diff --git a/include/private/media/AudioTrackShared.h b/include/private/media/AudioTrackShared.h
index 9cabd8b..200e92d 100644
--- a/include/private/media/AudioTrackShared.h
+++ b/include/private/media/AudioTrackShared.h
@@ -182,6 +182,7 @@
                 // This is set by AudioTrack.setBufferSizeInFrames().
                 // A write will not fill the buffer above this limit.
     volatile    uint32_t   mBufferSizeInFrames;  // effective size of the buffer
+    volatile    uint32_t   mStartThresholdInFrames; // min frames in buffer to start streaming
 
 public:
 
@@ -216,6 +217,8 @@
     };
 
     size_t frameCount() const { return mFrameCount; }
+    uint32_t getStartThresholdInFrames() const;
+    uint32_t setStartThresholdInFrames(uint32_t startThresholdInFrames);
 
 protected:
     // These refer to shared memory, and are virtual addresses with respect to the current process.
diff --git a/media/libaudioclient/AudioTrack.cpp b/media/libaudioclient/AudioTrack.cpp
index 19d1d1a..e37cc12 100644
--- a/media/libaudioclient/AudioTrack.cpp
+++ b/media/libaudioclient/AudioTrack.cpp
@@ -1249,6 +1249,46 @@
     return finalBufferSize;
 }
 
+ssize_t AudioTrack::getStartThresholdInFrames() const
+{
+    AutoMutex lock(mLock);
+    if (mOutput == AUDIO_IO_HANDLE_NONE || mProxy.get() == 0) {
+        return NO_INIT;
+    }
+    return (ssize_t) mProxy->getStartThresholdInFrames();
+}
+
+ssize_t AudioTrack::setStartThresholdInFrames(size_t startThresholdInFrames)
+{
+    if (startThresholdInFrames > INT32_MAX || startThresholdInFrames == 0) {
+        // contractually we could simply return the current threshold in frames
+        // to indicate the request was ignored, but we return an error here.
+        return BAD_VALUE;
+    }
+    AutoMutex lock(mLock);
+    // We do not permit calling setStartThresholdInFrames() between the AudioTrack
+    // default ctor AudioTrack() and set(...) but rather fail such an attempt.
+    // (To do so would require a cached mOrigStartThresholdInFrames and we may
+    // not have proper validation for the actual set value).
+    if (mOutput == AUDIO_IO_HANDLE_NONE || mProxy.get() == 0) {
+        return NO_INIT;
+    }
+    const uint32_t original = mProxy->getStartThresholdInFrames();
+    const uint32_t final = mProxy->setStartThresholdInFrames(startThresholdInFrames);
+    if (original != final) {
+        android::mediametrics::LogItem(mMetricsId)
+                .set(AMEDIAMETRICS_PROP_EVENT, AMEDIAMETRICS_PROP_EVENT_VALUE_SETSTARTTHRESHOLD)
+                .set(AMEDIAMETRICS_PROP_STARTTHRESHOLDFRAMES, (int32_t)final)
+                .record();
+        if (original > final) {
+            // restart track if it was disabled by audioflinger due to previous underrun
+            // and we reduced the number of frames for the threshold.
+            restartIfDisabled();
+        }
+    }
+    return final;
+}
+
 status_t AudioTrack::setLoop(uint32_t loopStart, uint32_t loopEnd, int loopCount)
 {
     if (mSharedBuffer == 0 || isOffloadedOrDirect()) {
@@ -2562,6 +2602,10 @@
         staticPosition = mStaticProxy->getPosition().unsignedValue();
     }
 
+    // save the old startThreshold and framecount
+    const uint32_t originalStartThresholdInFrames = mProxy->getStartThresholdInFrames();
+    const uint32_t originalFrameCount = mProxy->frameCount();
+
     // See b/74409267. Connecting to a BT A2DP device supporting multiple codecs
     // causes a lot of churn on the service side, and it can reject starting
     // playback of a previously created track. May also apply to other cases.
@@ -2616,6 +2660,18 @@
             return mAudioTrack->applyVolumeShaper(shaper.mConfiguration, operationToEnd);
         });
 
