audio HWL: removed unused code.

Removed C++ implementations of echo reference and resampler not needed
anymore now that libaudioutils is used.

Change-Id: Ibedf96fbaeeb38ea06b35adf7c95ed49cbafa916

4 files changed, 0 insertions, 745 deletions

diff --git a/libaudio/EchoReference.cpp b/libaudio/EchoReference.cpp
deleted file mode 100644
index 2c10062..0000000
--- a/libaudio/EchoReference.cpp
+++ /dev/null
@@ -1,376 +0,0 @@
-/*
-** Copyright 2011, 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_NDEBUG 0
-#define LOG_TAG "EchoReference"
-
-#include <utils/Log.h>
-#include "EchoReference.h"
-
-namespace android_audio_legacy {
-
-//------------------------------------------------------------------------------
-// Echo reference buffer
-//------------------------------------------------------------------------------
-
-EchoReference::EchoReference(audio_format_t rdFormat,
-                                            uint32_t rdChannelCount,
-                                            uint32_t rdSamplingRate,
-                                            audio_format_t wrFormat,
-                                            uint32_t wrChannelCount,
-                                            uint32_t wrSamplingRate)
-: mStatus (NO_INIT), mState(ECHOREF_IDLE),
-  mRdFormat(rdFormat), mRdChannelCount(rdChannelCount), mRdSamplingRate(rdSamplingRate),
-  mWrFormat(wrFormat), mWrChannelCount(wrChannelCount), mWrSamplingRate(wrSamplingRate),
-  mBuffer(NULL), mBufSize(0), mFramesIn(0), mWrBuf(NULL), mWrBufSize(0), mWrFramesIn(0),
-  mDownSampler(NULL)
-{
-    LOGV("EchoReference cstor");
-    if (rdFormat != AUDIO_FORMAT_PCM_16_BIT ||
-            rdFormat != wrFormat) {
-        LOGW("EchoReference cstor bad format rd %d, wr %d", rdFormat, wrFormat);
-        mStatus = BAD_VALUE;
-        return;
-    }
-    if ((rdChannelCount != 1 && rdChannelCount != 2) ||
-            wrChannelCount != 2) {
-        LOGW("EchoReference cstor bad channel count rd %d, wr %d", rdChannelCount, wrChannelCount);
-        mStatus = BAD_VALUE;
-        return;
-    }
-
-    if (wrSamplingRate < rdSamplingRate) {
-        LOGW("EchoReference cstor bad smp rate rd %d, wr %d", rdSamplingRate, wrSamplingRate);
-        mStatus = BAD_VALUE;
-        return;
-    }
-
-    mRdFrameSize = audio_bytes_per_sample(rdFormat) * rdChannelCount;
-    mWrFrameSize = audio_bytes_per_sample(wrFormat) * wrChannelCount;
-    mStatus = NO_ERROR;
-}
-
-
-EchoReference::~EchoReference() {
-    LOGV("EchoReference dstor");
-    reset_l();
-    delete mDownSampler;
-}
-
-status_t EchoReference::write(Buffer *buffer)
-{
-    if (mStatus != NO_ERROR) {
-        LOGV("EchoReference::write() ERROR, exiting early");
-        return mStatus;
-    }
-
-    AutoMutex _l(mLock);
-
-    if (buffer == NULL) {
-        LOGV("EchoReference::write() stop write");
-        mState &= ~ECHOREF_WRITING;
-        reset_l();
-        return NO_ERROR;
-    }
-
-    LOGV("EchoReference::write() START trying to write %d frames", buffer->frameCount);
-    LOGV("EchoReference::write() playbackTimestamp:[%lld].[%lld], mPlaybackDelay:[%ld]",
-    (int64_t)buffer->timeStamp.tv_sec,
-    (int64_t)buffer->timeStamp.tv_nsec, mPlaybackDelay);
-
-    //LOGV("EchoReference::write() %d frames", buffer->frameCount);
-    // discard writes until a valid time stamp is provided.
