move native services under services/
moved surfaceflinger, audioflinger, cameraservice
all native services should now reside in this location.
Change-Id: Iee42b83dd2a94c3bf5107ab0895fe2dfcd5337a8
diff --git a/services/audioflinger/AudioResamplerCubic.cpp b/services/audioflinger/AudioResamplerCubic.cpp
new file mode 100644
index 0000000..1d247bd
--- /dev/null
+++ b/services/audioflinger/AudioResamplerCubic.cpp
@@ -0,0 +1,184 @@
+/*
+ * Copyright (C) 2007 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 <stdint.h>
+#include <string.h>
+#include <sys/types.h>
+#include <cutils/log.h>
+
+#include "AudioResampler.h"
+#include "AudioResamplerCubic.h"
+
+#define LOG_TAG "AudioSRC"
+
+namespace android {
+// ----------------------------------------------------------------------------
+
+void AudioResamplerCubic::init() {
+ memset(&left, 0, sizeof(state));
+ memset(&right, 0, sizeof(state));
+}
+
+void AudioResamplerCubic::resample(int32_t* out, size_t outFrameCount,
+ AudioBufferProvider* provider) {
+
+ // should never happen, but we overflow if it does
+ // LOG_ASSERT(outFrameCount < 32767);
+
+ // select the appropriate resampler
+ switch (mChannelCount) {
+ case 1:
+ resampleMono16(out, outFrameCount, provider);
+ break;
+ case 2:
+ resampleStereo16(out, outFrameCount, provider);
+ break;
+ }
+}
+
+void AudioResamplerCubic::resampleStereo16(int32_t* out, size_t outFrameCount,
+ AudioBufferProvider* provider) {
+
+ int32_t vl = mVolume[0];
+ int32_t vr = mVolume[1];
+
+ size_t inputIndex = mInputIndex;
+ uint32_t phaseFraction = mPhaseFraction;
+ uint32_t phaseIncrement = mPhaseIncrement;
+ size_t outputIndex = 0;
+ size_t outputSampleCount = outFrameCount * 2;
+ size_t inFrameCount = (outFrameCount*mInSampleRate)/mSampleRate;
+
+ // fetch first buffer
+ if (mBuffer.frameCount == 0) {
+ mBuffer.frameCount = inFrameCount;
+ provider->getNextBuffer(&mBuffer);
+ if (mBuffer.raw == NULL)
+ return;
+ // LOGW("New buffer: offset=%p, frames=%dn", mBuffer.raw, mBuffer.frameCount);
+ }
+ int16_t *in = mBuffer.i16;
+
+ while (outputIndex < outputSampleCount) {
+ int32_t sample;
+ int32_t x;
+
+ // calculate output sample
+ x = phaseFraction >> kPreInterpShift;
+ out[outputIndex++] += vl * interp(&left, x);
+ out[outputIndex++] += vr * interp(&right, x);
+ // out[outputIndex++] += vr * in[inputIndex*2];
+
+ // increment phase
+ phaseFraction += phaseIncrement;
+ uint32_t indexIncrement = (phaseFraction >> kNumPhaseBits);
+ phaseFraction &= kPhaseMask;
+
+ // time to fetch another sample
+ while (indexIncrement--) {
+
+ inputIndex++;
+ if (inputIndex == mBuffer.frameCount) {
+ inputIndex = 0;
+ provider->releaseBuffer(&mBuffer);
+ mBuffer.frameCount = inFrameCount;
+ provider->getNextBuffer(&mBuffer);
+ if (mBuffer.raw == NULL)
+ goto save_state; // ugly, but efficient
+ in = mBuffer.i16;
+ // LOGW("New buffer: offset=%p, frames=%d\n", mBuffer.raw, mBuffer.frameCount);
+ }
+
+ // advance sample state
+ advance(&left, in[inputIndex*2]);
+ advance(&right, in[inputIndex*2+1]);
+ }
+ }
+
+save_state:
+ // LOGW("Done: index=%d, fraction=%u", inputIndex, phaseFraction);
+ mInputIndex = inputIndex;
+ mPhaseFraction = phaseFraction;
+}
+
+void AudioResamplerCubic::resampleMono16(int32_t* out, size_t outFrameCount,
+ AudioBufferProvider* provider) {
+
+ int32_t vl = mVolume[0];
+ int32_t vr = mVolume[1];
+
+ size_t inputIndex = mInputIndex;
+ uint32_t phaseFraction = mPhaseFraction;
+ uint32_t phaseIncrement = mPhaseIncrement;
+ size_t outputIndex = 0;
+ size_t outputSampleCount = outFrameCount * 2;
+ size_t inFrameCount = (outFrameCount*mInSampleRate)/mSampleRate;
+
+ // fetch first buffer
+ if (mBuffer.frameCount == 0) {
+ mBuffer.frameCount = inFrameCount;
+ provider->getNextBuffer(&mBuffer);
+ if (mBuffer.raw == NULL)
+ return;
+ // LOGW("New buffer: offset=%p, frames=%d\n", mBuffer.raw, mBuffer.frameCount);
+ }
+ int16_t *in = mBuffer.i16;
+
+ while (outputIndex < outputSampleCount) {
+ int32_t sample;
+ int32_t x;
+
+ // calculate output sample
+ x = phaseFraction >> kPreInterpShift;
+ sample = interp(&left, x);
+ out[outputIndex++] += vl * sample;
+ out[outputIndex++] += vr * sample;
+
+ // increment phase
+ phaseFraction += phaseIncrement;
+ uint32_t indexIncrement = (phaseFraction >> kNumPhaseBits);
+ phaseFraction &= kPhaseMask;
+
+ // time to fetch another sample
+ while (indexIncrement--) {
+
+ inputIndex++;
+ if (inputIndex == mBuffer.frameCount) {
+ inputIndex = 0;
+ provider->releaseBuffer(&mBuffer);
+ mBuffer.frameCount = inFrameCount;
+ provider->getNextBuffer(&mBuffer);
+ if (mBuffer.raw == NULL)
+ goto save_state; // ugly, but efficient
+ // LOGW("New buffer: offset=%p, frames=%dn", mBuffer.raw, mBuffer.frameCount);
+ in = mBuffer.i16;
+ }
+
+ // advance sample state
+ advance(&left, in[inputIndex]);
+ }
+ }
+
+save_state:
+ // LOGW("Done: index=%d, fraction=%u", inputIndex, phaseFraction);
+ mInputIndex = inputIndex;
+ mPhaseFraction = phaseFraction;
+}
+
+// ----------------------------------------------------------------------------
+}
+; // namespace android
+