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

2 files changed, 35 insertions, 22 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;