1 files changed, 0 insertions, 93 deletions
diff --git a/libs/audioflinger/AudioResampler.h b/libs/audioflinger/AudioResampler.h
deleted file mode 100644
index 39656c0..0000000
--- a/libs/audioflinger/AudioResampler.h
+++ /dev/null
@@ -1,93 +0,0 @@
-/*
- * 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.
- */
-
-#ifndef ANDROID_AUDIO_RESAMPLER_H
-#define ANDROID_AUDIO_RESAMPLER_H
-
-#include <stdint.h>
-#include <sys/types.h>
-
-#include "AudioBufferProvider.h"
-
-namespace android {
-// ----------------------------------------------------------------------------
-
-class AudioResampler {
-public:
-    // Determines quality of SRC.
-    //  LOW_QUALITY: linear interpolator (1st order)
-    //  MED_QUALITY: cubic interpolator (3rd order)
-    //  HIGH_QUALITY: fixed multi-tap FIR (e.g. 48KHz->44.1KHz)
-    // NOTE: high quality SRC will only be supported for
-    // certain fixed rate conversions. Sample rate cannot be
-    // changed dynamically. 
-    enum src_quality {
-        DEFAULT=0,
-        LOW_QUALITY=1,
-        MED_QUALITY=2,
-        HIGH_QUALITY=3
-    };
-
-    static AudioResampler* create(int bitDepth, int inChannelCount,
-            int32_t sampleRate, int quality=DEFAULT);
-
-    virtual ~AudioResampler();
-
-    virtual void init() = 0;
-    virtual void setSampleRate(int32_t inSampleRate);
-    virtual void setVolume(int16_t left, int16_t right);
-
-    virtual void resample(int32_t* out, size_t outFrameCount,
-            AudioBufferProvider* provider) = 0;
-
-protected:
-    // number of bits for phase fraction - 30 bits allows nearly 2x downsampling
-    static const int kNumPhaseBits = 30;
-
-    // phase mask for fraction
-    static const uint32_t kPhaseMask = (1LU<<kNumPhaseBits)-1;
-
-    // multiplier to calculate fixed point phase increment
-    static const double kPhaseMultiplier = 1L << kNumPhaseBits;
-
-    enum format {MONO_16_BIT, STEREO_16_BIT};
-    AudioResampler(int bitDepth, int inChannelCount, int32_t sampleRate);
-
-    // prevent copying
-    AudioResampler(const AudioResampler&);
-    AudioResampler& operator=(const AudioResampler&);
-
-    int32_t mBitDepth;
-    int32_t mChannelCount;
-    int32_t mSampleRate;
-    int32_t mInSampleRate;
-    AudioBufferProvider::Buffer mBuffer;
-    union {
-    	int16_t mVolume[2];
-    	uint32_t mVolumeRL;
-    };
-    int16_t mTargetVolume[2];
-    format mFormat;
-    size_t mInputIndex;
-    int32_t mPhaseIncrement;
-    uint32_t mPhaseFraction;
-};
-
-// ----------------------------------------------------------------------------
-}
-; // namespace android
-
-#endif // ANDROID_AUDIO_RESAMPLER_H