-
-    if ((buffer->timeStamp.tv_sec == 0) && (buffer->timeStamp.tv_nsec == 0) &&
-        (mWrRenderTime.tv_sec     == 0) && (mWrRenderTime.tv_nsec     == 0)) {
-            return NO_ERROR;
-    }
-
-    if ((mState & ECHOREF_WRITING) == 0) {
-        LOGV("EchoReference::write() start write");
-        if (mDownSampler != NULL) {
-            mDownSampler->reset();
-        }
-        mState |= ECHOREF_WRITING;
-    }
-
-    if ((mState & ECHOREF_READING) == 0) {
-        return NO_ERROR;
-    }
-
-    mWrRenderTime.tv_sec  = buffer->timeStamp.tv_sec;
-    mWrRenderTime.tv_nsec = buffer->timeStamp.tv_nsec;
-
-    mPlaybackDelay = buffer->delayNs;
-
-    void *srcBuf;
-    size_t inFrames;
-    // do stereo to mono and down sampling if necessary
-    if (mRdChannelCount != mWrChannelCount ||
-            mRdSamplingRate != mWrSamplingRate) {
-        if (mWrBufSize < buffer->frameCount) {
-            mWrBufSize = buffer->frameCount;
-            //max buffer size is normally function of read sampling rate but as write sampling rate
-            //is always more than read sampling rate this works
-            mWrBuf = realloc(mWrBuf, mWrBufSize * mRdFrameSize);
-        }
-
-        inFrames = buffer->frameCount;
-        if (mRdChannelCount != mWrChannelCount) {
-            // must be stereo to mono
-            int16_t *src16 = (int16_t *)buffer->raw;
-            int16_t *dst16 = (int16_t *)mWrBuf;
-            size_t frames = buffer->frameCount;
-            while (frames--) {
-                *dst16++ = (int16_t)(((int32_t)*src16 + (int32_t)*(src16 + 1)) >> 1);
-                src16 += 2;
-            }
-        }
-        if (mWrSamplingRate != mRdSamplingRate) {
-            if (mDownSampler == NULL) {
-                LOGV("EchoReference::write() new ReSampler(%d, %d)",
-                      mWrSamplingRate, mRdSamplingRate);
-                mDownSampler = new ReSampler(mWrSamplingRate,
-                                             mRdSamplingRate,
-                                             mRdChannelCount,
-                                             this);
-
-            }
-            // mWrSrcBuf and mWrFramesIn are used by getNexBuffer() called by the resampler
-            // to get new frames
-            if (mRdChannelCount != mWrChannelCount) {
-                mWrSrcBuf = mWrBuf;
-            } else {
-                mWrSrcBuf = buffer->raw;
-            }
-            mWrFramesIn = buffer->frameCount;
-            // inFrames is always more than we need here to get frames remaining from previous runs
-            // inFrames is updated by resample() with the number of frames produced
-            LOGV("EchoReference::write() ReSampling(%d, %d)",
-                  mWrSamplingRate, mRdSamplingRate);
-            mDownSampler->resample((int16_t *)mWrBuf, &inFrames);
-            LOGV_IF(mWrFramesIn != 0,
-                    "EchoReference::write() mWrFramesIn not 0 (%d) after resampler",
-                    mWrFramesIn);
-        }
-        srcBuf = mWrBuf;
-    } else {
-        inFrames = buffer->frameCount;
-        srcBuf = buffer->raw;
-    }
-
-    if (mFramesIn + inFrames > mBufSize) {
-        LOGV("EchoReference::write() increasing buffer size from %d to %d",
-        mBufSize, mFramesIn + inFrames);
-        mBufSize = mFramesIn + inFrames;
-        mBuffer = realloc(mBuffer, mBufSize * mRdFrameSize);
-    }
-    memcpy((char *)mBuffer + mFramesIn * mRdFrameSize,
-           srcBuf,
-           inFrames * mRdFrameSize);
-    mFramesIn += inFrames;
-
-    LOGV("EchoReference::write_log() inFrames:[%d], mFramesInOld:[%d], "\
-         "mFramesInNew:[%d], mBufSize:[%d], mWrRenderTime:[%lld].[%lld], mPlaybackDelay:[%ld]",
-         inFrames, mFramesIn - inFrames, mFramesIn, mBufSize,  (int64_t)mWrRenderTime.tv_sec,
-         (int64_t)mWrRenderTime.tv_nsec, mPlaybackDelay);
-
-    mCond.signal();
-    LOGV("EchoReference::write() END");
-    return NO_ERROR;
-}
-
-status_t EchoReference::read(EchoReference::Buffer *buffer)
-{
-    if (mStatus != NO_ERROR) {
-        return mStatus;
-    }