+        // restore the original start threshold if different than frameCount.
+        if (originalStartThresholdInFrames != originalFrameCount) {
+            // Note: mProxy->setStartThresholdInFrames() call is in the Proxy
+            // and does not trigger a restart.
+            // (Also CBLK_DISABLED is not set, buffers are empty after track recreation).
+            // Any start would be triggered on the mState == ACTIVE check below.
+            const uint32_t currentThreshold =
+                    mProxy->setStartThresholdInFrames(originalStartThresholdInFrames);
+            ALOGD_IF(originalStartThresholdInFrames != currentThreshold,
+                    "%s(%d) startThresholdInFrames changing from %u to %u",
+                    __func__, mPortId, originalStartThresholdInFrames, currentThreshold);
+        }
         if (mState == STATE_ACTIVE) {
             result = mAudioTrack->start();
         }
diff --git a/media/libaudioclient/AudioTrackShared.cpp b/media/libaudioclient/AudioTrackShared.cpp
index f1f8f9c..35719be 100644
--- a/media/libaudioclient/AudioTrackShared.cpp
+++ b/media/libaudioclient/AudioTrackShared.cpp
@@ -17,6 +17,7 @@
 #define LOG_TAG "AudioTrackShared"
 //#define LOG_NDEBUG 0
 
+#include <atomic>
 #include <android-base/macros.h>
 #include <private/media/AudioTrackShared.h>
 #include <utils/Log.h>
@@ -33,6 +34,21 @@
     return sizeof(T) > sizeof(size_t) && x > (T) SIZE_MAX ? SIZE_MAX : x < 0 ? 0 : (size_t) x;
 }
 
+// compile-time safe atomics. TODO: update all methods to use it
+template <typename T>
+T android_atomic_load(const volatile T* addr) {
+    static_assert(sizeof(T) == sizeof(std::atomic<T>)); // no extra sync data required.
+    static_assert(std::atomic<T>::is_always_lock_free); // no hash lock somewhere.
+    return atomic_load((std::atomic<T>*)addr);          // memory_order_seq_cst
+}
+
+template <typename T>
+void android_atomic_store(const volatile T* addr, T value) {
+    static_assert(sizeof(T) == sizeof(std::atomic<T>)); // no extra sync data required.
+    static_assert(std::atomic<T>::is_always_lock_free); // no hash lock somewhere.
+    atomic_store((std::atomic<T>*)addr, value);         // memory_order_seq_cst
+}
+
 // incrementSequence is used to determine the next sequence value
 // for the loop and position sequence counters.  It should return
 // a value between "other" + 1 and "other" + INT32_MAX, the choice of
@@ -51,6 +67,7 @@
     : mServer(0), mFutex(0), mMinimum(0)
     , mVolumeLR(GAIN_MINIFLOAT_PACKED_UNITY), mSampleRate(0), mSendLevel(0)
     , mBufferSizeInFrames(0)
+    , mStartThresholdInFrames(0) // filled in by the server.
     , mFlags(0)
 {
     memset(&u, 0, sizeof(u));
@@ -66,6 +83,26 @@
 {
 }
 
+uint32_t Proxy::getStartThresholdInFrames() const
+{
+    const uint32_t startThresholdInFrames =
+           android_atomic_load(&mCblk->mStartThresholdInFrames);
+    if (startThresholdInFrames == 0 || startThresholdInFrames > mFrameCount) {
+        ALOGD("%s: startThresholdInFrames %u not between 1 and frameCount %zu, "
+                "setting to frameCount",
+                __func__, startThresholdInFrames, mFrameCount);
+        return mFrameCount;
+    }
+    return startThresholdInFrames;
+}
+
+uint32_t Proxy::setStartThresholdInFrames(uint32_t startThresholdInFrames)
+{
+    const uint32_t actual = std::min((size_t)startThresholdInFrames, frameCount());
+    android_atomic_store(&mCblk->mStartThresholdInFrames, actual);
+    return actual;
+}
+
 // ---------------------------------------------------------------------------
 
 ClientProxy::ClientProxy(audio_track_cblk_t* cblk, void *buffers, size_t frameCount,
@@ -663,6 +700,7 @@
     , mTimestampMutator(&cblk->mExtendedTimestampQueue)
 {
     cblk->mBufferSizeInFrames = frameCount;
+    cblk->mStartThresholdInFrames = frameCount;
 }
 
 __attribute__((no_sanitize("integer")))
@@ -900,11 +938,8 @@
     }
     audio_track_cblk_t* cblk = mCblk;
 
-    int32_t flush = cblk->u.mStreaming.mFlush;
-    if (flush != mFlush) {
-        // FIXME should return an accurate value, but over-estimate is better than under-estimate
-        return mFrameCount;
-    }
+    flushBufferIfNeeded();
+
     const int32_t rear = getRear();
     ssize_t filled = audio_utils::safe_sub_overflow(rear, cblk->u.mStreaming.mFront);
     // pipe should not already be overfull
diff --git a/media/libaudioclient/include/media/AudioTrack.h b/media/libaudioclient/include/media/AudioTrack.h
index fac4c83..4afa9c9 100644
--- a/media/libaudioclient/include/media/AudioTrack.h
+++ b/media/libaudioclient/include/media/AudioTrack.h
@@ -429,6 +429,19 @@
      */
             ssize_t     setBufferSizeInFrames(size_t size);
 
