoid resetting BufferProviders in mixer unnecessarily

    Change-Id: Iad85c4dfd21be1dbf89dc11906106b34219376f8

commit 7f1a6d6da21c616f80cf9ba21bea11b419ec561b
Author: Andy Hung <hunga@google.com>
Date:   Tue May 27 12:32:17 2014 -0700

    Update dynamic resampler buffer fetching

    Make the criteria tight for fetching to avoid storing
    excessive frame data internal to the resampler.
    This should reduce jitter in frame delivery computation.

    Bug: 14962343
    Change-Id: I7adaf714d11c272696ccdbf218bda994c7217477

commit b5e4aac07b9a02f0c803c090058602b03ac09ebb
Author: Glenn Kasten <gkasten@google.com>
Date:   Tue May 27 12:30:54 2014 -0700

    Allow kFastTrackMultiplier to be specified per device

    Change-Id: I4eaaaf038df720cec4f5d9221d1b632970f9e3dd

commit b93cd97a52af31122df2da2cc0415cda888c8c73
Author: Andy Hung <hunga@google.com>
Date:   Fri May 23 21:13:31 2014 -0700

    Rename mixBuffer to mMixerBuffer in FastMixer

    Likewise mixBufferState becomes mMixerBufferState.
    This harmonizes with the naming in AF::MixerThread.

    Change-Id: I1255d7c07cc2c6ee925d7430925236d2bd163122

commit 8340758622b9711365a8801806cbdf934803c63f
Author: Andy Hung <hunga@google.com>
Date:   Mon May 12 16:51:41 2014 -0700

    Add multiple format capability to AudioMixer

    Change-Id: I04ac1cafd90b6ed652f8d51888ad07576678f0bc
    Signed-off-by: Andy Hung <hunga@google.com>

commit 6b695b9d094820c232a897a3fabbe83d2b7193fe
Author: Glenn Kasten <gkasten@google.com>
Date:   Thu Mar 13 14:59:31 2014 -0700

    Start adding FastCapture based on FastThread WIP

    This version supports at most one fast capture client.

    Change-Id: Idf609bfc80ae22433433d66a5232c043c65506df

commit e951ad05a2c388471d7e2806d91e7d51325a150a
Author: Glenn Kasten <gkasten@google.com>
Date:   Mon May 12 11:06:26 2014 -0700

    Move validation of frameCount from set to openRecord_l

    This move is needed because frameCount is validated on server side for
    fast tracks (as should be done for normal tracks too).

    Change-Id: I6d99e80869fd90fab373cf60ef348c01f075fbca

commit 73e76992dbba794894837c38e5472312ea829cf3
Author: Glenn Kasten <gkasten@google.com>
Date:   Tue May 13 10:41:52 2014 -0700

    Allow track buffer "allocation" to be from pipe

    Change-Id: Ib9ac170f8e8b7746b3588157a56cbee3b753a1cb

commit 60de1d7ded05c6304037d4858b401094b1d2b4d3
Author: Andy Hung <hunga@google.com>
Date:   Fri May 9 15:02:21 2014 -0700

    Add format parameter to getTrackName() and track_t

    Change-Id: Ia152a839014e235fbfb656104c15d7c1b456d02e
    Signed-off-by: Andy Hung <hunga@google.com>

Change-Id: Ied0ade8b25d23e89bb03319a7e3135c238f735b9

21 files changed, 1228 insertions, 204 deletions

diff --git a/media/libmedia/AudioRecord.cpp b/media/libmedia/AudioRecord.cpp
index 1c808d0..db61e85 100644
--- a/media/libmedia/AudioRecord.cpp
+++ b/media/libmedia/AudioRecord.cpp
@@ -203,23 +203,6 @@ status_t AudioRecord::set(
         mFrameSize = sizeof(uint8_t);
     }
 
-    // validate framecount
-    size_t minFrameCount;
-    status_t status = AudioRecord::getMinFrameCount(&minFrameCount,
-            sampleRate, format, channelMask);
-    if (status != NO_ERROR) {
-        ALOGE("getMinFrameCount() failed for sampleRate %u, format %#x, channelMask %#x; status %d",
-                sampleRate, format, channelMask, status);
-        return status;
-    }
-    ALOGV("AudioRecord::set() minFrameCount = %d", minFrameCount);
-
-    if (frameCount == 0) {
-        frameCount = minFrameCount;
-    } else if (frameCount < minFrameCount) {
-        ALOGE("frameCount %u < minFrameCount %u", frameCount, minFrameCount);
-        return BAD_VALUE;
-    }
     // mFrameCount is initialized in openRecord_l
     mReqFrameCount = frameCount;
 
@@ -242,7 +225,7 @@ status_t AudioRecord::set(
     }
 
     // create the IAudioRecord
-    status = openRecord_l(0 /*epoch*/);
+    status_t status = openRecord_l(0 /*epoch*/);
 
     if (status != NO_ERROR) {
         if (mAudioRecordThread != 0) {
@@ -464,6 +447,29 @@ status_t AudioRecord::openRecord_l(size_t epoch)
     size_t frameCount = mReqFrameCount;
 
     if (!(mFlags & AUDIO_INPUT_FLAG_FAST)) {
+        // validate framecount
+        // If fast track was not requested, this preserves
+        // the old behavior of validating on client side.
+        // FIXME Eventually the validation should be done on server side
+        // regardless of whether it's a fast or normal track.  It's debatable
+        // whether to account for the input latency to provision buffers appropriately.
+        size_t minFrameCount;
+        status = AudioRecord::getMinFrameCount(&minFrameCount,
+                mSampleRate, mFormat, mChannelMask);
+        if (status != NO_ERROR) {
+            ALOGE("getMinFrameCount() failed for sampleRate %u, format %#x, channelMask %#x; "
+                    "status %d",
+                    mSampleRate, mFormat, mChannelMask, status);
+            return status;
+        }
+
+        if (frameCount == 0) {
+            frameCount = minFrameCount;
+        } else if (frameCount < minFrameCount) {
+            ALOGE("frameCount %u < minFrameCount %u", frameCount, minFrameCount);
+            return BAD_VALUE;
+        }
+
         // Make sure that application is notified with sufficient margin before overrun
         if (mNotificationFramesAct == 0 || mNotificationFramesAct > frameCount/2) {
             mNotificationFramesAct = frameCount/2;
diff --git a/media/libmedia/AudioTrackShared.cpp b/media/libmedia/AudioTrackShared.cpp
index 219dbfd..0dbfa62 100644
--- a/media/libmedia/AudioTrackShared.cpp
+++ b/media/libmedia/AudioTrackShared.cpp
@@ -134,10 +134,17 @@ status_t ClientProxy::obtainBuffer(Buffer* buffer, const struct timespec *reques
         ssize_t filled = rear - front;
         // pipe should not be overfull
         if (!(0 <= filled && (size_t) filled <= mFrameCount)) {
-            ALOGE("Shared memory control block is corrupt (filled=%d); shutting down", filled);
-            mIsShutdown = true;
-            status = NO_INIT;
-            goto end;
+            if (mIsOut) {
+                ALOGE("Shared memory control block is corrupt (filled=%d, mFrameCount=%u); "
+                        "shutting down", filled, mFrameCount);
+                mIsShutdown = true;
+                status = NO_INIT;
+                goto end;
+            }
+            // for input, sync up on overrun
+            filled = 0;
+            cblk->u.mStreaming.mFront = rear;
+            (void) android_atomic_or(CBLK_OVERRUN, &cblk->mFlags);
         }
         // don't allow filling pipe beyond the nominal size
         size_t avail = mIsOut ? mFrameCount - filled : filled;
diff --git a/services/audioflinger/Android.mk b/services/audioflinger/Android.mk
index f7b6f64..0bdf5a3 100644
--- a/services/audioflinger/Android.mk
+++ b/services/audioflinger/Android.mk
@@ -64,6 +64,7 @@ LOCAL_32_BIT_ONLY := true
 
 LOCAL_SRC_FILES += FastMixer.cpp FastMixerState.cpp AudioWatchdog.cpp
 LOCAL_SRC_FILES += FastThread.cpp FastThreadState.cpp
+LOCAL_SRC_FILES += FastCapture.cpp FastCaptureState.cpp
 
 LOCAL_CFLAGS += -DSTATE_QUEUE_INSTANTIATIONS='"StateQueueInstantiations.cpp"'
 
diff --git a/services/audioflinger/AudioFlinger.cpp b/services/audioflinger/AudioFlinger.cpp
index 457ac3d..500e2b8 100644
--- a/services/audioflinger/AudioFlinger.cpp
+++ b/services/audioflinger/AudioFlinger.cpp
@@ -169,7 +169,8 @@ AudioFlinger::AudioFlinger()
       mBtNrecIsOff(false),
       mIsLowRamDevice(true),
       mIsDeviceTypeKnown(false),
-      mGlobalEffectEnableTime(0)
+      mGlobalEffectEnableTime(0),
+      mPrimaryOutputSampleRate(0)
 {
     getpid_cached = getpid();
     char value[PROPERTY_VALUE_MAX];
@@ -1679,6 +1680,8 @@ audio_io_handle_t AudioFlinger::openOutput(audio_module_handle_t module,
             mHardwareStatus = AUDIO_HW_SET_MODE;
             hwDevHal->set_mode(hwDevHal, mMode);
             mHardwareStatus = AUDIO_HW_IDLE;
+
+            mPrimaryOutputSampleRate = config.sample_rate;
         }
         return id;
     }
diff --git a/services/audioflinger/AudioFlinger.h b/services/audioflinger/AudioFlinger.h
index 29dc6b2..6e73a14 100644
--- a/services/audioflinger/AudioFlinger.h
+++ b/services/audioflinger/AudioFlinger.h
@@ -50,6 +50,8 @@
 
 #include <media/AudioBufferProvider.h>
 #include <media/ExtendedAudioBufferProvider.h>
+
+#include "FastCapture.h"
 #include "FastMixer.h"
 #include <media/nbaio/NBAIO.h>
 #include "AudioWatchdog.h"
@@ -691,6 +693,9 @@ private:
     nsecs_t mGlobalEffectEnableTime;  // when a global effect was last enabled
 
     sp<PatchPanel> mPatchPanel;
+
+    uint32_t    mPrimaryOutputSampleRate;   // sample rate of the primary output, or zero if none
+                                            // protected by mHardwareLock
 };
 
 #undef INCLUDING_FROM_AUDIOFLINGER_H
diff --git a/services/audioflinger/AudioMixer.cpp b/services/audioflinger/AudioMixer.cpp
index 805eaa4..d73292e 100644
--- a/services/audioflinger/AudioMixer.cpp
+++ b/services/audioflinger/AudioMixer.cpp
@@ -34,6 +34,7 @@
 #include <system/audio.h>
 
 #include <audio_utils/primitives.h>
+#include <audio_utils/format.h>
 #include <common_time/local_clock.h>
 #include <common_time/cc_helper.h>
 
@@ -88,6 +89,103 @@ void AudioMixer::DownmixerBufferProvider::releaseBuffer(AudioBufferProvider::Buf
     }
 }
 
+template <typename T>
+T min(const T& a, const T& b)
+{
+    return a < b ? a : b;
+}
+
+AudioMixer::ReformatBufferProvider::ReformatBufferProvider(int32_t channels,
+        audio_format_t inputFormat, audio_format_t outputFormat) :
+        mTrackBufferProvider(NULL),
+        mChannels(channels),
+        mInputFormat(inputFormat),
+        mOutputFormat(outputFormat),
+        mInputFrameSize(channels * audio_bytes_per_sample(inputFormat)),
+        mOutputFrameSize(channels * audio_bytes_per_sample(outputFormat)),
+        mOutputData(NULL),
+        mOutputCount(0),
+        mConsumed(0)
+{
+    ALOGV("ReformatBufferProvider(%p)(%d, %#x, %#x)", this, channels, inputFormat, outputFormat);
+    if (requiresInternalBuffers()) {
+        mOutputCount = 256;
+        (void)posix_memalign(&mOutputData, 32, mOutputCount * mOutputFrameSize);
+    }
+    mBuffer.frameCount = 0;
+}
+
+AudioMixer::ReformatBufferProvider::~ReformatBufferProvider()
+{
+    ALOGV("~ReformatBufferProvider(%p)", this);
+    if (mBuffer.frameCount != 0) {
+        mTrackBufferProvider->releaseBuffer(&mBuffer);
+    }
+    free(mOutputData);
+}
+
+status_t AudioMixer::ReformatBufferProvider::getNextBuffer(AudioBufferProvider::Buffer *pBuffer,
+        int64_t pts) {
+    //ALOGV("ReformatBufferProvider(%p)::getNextBuffer(%p (%zu), %lld)",
+    //        this, pBuffer, pBuffer->frameCount, pts);
+    if (!requiresInternalBuffers()) {
+        status_t res = mTrackBufferProvider->getNextBuffer(pBuffer, pts);
+        if (res == OK) {
+            memcpy_by_audio_format(pBuffer->raw, mOutputFormat, pBuffer->raw, mInputFormat,
+                    pBuffer->frameCount * mChannels);
+        }
+        return res;
+    }
+    if (mBuffer.frameCount == 0) {
+        mBuffer.frameCount = pBuffer->frameCount;
+        status_t res = mTrackBufferProvider->getNextBuffer(&mBuffer, pts);
+        // TODO: Track down a bug in the upstream provider
+        // LOG_ALWAYS_FATAL_IF(res == OK && mBuffer.frameCount == 0,
+        //        "ReformatBufferProvider::getNextBuffer():"
+        //        " Invalid zero framecount returned from getNextBuffer()");
+        if (res != OK || mBuffer.frameCount == 0) {
+            pBuffer->raw = NULL;
+            pBuffer->frameCount = 0;
+            return res;
+        }
+    }
+    ALOG_ASSERT(mConsumed < mBuffer.frameCount);
+    size_t count = min(mOutputCount, mBuffer.frameCount - mConsumed);
+    count = min(count, pBuffer->frameCount);
+    pBuffer->raw = mOutputData;
+    pBuffer->frameCount = count;
+    //ALOGV("reformatting %d frames from %#x to %#x, %d chan",
+    //        pBuffer->frameCount, mInputFormat, mOutputFormat, mChannels);
+    memcpy_by_audio_format(pBuffer->raw, mOutputFormat,
+            (uint8_t*)mBuffer.raw + mConsumed * mInputFrameSize, mInputFormat,
+            pBuffer->frameCount * mChannels);
+    return OK;
+}
+
+void AudioMixer::ReformatBufferProvider::releaseBuffer(AudioBufferProvider::Buffer *pBuffer) {
+    //ALOGV("ReformatBufferProvider(%p)::releaseBuffer(%p(%zu))",
+    //        this, pBuffer, pBuffer->frameCount);
+    if (!requiresInternalBuffers()) {
+        mTrackBufferProvider->releaseBuffer(pBuffer);
+        return;
+    }
+    // LOG_ALWAYS_FATAL_IF(pBuffer->frameCount == 0, "Invalid framecount");
+    mConsumed += pBuffer->frameCount; // TODO: update for efficiency to reuse existing content
+    if (mConsumed != 0 && mConsumed >= mBuffer.frameCount) {
+        mConsumed = 0;
+        mTrackBufferProvider->releaseBuffer(&mBuffer);
+        // ALOG_ASSERT(mBuffer.frameCount == 0);
+    }
+    pBuffer->raw = NULL;
+    pBuffer->frameCount = 0;
+}
+
+void AudioMixer::ReformatBufferProvider::reset() {
+    if (mBuffer.frameCount != 0) {
+        mTrackBufferProvider->releaseBuffer(&mBuffer);
+    }
+    mConsumed = 0;
+}
 