-    AutoMutex _l(mLock);
-
-    if (buffer == NULL) {
-        LOGV("EchoReference::read() stop read");
-        mState &= ~ECHOREF_READING;
-        return NO_ERROR;
-    }
-
-    LOGV("EchoReference::read() START, delayCapture:[%ld],mFramesIn:[%d],buffer->frameCount:[%d]",
-    buffer->delayNs, mFramesIn, buffer->frameCount);
-
-    if ((mState & ECHOREF_READING) == 0) {
-        LOGV("EchoReference::read() start read");
-        reset_l();
-        mState |= ECHOREF_READING;
-    }
-
-    if ((mState & ECHOREF_WRITING) == 0) {
-        memset(buffer->raw, 0, mRdFrameSize * buffer->frameCount);
-        buffer->delayNs = 0;
-        return NO_ERROR;
-    }
-
-//    LOGV("EchoReference::read() %d frames", buffer->frameCount);
-
-    // allow some time for new frames to arrive if not enough frames are ready for read
-    if (mFramesIn < buffer->frameCount) {
-        uint32_t timeoutMs = (uint32_t)((1000 * buffer->frameCount) / mRdSamplingRate / 2);
-
-        mCond.waitRelative(mLock, milliseconds(timeoutMs));
-        if (mFramesIn < buffer->frameCount) {
-            LOGV("EchoReference::read() waited %d ms but still not enough frames"\
-                 " mFramesIn: %d, buffer->frameCount = %d",
-                timeoutMs, mFramesIn, buffer->frameCount);
-            buffer->frameCount = mFramesIn;
-        }
-    }
-
-    int64_t timeDiff;
-    struct timespec tmp;
-
-    if ((mWrRenderTime.tv_sec == 0 && mWrRenderTime.tv_nsec == 0) ||
-        (buffer->timeStamp.tv_sec == 0 && buffer->timeStamp.tv_nsec == 0)) {
-        LOGV("read: NEW:timestamp is zero---------setting timeDiff = 0, "\
-             "not updating delay this time");
-        timeDiff = 0;
-    } else {
-        if (buffer->timeStamp.tv_nsec < mWrRenderTime.tv_nsec) {
-            tmp.tv_sec = buffer->timeStamp.tv_sec - mWrRenderTime.tv_sec - 1;
-            tmp.tv_nsec = 1000000000 + buffer->timeStamp.tv_nsec - mWrRenderTime.tv_nsec;
-        } else {
-            tmp.tv_sec = buffer->timeStamp.tv_sec - mWrRenderTime.tv_sec;
-            tmp.tv_nsec = buffer->timeStamp.tv_nsec - mWrRenderTime.tv_nsec;
-        }
-        timeDiff = (((int64_t)tmp.tv_sec * 1000000000 + tmp.tv_nsec));
-
-        int64_t expectedDelayNs =  mPlaybackDelay + buffer->delayNs - timeDiff;
-
-        LOGV("expectedDelayNs[%lld] =  mPlaybackDelay[%ld] + delayCapture[%ld] - timeDiff[%lld]",
-        expectedDelayNs, mPlaybackDelay, buffer->delayNs, timeDiff);
-
-        if (expectedDelayNs > 0) {
-            int64_t delayNs = ((int64_t)mFramesIn * 1000000000) / mRdSamplingRate;
-
-            delayNs -= expectedDelayNs;
-
-            if (abs(delayNs) >= sMinDelayUpdate) {
-                if (delayNs < 0) {
-                    size_t previousFrameIn = mFramesIn;
-                    mFramesIn = (expectedDelayNs * mRdSamplingRate)/1000000000;
-                    int    offset = mFramesIn - previousFrameIn;
-                    LOGV("EchoReference::readlog: delayNs = NEGATIVE and ENOUGH : "\
-                         "setting %d frames to zero mFramesIn: %d, previousFrameIn = %d",
-                         offset, mFramesIn, previousFrameIn);
-
-                    if (mFramesIn > mBufSize) {
-                        mBufSize = mFramesIn;
-                        mBuffer  = realloc(mBuffer, mFramesIn * mRdFrameSize);
-                        LOGV("EchoReference::read: increasing buffer size to %d", mBufSize);
-                    }
-
-                    if (offset > 0)
-                        memset((char *)mBuffer + previousFrameIn * mRdFrameSize,
-                               0, offset * mRdFrameSize);
-                } else {
-                    size_t  previousFrameIn = mFramesIn;
-                    int     framesInInt = (int)(((int64_t)expectedDelayNs *
-                                           (int64_t)mRdSamplingRate)/1000000000);
-                    int     offset = previousFrameIn - framesInInt;
-
-                    LOGV("EchoReference::readlog: delayNs = POSITIVE/ENOUGH :previousFrameIn: %d,"\
-                         "framesInInt: [%d], offset:[%d], buffer->frameCount:[%d]",