+    /* Returns the start threshold on the buffer for audio streaming
+     * or a negative value if the AudioTrack is not initialized.
+     */
+            ssize_t     getStartThresholdInFrames() const;
+
+    /* Sets the start threshold in frames on the buffer for audio streaming.
+     *
+     * May be clamped internally. Returns the actual value set, or a negative
+     * value if the AudioTrack is not initialized or if the input
+     * is zero or greater than INT_MAX.
+     */
+            ssize_t     setStartThresholdInFrames(size_t startThresholdInFrames);
+
     /* Return the static buffer specified in constructor or set(), or 0 for streaming mode */
             sp<IMemory> sharedBuffer() const { return mSharedBuffer; }
 
diff --git a/media/libaudiohal/impl/DeviceHalHidl.cpp b/media/libaudiohal/impl/DeviceHalHidl.cpp
index da16477..0d5fe59 100644
--- a/media/libaudiohal/impl/DeviceHalHidl.cpp
+++ b/media/libaudiohal/impl/DeviceHalHidl.cpp
@@ -246,6 +246,10 @@
         return status;
     }
     CoreUtils::AudioInputFlags hidlFlags;
+#if MAJOR_VERSION <= 5
+    // Some flags were specific to framework and must not leak to the HAL.
+    flags = static_cast<audio_input_flags_t>(flags & ~AUDIO_INPUT_FLAG_DIRECT);
+#endif
     if (status_t status = CoreUtils::audioInputFlagsFromHal(flags, &hidlFlags); status != OK) {
         return status;
     }
@@ -278,10 +282,6 @@
         sinkMetadata.tracks[0].destination.device(std::move(hidlOutputDevice));
     }
 #endif
-#if MAJOR_VERSION <= 5
-    // Some flags were specific to framework and must not leak to the HAL.
-    flags = static_cast<audio_input_flags_t>(flags & ~AUDIO_INPUT_FLAG_DIRECT);
-#endif
     Return<void> ret = mDevice->openInputStream(
             handle, hidlDevice, hidlConfig, hidlFlags, sinkMetadata,
             [&](Result r, const sp<IStreamIn>& result, const AudioConfig& suggestedConfig) {
diff --git a/media/libeffects/preprocessing/Android.bp b/media/libeffects/preprocessing/Android.bp
index 87ed8b6..c6e036a 100644
--- a/media/libeffects/preprocessing/Android.bp
+++ b/media/libeffects/preprocessing/Android.bp
@@ -18,15 +18,10 @@
     ],
 }
 
-cc_library {
-    name: "libaudiopreprocessing",
+cc_defaults {
+    name: "libaudiopreprocessing-defaults",
     vendor: true,
-    relative_install_path: "soundfx",
     host_supported: true,
-    srcs: ["PreProcessing.cpp"],
-    local_include_dirs: [
-        ".",
-    ],
     cflags: [
         "-Wall",
         "-Werror",
@@ -46,7 +41,6 @@
     header_libs: [
         "libaudioeffects",
         "libhardware_headers",
-        "libwebrtc_absl_headers",
     ],
     target: {
         darwin: {
@@ -54,3 +48,13 @@
         },
     },
 }
+
+cc_library {
+    name: "libaudiopreprocessing",
+    defaults: ["libaudiopreprocessing-defaults"],
+    relative_install_path: "soundfx",
+    srcs: ["PreProcessing.cpp"],
+    header_libs: [
+        "libwebrtc_absl_headers",
+    ],
+}
diff --git a/media/libeffects/preprocessing/PreProcessing.cpp b/media/libeffects/preprocessing/PreProcessing.cpp
index 3b0b6d6..19a8b2f 100644
--- a/media/libeffects/preprocessing/PreProcessing.cpp
+++ b/media/libeffects/preprocessing/PreProcessing.cpp
@@ -105,9 +105,8 @@
     webrtc::AudioProcessing* apm;  // handle on webRTC audio processing module (APM)
     // Audio Processing module builder
     webrtc::AudioProcessingBuilder ap_builder;
-    size_t apmFrameCount;      // buffer size for webRTC process (10 ms)
-    uint32_t apmSamplingRate;  // webRTC APM sampling rate (8/16 or 32 kHz)
-    size_t frameCount;         // buffer size before input resampler ( <=> apmFrameCount)
+    // frameCount represents the size of the buffers used for processing, and must represent 10ms.
+    size_t frameCount;
     uint32_t samplingRate;     // sampling rate at effect process interface
     uint32_t inChannelCount;   // input channel count
     uint32_t outChannelCount;  // output channel count
@@ -119,21 +118,12 @@
     webrtc::AudioProcessing::Config config;
     webrtc::StreamConfig inputConfig;   // input stream configuration
     webrtc::StreamConfig outputConfig;  // output stream configuration
-    int16_t* inBuf;    // input buffer used when resampling
-    size_t inBufSize;  // input buffer size in frames
-    size_t framesIn;   // number of frames in input buffer
-    int16_t* outBuf;    // output buffer used when resampling
-    size_t outBufSize;  // output buffer size in frames
-    size_t framesOut;   // number of frames in output buffer
     uint32_t revChannelCount;  // number of channels on reverse stream
     uint32_t revEnabledMsk;    // bit field containing IDs of enabled pre processors
                                // with reverse channel
     uint32_t revProcessedMsk;  // bit field containing IDs of pre processors with reverse
                                // channel already processed in current round
     webrtc::StreamConfig revConfig;     // reverse stream configuration.
-    int16_t* revBuf;    // reverse channel input buffer
-    size_t revBufSize;  // reverse channel input buffer size
-    size_t framesRev;   // number of frames in reverse channel input buffer
 };
 