 // ----------------------------------------------------------------------------
 bool AudioMixer::sIsMultichannelCapable = false;
@@ -153,8 +251,13 @@ void AudioMixer::setLog(NBLog::Writer *log)
     mState.mLog = log;
 }
 
-int AudioMixer::getTrackName(audio_channel_mask_t channelMask, int sessionId)
+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);
@@ -162,8 +265,8 @@ int AudioMixer::getTrackName(audio_channel_mask_t channelMask, int sessionId)
         // assume default parameters for the track, except where noted below
         track_t* t = &mState.tracks[n];
         t->needs = 0;
-        t->volume[0] = UNITY_GAIN;
-        t->volume[1] = UNITY_GAIN;
+        t->volume[0] = UNITY_GAIN_INT;
+        t->volume[1] = UNITY_GAIN_INT;
         // no initialization needed
         // t->prevVolume[0]
         // t->prevVolume[1]
@@ -176,7 +279,8 @@ int AudioMixer::getTrackName(audio_channel_mask_t channelMask, int sessionId)
         // t->frameCount
         t->channelCount = audio_channel_count_from_out_mask(channelMask);
         t->enabled = false;
-        t->format = 16;
+        ALOGV_IF(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)
@@ -191,9 +295,15 @@ int AudioMixer::getTrackName(audio_channel_mask_t channelMask, int sessionId)
         // 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->mMixerFormat = AUDIO_FORMAT_PCM_16_BIT;
-
+        t->mFormat = format;
+        t->mMixerInFormat = AUDIO_FORMAT_PCM_16_BIT;
+        if (t->mFormat != t->mMixerInFormat) {
+            prepareTrackForReformat(t, n);
+        }
         status_t status = initTrackDownmix(&mState.tracks[n], n, channelMask);
         if (status != OK) {
             ALOGE("AudioMixer::getTrackName invalid channelMask (%#x)", channelMask);
@@ -237,9 +347,9 @@ void AudioMixer::unprepareTrackForDownmix(track_t* pTrack, int trackName __unuse
     if (pTrack->downmixerBufferProvider != NULL) {
         // this track had previously been configured with a downmixer, delete it
         ALOGV(" deleting old downmixer");
-        pTrack->bufferProvider = pTrack->downmixerBufferProvider->mTrackBufferProvider;
         delete pTrack->downmixerBufferProvider;
         pTrack->downmixerBufferProvider = NULL;
+        reconfigureBufferProviders(pTrack);
     } else {
         ALOGV(" nothing to do, no downmixer to delete");
     }
@@ -333,21 +443,51 @@ status_t AudioMixer::prepareTrackForDownmix(track_t* pTrack, int trackName)
     }// end of scope for local variables that are not used in goto label "noDownmixForActiveTrack"
 
     // initialization successful:
-    // - keep track of the real buffer provider in case it was set before
-    pDbp->mTrackBufferProvider = pTrack->bufferProvider;
-    // - we'll use the downmix effect integrated inside this
-    //    track's buffer provider, and we'll use it as the track's buffer provider
     pTrack->downmixerBufferProvider = pDbp;
-    pTrack->bufferProvider = pDbp;
-
+    reconfigureBufferProviders(pTrack);
     return NO_ERROR;
 
 noDownmixForActiveTrack:
     delete pDbp;
     pTrack->downmixerBufferProvider = NULL;
+    reconfigureBufferProviders(pTrack);
     return NO_INIT;
 }
 
+void AudioMixer::unprepareTrackForReformat(track_t* pTrack, int trackName __unused) {
+    ALOGV("AudioMixer::unprepareTrackForReformat(%d)", trackName);
+    if (pTrack->mReformatBufferProvider != NULL) {
+        delete pTrack->mReformatBufferProvider;
+        pTrack->mReformatBufferProvider = NULL;
+        reconfigureBufferProviders(pTrack);
+    }
+}
+
+status_t AudioMixer::prepareTrackForReformat(track_t* pTrack, int trackName)
+{
+    ALOGV("AudioMixer::prepareTrackForReformat(%d) with format %#x", trackName, pTrack->mFormat);
+    // discard the previous reformatter if there was one
+     unprepareTrackForReformat(pTrack, trackName);
+     pTrack->mReformatBufferProvider = new ReformatBufferProvider(
+             audio_channel_count_from_out_mask(pTrack->channelMask),
+             pTrack->mFormat, pTrack->mMixerInFormat);
+     reconfigureBufferProviders(pTrack);
+     return NO_ERROR;
+}
+
+void AudioMixer::reconfigureBufferProviders(track_t* pTrack)
+{
+    pTrack->bufferProvider = pTrack->mInputBufferProvider;
+    if (pTrack->mReformatBufferProvider) {
+        pTrack->mReformatBufferProvider->mTrackBufferProvider = pTrack->bufferProvider;
+        pTrack->bufferProvider = pTrack->mReformatBufferProvider;
+    }
+    if (pTrack->downmixerBufferProvider) {
+        pTrack->downmixerBufferProvider->mTrackBufferProvider = pTrack->bufferProvider;
+        pTrack->bufferProvider = pTrack->downmixerBufferProvider;
+    }
+}
+
 void AudioMixer::deleteTrackName(int name)
 {
     ALOGV("AudioMixer::deleteTrackName(%d)", name);
@@ -364,6 +504,8 @@ void AudioMixer::deleteTrackName(int name)
     track.resampler = NULL;
     // delete the downmixer
     unprepareTrackForDownmix(&mState.tracks[name], name);
+    // delete the reformatter
+    unprepareTrackForReformat(&mState.tracks[name], name);
 
     mTrackNames &= ~(1<<name);
 }
@@ -394,6 +536,44 @@ void AudioMixer::disable(int name)
     }
 }
 
+/* Sets the volume ramp variables for the AudioMixer.
+ *
+ * The volume ramp variables are used to transition between the previous
+ * volume to the target volume.  The duration of the transition is
+ * set by ramp, which is either 0 for immediate, or typically one state
+ * framecount period.
+ *
+ * @param newFloatValue new volume target in float [0.0, 1.0].
+ * @param ramp number of frames to increment over. ramp is 0 if the volume
+ * should be set immediately.
+ * @param volume reference to the U4.12 target volume, set on return.
+ * @param prevVolume reference to the U4.27 previous volume, set on return.
+ * @param volumeInc reference to the 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 newFloatValue, int32_t ramp,
+        int16_t &volume, int32_t &prevVolume, int32_t &volumeInc) {
+    int32_t newValue = newFloatValue * AudioMixer::UNITY_GAIN_INT;
+    if (newValue > AudioMixer::UNITY_GAIN_INT) {
+        newValue = AudioMixer::UNITY_GAIN_INT;
+    } else if (newValue < 0) {
+        ALOGE("negative volume %.7g", newFloatValue);
+        newValue = 0; // should never happen, but for safety check.
+    }
+    if (newValue == volume) {
+        return false;
+    }
+    if (ramp != 0) {
+        volumeInc = ((newValue - volume) << 16) / ramp;
+        prevVolume = (volumeInc == 0 ? newValue : volume) << 16;
+    } else {
+        volumeInc = 0;
+        prevVolume = newValue << 16;
+    }
+    volume = newValue;
+    return true;
+}
+
 void AudioMixer::setParameter(int name, int target, int param, void *value)
 {
     name -= TRACK0;
@@ -435,9 +615,20 @@ void AudioMixer::setParameter(int name, int target, int param, void *value)
                 invalidateState(1 << name);
             }
             break;
-        case FORMAT:
-            ALOG_ASSERT(valueInt == AUDIO_FORMAT_PCM_16_BIT);
-            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);
+                //if (track.mFormat != track.mMixerInFormat)
+                {
+                    ALOGD("Reformatting!");
+                    prepareTrackForReformat(&track, name);
+                }
+                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:
@@ -484,41 +675,23 @@ void AudioMixer::setParameter(int name, int target, int param, void *value)
         switch (param) {
         case VOLUME0:
         case VOLUME1:
-            if (track.volume[param-VOLUME0] != valueInt) {
-                ALOGV("setParameter(VOLUME, VOLUME0/1: %04x)", valueInt);
-                track.prevVolume[param-VOLUME0] = track.volume[param-VOLUME0] << 16;
-                track.volume[param-VOLUME0] = valueInt;
-                if (target == VOLUME) {
-                    track.prevVolume[param-VOLUME0] = valueInt << 16;
-                    track.volumeInc[param-VOLUME0] = 0;
-                } else {
-                    int32_t d = (valueInt<<16) - track.prevVolume[param-VOLUME0];
-                    int32_t volInc = d / int32_t(mState.frameCount);
-                    track.volumeInc[param-VOLUME0] = volInc;
-                    if (volInc == 0) {
-                        track.prevVolume[param-VOLUME0] = valueInt << 16;
-                    }
-                }
+            if (setVolumeRampVariables(*reinterpret_cast<float*>(value),
+                    target == RAMP_VOLUME ? mState.frameCount : 0,
+                    track.volume[param - VOLUME0], track.prevVolume[param - VOLUME0],
+                    track.volumeInc[param - VOLUME0])) {
+                ALOGV("setParameter(%s, VOLUME%d: %04x)",
+                        target == VOLUME ? "VOLUME" : "RAMP_VOLUME", param - VOLUME0,
+                                track.volume[param - VOLUME0]);
                 invalidateState(1 << name);
             }
             break;
         case AUXLEVEL:
             //ALOG_ASSERT(0 <= valueInt && valueInt <= MAX_GAIN_INT, "bad aux level %d", valueInt);
-            if (track.auxLevel != valueInt) {
-                ALOGV("setParameter(VOLUME, AUXLEVEL: %04x)", valueInt);
-                track.prevAuxLevel = track.auxLevel << 16;
-                track.auxLevel = valueInt;
-                if (target == VOLUME) {
-                    track.prevAuxLevel = valueInt << 16;
-                    track.auxInc = 0;
-                } else {
-                    int32_t d = (valueInt<<16) - track.prevAuxLevel;
-                    int32_t volInc = d / int32_t(mState.frameCount);
-                    track.auxInc = volInc;
-                    if (volInc == 0) {
-                        track.prevAuxLevel = valueInt << 16;
-                    }
-                }
+            if (setVolumeRampVariables(*reinterpret_cast<float*>(value),
+                    target == RAMP_VOLUME ? mState.frameCount : 0,
+                    track.auxLevel, track.prevAuxLevel, track.auxInc)) {
+                ALOGV("setParameter(%s, AUXLEVEL: %04x)",
+                        target == VOLUME ? "VOLUME" : "RAMP_VOLUME", track.auxLevel);
                 invalidateState(1 << name);
             }
             break;
@@ -550,8 +723,9 @@ bool AudioMixer::track_t::setResampler(uint32_t value, uint32_t devSampleRate)
                 } else {
                     quality = AudioResampler::DEFAULT_QUALITY;
                 }
+                const int bits = mMixerInFormat == AUDIO_FORMAT_PCM_16_BIT ? 16 : /* FLOAT */ 32;
                 resampler = AudioResampler::create(
-                        format,
+                        bits,
                         // the resampler sees the number of channels after the downmixer, if any
                         (int) (downmixerBufferProvider != NULL ? MAX_NUM_CHANNELS : channelCount),
                         devSampleRate, quality);
@@ -596,21 +770,16 @@ 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].downmixerBufferProvider != NULL) {
-        // update required?
-        if (mState.tracks[name].downmixerBufferProvider->mTrackBufferProvider != bufferProvider) {
-            ALOGV("AudioMixer::setBufferProvider(%p) for downmix", bufferProvider);
-            // setting the buffer provider for a track that gets downmixed consists in:
-            //  1/ setting the buffer provider to the "downmix / buffer provider" wrapper
-            //     so it's the one that gets called when the buffer provider is needed,
-            mState.tracks[name].bufferProvider = mState.tracks[name].downmixerBufferProvider;
-            //  2/ saving the buffer provider for the track so the wrapper can use it
-            //     when it downmixes.
-            mState.tracks[name].downmixerBufferProvider->mTrackBufferProvider = bufferProvider;
-        }
-    } else {
-        mState.tracks[name].bufferProvider = bufferProvider;
+    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].mInputBufferProvider = bufferProvider;
+    reconfigureBufferProviders(&mState.tracks[name]);
 }
 