-                         previousFrameIn, framesInInt, offset, buffer->frameCount);
-
-                    if (framesInInt < (int)buffer->frameCount) {
-                        if (framesInInt > 0) {
-                            memset((char *)mBuffer + framesInInt * mRdFrameSize,
-                                   0, (buffer->frameCount-framesInInt) * mRdFrameSize);
-                            LOGV("EchoReference::read: pushing [%d] zeros into ref buffer",
-                                 (buffer->frameCount-framesInInt));
-                        } else {
-                            LOGV("framesInInt = %d", framesInInt);
-                        }
-                        framesInInt = buffer->frameCount;
-                    } else {
-                        if (offset > 0) {
-                            memcpy(mBuffer, (char *)mBuffer + (offset * mRdFrameSize),
-                                   framesInInt * mRdFrameSize);
-                            LOGV("EchoReference::read: shifting ref buffer by [%d]",framesInInt);
-                        }
-                    }
-                    mFramesIn = (size_t)framesInInt;
-                }
-            } else {
-                LOGV("EchoReference::read: NOT ENOUGH samples to update %lld", delayNs);
-            }
-        } else {
-            LOGV("NEGATIVE expectedDelayNs[%lld] =  "\
-                 "mPlaybackDelay[%ld] + delayCapture[%ld] - timeDiff[%lld]",
-                 expectedDelayNs, mPlaybackDelay, buffer->delayNs, timeDiff);
-        }
-    }
-
-    memcpy(buffer->raw,
-           (char *)mBuffer,
-           buffer->frameCount * mRdFrameSize);
-
-    mFramesIn -= buffer->frameCount;
-    memcpy(mBuffer,
-           (char *)mBuffer + buffer->frameCount * mRdFrameSize,
-           mFramesIn * mRdFrameSize);
-
-    // As the reference buffer is now time aligned to the microphone signal there is a zero delay
-    buffer->delayNs = 0;
-
-    LOGV("EchoReference::read() END %d frames, total frames in %d",
-          buffer->frameCount, mFramesIn);
-
-    mCond.signal();
-    return NO_ERROR;
-}
-
-void EchoReference::reset_l() {
-    LOGV("EchoReference::reset_l()");
-    free(mBuffer);
-    mBuffer = NULL;
-    mBufSize = 0;
-    mFramesIn = 0;
-    free(mWrBuf);
-    mWrBuf = NULL;
-    mWrBufSize = 0;
-    mWrRenderTime.tv_sec = 0;
-    mWrRenderTime.tv_nsec = 0;
-}
-
-status_t EchoReference::getNextBuffer(ReSampler::BufferProvider::Buffer* buffer)
-{
-    if (mWrSrcBuf == NULL || mWrFramesIn == 0) {
-        buffer->raw = NULL;
-        buffer->frameCount = 0;
-        return NOT_ENOUGH_DATA;
-    }
-
-    buffer->frameCount = (buffer->frameCount > mWrFramesIn) ? mWrFramesIn : buffer->frameCount;
-    // this is mRdChannelCount here as we resample after stereo to mono conversion if any
-    buffer->i16 = (int16_t *)mWrSrcBuf + (mWrBufSize - mWrFramesIn) * mRdChannelCount;
-
-    return 0;
-}
-
-void EchoReference::releaseBuffer(ReSampler::BufferProvider::Buffer* buffer)
-{
-    mWrFramesIn -= buffer->frameCount;
-}
-
-}; // namespace android_audio_legacy
diff --git a/libaudio/EchoReference.h b/libaudio/EchoReference.h
deleted file mode 100644
index b82cd11..0000000
--- a/libaudio/EchoReference.h
+++ /dev/null
@@ -1,118 +0,0 @@
-/*
-** Copyright 2011, 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_ECHO_REFERENCE_H
-#define ANDROID_ECHO_REFERENCE_H
-
-#include <stdint.h>
-#include <sys/types.h>
-#include <utils/threads.h>
-#include <hardware_legacy/AudioSystemLegacy.h>
-#include "ReSampler.h"
-
-namespace android_audio_legacy {
-using android::Mutex;
-using android::AutoMutex;
-
-class EchoReference : public ReSampler::BufferProvider {
-
-public:
-
-    EchoReference(audio_format_t rdFormat,
-                  uint32_t rdChannelCount,
-                  uint32_t rdSamplingRate,
-                  audio_format_t wrFormat,
-                  uint32_t wrChannelCount,
-                  uint32_t wrSamplingRate);
-    virtual ~EchoReference();
-
-    // echo reference state: it field indicating if read, write or both are active.