 #ifdef DUAL_MIC_TEST
@@ -862,9 +852,7 @@
             ALOGW("Session_CreateEffect could not get apm engine");
             goto error;
         }
-        session->apmSamplingRate = kPreprocDefaultSr;
-        session->apmFrameCount = (kPreprocDefaultSr) / 100;
-        session->frameCount = session->apmFrameCount;
+        session->frameCount = kPreprocDefaultSr / 100;
         session->samplingRate = kPreprocDefaultSr;
         session->inChannelCount = kPreProcDefaultCnl;
         session->outChannelCount = kPreProcDefaultCnl;
@@ -879,12 +867,6 @@
         session->processedMsk = 0;
         session->revEnabledMsk = 0;
         session->revProcessedMsk = 0;
-        session->inBuf = NULL;
-        session->inBufSize = 0;
-        session->outBuf = NULL;
-        session->outBufSize = 0;
-        session->revBuf = NULL;
-        session->revBufSize = 0;
     }
     status = Effect_Create(&session->effects[procId], session, interface);
     if (status < 0) {
@@ -908,13 +890,6 @@
     if (session->createdMsk == 0) {
         delete session->apm;
         session->apm = NULL;
-        delete session->inBuf;
-        session->inBuf = NULL;
-        free(session->outBuf);
-        session->outBuf = NULL;
-        delete session->revBuf;
-        session->revBuf = NULL;
-
         session->id = 0;
     }
 
@@ -934,24 +909,8 @@
     ALOGV("Session_SetConfig sr %d cnl %08x", config->inputCfg.samplingRate,
           config->inputCfg.channels);
 
-    // AEC implementation is limited to 16kHz
-    if (config->inputCfg.samplingRate >= 32000 && !(session->createdMsk & (1 << PREPROC_AEC))) {
-        session->apmSamplingRate = 32000;
-    } else if (config->inputCfg.samplingRate >= 16000) {
-        session->apmSamplingRate = 16000;
-    } else if (config->inputCfg.samplingRate >= 8000) {
-        session->apmSamplingRate = 8000;
-    }
-
-
     session->samplingRate = config->inputCfg.samplingRate;
-    session->apmFrameCount = session->apmSamplingRate / 100;
-    if (session->samplingRate == session->apmSamplingRate) {
-        session->frameCount = session->apmFrameCount;
-    } else {
-        session->frameCount =
-                (session->apmFrameCount * session->samplingRate) / session->apmSamplingRate;
-    }
+    session->frameCount = session->samplingRate / 100;
     session->inChannelCount = inCnl;
     session->outChannelCount = outCnl;
     session->inputConfig.set_sample_rate_hz(session->samplingRate);
@@ -963,13 +922,6 @@
     session->revConfig.set_sample_rate_hz(session->samplingRate);
     session->revConfig.set_num_channels(inCnl);
 
