From 65ab47156e1c7dfcd8cc4266253a5ff30219e7f0 Mon Sep 17 00:00:00 2001 From: Mathias Agopian Date: Wed, 14 Jul 2010 17:59:35 -0700 Subject: move native services under services/ moved surfaceflinger, audioflinger, cameraservice all native services should now reside in this location. Change-Id: Iee42b83dd2a94c3bf5107ab0895fe2dfcd5337a8 --- services/audioflinger/AudioResampler.h | 93 ++++++++++++++++++++++++++++++++++ 1 file changed, 93 insertions(+) create mode 100644 services/audioflinger/AudioResampler.h (limited to 'services/audioflinger/AudioResampler.h') diff --git a/services/audioflinger/AudioResampler.h b/services/audioflinger/AudioResampler.h new file mode 100644 index 0000000..2dfac76 --- /dev/null +++ b/services/audioflinger/AudioResampler.h @@ -0,0 +1,93 @@ +/* + * 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 +#include + +#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<