-    enum state {
-        ECHOREF_IDLE = 0x00,        // idle
-        ECHOREF_READING = 0x01,     // reading is active
-        ECHOREF_WRITING = 0x02      // writing is active
-    };
-
-    // Buffer descriptor used by read() and write() methods, including the time stamp and delay.
-    class Buffer {
-     public:
-        void *raw;                  // pointer to audio frame
-        size_t frameCount;          // number of frames in buffer
-        int32_t delayNs;            // delay for this buffer (see comment below)
-        struct timespec timeStamp;  // time stamp for this buffer (see comment below)
-    };
-    // when used for EchoReference::write():
-    // + as input:
-    //      - delayNs is the delay introduced by playback buffers
-    //      - timeStamp is the time stamp corresponding to the delay calculation
-    // + as output:
-    //      unused
-    // when used for EchoReference::read():
-    // + as input:
-    //      - delayNs is the delay introduced by capture buffers
-    //      - timeStamp is the time stamp corresponding to the delay calculation
-    // + as output:
-    //      - delayNs is the delay between the returned frames and the capture time derived from
-    //      delay and time stamp indicated as input. This delay is to be communicated to the AEC.
-    //      - frameCount is updated with the actual number of frames returned
-
-
-    // BufferProvider
-    status_t getNextBuffer(ReSampler::BufferProvider::Buffer* buffer);
-    void releaseBuffer(ReSampler::BufferProvider::Buffer* buffer);
-
-    status_t initCheck() { return mStatus; }
-
-    status_t read(Buffer *buffer);
-    status_t write(Buffer *buffer);
-
-
-private:
-
-    void reset_l();
-
-    status_t mStatus;               // init status
-    uint32_t mState;                // active state: reading, writing or both
-    audio_format_t mRdFormat;       // read sample format
-    uint32_t mRdChannelCount;       // read number of channels
-    uint32_t mRdSamplingRate;       // read sampling rate
-    size_t mRdFrameSize;            // read frame size (bytes per sample)
-    audio_format_t mWrFormat;       // write sample format
-    uint32_t mWrChannelCount;       // write number of channels
-    uint32_t mWrSamplingRate;       // write sampling rate
-    size_t mWrFrameSize;            // write frame size (bytes per sample)
-    void *mBuffer;                  // main buffer
-    size_t mBufSize;                // main buffer size in frames
-    size_t mFramesIn;               // number of frames in main buffer
-    void *mWrBuf;                   // buffer for input conversions
-    size_t mWrBufSize;              // size of conversion buffer in frames
-    size_t mWrFramesIn;             // number of frames in conversion buffer
-    void *mWrSrcBuf;                // resampler input buf (either mWrBuf or buffer used by write()
-    struct timespec mWrRenderTime;  // latest render time indicated by write()
-    int32_t  mPlaybackDelay;
-
-    uint32_t mRdDurationUs;         // Duration of last buffer read (used for mCond wait timeout)