-    // force process buffer reallocation
-    session->inBufSize = 0;
-    session->outBufSize = 0;
-    session->framesIn = 0;
-    session->framesOut = 0;
-
-
     session->state = PREPROC_SESSION_STATE_CONFIG;
     return 0;
 }
@@ -1004,9 +956,6 @@
     }
     uint32_t inCnl = audio_channel_count_from_out_mask(config->inputCfg.channels);
     session->revChannelCount = inCnl;
-    // force process buffer reallocation
-    session->revBufSize = 0;
-    session->framesRev = 0;
 
     return 0;
 }
@@ -1023,12 +972,8 @@
 
 void Session_SetProcEnabled(preproc_session_t* session, uint32_t procId, bool enabled) {
     if (enabled) {
-        if (session->enabledMsk == 0) {
-            session->framesIn = 0;
-        }
         session->enabledMsk |= (1 << procId);
         if (HasReverseStream(procId)) {
-            session->framesRev = 0;
             session->revEnabledMsk |= (1 << procId);
         }
     } else {
@@ -1117,43 +1062,24 @@
         return -EINVAL;
     }
 
+    if (inBuffer->frameCount != outBuffer->frameCount) {
+        ALOGW("inBuffer->frameCount %zu is not equal to outBuffer->frameCount %zu",
+              inBuffer->frameCount, outBuffer->frameCount);
+        return -EINVAL;
+    }
+
+    if (inBuffer->frameCount != session->frameCount) {
+        ALOGW("inBuffer->frameCount %zu != %zu representing 10ms at sampling rate %d",
+              inBuffer->frameCount, session->frameCount, session->samplingRate);
+        return -EINVAL;
+    }
+
     session->processedMsk |= (1 << effect->procId);
 
     //    ALOGV("PreProcessingFx_Process In %d frames enabledMsk %08x processedMsk %08x",
     //         inBuffer->frameCount, session->enabledMsk, session->processedMsk);
-
     if ((session->processedMsk & session->enabledMsk) == session->enabledMsk) {
         effect->session->processedMsk = 0;
-        size_t framesRq = outBuffer->frameCount;
-        size_t framesWr = 0;
-        if (session->framesOut) {
-            size_t fr = session->framesOut;
-            if (outBuffer->frameCount < fr) {
-                fr = outBuffer->frameCount;
-            }
-            memcpy(outBuffer->s16, session->outBuf,
-                   fr * session->outChannelCount * sizeof(int16_t));
-            memmove(session->outBuf, session->outBuf + fr * session->outChannelCount,
-                    (session->framesOut - fr) * session->outChannelCount * sizeof(int16_t));
-            session->framesOut -= fr;
-            framesWr += fr;
-        }
-        outBuffer->frameCount = framesWr;
-        if (framesWr == framesRq) {
-            inBuffer->frameCount = 0;
-            return 0;
-        }
-
-        size_t fr = session->frameCount - session->framesIn;
-        if (inBuffer->frameCount < fr) {
-            fr = inBuffer->frameCount;
-        }
-        session->framesIn += fr;
-        inBuffer->frameCount = fr;
-        if (session->framesIn < session->frameCount) {
-            return 0;
-        }
-        session->framesIn = 0;
         if (int status = effect->session->apm->ProcessStream(
                     (const int16_t* const)inBuffer->s16,
                     (const webrtc::StreamConfig)effect->session->inputConfig,
@@ -1163,34 +1089,6 @@
             ALOGE("Process Stream failed with error %d\n", status);
             return status;
         }
-        outBuffer->frameCount = inBuffer->frameCount;
-
-        if (session->outBufSize < session->framesOut + session->frameCount) {
-            int16_t* buf;
-            session->outBufSize = session->framesOut + session->frameCount;
-            buf = (int16_t*)realloc(
-                    session->outBuf,
-                    session->outBufSize * session->outChannelCount * sizeof(int16_t));
-            if (buf == NULL) {
-                session->framesOut = 0;
-                free(session->outBuf);
-                session->outBuf = NULL;
-                return -ENOMEM;
-            }
-            session->outBuf = buf;
-        }
-
-        fr = session->framesOut;
-        if (framesRq - framesWr < fr) {
-            fr = framesRq - framesWr;
-        }
-        memcpy(outBuffer->s16 + framesWr * session->outChannelCount, session->outBuf,
-               fr * session->outChannelCount * sizeof(int16_t));
-        memmove(session->outBuf, session->outBuf + fr * session->outChannelCount,
-                (session->framesOut - fr) * session->outChannelCount * sizeof(int16_t));
-        session->framesOut -= fr;
-        outBuffer->frameCount += fr;
-
         return 0;
     } else {
         return -ENODATA;
@@ -1565,6 +1463,18 @@
         return -EINVAL;
     }
 