 
@@ -769,7 +938,7 @@ void AudioMixer::track__genericResample(track_t* t, int32_t* out, size_t outFram
         // always resample with unity gain when sending to auxiliary buffer to be able
         // to apply send level after resampling
         // TODO: modify each resampler to support aux channel?
-        t->resampler->setVolume(UNITY_GAIN, UNITY_GAIN);
+        t->resampler->setVolume(UNITY_GAIN_INT, UNITY_GAIN_INT);
         memset(temp, 0, outFrameCount * MAX_NUM_CHANNELS * sizeof(int32_t));
         t->resampler->resample(temp, outFrameCount, t->bufferProvider);
         if (CC_UNLIKELY(t->volumeInc[0]|t->volumeInc[1]|t->auxInc)) {
@@ -779,7 +948,7 @@ void AudioMixer::track__genericResample(track_t* t, int32_t* out, size_t outFram
         }
     } else {
         if (CC_UNLIKELY(t->volumeInc[0]|t->volumeInc[1])) {
-            t->resampler->setVolume(UNITY_GAIN, UNITY_GAIN);
+            t->resampler->setVolume(UNITY_GAIN_INT, UNITY_GAIN_INT);
             memset(temp, 0, outFrameCount * MAX_NUM_CHANNELS * sizeof(int32_t));
             t->resampler->resample(temp, outFrameCount, t->bufferProvider);
             volumeRampStereo(t, out, outFrameCount, temp, aux);
@@ -1301,6 +1470,7 @@ void AudioMixer::process__OneTrack16BitsStereoNoResampling(state_t* state,
     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];
@@ -1314,9 +1484,10 @@ void AudioMixer::process__OneTrack16BitsStereoNoResampling(state_t* state,
 
         // in == NULL can happen if the track was flushed just after having
         // been enabled for mixing.
-        if (in == NULL || ((unsigned long)in & 3)) {
-            memset(out, 0, numFrames*MAX_NUM_CHANNELS*sizeof(int16_t));
-            ALOGE_IF(((unsigned long)in & 3), "process stereo track: input buffer alignment pb: "
+        if (in == NULL || (((uintptr_t)in) & 3)) {
+            memset(out, 0, numFrames
+                    * MAX_NUM_CHANNELS * audio_bytes_per_sample(t.mMixerFormat));
+            ALOGE_IF((((uintptr_t)in) & 3), "process stereo track: input buffer alignment pb: "
                                               "buffer %p track %d, channels %d, needs %08x",
                     in, i, t.channelCount, t.needs);
             return;
@@ -1324,8 +1495,7 @@ void AudioMixer::process__OneTrack16BitsStereoNoResampling(state_t* state,
         size_t outFrames = b.frameCount;
 
         switch (t.mMixerFormat) {
-        case AUDIO_FORMAT_PCM_FLOAT: {
-            float *fout = reinterpret_cast<float*>(out);
+        case AUDIO_FORMAT_PCM_FLOAT:
             do {
                 uint32_t rl = *reinterpret_cast<const uint32_t *>(in);
                 in += 2;
@@ -1336,9 +1506,9 @@ void AudioMixer::process__OneTrack16BitsStereoNoResampling(state_t* state,
                 // Note: In case of later int16_t sink output,
                 // conversion and clamping is done by memcpy_to_i16_from_float().
             } while (--outFrames);
-            } break;
+            break;
         case AUDIO_FORMAT_PCM_16_BIT:
-            if (CC_UNLIKELY(uint32_t(vl) > UNITY_GAIN || uint32_t(vr) > UNITY_GAIN)) {
+            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 {
diff --git a/services/audioflinger/AudioMixer.h b/services/audioflinger/AudioMixer.h
index 09e63a6..766ff60 100644
--- a/services/audioflinger/AudioMixer.h
+++ b/services/audioflinger/AudioMixer.h
@@ -31,7 +31,7 @@
 #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
+#define MAX_GAIN_INT AudioMixer::UNITY_GAIN_INT
 
 namespace android {
 
@@ -58,7 +58,8 @@ public:
     // maximum number of channels supported for the content
     static const uint32_t MAX_NUM_CHANNELS_TO_DOWNMIX = 8;
 
-    static const uint16_t UNITY_GAIN = 0x1000;
+    static const uint16_t UNITY_GAIN_INT = 0x1000;
+    static const float    UNITY_GAIN_FLOAT = 1.0f;
 
     enum { // names
 
@@ -104,7 +105,10 @@ public:
     // For all APIs with "name": TRACK0 <= name < TRACK0 + MAX_NUM_TRACKS
 
     // Allocate a track name.  Returns new track name if successful, -1 on failure.
-    int         getTrackName(audio_channel_mask_t channelMask, int sessionId);
+    // 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);
@@ -122,6 +126,13 @@ public:
 
     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 {
@@ -143,6 +154,7 @@ private:
     struct state_t;
     struct track_t;
     class DownmixerBufferProvider;
+    class ReformatBufferProvider;
 
     typedef void (*hook_t)(track_t* t, int32_t* output, size_t numOutFrames, int32_t* temp,
                            int32_t* aux);
@@ -170,7 +182,7 @@ private:
         uint16_t    frameCount;
 
         uint8_t     channelCount;   // 1 or 2, redundant with (needs & NEEDS_CHANNEL_COUNT__MASK)
-        uint8_t     format;         // always 16
+        uint8_t     unused_padding; // formerly format, was always 16
         uint16_t    enabled;        // actually bool
         audio_channel_mask_t channelMask;
 
@@ -193,14 +205,19 @@ private:
         int32_t*           auxBuffer;
 
         // 16-byte boundary
-
+        AudioBufferProvider*     mInputBufferProvider;    // 4 bytes
+        ReformatBufferProvider*  mReformatBufferProvider; // 4 bytes
         DownmixerBufferProvider* downmixerBufferProvider; // 4 bytes
 
         int32_t     sessionId;
 
-        audio_format_t mMixerFormat; // at this time: AUDIO_FORMAT_PCM_(FLOAT|16_BIT)
+        // 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.
 
-        int32_t     padding[1];
+        int32_t        mUnused[1];       // alignment padding
 
         // 16-byte boundary
 
@@ -239,6 +256,35 @@ private:
         effect_config_t    mDownmixConfig;
     };
 
+    // AudioBufferProvider wrapper that reformats track to acceptable mixer input type
+    class ReformatBufferProvider : public AudioBufferProvider {
+    public:
+        ReformatBufferProvider(int32_t channels,
+                audio_format_t inputFormat, audio_format_t outputFormat);
+        virtual ~ReformatBufferProvider();
+
+        // overrides AudioBufferProvider methods
+        virtual status_t getNextBuffer(Buffer* buffer, int64_t pts);
+        virtual void releaseBuffer(Buffer* buffer);
+
+        void reset();
+        inline bool requiresInternalBuffers() {
+            return true; //mInputFrameSize < mOutputFrameSize;
+        }
+
+        AudioBufferProvider* mTrackBufferProvider;
+        int32_t              mChannels;
+        audio_format_t       mInputFormat;
+        audio_format_t       mOutputFormat;
+        size_t               mInputFrameSize;
+        size_t               mOutputFrameSize;
+        // (only) required for reformatting to a larger size.
+        AudioBufferProvider::Buffer mBuffer;
+        void*                mOutputData;
+        size_t               mOutputCount;
+        size_t               mConsumed;
+    };
+
     // bitmask of allocated track names, where bit 0 corresponds to TRACK0 etc.
     uint32_t        mTrackNames;
 
@@ -266,6 +312,9 @@ private:
     static status_t initTrackDownmix(track_t* pTrack, int trackNum, audio_channel_mask_t mask);
     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);
diff --git a/services/audioflinger/AudioResamplerDyn.cpp b/services/audioflinger/AudioResamplerDyn.cpp
index 3abe8fd..a4446a4 100644
--- a/services/audioflinger/AudioResamplerDyn.cpp
+++ b/services/audioflinger/AudioResamplerDyn.cpp
@@ -455,12 +455,13 @@ void AudioResamplerDyn<TC, TI, TO>::resample(TO* out, size_t outFrameCount,
     const Constants& c(mConstants);
     const TC* const coefs = mConstants.mFirCoefs;
     TI* impulse = mInBuffer.getImpulse();
-    size_t inputIndex = mInputIndex;
+    size_t inputIndex = 0;
     uint32_t phaseFraction = mPhaseFraction;
     const uint32_t phaseIncrement = mPhaseIncrement;
     size_t outputIndex = 0;
     size_t outputSampleCount = outFrameCount * 2;   // stereo output
-    size_t inFrameCount = getInFrameCountRequired(outFrameCount);
+    size_t inFrameCount = getInFrameCountRequired(outFrameCount) + (phaseFraction != 0);
+    ALOG_ASSERT(0 < inFrameCount && inFrameCount < (1U << 31));
     const uint32_t phaseWrapLimit = c.mL << c.mShift;
 
     // NOTE: be very careful when modifying the code here. register
@@ -474,11 +475,13 @@ void AudioResamplerDyn<TC, TI, TO>::resample(TO* out, size_t outFrameCount,
         // buffer is empty, fetch a new one
         while (mBuffer.frameCount == 0) {
             mBuffer.frameCount = inFrameCount;
+            ALOG_ASSERT(inFrameCount > 0);
             provider->getNextBuffer(&mBuffer,
                     calculateOutputPTS(outputIndex / 2));
             if (mBuffer.raw == NULL) {
                 goto resample_exit;
             }
+            inFrameCount -= mBuffer.frameCount;
             if (phaseFraction >= phaseWrapLimit) { // read in data
                 mInBuffer.template readAdvance<CHANNELS>(
                         impulse, c.mHalfNumCoefs,
@@ -487,7 +490,7 @@ void AudioResamplerDyn<TC, TI, TO>::resample(TO* out, size_t outFrameCount,
                 while (phaseFraction >= phaseWrapLimit) {
                     inputIndex++;
                     if (inputIndex >= mBuffer.frameCount) {
-                        inputIndex -= mBuffer.frameCount;
+                        inputIndex = 0;
                         provider->releaseBuffer(&mBuffer);
                         break;
                     }
@@ -535,15 +538,22 @@ void AudioResamplerDyn<TC, TI, TO>::resample(TO* out, size_t outFrameCount,
 done:
         // often arrives here when input buffer runs out
         if (inputIndex >= frameCount) {
-            inputIndex -= frameCount;
+            inputIndex = 0;
             provider->releaseBuffer(&mBuffer);
-            // mBuffer.frameCount MUST be zero here.
+            ALOG_ASSERT(mBuffer.frameCount == 0);
         }
     }
 
 resample_exit:
+    // Release frames to avoid the count being inaccurate for pts timing.
+    // TODO: Avoid this extra check by making fetch count exact. This is tricky
+    // due to the overfetching mechanism which loads unnecessarily when
+    // mBuffer.frameCount == 0.
+    if (inputIndex) {
+        mBuffer.frameCount = inputIndex;
+        provider->releaseBuffer(&mBuffer);
+    }
     mInBuffer.setImpulse(impulse);
-    mInputIndex = inputIndex;
     mPhaseFraction = phaseFraction;
 }
 