-    android::Mutex   mLock;         // Mutex protecting read/write concurrency
-    android::Condition mCond;       // Condition signaled when data is ready to read
-    ReSampler *mDownSampler;        // input resampler
-
-    static const int sMinDelayUpdate = 62500; // delay jump threshold to update ref buffer
-                                              // 0.5 samples at 8kHz in nsecs
-};
-
-}; // namespace android
-
-#endif // ANDROID_ECHO_REFERENCE_H
diff --git a/libaudio/ReSampler.cpp b/libaudio/ReSampler.cpp
deleted file mode 100644
index aa1d5e0..0000000
--- a/libaudio/ReSampler.cpp
+++ /dev/null
@@ -1,171 +0,0 @@
-/*
-** Copyright 2011, 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_NDEBUG 0
-#define LOG_TAG "ReSampler"
-
-#include <utils/Log.h>
-#include "ReSampler.h"
-
-namespace android_audio_legacy {
-
-
-//------------------------------------------------------------------------------
-// speex based resampler
-//------------------------------------------------------------------------------
-
-#define RESAMPLER_QUALITY 2
-
-ReSampler::ReSampler(uint32_t inSampleRate,
-                                    uint32_t outSampleRate,
-                                    uint32_t channelCount,
-                                    BufferProvider* provider)
-    :  mStatus(NO_INIT), mSpeexResampler(NULL), mProvider(provider),
-       mInSampleRate(inSampleRate), mOutSampleRate(outSampleRate), mChannelCount(channelCount),
-       mInBuf(NULL), mInBufSize(0)
-{
-    LOGV("ReSampler() cstor %p In SR %d Out SR %d channels %d",
-         this, mInSampleRate, mOutSampleRate, mChannelCount);
-
-    if (mProvider == NULL) {
-        return;
-    }
-
-    int error;
-    mSpeexResampler = speex_resampler_init(channelCount,
-                                      inSampleRate,
-                                      outSampleRate,
-                                      RESAMPLER_QUALITY,
-                                      &error);
-    if (mSpeexResampler == NULL) {
-        LOGW("ReSampler: Cannot create speex resampler: %s", speex_resampler_strerror(error));
-        return;
-    }
-
-    reset();
-
-    int frames = speex_resampler_get_input_latency(mSpeexResampler);
-    mSpeexDelayNs = (int32_t)((1000000000 * (int64_t)frames) / mInSampleRate);
-    frames = speex_resampler_get_output_latency(mSpeexResampler);
-    mSpeexDelayNs += (int32_t)((1000000000 * (int64_t)frames) / mOutSampleRate);
-
-    mStatus = NO_ERROR;
-}
-
-ReSampler::~ReSampler()
-{
-    free(mInBuf);
-
-    if (mSpeexResampler != NULL) {
-        speex_resampler_destroy(mSpeexResampler);
-    }
-}
-
-void ReSampler::reset()
-{
-    mFramesIn = 0;
-    mFramesRq = 0;
-
-    if (mSpeexResampler != NULL) {
-        speex_resampler_reset_mem(mSpeexResampler);
-    }
-}
-
-int32_t ReSampler::delayNs()
-{
-    int32_t delay = (int32_t)((1000000000 * (int64_t)mFramesIn) / mInSampleRate);
-    delay += mSpeexDelayNs;
-
-    return delay;
-}
-
-// outputs a number of frames less or equal to *outFrameCount and updates *outFrameCount
-// with the actual number of frames produced.