+    if (inBuffer->frameCount != outBuffer->frameCount) {
+        ALOGW("inBuffer->frameCount %zu is not equal to outBuffer->frameCount %zu",
+              inBuffer->frameCount, outBuffer->frameCount);
+        return -EINVAL;
+    }
+
+    if (inBuffer->frameCount != session->frameCount) {
+        ALOGW("inBuffer->frameCount %zu != %zu representing 10ms at sampling rate %d",
+              inBuffer->frameCount, session->frameCount, session->samplingRate);
+        return -EINVAL;
+    }
+
     session->revProcessedMsk |= (1 << effect->procId);
 
     //    ALOGV("PreProcessingFx_ProcessReverse In %d frames revEnabledMsk %08x revProcessedMsk
@@ -1573,16 +1483,6 @@
 
     if ((session->revProcessedMsk & session->revEnabledMsk) == session->revEnabledMsk) {
         effect->session->revProcessedMsk = 0;
-        size_t fr = session->frameCount - session->framesRev;
-        if (inBuffer->frameCount < fr) {
-            fr = inBuffer->frameCount;
-        }
-        session->framesRev += fr;
-        inBuffer->frameCount = fr;
-        if (session->framesRev < session->frameCount) {
-            return 0;
-        }
-        session->framesRev = 0;
         if (int status = effect->session->apm->ProcessReverseStream(
                     (const int16_t* const)inBuffer->s16,
                     (const webrtc::StreamConfig)effect->session->revConfig,
diff --git a/media/libeffects/preprocessing/README.md b/media/libeffects/preprocessing/README.md
new file mode 100644
index 0000000..af46376
--- /dev/null
+++ b/media/libeffects/preprocessing/README.md
@@ -0,0 +1,7 @@
+# Preprocessing effects
+
+## Limitations
+- Preprocessing effects currently work on 10ms worth of data and do not support
+  arbitrary frame counts. This limiation comes from the underlying effects in
+  webrtc modules
+- There is currently no api to communicate this requirement
diff --git a/media/libeffects/preprocessing/benchmarks/Android.bp b/media/libeffects/preprocessing/benchmarks/Android.bp
index c1b2295..fbbcab4 100644
--- a/media/libeffects/preprocessing/benchmarks/Android.bp
+++ b/media/libeffects/preprocessing/benchmarks/Android.bp
@@ -11,27 +11,10 @@
 
 cc_benchmark {
     name: "preprocessing_benchmark",
-    vendor: true,
+    defaults: ["libaudiopreprocessing-defaults"],
     srcs: ["preprocessing_benchmark.cpp"],
-    shared_libs: [
-        "libaudioutils",
-        "liblog",
-        "libutils",
-    ],
     static_libs: [
         "libaudiopreprocessing",
-        "webrtc_audio_processing",
-    ],
-    cflags: [
-        "-DWEBRTC_POSIX",
-        "-fvisibility=default",
-        "-Wall",
-        "-Werror",
-        "-Wextra",
-    ],
-    header_libs: [
-        "libaudioeffects",
-        "libhardware_headers",
-        "libwebrtc_absl_headers",
+        "libaudioutils",
     ],
 }
diff --git a/media/libeffects/preprocessing/tests/Android.bp b/media/libeffects/preprocessing/tests/Android.bp
index 18c6c98..d80b135 100644
--- a/media/libeffects/preprocessing/tests/Android.bp
+++ b/media/libeffects/preprocessing/tests/Android.bp
@@ -12,9 +12,8 @@
 
 cc_test {
     name: "EffectPreprocessingTest",
-    vendor: true,
+    defaults: ["libaudiopreprocessing-defaults"],
     gtest: true,
-    host_supported: true,
     test_suites: ["device-tests"],
     srcs: [
         "EffectPreprocessingTest.cpp",
@@ -23,46 +22,18 @@
     static_libs: [
         "libaudiopreprocessing",
         "libaudioutils",
-        "webrtc_audio_processing",
     ],
-    shared_libs: [
-        "liblog",
-    ],
-    header_libs: [
-        "libaudioeffects",
-        "libhardware_headers",
-    ],
-    target: {
-        darwin: {
-            enabled: false,
-        },
-    },
 }
 
 cc_test {
     name: "AudioPreProcessingTest",
-    vendor: true,
-    host_supported: true,
+    defaults: ["libaudiopreprocessing-defaults"],
     gtest: false,
     srcs: ["PreProcessingTest.cpp"],
-    shared_libs: [
-        "libaudioutils",
-        "liblog",
-        "libutils",
-    ],
     static_libs: [
         "libaudiopreprocessing",
-        "webrtc_audio_processing",
+        "libaudioutils",
     ],
-    header_libs: [
-        "libaudioeffects",
-        "libhardware_headers",
-    ],
-    target: {
-        darwin: {
-            enabled: false,
-        },
-    },
 }
 