diff --git a/services/audioflinger/FastCapture.cpp b/services/audioflinger/FastCapture.cpp
new file mode 100644
index 0000000..0c9b976
--- /dev/null
+++ b/services/audioflinger/FastCapture.cpp
@@ -0,0 +1,222 @@
+/*
+ * 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.
+ */
+
+#define LOG_TAG "FastCapture"
+//#define LOG_NDEBUG 0
+
+#define ATRACE_TAG ATRACE_TAG_AUDIO
+
+#include "Configuration.h"
+#include <linux/futex.h>
+#include <sys/syscall.h>
+#include <media/AudioBufferProvider.h>
+#include <utils/Log.h>
+#include <utils/Trace.h>
+#include "FastCapture.h"
+
+namespace android {
+
+/*static*/ const FastCaptureState FastCapture::initial;
+
+FastCapture::FastCapture() : FastThread(),
+    inputSource(NULL), inputSourceGen(0), pipeSink(NULL), pipeSinkGen(0),
+    readBuffer(NULL), readBufferState(-1), format(Format_Invalid), sampleRate(0),
+    // dummyDumpState
+    totalNativeFramesRead(0)
+{
+    previous = &initial;
+    current = &initial;
+
+    mDummyDumpState = &dummyDumpState;
+}
+
+FastCapture::~FastCapture()
+{
+}
+
+FastCaptureStateQueue* FastCapture::sq()
+{
+    return &mSQ;
+}
+
+const FastThreadState *FastCapture::poll()
+{
+    return mSQ.poll();
+}
+
+void FastCapture::setLog(NBLog::Writer *logWriter __unused)
+{
+}
+
+void FastCapture::onIdle()
+{
+    preIdle = *(const FastCaptureState *)current;
+    current = &preIdle;
+}
+
+void FastCapture::onExit()
+{
+    delete[] readBuffer;
+}
+
+bool FastCapture::isSubClassCommand(FastThreadState::Command command)
+{
+    switch ((FastCaptureState::Command) command) {
+    case FastCaptureState::READ:
+    case FastCaptureState::WRITE:
+    case FastCaptureState::READ_WRITE:
+        return true;
+    default:
+        return false;
+    }
+}
+
+void FastCapture::onStateChange()
+{
+    const FastCaptureState * const current = (const FastCaptureState *) this->current;
+    const FastCaptureState * const previous = (const FastCaptureState *) this->previous;
+    FastCaptureDumpState * const dumpState = (FastCaptureDumpState *) this->dumpState;
+    const size_t frameCount = current->mFrameCount;
+
+    bool eitherChanged = false;
+
+    // check for change in input HAL configuration
+    NBAIO_Format previousFormat = format;
+    if (current->mInputSourceGen != inputSourceGen) {
+        inputSource = current->mInputSource;
+        inputSourceGen = current->mInputSourceGen;
+        if (inputSource == NULL) {
+            format = Format_Invalid;
+            sampleRate = 0;
+        } else {
+            format = inputSource->format();
+            sampleRate = Format_sampleRate(format);
+            unsigned channelCount = Format_channelCount(format);
+            ALOG_ASSERT(channelCount == 1 || channelCount == 2);
+        }
+        dumpState->mSampleRate = sampleRate;
+        eitherChanged = true;
+    }
+
+    // check for change in pipe
+    if (current->mPipeSinkGen != pipeSinkGen) {
+        pipeSink = current->mPipeSink;
+        pipeSinkGen = current->mPipeSinkGen;
+        eitherChanged = true;
+    }
+
+    // input source and pipe sink must be compatible
+    if (eitherChanged && inputSource != NULL && pipeSink != NULL) {
+        ALOG_ASSERT(Format_isEqual(format, pipeSink->format()));
+    }
+
+    if ((!Format_isEqual(format, previousFormat)) || (frameCount != previous->mFrameCount)) {
+        // FIXME to avoid priority inversion, don't delete here
+        delete[] readBuffer;
+        readBuffer = NULL;
+        if (frameCount > 0 && sampleRate > 0) {
+            // FIXME new may block for unbounded time at internal mutex of the heap
+            //       implementation; it would be better to have normal capture thread allocate for
+            //       us to avoid blocking here and to prevent possible priority inversion
+            unsigned channelCount = Format_channelCount(format);
+            // FIXME frameSize
+            readBuffer = new short[frameCount * channelCount];
+            periodNs = (frameCount * 1000000000LL) / sampleRate;    // 1.00
+            underrunNs = (frameCount * 1750000000LL) / sampleRate;  // 1.75
+            overrunNs = (frameCount * 500000000LL) / sampleRate;    // 0.50
+            forceNs = (frameCount * 950000000LL) / sampleRate;      // 0.95
+            warmupNs = (frameCount * 500000000LL) / sampleRate;     // 0.50
+        } else {
+            periodNs = 0;
+            underrunNs = 0;
+            overrunNs = 0;
+            forceNs = 0;
+            warmupNs = 0;
+        }
+        readBufferState = -1;
+        dumpState->mFrameCount = frameCount;
+    }
+
+}
+
+void FastCapture::onWork()
+{
+    const FastCaptureState * const current = (const FastCaptureState *) this->current;
+    FastCaptureDumpState * const dumpState = (FastCaptureDumpState *) this->dumpState;
+    const FastCaptureState::Command command = this->command;
+    const size_t frameCount = current->mFrameCount;
+
+    if ((command & FastCaptureState::READ) /*&& isWarm*/) {
+        ALOG_ASSERT(inputSource != NULL);
+        ALOG_ASSERT(readBuffer != NULL);
+        dumpState->mReadSequence++;
+        ATRACE_BEGIN("read");
+        ssize_t framesRead = inputSource->read(readBuffer, frameCount,
+                AudioBufferProvider::kInvalidPTS);
+        ATRACE_END();
+        dumpState->mReadSequence++;
+        if (framesRead >= 0) {
+            LOG_ALWAYS_FATAL_IF((size_t) framesRead > frameCount);
+            totalNativeFramesRead += framesRead;
+            dumpState->mFramesRead = totalNativeFramesRead;
+            readBufferState = framesRead;
+        } else {
+            dumpState->mReadErrors++;
+            readBufferState = 0;
+        }
+        // FIXME rename to attemptedIO
+        attemptedWrite = true;
+    }
+
+    if (command & FastCaptureState::WRITE) {
+        ALOG_ASSERT(pipeSink != NULL);
+        ALOG_ASSERT(readBuffer != NULL);
+        if (readBufferState < 0) {
+            unsigned channelCount = Format_channelCount(format);
+            // FIXME frameSize
+            memset(readBuffer, 0, frameCount * channelCount * sizeof(short));
+            readBufferState = frameCount;
+        }
+        if (readBufferState > 0) {
+            ssize_t framesWritten = pipeSink->write(readBuffer, readBufferState);
+            // FIXME This supports at most one fast capture client.
+            //       To handle multiple clients this could be converted to an array,
+            //       or with a lot more work the control block could be shared by all clients.
+            audio_track_cblk_t* cblk = current->mCblk;
+            if (cblk != NULL && framesWritten > 0) {
+                int32_t rear = cblk->u.mStreaming.mRear;
+                android_atomic_release_store(framesWritten + rear, &cblk->u.mStreaming.mRear);
+                cblk->mServer += framesWritten;
+                int32_t old = android_atomic_or(CBLK_FUTEX_WAKE, &cblk->mFutex);
+                if (!(old & CBLK_FUTEX_WAKE)) {
+                    // client is never in server process, so don't use FUTEX_WAKE_PRIVATE
+                    (void) syscall(__NR_futex, &cblk->mFutex, FUTEX_WAKE, 1);
+                }
+            }
+        }
+    }
+}
+
+FastCaptureDumpState::FastCaptureDumpState() : FastThreadDumpState(),
+    mReadSequence(0), mFramesRead(0), mReadErrors(0), mSampleRate(0), mFrameCount(0)
+{
+}
+
+FastCaptureDumpState::~FastCaptureDumpState()
+{
+}
+
+}   // namespace android
diff --git a/services/audioflinger/FastCapture.h b/services/audioflinger/FastCapture.h
new file mode 100644
index 0000000..e535b9d
--- /dev/null
+++ b/services/audioflinger/FastCapture.h
@@ -0,0 +1,78 @@
+/*
+ * 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_FAST_CAPTURE_H
+#define ANDROID_AUDIO_FAST_CAPTURE_H
+
+#include "FastThread.h"
+#include "StateQueue.h"
+#include "FastCaptureState.h"
+
+namespace android {
+
+typedef StateQueue<FastCaptureState> FastCaptureStateQueue;
+
+struct FastCaptureDumpState : FastThreadDumpState {
+    FastCaptureDumpState();
+    /*virtual*/ ~FastCaptureDumpState();
+
+    // FIXME by renaming, could pull up many of these to FastThreadDumpState
+    uint32_t mReadSequence;     // incremented before and after each read()
+    uint32_t mFramesRead;       // total number of frames read successfully
+    uint32_t mReadErrors;       // total number of read() errors
+    uint32_t mSampleRate;
+    size_t   mFrameCount;
+};
+
+class FastCapture : public FastThread {
+
+public:
+            FastCapture();
+    virtual ~FastCapture();
+
+            FastCaptureStateQueue*  sq();
+
+private:
+            FastCaptureStateQueue   mSQ;
+
+    // callouts
+    virtual const FastThreadState *poll();
+    virtual void setLog(NBLog::Writer *logWriter);
+    virtual void onIdle();
+    virtual void onExit();
+    virtual bool isSubClassCommand(FastThreadState::Command command);
+    virtual void onStateChange();
+    virtual void onWork();
+
+    static const FastCaptureState initial;
+    FastCaptureState preIdle; // copy of state before we went into idle
+    // FIXME by renaming, could pull up many of these to FastThread
+    NBAIO_Source *inputSource;
+    int inputSourceGen;
+    NBAIO_Sink *pipeSink;
+    int pipeSinkGen;
+    short *readBuffer;
+    ssize_t readBufferState;    // number of initialized frames in readBuffer, or -1 to clear
+    NBAIO_Format format;
+    unsigned sampleRate;
+    FastCaptureDumpState dummyDumpState;
+    uint32_t totalNativeFramesRead; // copied to dumpState->mFramesRead
+
+};  // class FastCapture
+
+}   // namespace android
+
+#endif  // ANDROID_AUDIO_FAST_CAPTURE_H
diff --git a/services/audioflinger/FastCaptureState.cpp b/services/audioflinger/FastCaptureState.cpp
new file mode 100644
index 0000000..1d029b7
--- /dev/null
+++ b/services/audioflinger/FastCaptureState.cpp
@@ -0,0 +1,30 @@
+/*
+ * 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 "FastCaptureState.h"
+
+namespace android {
+
+FastCaptureState::FastCaptureState() : FastThreadState(),
+    mInputSource(NULL), mInputSourceGen(0), mPipeSink(NULL), mPipeSinkGen(0), mFrameCount(0)
+{
+}
+
+FastCaptureState::~FastCaptureState()
+{
+}
+
+}   // android
diff --git a/services/audioflinger/FastCaptureState.h b/services/audioflinger/FastCaptureState.h
new file mode 100644
index 0000000..29c865a
--- /dev/null
+++ b/services/audioflinger/FastCaptureState.h
@@ -0,0 +1,51 @@
+/*
+ * 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_FAST_CAPTURE_STATE_H
+#define ANDROID_AUDIO_FAST_CAPTURE_STATE_H
+
+#include <media/nbaio/NBAIO.h>
+#include "FastThreadState.h"
+#include <private/media/AudioTrackShared.h>
+
+namespace android {
+
+// Represent a single state of the fast capture
+struct FastCaptureState : FastThreadState {
+                FastCaptureState();
+    /*virtual*/ ~FastCaptureState();
+
+    // all pointer fields use raw pointers; objects are owned and ref-counted by RecordThread
+    NBAIO_Source    *mInputSource;      // HAL input device, must already be negotiated
+    // FIXME by renaming, could pull up these fields to FastThreadState
+    int             mInputSourceGen;    // increment when mInputSource is assigned
+    NBAIO_Sink      *mPipeSink;         // after reading from input source, write to this pipe sink
+    int             mPipeSinkGen;       // increment when mPipeSink is assigned
+    size_t          mFrameCount;        // number of frames per fast capture buffer
+    audio_track_cblk_t  *mCblk;         // control block for the single fast client, or NULL
+
+    // Extends FastThreadState::Command
+    static const Command
+        // The following commands also process configuration changes, and can be "or"ed:
+        READ = 0x8,             // read from input source
+        WRITE = 0x10,           // write to pipe sink
+        READ_WRITE = 0x18;      // read from input source and write to pipe sink
+
+};  // struct FastCaptureState
+
+}   // namespace android
+
+#endif  // ANDROID_AUDIO_FAST_CAPTURE_STATE_H
diff --git a/services/audioflinger/FastMixer.cpp b/services/audioflinger/FastMixer.cpp
index 1caed11..c9a3f10 100644
--- a/services/audioflinger/FastMixer.cpp
+++ b/services/audioflinger/FastMixer.cpp
@@ -37,6 +37,7 @@
 #include <cpustats/ThreadCpuUsage.h>
 #endif
 #endif
+#include <audio_utils/format.h>
 #include "AudioMixer.h"
 #include "FastMixer.h"
 
