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authorMathias Agopian <mathias@google.com>2010-07-14 17:59:35 -0700
committerMathias Agopian <mathias@google.com>2010-07-14 17:59:35 -0700
commit65ab47156e1c7dfcd8cc4266253a5ff30219e7f0 (patch)
treeda088916c7cadadcddf47467f20cc8b0b6ff7f41 /services/audioflinger/AudioResampler.h
parent98fe12aba55e8b8bb14b23c4c3cdba9a99a9753c (diff)
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move native services under services/
moved surfaceflinger, audioflinger, cameraservice all native services should now reside in this location. Change-Id: Iee42b83dd2a94c3bf5107ab0895fe2dfcd5337a8
Diffstat (limited to 'services/audioflinger/AudioResampler.h')
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diff --git a/services/audioflinger/AudioResampler.h b/services/audioflinger/AudioResampler.h
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+/*
+ * 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