 cc_test {
diff --git a/media/libeffects/preprocessing/tests/EffectTestHelper.h b/media/libeffects/preprocessing/tests/EffectTestHelper.h
index db06823..117cf7b 100644
--- a/media/libeffects/preprocessing/tests/EffectTestHelper.h
+++ b/media/libeffects/preprocessing/tests/EffectTestHelper.h
@@ -88,7 +88,8 @@
 
     static constexpr size_t kNumChMasks = std::size(kChMasks);
 
-    static constexpr size_t kSampleRates[] = {8000, 16000, 24000, 32000, 48000};
+    static constexpr size_t kSampleRates[] = {8000,  11025, 12000, 16000, 22050,
+                                              24000, 32000, 44100, 48000};
 
     static constexpr size_t kNumSampleRates = std::size(kSampleRates);
 
diff --git a/media/libeffects/preprocessing/tests/PreProcessingTest.cpp b/media/libeffects/preprocessing/tests/PreProcessingTest.cpp
index e0025fe..3bd93f8 100644
--- a/media/libeffects/preprocessing/tests/PreProcessingTest.cpp
+++ b/media/libeffects/preprocessing/tests/PreProcessingTest.cpp
@@ -451,8 +451,8 @@
         }
         audio_buffer_t inputBuffer, outputBuffer;
         audio_buffer_t farInBuffer{};
-        inputBuffer.frameCount = samplesRead;
-        outputBuffer.frameCount = samplesRead;
+        inputBuffer.frameCount = frameLength;
+        outputBuffer.frameCount = frameLength;
         inputBuffer.s16 = in.data();
         outputBuffer.s16 = out.data();
 
@@ -472,7 +472,7 @@
                 }
             }
 
-            farInBuffer.frameCount = samplesRead;
+            farInBuffer.frameCount = frameLength;
             farInBuffer.s16 = farIn.data();
         }
 
@@ -519,6 +519,7 @@
         }
         frameCounter += frameLength;
     }
+    printf("frameCounter: [%d]\n", frameCounter);
     // Release all the effect handles created
     for (int i = 0; i < PREPROC_NUM_EFFECTS; i++) {
         if (int status = AUDIO_EFFECT_LIBRARY_INFO_SYM.release_effect(effectHandle[i]);
diff --git a/media/libeffects/preprocessing/tests/build_and_run_all_unit_tests.sh b/media/libeffects/preprocessing/tests/build_and_run_all_unit_tests.sh
index 942f2ec..35da13e 100755
--- a/media/libeffects/preprocessing/tests/build_and_run_all_unit_tests.sh
+++ b/media/libeffects/preprocessing/tests/build_and_run_all_unit_tests.sh
@@ -59,9 +59,13 @@
 
 fs_arr=(
     8000
+    11025
+    12000
     16000
+    22050
     24000
     32000
+    44100
     48000
 )
 
