if (framesAvailable != NULL) {
        *framesAvailable = framesIn;
    }
    if (hasOverrun != NULL) {
        *hasOverrun = overrun;
    }
}

// AudioBufferProvider interface
status_t ResamplerBufferProvider::getNextBuffer(
        AudioBufferProvider::Buffer* buffer)
{
    const auto threadBase = mRecordTrack->thread().promote();
    if (threadBase == 0) {
        buffer->frameCount = 0;
        buffer->raw = NULL;
        return NOT_ENOUGH_DATA;
    }
    auto* const recordThread = static_cast<RecordThread *>(threadBase->asIAfRecordThread().get());
    int32_t rear = recordThread->mRsmpInRear;
    int32_t front = mRsmpInFront;
    ssize_t filled = audio_utils::safe_sub_overflow(rear, front);
    // FIXME should not be P2 (don't want to increase latency)
    // FIXME if client not keeping up, discard
    LOG_ALWAYS_FATAL_IF(!(0 <= filled && (size_t) filled <= recordThread->mRsmpInFrames));
    // 'filled' may be non-contiguous, so return only the first contiguous chunk

    front &= recordThread->mRsmpInFramesP2 - 1;
    size_t part1 = recordThread->mRsmpInFramesP2 - front;
    if (part1 > (size_t) filled) {
        part1 = filled;
    }
    size_t ask = buffer->frameCount;
    ALOG_ASSERT(ask > 0);
    if (part1 > ask) {
        part1 = ask;
    }
    if (part1 == 0) {
        // out of data is fine since the resampler will return a short-count.
        buffer->raw = NULL;
        buffer->frameCount = 0;
        mRsmpInUnrel = 0;
        return NOT_ENOUGH_DATA;
    }

    buffer->raw = (uint8_t*)recordThread->mRsmpInBuffer + front * recordThread->mFrameSize;
    buffer->frameCount = part1;
    mRsmpInUnrel = part1;
    return NO_ERROR;
}

// AudioBufferProvider interface
void ResamplerBufferProvider::releaseBuffer(
        AudioBufferProvider::Buffer* buffer)
{
    int32_t stepCount = static_cast<int32_t>(buffer->frameCount);
    if (stepCount == 0) {
        return;
    }
    ALOG_ASSERT(stepCount <= (int32_t)mRsmpInUnrel);
    mRsmpInUnrel -= stepCount;
    mRsmpInFront = audio_utils::safe_add_overflow(mRsmpInFront, stepCount);
    buffer->raw = NULL;
    buffer->frameCount = 0;
}

void RecordThread::checkBtNrec()
{
    audio_utils::lock_guard _l(mutex());
    checkBtNrec_l();
}

void RecordThread::checkBtNrec_l()
{
    // disable AEC and NS if the device is a BT SCO headset supporting those
    // pre processings
    bool suspend = audio_is_bluetooth_sco_device(inDeviceType_l()) &&
                        mAfThreadCallback->btNrecIsOff();
    if (mBtNrecSuspended.exchange(suspend) != suspend) {
        for (size_t i = 0; i < mEffectChains.size(); i++) {
            setEffectSuspended_l(FX_IID_AEC, suspend, mEffectChains[i]->sessionId());
            setEffectSuspended_l(FX_IID_NS, suspend, mEffectChains[i]->sessionId());
        }
    }
}


bool RecordThread::checkForNewParameter_l(const String8& keyValuePair,
                                                        status_t& status)
{
    bool reconfig = false;

    status = NO_ERROR;

    audio_format_t reqFormat = mFormat;
    uint32_t samplingRate = mSampleRate;
    // TODO this may change if we want to support capture from HDMI PCM multi channel (e.g on TVs).
    [[maybe_unused]] audio_channel_mask_t channelMask =
                                audio_channel_in_mask_from_count(mChannelCount);

    AudioParameter param = AudioParameter(keyValuePair);
    int value;

    // scope for AutoPark extends to end of method
    AutoPark<FastCapture> park(mFastCapture);

    // TODO Investigate when this code runs. Check with audio policy when a sample rate and
    //      channel count change can be requested. Do we mandate the first client defines the
    //      HAL sampling rate and channel count or do we allow changes on the fly?
    if (param.getInt(String8(AudioParameter::keySamplingRate), value) == NO_ERROR) {
        samplingRate = value;