@@ -53,8 +54,12 @@ FastMixer::FastMixer() : FastThread(),
     outputSink(NULL),
     outputSinkGen(0),
     mixer(NULL),
-    mixBuffer(NULL),
-    mixBufferState(UNDEFINED),
+    mSinkBuffer(NULL),
+    mSinkBufferSize(0),
+    mMixerBuffer(NULL),
+    mMixerBufferSize(0),
+    mMixerBufferFormat(AUDIO_FORMAT_PCM_16_BIT),
+    mMixerBufferState(UNDEFINED),
     format(Format_Invalid),
     sampleRate(0),
     fastTracksGen(0),
@@ -109,7 +114,8 @@ void FastMixer::onIdle()
 void FastMixer::onExit()
 {
     delete mixer;
-    delete[] mixBuffer;
+    free(mMixerBuffer);
+    free(mSinkBuffer);
 }
 
 bool FastMixer::isSubClassCommand(FastThreadState::Command command)
@@ -155,14 +161,23 @@ void FastMixer::onStateChange()
         // FIXME to avoid priority inversion, don't delete here
         delete mixer;
         mixer = NULL;
-        delete[] mixBuffer;
-        mixBuffer = NULL;
+        free(mMixerBuffer);
+        mMixerBuffer = NULL;
+        free(mSinkBuffer);
+        mSinkBuffer = NULL;
         if (frameCount > 0 && sampleRate > 0) {
             // FIXME new may block for unbounded time at internal mutex of the heap
             //       implementation; it would be better to have normal mixer allocate for us
             //       to avoid blocking here and to prevent possible priority inversion
             mixer = new AudioMixer(frameCount, sampleRate, FastMixerState::kMaxFastTracks);
-            mixBuffer = new short[frameCount * FCC_2];
+            const size_t mixerFrameSize = FCC_2 * audio_bytes_per_sample(mMixerBufferFormat);
+            mMixerBufferSize = mixerFrameSize * frameCount;
+            (void)posix_memalign(&mMixerBuffer, 32, mMixerBufferSize);
+            const size_t sinkFrameSize = FCC_2 * audio_bytes_per_sample(format.mFormat);
+            if (sinkFrameSize > mixerFrameSize) { // need a sink buffer
+                mSinkBufferSize = sinkFrameSize * frameCount;
+                (void)posix_memalign(&mSinkBuffer, 32, mSinkBufferSize);
+            }
             periodNs = (frameCount * 1000000000LL) / sampleRate;    // 1.00
             underrunNs = (frameCount * 1750000000LL) / sampleRate;  // 1.75
             overrunNs = (frameCount * 500000000LL) / sampleRate;    // 0.50
@@ -175,7 +190,7 @@ void FastMixer::onStateChange()
             forceNs = 0;
             warmupNs = 0;
         }
-        mixBufferState = UNDEFINED;
+        mMixerBufferState = UNDEFINED;
 #if !LOG_NDEBUG
         for (unsigned i = 0; i < FastMixerState::kMaxFastTracks; ++i) {
             fastTrackNames[i] = -1;
@@ -193,7 +208,7 @@ void FastMixer::onStateChange()
     const unsigned currentTrackMask = current->mTrackMask;
     dumpState->mTrackMask = currentTrackMask;
     if (current->mFastTracksGen != fastTracksGen) {
-        ALOG_ASSERT(mixBuffer != NULL);
+        ALOG_ASSERT(mMixerBuffer != NULL);
         int name;
 
         // process removed tracks first to avoid running out of track names
@@ -224,13 +239,20 @@ void FastMixer::onStateChange()
             AudioBufferProvider *bufferProvider = fastTrack->mBufferProvider;
             ALOG_ASSERT(bufferProvider != NULL && fastTrackNames[i] == -1);
             if (mixer != NULL) {
-                name = mixer->getTrackName(fastTrack->mChannelMask, AUDIO_SESSION_OUTPUT_MIX);
+                name = mixer->getTrackName(fastTrack->mChannelMask,
+                        fastTrack->mFormat, AUDIO_SESSION_OUTPUT_MIX);
                 ALOG_ASSERT(name >= 0);
                 fastTrackNames[i] = name;
                 mixer->setBufferProvider(name, bufferProvider);
                 mixer->setParameter(name, AudioMixer::TRACK, AudioMixer::MAIN_BUFFER,
-                        (void *) mixBuffer);
+                        (void *) mMixerBuffer);
                 // newly allocated track names default to full scale volume
+                mixer->setParameter(
+                        name,
+                        AudioMixer::TRACK,
+                        AudioMixer::MIXER_FORMAT, (void *)mMixerBufferFormat);
+                mixer->setParameter(name, AudioMixer::TRACK, AudioMixer::FORMAT,
+                        (void *)(uintptr_t)fastTrack->mFormat);
                 mixer->enable(name);
             }
             generations[i] = fastTrack->mGeneration;
@@ -252,13 +274,18 @@ void FastMixer::onStateChange()
                     ALOG_ASSERT(name >= 0);
                     mixer->setBufferProvider(name, bufferProvider);
                     if (fastTrack->mVolumeProvider == NULL) {
-                        mixer->setParameter(name, AudioMixer::VOLUME, AudioMixer::VOLUME0,
-                                (void *) MAX_GAIN_INT);
-                        mixer->setParameter(name, AudioMixer::VOLUME, AudioMixer::VOLUME1,
-                                (void *) MAX_GAIN_INT);
+                        float f = AudioMixer::UNITY_GAIN_FLOAT;
+                        mixer->setParameter(name, AudioMixer::VOLUME, AudioMixer::VOLUME0, &f);
+                        mixer->setParameter(name, AudioMixer::VOLUME, AudioMixer::VOLUME1, &f);
                     }
                     mixer->setParameter(name, AudioMixer::RESAMPLE,
                             AudioMixer::REMOVE, NULL);
+                    mixer->setParameter(
+                            name,
+                            AudioMixer::TRACK,
+                            AudioMixer::MIXER_FORMAT, (void *)mMixerBufferFormat);
+                    mixer->setParameter(name, AudioMixer::TRACK, AudioMixer::FORMAT,
+                            (void *)(uintptr_t)fastTrack->mFormat);
                     mixer->setParameter(name, AudioMixer::TRACK, AudioMixer::CHANNEL_MASK,
                             (void *)(uintptr_t) fastTrack->mChannelMask);
                     // already enabled
@@ -281,7 +308,7 @@ void FastMixer::onWork()
     const size_t frameCount = current->mFrameCount;
 
     if ((command & FastMixerState::MIX) && (mixer != NULL) && isWarm) {
-        ALOG_ASSERT(mixBuffer != NULL);
+        ALOG_ASSERT(mMixerBuffer != NULL);
         // for each track, update volume and check for underrun
         unsigned currentTrackMask = current->mTrackMask;
         while (currentTrackMask != 0) {
@@ -309,12 +336,11 @@ void FastMixer::onWork()
             ALOG_ASSERT(name >= 0);
             if (fastTrack->mVolumeProvider != NULL) {
                 gain_minifloat_packed_t vlr = fastTrack->mVolumeProvider->getVolumeLR();
-                mixer->setParameter(name, AudioMixer::VOLUME, AudioMixer::VOLUME0,
-                        (void *) (uintptr_t)
-                            (float_from_gain(gain_minifloat_unpack_left(vlr)) * MAX_GAIN_INT));
-                mixer->setParameter(name, AudioMixer::VOLUME, AudioMixer::VOLUME1,
-                        (void *) (uintptr_t)
-                            (float_from_gain(gain_minifloat_unpack_right(vlr)) * MAX_GAIN_INT));
+                float vlf = float_from_gain(gain_minifloat_unpack_left(vlr));
+                float vrf = float_from_gain(gain_minifloat_unpack_right(vlr));
+
+                mixer->setParameter(name, AudioMixer::VOLUME, AudioMixer::VOLUME0, &vlf);
+                mixer->setParameter(name, AudioMixer::VOLUME, AudioMixer::VOLUME1, &vrf);
             }
             // FIXME The current implementation of framesReady() for fast tracks
             // takes a tryLock, which can block
@@ -358,26 +384,31 @@ void FastMixer::onWork()
 
         // process() is CPU-bound
         mixer->process(pts);
-        mixBufferState = MIXED;
-    } else if (mixBufferState == MIXED) {
-        mixBufferState = UNDEFINED;
+        mMixerBufferState = MIXED;
+    } else if (mMixerBufferState == MIXED) {
+        mMixerBufferState = UNDEFINED;
     }
     //bool didFullWrite = false;    // dumpsys could display a count of partial writes
-    if ((command & FastMixerState::WRITE) && (outputSink != NULL) && (mixBuffer != NULL)) {
-        if (mixBufferState == UNDEFINED) {
-            memset(mixBuffer, 0, frameCount * FCC_2 * sizeof(short));
-            mixBufferState = ZEROED;
+    if ((command & FastMixerState::WRITE) && (outputSink != NULL) && (mMixerBuffer != NULL)) {
+        if (mMixerBufferState == UNDEFINED) {
+            memset(mMixerBuffer, 0, mMixerBufferSize);
+            mMixerBufferState = ZEROED;
+        }
+        void *buffer = mSinkBuffer != NULL ? mSinkBuffer : mMixerBuffer;
+        if (format.mFormat != mMixerBufferFormat) { // sink format not the same as mixer format
+            memcpy_by_audio_format(buffer, format.mFormat, mMixerBuffer, mMixerBufferFormat,
+                    frameCount * Format_channelCount(format));
         }
         // if non-NULL, then duplicate write() to this non-blocking sink
         NBAIO_Sink* teeSink;
         if ((teeSink = current->mTeeSink) != NULL) {
-            (void) teeSink->write(mixBuffer, frameCount);
+            (void) teeSink->write(mMixerBuffer, frameCount);
         }
         // FIXME write() is non-blocking and lock-free for a properly implemented NBAIO sink,
         //       but this code should be modified to handle both non-blocking and blocking sinks
         dumpState->mWriteSequence++;
         ATRACE_BEGIN("write");
-        ssize_t framesWritten = outputSink->write(mixBuffer, frameCount);
+        ssize_t framesWritten = outputSink->write(buffer, frameCount);
         ATRACE_END();
         dumpState->mWriteSequence++;
         if (framesWritten >= 0) {
diff --git a/services/audioflinger/FastMixer.h b/services/audioflinger/FastMixer.h
index db89ef4..4671670 100644
--- a/services/audioflinger/FastMixer.h
+++ b/services/audioflinger/FastMixer.h
@@ -61,8 +61,16 @@ private:
     NBAIO_Sink *outputSink;
     int outputSinkGen;
     AudioMixer* mixer;
-    short *mixBuffer;
-    enum {UNDEFINED, MIXED, ZEROED} mixBufferState;
+
+    // mSinkBuffer audio format is stored in format.mFormat.
+    void* mSinkBuffer;                  // used for mixer output format translation
+                                        // if sink format is different than mixer output.
+    size_t mSinkBufferSize;
+    void* mMixerBuffer;                 // mixer output buffer.
+    size_t mMixerBufferSize;
+    audio_format_t mMixerBufferFormat;  // mixer output format: AUDIO_FORMAT_PCM_(16_BIT|FLOAT).
+
+    enum {UNDEFINED, MIXED, ZEROED} mMixerBufferState;
     NBAIO_Format format;
     unsigned sampleRate;
     int fastTracksGen;
diff --git a/services/audioflinger/FastMixerState.cpp b/services/audioflinger/FastMixerState.cpp
index 8e6d0d4..3aa8dad 100644
--- a/services/audioflinger/FastMixerState.cpp
+++ b/services/audioflinger/FastMixerState.cpp
@@ -20,7 +20,7 @@ namespace android {
 
 FastTrack::FastTrack() :
     mBufferProvider(NULL), mVolumeProvider(NULL),
-    mChannelMask(AUDIO_CHANNEL_OUT_STEREO), mGeneration(0)
+    mChannelMask(AUDIO_CHANNEL_OUT_STEREO), mFormat(AUDIO_FORMAT_INVALID), mGeneration(0)
 {
 }
 
diff --git a/services/audioflinger/FastMixerState.h b/services/audioflinger/FastMixerState.h
index e388fb3..661c9ca 100644
--- a/services/audioflinger/FastMixerState.h
+++ b/services/audioflinger/FastMixerState.h
@@ -45,6 +45,7 @@ struct FastTrack {
     ExtendedAudioBufferProvider* mBufferProvider; // must be NULL if inactive, or non-NULL if active
     VolumeProvider*         mVolumeProvider; // optional; if NULL then full-scale
     audio_channel_mask_t    mChannelMask;    // AUDIO_CHANNEL_OUT_MONO or AUDIO_CHANNEL_OUT_STEREO
+    audio_format_t          mFormat;         // track format
     int                     mGeneration;     // increment when any field is assigned
 };
 
diff --git a/services/audioflinger/StateQueueInstantiations.cpp b/services/audioflinger/StateQueueInstantiations.cpp
index 0d5cd0c..6f4505e 100644
--- a/services/audioflinger/StateQueueInstantiations.cpp
+++ b/services/audioflinger/StateQueueInstantiations.cpp
@@ -16,12 +16,14 @@
 
 #include "Configuration.h"
 #include "FastMixerState.h"
+#include "FastCaptureState.h"
 #include "StateQueue.h"
 
 // FIXME hack for gcc
 
 namespace android {
 
-template class StateQueue<FastMixerState>;  // typedef FastMixerStateQueue
+template class StateQueue<FastMixerState>;      // typedef FastMixerStateQueue
+template class StateQueue<FastCaptureState>;    // typedef FastCaptureStateQueue
 
 }
diff --git a/services/audioflinger/Threads.cpp b/services/audioflinger/Threads.cpp
index 4972c7a..d6333be 100644
--- a/services/audioflinger/Threads.cpp
+++ b/services/audioflinger/Threads.cpp
@@ -38,6 +38,7 @@
 #include <audio_utils/minifloat.h>
 
 // NBAIO implementations
+#include <media/nbaio/AudioStreamInSource.h>
 #include <media/nbaio/AudioStreamOutSink.h>
 #include <media/nbaio/MonoPipe.h>
 #include <media/nbaio/MonoPipeReader.h>
@@ -53,6 +54,7 @@
 #include "AudioFlinger.h"
 #include "AudioMixer.h"
 #include "FastMixer.h"
+#include "FastCapture.h"
 #include "ServiceUtilities.h"
 #include "SchedulingPolicyService.h"
 
@@ -131,9 +133,17 @@ static const enum {
     //  up large writes into smaller ones, and the wrapper would need to deal with scheduler.
 } kUseFastMixer = FastMixer_Static;
 
+// Whether to use fast capture
+static const enum {
+    FastCapture_Never,  // never initialize or use: for debugging only
+    FastCapture_Always, // always initialize and use, even if not needed: for debugging only
+    FastCapture_Static, // initialize if needed, then use all the time if initialized
+} kUseFastCapture = FastCapture_Static;
+
 // Priorities for requestPriority
 static const int kPriorityAudioApp = 2;
 static const int kPriorityFastMixer = 3;
+static const int kPriorityFastCapture = 3;
 