-int ReSampler::resample(int16_t *out, size_t *outFrameCount)
-{
-    if (mStatus != NO_ERROR) {
-        return mStatus;
-    }
-
-    if (out == NULL || outFrameCount == NULL) {
-        return BAD_VALUE;
-    }
-
-    size_t framesRq = *outFrameCount;
-    // update and cache the number of frames needed at the input sampling rate to produce
-    // the number of frames requested at the output sampling rate
-    if (framesRq != mFramesRq) {
-        mFramesNeeded = (framesRq * mOutSampleRate) / mInSampleRate + 1;
-        mFramesRq = framesRq;
-    }
-
-    size_t framesWr = 0;
-    size_t inFrames = 0;
-    while (framesWr < framesRq) {
-        if (mFramesIn < mFramesNeeded) {
-            // make sure that the number of frames present in mInBuf (mFramesIn) is at least
-            // the number of frames needed to produce the number of frames requested at
-            // the output sampling rate
-            if (mInBufSize < mFramesNeeded) {
-                mInBufSize = mFramesNeeded;
-                mInBuf = (int16_t *)realloc(mInBuf, mInBufSize * mChannelCount * sizeof(int16_t));
-            }
-            BufferProvider::Buffer buf;
-            buf.frameCount = mFramesNeeded - mFramesIn;
-            mProvider->getNextBuffer(&buf);
-            if (buf.raw == NULL) {
-                break;
-            }
-            memcpy(mInBuf + mFramesIn * mChannelCount,
-                    buf.raw,
-                    buf.frameCount * mChannelCount * sizeof(int16_t));
-            mFramesIn += buf.frameCount;
-            mProvider->releaseBuffer(&buf);
-        }
-
-        size_t outFrames = framesRq - framesWr;
-        inFrames = mFramesIn;
-        if (mChannelCount == 1) {
-            speex_resampler_process_int(mSpeexResampler,
-                                        0,
-                                        mInBuf,
-                                        &inFrames,
-                                        out + framesWr * mChannelCount,
-                                        &outFrames);
-        } else {
-            speex_resampler_process_interleaved_int(mSpeexResampler,
-                                        mInBuf,
-                                        &inFrames,
-                                        out + framesWr * mChannelCount,
-                                        &outFrames);
-        }
-        framesWr += outFrames;
-        mFramesIn -= inFrames;
-        LOGW_IF((framesWr != framesRq) && (mFramesIn != 0),
-                "ReSampler::resample() remaining %d frames in and %d frames out",
-                mFramesIn, (framesRq - framesWr));
-    }
-    if (mFramesIn) {
-        memmove(mInBuf,
-                mInBuf + inFrames * mChannelCount,
-                mFramesIn * mChannelCount * sizeof(int16_t));
-    }
-    *outFrameCount = framesWr;
-
-    return NO_ERROR;
-}
-
-}; // namespace android_audio_legacy
diff --git a/libaudio/ReSampler.h b/libaudio/ReSampler.h
deleted file mode 100644
index f531900..0000000
--- a/libaudio/ReSampler.h
+++ /dev/null
@@ -1,80 +0,0 @@
-/*
-** Copyright 2008, 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_RESAMPLER_H
-#define ANDROID_RESAMPLER_H
-
-#include <stdint.h>
-#include <sys/types.h>
-#include <hardware_legacy/AudioSystemLegacy.h>
-#include "speex/speex_resampler.h"
-
-namespace android_audio_legacy {
-
-class ReSampler {
-public:
-
-    class BufferProvider
-    {
-    public:
-
-        struct Buffer {
-            union {
-                void*       raw;
-                short*      i16;
-                int8_t*     i8;
-            };
-            size_t frameCount;
-        };
-
-        virtual ~BufferProvider() {}
-
-        virtual status_t getNextBuffer(Buffer* buffer) = 0;
-        virtual void releaseBuffer(Buffer* buffer) = 0;
-    };
-
-    ReSampler(uint32_t inSampleRate,
-              uint32_t outSampleRate,
-              uint32_t channelCount,
-              BufferProvider* provider);
-
-    virtual ~ReSampler();
-
-            status_t initCheck() { return mStatus; }
-            void reset();
-            int resample(int16_t* out, size_t *outFrameCount);
-            int32_t delayNs();
-
-
-private:
-    status_t    mStatus;                    // init status
-    SpeexResamplerState *mSpeexResampler;   // handle on speex resampler
-    BufferProvider* mProvider;              // buffer provider installed by client
-    uint32_t mInSampleRate;                 // input sampling rate
-    uint32_t mOutSampleRate;                // output sampling rate
-    uint32_t mChannelCount;                 // number of channels
-    int16_t *mInBuf;                        // input buffer
-    size_t mInBufSize;                      // input buffer size
-    size_t mFramesIn;                       // number of frames in input buffer
-    size_t mFramesRq;                       // cached number of output frames
-    size_t mFramesNeeded;                   // minimum number of input frames to produce mFramesRq
-                                            // output frames
-    int32_t mSpeexDelayNs;                  // delay introduced by speex resampler in ns
-};
-
-}; // namespace android_audio_legacy
-
-#endif // ANDROID_RESAMPLER_H