diff --git a/media/libeffects/proxy/EffectProxy.cpp b/media/libeffects/proxy/EffectProxy.cpp
index c010d68..be9f8c0 100644
--- a/media/libeffects/proxy/EffectProxy.cpp
+++ b/media/libeffects/proxy/EffectProxy.cpp
@@ -116,6 +116,16 @@
         pContext->sube[SUB_FX_OFFLOAD] = sube[1];
         pContext->desc[SUB_FX_OFFLOAD] = desc[1];
         pContext->aeli[SUB_FX_OFFLOAD] = aeli[1];
+    } else {
+        ALOGE("Both effects have (or don't have) EFFECT_FLAG_HW_ACC_TUNNEL flag");
+        delete[] sube;
+        delete[] desc;
+        delete[] aeli;
+        delete[] pContext->sube;
+        delete[] pContext->desc;
+        delete[] pContext->aeli;
+        delete pContext;
+        return -EINVAL;
     }
     delete[] desc;
     delete[] aeli;
diff --git a/media/libmediametrics/include/MediaMetricsConstants.h b/media/libmediametrics/include/MediaMetricsConstants.h
index 84388c9..674df17 100644
--- a/media/libmediametrics/include/MediaMetricsConstants.h
+++ b/media/libmediametrics/include/MediaMetricsConstants.h
@@ -139,6 +139,7 @@
 #define AMEDIAMETRICS_PROP_SESSIONID      "sessionId"      // int32
 #define AMEDIAMETRICS_PROP_SHARINGMODE    "sharingMode"    // string value, "exclusive", shared"
 #define AMEDIAMETRICS_PROP_SOURCE         "source"         // string (AudioAttributes)
+#define AMEDIAMETRICS_PROP_STARTTHRESHOLDFRAMES "startThresholdFrames" // int32 (AudioTrack)
 #define AMEDIAMETRICS_PROP_STARTUPMS      "startupMs"      // double value
 // State is "ACTIVE" or "STOPPED" for AudioRecord
 #define AMEDIAMETRICS_PROP_STATE          "state"          // string
@@ -181,6 +182,7 @@
 #define AMEDIAMETRICS_PROP_EVENT_VALUE_SETMODE    "setMode" // AudioFlinger
 #define AMEDIAMETRICS_PROP_EVENT_VALUE_SETBUFFERSIZE    "setBufferSize" // AudioTrack
 #define AMEDIAMETRICS_PROP_EVENT_VALUE_SETPLAYBACKPARAM "setPlaybackParam" // AudioTrack
+#define AMEDIAMETRICS_PROP_EVENT_VALUE_SETSTARTTHRESHOLD "setStartThreshold" // AudioTrack
 #define AMEDIAMETRICS_PROP_EVENT_VALUE_SETVOICEVOLUME   "setVoiceVolume" // AudioFlinger
 #define AMEDIAMETRICS_PROP_EVENT_VALUE_SETVOLUME  "setVolume"  // AudioTrack
 #define AMEDIAMETRICS_PROP_EVENT_VALUE_START      "start"  // AudioTrack, AudioRecord
diff --git a/services/audioflinger/PlaybackTracks.h b/services/audioflinger/PlaybackTracks.h
index 7804822..472c359 100644
--- a/services/audioflinger/PlaybackTracks.h
+++ b/services/audioflinger/PlaybackTracks.h
@@ -284,8 +284,6 @@
     };
     sp<AudioVibrationController> mAudioVibrationController;
     sp<os::ExternalVibration>    mExternalVibration;
-    /** How many frames should be in the buffer before the track is considered ready */
-    const size_t        mFrameCountToBeReady;
 
     audio_dual_mono_mode_t mDualMonoMode = AUDIO_DUAL_MONO_MODE_OFF;
     float               mAudioDescriptionMixLevel = -std::numeric_limits<float>::infinity();
diff --git a/services/audioflinger/Tracks.cpp b/services/audioflinger/Tracks.cpp
index ee886d5..fb43a6e 100644
--- a/services/audioflinger/Tracks.cpp
+++ b/services/audioflinger/Tracks.cpp
@@ -638,7 +638,6 @@
     mOpPlayAudioMonitor(OpPlayAudioMonitor::createIfNeeded(
             uid, attr, id(), streamType, opPackageName)),
     // mSinkTimestamp
-    mFrameCountToBeReady(frameCountToBeReady),
     mFastIndex(-1),
     mCachedVolume(1.0),
     /* The track might not play immediately after being active, similarly as if its volume was 0.
@@ -672,6 +671,7 @@
                 mFrameSize, sampleRate);
     }
     mServerProxy = mAudioTrackServerProxy;
+    mServerProxy->setStartThresholdInFrames(frameCountToBeReady); // update the Cblk value
 
     // only allocate a fast track index if we were able to allocate a normal track name
     if (flags & AUDIO_OUTPUT_FLAG_FAST) {
@@ -999,7 +999,10 @@
     }
 
     size_t bufferSizeInFrames = mServerProxy->getBufferSizeInFrames();
-    size_t framesToBeReady = std::min(mFrameCountToBeReady, bufferSizeInFrames);
+    // Note: mServerProxy->getStartThresholdInFrames() is clamped.
+    const size_t startThresholdInFrames = mServerProxy->getStartThresholdInFrames();
+    const size_t framesToBeReady = std::clamp(  // clamp again to validate client values.
+            std::min(startThresholdInFrames, bufferSizeInFrames), size_t(1), mFrameCount);
 
     if (framesReady() >= framesToBeReady || (mCblk->mFlags & CBLK_FORCEREADY)) {
         ALOGV("%s(%d): consider track ready with %zu/%zu, target was %zu)",
@@ -1038,6 +1041,11 @@
         // initial state-stopping. next state-pausing.
         // What if resume is called ?
 
+        if (state == FLUSHED) {
+            // avoid underrun glitches when starting after flush
+            reset();
+        }
+
         if (state == PAUSED || state == PAUSING) {
             if (mResumeToStopping) {
                 // happened we need to resume to STOPPING_1