 // IAudioFlinger::createTrack() reports back to client the total size of shared memory area
 // for the track.  The client then sub-divides this into smaller buffers for its use.
@@ -142,8 +152,17 @@ static const int kPriorityFastMixer = 3;
 // FIXME It would be better for client to tell AudioFlinger the value of N,
 // so AudioFlinger could allocate the right amount of memory.
 // See the client's minBufCount and mNotificationFramesAct calculations for details.
+
+// This is the default value, if not specified by property.
 static const int kFastTrackMultiplier = 2;
 
+// The minimum and maximum allowed values
+static const int kFastTrackMultiplierMin = 1;
+static const int kFastTrackMultiplierMax = 2;
+
+// The actual value to use, which can be specified per-device via property af.fast_track_multiplier.
+static int sFastTrackMultiplier = kFastTrackMultiplier;
+
 // See Thread::readOnlyHeap().
 // Initially this heap is used to allocate client buffers for "fast" AudioRecord.
 // Eventually it will be the single buffer that FastCapture writes into via HAL read(),
@@ -152,6 +171,22 @@ static const size_t kRecordThreadReadOnlyHeapSize = 0x1000;
 
 // ----------------------------------------------------------------------------
 
+static pthread_once_t sFastTrackMultiplierOnce = PTHREAD_ONCE_INIT;
+
+static void sFastTrackMultiplierInit()
+{
+    char value[PROPERTY_VALUE_MAX];
+    if (property_get("af.fast_track_multiplier", value, NULL) > 0) {
+        char *endptr;
+        unsigned long ul = strtoul(value, &endptr, 0);
+        if (*endptr == '\0' && kFastTrackMultiplierMin <= ul && ul <= kFastTrackMultiplierMax) {
+            sFastTrackMultiplier = (int) ul;
+        }
+    }
+}
+
+// ----------------------------------------------------------------------------
+
 #ifdef ADD_BATTERY_DATA
 // To collect the amplifier usage
 static void addBatteryData(uint32_t params) {
@@ -1356,7 +1391,12 @@ sp<AudioFlinger::PlaybackThread::Track> AudioFlinger::PlaybackThread::createTrac
         ) {
         // if frameCount not specified, then it defaults to fast mixer (HAL) frame count
         if (frameCount == 0) {
-            frameCount = mFrameCount * kFastTrackMultiplier;
+            // read the fast track multiplier property the first time it is needed
+            int ok = pthread_once(&sFastTrackMultiplierOnce, sFastTrackMultiplierInit);
+            if (ok != 0) {
+                ALOGE("%s pthread_once failed: %d", __func__, ok);
+            }
+            frameCount = mFrameCount * sFastTrackMultiplier;
         }
         ALOGV("AUDIO_OUTPUT_FLAG_FAST accepted: frameCount=%d mFrameCount=%d",
                 frameCount, mFrameCount);
@@ -2715,9 +2755,27 @@ AudioFlinger::MixerThread::MixerThread(const sp<AudioFlinger>& audioFlinger, Aud
         break;
     }
     if (initFastMixer) {
+        audio_format_t fastMixerFormat;
+        if (mMixerBufferEnabled && mEffectBufferEnabled) {
+            fastMixerFormat = AUDIO_FORMAT_PCM_FLOAT;
+        } else {
+            fastMixerFormat = AUDIO_FORMAT_PCM_16_BIT;
+        }
+        if (mFormat != fastMixerFormat) {
+            // change our Sink format to accept our intermediate precision
+            mFormat = fastMixerFormat;
+            free(mSinkBuffer);
+            mFrameSize = mChannelCount * audio_bytes_per_sample(mFormat);
+            const size_t sinkBufferSize = mNormalFrameCount * mFrameSize;
+            (void)posix_memalign(&mSinkBuffer, 32, sinkBufferSize);
+        }
 
         // create a MonoPipe to connect our submix to FastMixer
         NBAIO_Format format = mOutputSink->format();
+        // adjust format to match that of the Fast Mixer
+        format.mFormat = fastMixerFormat;
+        format.mFrameSize = audio_bytes_per_sample(format.mFormat) * format.mChannelCount;
+
         // This pipe depth compensates for scheduling latency of the normal mixer thread.
         // When it wakes up after a maximum latency, it runs a few cycles quickly before
         // finally blocking.  Note the pipe implementation rounds up the request to a power of 2.
@@ -2758,6 +2816,8 @@ AudioFlinger::MixerThread::MixerThread(const sp<AudioFlinger>& audioFlinger, Aud
         // wrap the source side of the MonoPipe to make it an AudioBufferProvider
         fastTrack->mBufferProvider = new SourceAudioBufferProvider(new MonoPipeReader(monoPipe));
         fastTrack->mVolumeProvider = NULL;
+        fastTrack->mChannelMask = mChannelMask; // mPipeSink channel mask for audio to FastMixer
+        fastTrack->mFormat = mFormat; // mPipeSink format for audio to FastMixer
         fastTrack->mGeneration++;
         state->mFastTracksGen++;
         state->mTrackMask = 1;
@@ -3210,6 +3270,7 @@ AudioFlinger::PlaybackThread::mixer_state AudioFlinger::MixerThread::prepareTrac
                     fastTrack->mBufferProvider = eabp;
                     fastTrack->mVolumeProvider = vp;
                     fastTrack->mChannelMask = track->mChannelMask;
+                    fastTrack->mFormat = track->mFormat;
                     fastTrack->mGeneration++;
                     state->mTrackMask |= 1 << j;
                     didModify = true;
@@ -3319,9 +3380,11 @@ AudioFlinger::PlaybackThread::mixer_state AudioFlinger::MixerThread::prepareTrac
             }
 
             // compute volume for this track
-            uint32_t vl, vr, va;
+            uint32_t vl, vr;       // in U8.24 integer format
+            float vlf, vrf, vaf;   // in [0.0, 1.0] float format
             if (track->isPausing() || mStreamTypes[track->streamType()].mute) {
-                vl = vr = va = 0;
+                vl = vr = 0;
+                vlf = vrf = vaf = 0.;
                 if (track->isPausing()) {
                     track->setPaused();
                 }
@@ -3332,8 +3395,8 @@ AudioFlinger::PlaybackThread::mixer_state AudioFlinger::MixerThread::prepareTrac
                 float v = masterVolume * typeVolume;
                 AudioTrackServerProxy *proxy = track->mAudioTrackServerProxy;
                 gain_minifloat_packed_t vlr = proxy->getVolumeLR();
-                float vlf = float_from_gain(gain_minifloat_unpack_left(vlr));
-                float vrf = float_from_gain(gain_minifloat_unpack_right(vlr));
+                vlf = float_from_gain(gain_minifloat_unpack_left(vlr));
+                vrf = float_from_gain(gain_minifloat_unpack_right(vlr));
                 // track volumes come from shared memory, so can't be trusted and must be clamped
                 if (vlf > GAIN_FLOAT_UNITY) {
                     ALOGV("Track left volume out of range: %.3g", vlf);
@@ -3344,20 +3407,22 @@ AudioFlinger::PlaybackThread::mixer_state AudioFlinger::MixerThread::prepareTrac
                     vrf = GAIN_FLOAT_UNITY;
                 }
                 // now apply the master volume and stream type volume
-                // FIXME we're losing the wonderful dynamic range in the minifloat representation
-                float v8_24 = v * (MAX_GAIN_INT * MAX_GAIN_INT);
-                vl = (uint32_t) (v8_24 * vlf);
-                vr = (uint32_t) (v8_24 * vrf);
+                vlf *= v;
+                vrf *= v;
                 // assuming master volume and stream type volume each go up to 1.0,
-                // vl and vr are now in 8.24 format
-
+                // then derive vl and vr as U8.24 versions for the effect chain
+                const float scaleto8_24 = MAX_GAIN_INT * MAX_GAIN_INT;
+                vl = (uint32_t) (scaleto8_24 * vlf);
+                vr = (uint32_t) (scaleto8_24 * vrf);
+                // vl and vr are now in U8.24 format
                 uint16_t sendLevel = proxy->getSendLevel_U4_12();
                 // send level comes from shared memory and so may be corrupt
                 if (sendLevel > MAX_GAIN_INT) {
                     ALOGV("Track send level out of range: %04X", sendLevel);
                     sendLevel = MAX_GAIN_INT;
                 }
-                va = (uint32_t)(v * sendLevel);
+                // vaf is represented as [0.0, 1.0] float by rescaling sendLevel
+                vaf = v * sendLevel * (1. / MAX_GAIN_INT);
             }
 
             // Delegate volume control to effect in track effect chain if needed
@@ -3374,29 +3439,13 @@ AudioFlinger::PlaybackThread::mixer_state AudioFlinger::MixerThread::prepareTrac
                 track->mHasVolumeController = false;
             }
 
-            // FIXME Use float
-            // Convert volumes from 8.24 to 4.12 format
-            // This additional clamping is needed in case chain->setVolume_l() overshot
-            vl = (vl + (1 << 11)) >> 12;
-            if (vl > MAX_GAIN_INT) {
-                vl = MAX_GAIN_INT;
-            }
-            vr = (vr + (1 << 11)) >> 12;
-            if (vr > MAX_GAIN_INT) {
-                vr = MAX_GAIN_INT;
-            }
-
-            if (va > MAX_GAIN_INT) {
-                va = MAX_GAIN_INT;   // va is uint32_t, so no need to check for -
-            }
-
             // XXX: these things DON'T need to be done each time
             mAudioMixer->setBufferProvider(name, track);
             mAudioMixer->enable(name);
 
-            mAudioMixer->setParameter(name, param, AudioMixer::VOLUME0, (void *)(uintptr_t)vl);
-            mAudioMixer->setParameter(name, param, AudioMixer::VOLUME1, (void *)(uintptr_t)vr);
-            mAudioMixer->setParameter(name, param, AudioMixer::AUXLEVEL, (void *)(uintptr_t)va);
+            mAudioMixer->setParameter(name, param, AudioMixer::VOLUME0, &vlf);
+            mAudioMixer->setParameter(name, param, AudioMixer::VOLUME1, &vrf);
+            mAudioMixer->setParameter(name, param, AudioMixer::AUXLEVEL, &vaf);
             mAudioMixer->setParameter(
                 name,
                 AudioMixer::TRACK,
@@ -3601,9 +3650,10 @@ track_is_ready: ;
 }
 
 // getTrackName_l() must be called with ThreadBase::mLock held
-int AudioFlinger::MixerThread::getTrackName_l(audio_channel_mask_t channelMask, int sessionId)
+int AudioFlinger::MixerThread::getTrackName_l(audio_channel_mask_t channelMask,
+        audio_format_t format, int sessionId)
 {
-    return mAudioMixer->getTrackName(channelMask, sessionId);
+    return mAudioMixer->getTrackName(channelMask, format, sessionId);
 }
 
 // deleteTrackName_l() must be called with ThreadBase::mLock held
@@ -3716,7 +3766,8 @@ bool AudioFlinger::MixerThread::checkForNewParameter_l(const String8& keyValuePa
             delete mAudioMixer;
             mAudioMixer = new AudioMixer(mNormalFrameCount, mSampleRate);
             for (size_t i = 0; i < mTracks.size() ; i++) {
-                int name = getTrackName_l(mTracks[i]->mChannelMask, mTracks[i]->mSessionId);
+                int name = getTrackName_l(mTracks[i]->mChannelMask,
+                        mTracks[i]->mFormat, mTracks[i]->mSessionId);
                 if (name < 0) {
                     break;
                 }
@@ -4007,7 +4058,7 @@ void AudioFlinger::DirectOutputThread::threadLoop_sleepTime()
 
 // getTrackName_l() must be called with ThreadBase::mLock held
 int AudioFlinger::DirectOutputThread::getTrackName_l(audio_channel_mask_t channelMask __unused,
-        int sessionId __unused)
+        audio_format_t format __unused, int sessionId __unused)
 {
     return 0;
 }
@@ -4708,16 +4759,151 @@ AudioFlinger::RecordThread::RecordThread(const sp<AudioFlinger>& audioFlinger,
 #endif
     , mReadOnlyHeap(new MemoryDealer(kRecordThreadReadOnlyHeapSize,
             "RecordThreadRO", MemoryHeapBase::READ_ONLY))
+    // mFastCapture below
+    , mFastCaptureFutex(0)
+    // mInputSource
+    // mPipeSink
+    // mPipeSource
+    , mPipeFramesP2(0)
+    // mPipeMemory
+    // mFastCaptureNBLogWriter
+    , mFastTrackAvail(true)
 {
     snprintf(mName, kNameLength, "AudioIn_%X", id);
     mNBLogWriter = audioFlinger->newWriter_l(kLogSize, mName);
 
     readInputParameters_l();
+
+    // create an NBAIO source for the HAL input stream, and negotiate
+    mInputSource = new AudioStreamInSource(input->stream);
+    size_t numCounterOffers = 0;
+    const NBAIO_Format offers[1] = {Format_from_SR_C(mSampleRate, mChannelCount, mFormat)};
+    ssize_t index = mInputSource->negotiate(offers, 1, NULL, numCounterOffers);
+    ALOG_ASSERT(index == 0);
+
+    // initialize fast capture depending on configuration
+    bool initFastCapture;
+    switch (kUseFastCapture) {
+    case FastCapture_Never:
+        initFastCapture = false;
+        break;
+    case FastCapture_Always:
+        initFastCapture = true;
+        break;
+    case FastCapture_Static:
+        uint32_t primaryOutputSampleRate;
+        {
+            AutoMutex _l(audioFlinger->mHardwareLock);
+            primaryOutputSampleRate = audioFlinger->mPrimaryOutputSampleRate;
+        }
+        initFastCapture =
+                // either capture sample rate is same as (a reasonable) primary output sample rate
+                (((primaryOutputSampleRate == 44100 || primaryOutputSampleRate == 48000) &&
+                    (mSampleRate == primaryOutputSampleRate)) ||
+                // or primary output sample rate is unknown, and capture sample rate is reasonable
+                ((primaryOutputSampleRate == 0) &&
+                    ((mSampleRate == 44100 || mSampleRate == 48000)))) &&
+                // and the buffer size is < 10 ms
+                (mFrameCount * 1000) / mSampleRate < 10;
+        break;
+    // case FastCapture_Dynamic:
+    }
+
+    if (initFastCapture) {
+        // create a Pipe for FastMixer to write to, and for us and fast tracks to read from
+        NBAIO_Format format = mInputSource->format();
+        size_t pipeFramesP2 = roundup(mFrameCount * 8);
+        size_t pipeSize = pipeFramesP2 * Format_frameSize(format);
+        void *pipeBuffer;
+        const sp<MemoryDealer> roHeap(readOnlyHeap());
+        sp<IMemory> pipeMemory;
+        if ((roHeap == 0) ||
+                (pipeMemory = roHeap->allocate(pipeSize)) == 0 ||
+                (pipeBuffer = pipeMemory->pointer()) == NULL) {
+            ALOGE("not enough memory for pipe buffer size=%zu", pipeSize);
+            goto failed;
+        }
+        // pipe will be shared directly with fast clients, so clear to avoid leaking old information
+        memset(pipeBuffer, 0, pipeSize);
+        Pipe *pipe = new Pipe(pipeFramesP2, format, pipeBuffer);
+        const NBAIO_Format offers[1] = {format};
+        size_t numCounterOffers = 0;
+        ssize_t index = pipe->negotiate(offers, 1, NULL, numCounterOffers);
+        ALOG_ASSERT(index == 0);
+        mPipeSink = pipe;
+        PipeReader *pipeReader = new PipeReader(*pipe);
+        numCounterOffers = 0;
+        index = pipeReader->negotiate(offers, 1, NULL, numCounterOffers);
+        ALOG_ASSERT(index == 0);
+        mPipeSource = pipeReader;
+        mPipeFramesP2 = pipeFramesP2;
+        mPipeMemory = pipeMemory;
+
+        // create fast capture
+        mFastCapture = new FastCapture();
+        FastCaptureStateQueue *sq = mFastCapture->sq();
+#ifdef STATE_QUEUE_DUMP
+        // FIXME
+#endif
+        FastCaptureState *state = sq->begin();
+        state->mCblk = NULL;
+        state->mInputSource = mInputSource.get();
+        state->mInputSourceGen++;
+        state->mPipeSink = pipe;
+        state->mPipeSinkGen++;
+        state->mFrameCount = mFrameCount;
+        state->mCommand = FastCaptureState::COLD_IDLE;
+        // already done in constructor initialization list
+        //mFastCaptureFutex = 0;
+        state->mColdFutexAddr = &mFastCaptureFutex;
+        state->mColdGen++;
+        state->mDumpState = &mFastCaptureDumpState;
+#ifdef TEE_SINK
+        // FIXME
+#endif
+        mFastCaptureNBLogWriter = audioFlinger->newWriter_l(kFastCaptureLogSize, "FastCapture");
+        state->mNBLogWriter = mFastCaptureNBLogWriter.get();
+        sq->end();
+        sq->push(FastCaptureStateQueue::BLOCK_UNTIL_PUSHED);
+
+        // start the fast capture
+        mFastCapture->run("FastCapture", ANDROID_PRIORITY_URGENT_AUDIO);
+        pid_t tid = mFastCapture->getTid();
+        int err = requestPriority(getpid_cached, tid, kPriorityFastMixer);
+        if (err != 0) {
+            ALOGW("Policy SCHED_FIFO priority %d is unavailable for pid %d tid %d; error %d",
+                    kPriorityFastCapture, getpid_cached, tid, err);
+        }
+
+#ifdef AUDIO_WATCHDOG
+        // FIXME
+#endif
+
+    }
+failed: ;
+
+    // FIXME mNormalSource
 }
 
 
 AudioFlinger::RecordThread::~RecordThread()
 {
+    if (mFastCapture != 0) {
+        FastCaptureStateQueue *sq = mFastCapture->sq();
+        FastCaptureState *state = sq->begin();
+        if (state->mCommand == FastCaptureState::COLD_IDLE) {
+            int32_t old = android_atomic_inc(&mFastCaptureFutex);
+            if (old == -1) {
+                (void) syscall(__NR_futex, &mFastCaptureFutex, FUTEX_WAKE_PRIVATE, 1);
+            }
+        }
+        state->mCommand = FastCaptureState::EXIT;
+        sq->end();
+        sq->push(FastCaptureStateQueue::BLOCK_UNTIL_PUSHED);
+        mFastCapture->join();
+        mFastCapture.clear();
+    }
+    mAudioFlinger->unregisterWriter(mFastCaptureNBLogWriter);
     mAudioFlinger->unregisterWriter(mNBLogWriter);
     delete[] mRsmpInBuffer;
 }
@@ -4772,6 +4958,8 @@ reacquire_wakelock:
         // activeTracks accumulates a copy of a subset of mActiveTracks
         Vector< sp<RecordTrack> > activeTracks;
 
+        // reference to the (first and only) fast track
+        sp<RecordTrack> fastTrack;
 
         { // scope for mLock
             Mutex::Autolock _l(mLock);
@@ -4853,6 +5041,11 @@ reacquire_wakelock:
                 activeTracks.add(activeTrack);
                 i++;
 
+                if (activeTrack->isFastTrack()) {
+                    ALOG_ASSERT(!mFastTrackAvail);
+                    ALOG_ASSERT(fastTrack == 0);
+                    fastTrack = activeTrack;
+                }
             }
             if (doBroadcast) {
                 mStartStopCond.broadcast();
@@ -4878,6 +5071,36 @@ reacquire_wakelock:
             effectChains[i]->process_l();
         }
 
+        // Start the fast capture if it's not already running
+        if (mFastCapture != 0) {
+            FastCaptureStateQueue *sq = mFastCapture->sq();
+            FastCaptureState *state = sq->begin();
+            if (state->mCommand != FastCaptureState::READ_WRITE /* FIXME &&
+                    (kUseFastMixer != FastMixer_Dynamic || state->mTrackMask > 1)*/) {
+                if (state->mCommand == FastCaptureState::COLD_IDLE) {
+                    int32_t old = android_atomic_inc(&mFastCaptureFutex);
+                    if (old == -1) {
+                        (void) syscall(__NR_futex, &mFastCaptureFutex, FUTEX_WAKE_PRIVATE, 1);
+                    }
+                }
+                state->mCommand = FastCaptureState::READ_WRITE;
+#if 0   // FIXME
+                mFastCaptureDumpState.increaseSamplingN(mAudioFlinger->isLowRamDevice() ?
+                        FastCaptureDumpState::kSamplingNforLowRamDevice : FastMixerDumpState::kSamplingN);
+#endif
+                state->mCblk = fastTrack != 0 ? fastTrack->cblk() : NULL;
+                sq->end();
+                sq->push(FastCaptureStateQueue::BLOCK_UNTIL_PUSHED);
+#if 0
+                if (kUseFastCapture == FastCapture_Dynamic) {
+                    mNormalSource = mPipeSource;
+                }
+#endif
+            } else {
+                sq->end(false /*didModify*/);
+            }
+        }
+
         // Read from HAL to keep up with fastest client if multiple active tracks, not slowest one.
         // Only the client(s) that are too slow will overrun. But if even the fastest client is too
         // slow, then this RecordThread will overrun by not calling HAL read often enough.
@@ -4885,24 +5108,45 @@ reacquire_wakelock:
         // copy to the right place.  Permitted because mRsmpInBuffer was over-allocated.
 
         int32_t rear = mRsmpInRear & (mRsmpInFramesP2 - 1);
-        ssize_t bytesRead = mInput->stream->read(mInput->stream,
-                &mRsmpInBuffer[rear * mChannelCount], mBufferSize);
-        if (bytesRead <= 0) {
-            ALOGE("read failed: bytesRead=%d < %u", bytesRead, mBufferSize);
+        ssize_t framesRead;
+
+        // If an NBAIO source is present, use it to read the normal capture's data
+        if (mPipeSource != 0) {
+            size_t framesToRead = mBufferSize / mFrameSize;
+            framesRead = mPipeSource->read(&mRsmpInBuffer[rear * mChannelCount],
+                    framesToRead, AudioBufferProvider::kInvalidPTS);
+            if (framesRead == 0) {
+                // since pipe is non-blocking, simulate blocking input
+                sleepUs = (framesToRead * 1000000LL) / mSampleRate;
+            }
+        // otherwise use the HAL / AudioStreamIn directly
+        } else {
+            ssize_t bytesRead = mInput->stream->read(mInput->stream,
+                    &mRsmpInBuffer[rear * mChannelCount], mBufferSize);
+            if (bytesRead < 0) {
+                framesRead = bytesRead;
+            } else {
+                framesRead = bytesRead / mFrameSize;
+            }
+        }
+
+        if (framesRead < 0 || (framesRead == 0 && mPipeSource == 0)) {
+            ALOGE("read failed: framesRead=%d", framesRead);
             // Force input into standby so that it tries to recover at next read attempt
             inputStandBy();
             sleepUs = kRecordThreadSleepUs;
+        }
+        if (framesRead <= 0) {
             continue;
         }
-        ALOG_ASSERT((size_t) bytesRead <= mBufferSize);
-        size_t framesRead = bytesRead / mFrameSize;
         ALOG_ASSERT(framesRead > 0);
+
         if (mTeeSink != 0) {
             (void) mTeeSink->write(&mRsmpInBuffer[rear * mChannelCount], framesRead);
         }
         // If destination is non-contiguous, we now correct for reading past end of buffer.
         size_t part1 = mRsmpInFramesP2 - rear;
-        if (framesRead > part1) {
+        if ((size_t) framesRead > part1) {
             memcpy(mRsmpInBuffer, &mRsmpInBuffer[mRsmpInFramesP2 * mChannelCount],
                     (framesRead - part1) * mFrameSize);
         }
@@ -4913,6 +5157,11 @@ reacquire_wakelock:
         for (size_t i = 0; i < size; i++) {
             activeTrack = activeTracks[i];
 
+            // skip fast tracks, as those are handled directly by FastCapture
+            if (activeTrack->isFastTrack()) {
+                continue;
+            }
+
             enum {
                 OVERRUN_UNKNOWN,
                 OVERRUN_TRUE,
@@ -5141,6 +5390,30 @@ void AudioFlinger::RecordThread::standbyIfNotAlreadyInStandby()
 
 void AudioFlinger::RecordThread::inputStandBy()
 {
+    // Idle the fast capture if it's currently running
+    if (mFastCapture != 0) {
+        FastCaptureStateQueue *sq = mFastCapture->sq();
+        FastCaptureState *state = sq->begin();
+        if (!(state->mCommand & FastCaptureState::IDLE)) {
+            state->mCommand = FastCaptureState::COLD_IDLE;
+            state->mColdFutexAddr = &mFastCaptureFutex;
+            state->mColdGen++;
+            mFastCaptureFutex = 0;
+            sq->end();
+            // BLOCK_UNTIL_PUSHED would be insufficient, as we need it to stop doing I/O now
+            sq->push(FastCaptureStateQueue::BLOCK_UNTIL_ACKED);
+#if 0
+            if (kUseFastCapture == FastCapture_Dynamic) {
+                // FIXME
+            }
+#endif
+#ifdef AUDIO_WATCHDOG
+            // FIXME
+#endif
+        } else {
+            sq->end(false /*didModify*/);
+        }
+    }
     mInput->stream->common.standby(&mInput->stream->common);
 }
 
@@ -5167,42 +5440,47 @@ sp<AudioFlinger::RecordThread::RecordTrack> AudioFlinger::RecordThread::createRe
             // use case: callback handler and frame count is default or at least as large as HAL
             (
                 (tid != -1) &&
-                ((frameCount == 0) ||
+                ((frameCount == 0) /*||
+                // FIXME must be equal to pipe depth, so don't allow it to be specified by client
                 // FIXME not necessarily true, should be native frame count for native SR!
-                (frameCount >= mFrameCount))
+                (frameCount >= mFrameCount)*/)
             ) &&
             // PCM data
             audio_is_linear_pcm(format) &&
+            // native format
+            (format == mFormat) &&
             // mono or stereo
             ( (channelMask == AUDIO_CHANNEL_IN_MONO) ||
               (channelMask == AUDIO_CHANNEL_IN_STEREO) ) &&
-            // hardware sample rate
-            // FIXME actually the native hardware sample rate
+            // native channel mask
+            (channelMask == mChannelMask) &&
+            // native hardware sample rate
             (sampleRate == mSampleRate) &&
             // record thread has an associated fast capture
-            hasFastCapture()
-            // fast capture does not require slots
+            hasFastCapture() &&
+            // there are sufficient fast track slots available
+            mFastTrackAvail
         ) {
-        // if frameCount not specified, then it defaults to fast capture (HAL) frame count
+        // if frameCount not specified, then it defaults to pipe frame count
         if (frameCount == 0) {
-            // FIXME wrong mFrameCount
-            frameCount = mFrameCount * kFastTrackMultiplier;
+            frameCount = mPipeFramesP2;
         }
         ALOGV("AUDIO_INPUT_FLAG_FAST accepted: frameCount=%d mFrameCount=%d",
                 frameCount, mFrameCount);
       } else {
         ALOGV("AUDIO_INPUT_FLAG_FAST denied: frameCount=%d "
                 "mFrameCount=%d format=%d isLinear=%d channelMask=%#x sampleRate=%u mSampleRate=%u "
-                "hasFastCapture=%d tid=%d",
+                "hasFastCapture=%d tid=%d mFastTrackAvail=%d",
                 frameCount, mFrameCount, format,
                 audio_is_linear_pcm(format),
-                channelMask, sampleRate, mSampleRate, hasFastCapture(), tid);
+                channelMask, sampleRate, mSampleRate, hasFastCapture(), tid, mFastTrackAvail);
         *flags &= ~IAudioFlinger::TRACK_FAST;
         // FIXME It's not clear that we need to enforce this any more, since we have a pipe.
         // For compatibility with AudioRecord calculation, buffer depth is forced
         // to be at least 2 x the record thread frame count and cover audio hardware latency.
         // This is probably too conservative, but legacy application code may depend on it.
         // If you change this calculation, also review the start threshold which is related.
+        // FIXME It's not clear how input latency actually matters.  Perhaps this should be 0.
         uint32_t latencyMs = 50; // FIXME mInput->stream->get_latency(mInput->stream);
         size_t mNormalFrameCount = 2048; // FIXME
         uint32_t minBufCount = latencyMs / ((1000 * mNormalFrameCount) / mSampleRate);
@@ -5424,6 +5702,10 @@ void AudioFlinger::RecordThread::removeTrack_l(const sp<RecordTrack>& track)
 {
     mTracks.remove(track);
     // need anything related to effects here?
+    if (track->isFastTrack()) {
+        ALOG_ASSERT(!mFastTrackAvail);
+        mFastTrackAvail = true;
+    }
 }
 
 void AudioFlinger::RecordThread::dump(int fd, const Vector<String16>& args)
@@ -5442,6 +5724,7 @@ void AudioFlinger::RecordThread::dumpInternals(int fd, const Vector<String16>& a
     } else {
         fdprintf(fd, "  No active record clients\n");
     }
+    dprintf(fd, "  Fast track available: %s\n", mFastTrackAvail ? "yes" : "no");
 
     dumpBase(fd, args);
 }
diff --git a/services/audioflinger/Threads.h b/services/audioflinger/Threads.h
index f8037c6..07887fb 100644
--- a/services/audioflinger/Threads.h
+++ b/services/audioflinger/Threads.h
@@ -356,6 +356,8 @@ public:
                 // If a thread does not have such a heap, this method returns 0.
                 virtual sp<MemoryDealer>    readOnlyHeap() const { return 0; }
 
+                virtual sp<IMemory> pipeMemory() const { return 0; }
+
     mutable     Mutex                   mLock;
 
 protected:
@@ -674,7 +676,8 @@ protected:
 
     // Allocate a track name for a given channel mask.
     //   Returns name >= 0 if successful, -1 on failure.
-    virtual int             getTrackName_l(audio_channel_mask_t channelMask, int sessionId) = 0;
+    virtual int             getTrackName_l(audio_channel_mask_t channelMask,
+                                           audio_format_t format, int sessionId) = 0;
     virtual void            deleteTrackName_l(int name) = 0;
 
     // Time to sleep between cycles when:
@@ -831,7 +834,8 @@ public:
 
 protected:
     virtual     mixer_state prepareTracks_l(Vector< sp<Track> > *tracksToRemove);
-    virtual     int         getTrackName_l(audio_channel_mask_t channelMask, int sessionId);
+    virtual     int         getTrackName_l(audio_channel_mask_t channelMask,
+                                           audio_format_t format, int sessionId);
     virtual     void        deleteTrackName_l(int name);
     virtual     uint32_t    idleSleepTimeUs() const;
     virtual     uint32_t    suspendSleepTimeUs() const;
@@ -884,7 +888,8 @@ public:
                                                    status_t& status);
 
 protected:
-    virtual     int         getTrackName_l(audio_channel_mask_t channelMask, int sessionId);
+    virtual     int         getTrackName_l(audio_channel_mask_t channelMask,
+                                           audio_format_t format, int sessionId);
     virtual     void        deleteTrackName_l(int name);
     virtual     uint32_t    activeSleepTimeUs() const;
     virtual     uint32_t    idleSleepTimeUs() const;
@@ -1059,6 +1064,8 @@ public:
 
     virtual sp<MemoryDealer>    readOnlyHeap() const { return mReadOnlyHeap; }
 
+    virtual sp<IMemory> pipeMemory() const { return mPipeMemory; }
+
             sp<AudioFlinger::RecordThread::RecordTrack>  createRecordTrack_l(
                     const sp<AudioFlinger::Client>& client,
                     uint32_t sampleRate,
@@ -1110,7 +1117,7 @@ public:
     static void syncStartEventCallback(const wp<SyncEvent>& event);
 
     virtual size_t      frameCount() const { return mFrameCount; }
-            bool        hasFastCapture() const { return false; }
+            bool        hasFastCapture() const { return mFastCapture != 0; }
 
 private:
             // Enter standby if not already in standby, and set mStandby flag
@@ -1140,4 +1147,40 @@ private:
             const sp<NBAIO_Sink>                mTeeSink;
 
             const sp<MemoryDealer>              mReadOnlyHeap;
+
+            // one-time initialization, no locks required
+            sp<FastCapture>                     mFastCapture;   // non-0 if there is also a fast capture
+            // FIXME audio watchdog thread
+
+            // contents are not guaranteed to be consistent, no locks required
+            FastCaptureDumpState                mFastCaptureDumpState;
+#ifdef STATE_QUEUE_DUMP
+            // FIXME StateQueue observer and mutator dump fields
+#endif
+            // FIXME audio watchdog dump
+
+            // accessible only within the threadLoop(), no locks required
+            //          mFastCapture->sq()      // for mutating and pushing state
+            int32_t     mFastCaptureFutex;      // for cold idle
+
+            // The HAL input source is treated as non-blocking,
+            // but current implementation is blocking
+            sp<NBAIO_Source>                    mInputSource;
+            // The source for the normal capture thread to read from: mInputSource or mPipeSource
+            sp<NBAIO_Source>                    mNormalSource;
+            // If a fast capture is present, the non-blocking pipe sink written to by fast capture,
+            // otherwise clear
+            sp<NBAIO_Sink>                      mPipeSink;
+            // If a fast capture is present, the non-blocking pipe source read by normal thread,
+            // otherwise clear
+            sp<NBAIO_Source>                    mPipeSource;
+            // Depth of pipe from fast capture to normal thread and fast clients, always power of 2
+            size_t                              mPipeFramesP2;
+            // If a fast capture is present, the Pipe as IMemory, otherwise clear
+            sp<IMemory>                         mPipeMemory;
+
+            static const size_t                 kFastCaptureLogSize = 4 * 1024;
+            sp<NBLog::Writer>                   mFastCaptureNBLogWriter;
+
+            bool                                mFastTrackAvail;    // true if fast track available
 };
diff --git a/services/audioflinger/TrackBase.h b/services/audioflinger/TrackBase.h
index 5f13be3..4cba3fd 100644
--- a/services/audioflinger/TrackBase.h
+++ b/services/audioflinger/TrackBase.h
@@ -39,6 +39,13 @@ public:
         STARTING_2,     // for RecordTrack only
     };
 
+    // where to allocate the data buffer
+    enum alloc_type {
+        ALLOC_CBLK,     // allocate immediately after control block
+        ALLOC_READONLY, // allocate from a separate read-only heap per thread
+        ALLOC_PIPE,     // do not allocate; use the pipe buffer
+    };
+
                         TrackBase(ThreadBase *thread,
                                 const sp<Client>& client,
                                 uint32_t sampleRate,
@@ -50,7 +57,7 @@ public:
                                 int uid,
                                 IAudioFlinger::track_flags_t flags,
                                 bool isOut,
-                                bool useReadOnlyHeap = false);
+                                alloc_type alloc = ALLOC_CBLK);
     virtual             ~TrackBase();
     virtual status_t    initCheck() const { return getCblk() != 0 ? NO_ERROR : NO_MEMORY; }
 
diff --git a/services/audioflinger/Tracks.cpp b/services/audioflinger/Tracks.cpp
index f698fa2..8d5dc7b 100644
--- a/services/audioflinger/Tracks.cpp
+++ b/services/audioflinger/Tracks.cpp
@@ -73,7 +73,7 @@ AudioFlinger::ThreadBase::TrackBase::TrackBase(
             int clientUid,
             IAudioFlinger::track_flags_t flags,
             bool isOut,
-            bool useReadOnlyHeap)
+            alloc_type alloc)
     :   RefBase(),
         mThread(thread),
         mClient(client),
@@ -117,7 +117,7 @@ AudioFlinger::ThreadBase::TrackBase::TrackBase(
     // ALOGD("Creating track with %d buffers @ %d bytes", bufferCount, bufferSize);
     size_t size = sizeof(audio_track_cblk_t);
     size_t bufferSize = (sharedBuffer == 0 ? roundup(frameCount) : frameCount) * mFrameSize;
-    if (sharedBuffer == 0 && !useReadOnlyHeap) {
+    if (sharedBuffer == 0 && alloc == ALLOC_CBLK) {
         size += bufferSize;
     }
 
@@ -139,7 +139,8 @@ AudioFlinger::ThreadBase::TrackBase::TrackBase(
     // construct the shared structure in-place.
     if (mCblk != NULL) {
         new(mCblk) audio_track_cblk_t();
-        if (useReadOnlyHeap) {
+        switch (alloc) {
+        case ALLOC_READONLY: {
             const sp<MemoryDealer> roHeap(thread->readOnlyHeap());
             if (roHeap == 0 ||
                     (mBufferMemory = roHeap->allocate(bufferSize)) == 0 ||
@@ -153,7 +154,17 @@ AudioFlinger::ThreadBase::TrackBase::TrackBase(
                 return;
             }
             memset(mBuffer, 0, bufferSize);
-        } else {
+            } break;
+        case ALLOC_PIPE:
+            mBufferMemory = thread->pipeMemory();
+            // mBuffer is the virtual address as seen from current process (mediaserver),
+            // and should normally be coming from mBufferMemory->pointer().
+            // However in this case the TrackBase does not reference the buffer directly.
+            // It should references the buffer via the pipe.
+            // Therefore, to detect incorrect usage of the buffer, we set mBuffer to NULL.
+            mBuffer = NULL;
+            break;
+        case ALLOC_CBLK:
             // clear all buffers
             if (sharedBuffer == 0) {
                 mBuffer = (char*)mCblk + sizeof(audio_track_cblk_t);
@@ -164,6 +175,7 @@ AudioFlinger::ThreadBase::TrackBase::TrackBase(
                 mCblk->mFlags = CBLK_FORCEREADY;    // FIXME hack, need to fix the track ready logic
 #endif
             }
+            break;
         }
 
 #ifdef TEE_SINK
@@ -385,7 +397,7 @@ AudioFlinger::PlaybackThread::Track::Track(
     }
     mServerProxy = mAudioTrackServerProxy;
 
-    mName = thread->getTrackName_l(channelMask, sessionId);
+    mName = thread->getTrackName_l(channelMask, format, sessionId);
     if (mName < 0) {
         ALOGE("no more track names available");
         return;
@@ -1842,7 +1854,7 @@ AudioFlinger::RecordThread::RecordTrack::RecordTrack(
     :   TrackBase(thread, client, sampleRate, format,
                   channelMask, frameCount, 0 /*sharedBuffer*/, sessionId, uid,
                   flags, false /*isOut*/,
-                  (flags & IAudioFlinger::TRACK_FAST) != 0 /*useReadOnlyHeap*/),
+                  flags & IAudioFlinger::TRACK_FAST ? ALLOC_PIPE : ALLOC_CBLK),
         mOverflow(false), mResampler(NULL), mRsmpOutBuffer(NULL), mRsmpOutFrameCount(0),
         // See real initialization of mRsmpInFront at RecordThread::start()
         mRsmpInUnrel(0), mRsmpInFront(0), mFramesToDrop(0), mResamplerBufferProvider(NULL)
@@ -1861,9 +1873,14 @@ AudioFlinger::RecordThread::RecordTrack::RecordTrack(
         mResampler = AudioResampler::create(16, thread->mChannelCount, sampleRate);
         // source SR
         mResampler->setSampleRate(thread->mSampleRate);
-        mResampler->setVolume(AudioMixer::UNITY_GAIN, AudioMixer::UNITY_GAIN);
+        mResampler->setVolume(AudioMixer::UNITY_GAIN_INT, AudioMixer::UNITY_GAIN_INT);
         mResamplerBufferProvider = new ResamplerBufferProvider(this);
     }
+
+    if (flags & IAudioFlinger::TRACK_FAST) {
+        ALOG_ASSERT(thread->mFastTrackAvail);
+        thread->mFastTrackAvail = false;
+    }
 }
 
 AudioFlinger::RecordThread::RecordTrack::~RecordTrack()