AudioResampler.h revision e53b9ead781c36e96d6b6f012ddffc93a3d80f0d
1/* 2 * Copyright (C) 2007 The Android Open Source Project 3 * 4 * Licensed under the Apache License, Version 2.0 (the "License"); 5 * you may not use this file except in compliance with the License. 6 * You may obtain a copy of the License at 7 * 8 * http://www.apache.org/licenses/LICENSE-2.0 9 * 10 * Unless required by applicable law or agreed to in writing, software 11 * distributed under the License is distributed on an "AS IS" BASIS, 12 * WITHOUT WARRANTIES OR CONDITIONS OF ANY KIND, either express or implied. 13 * See the License for the specific language governing permissions and 14 * limitations under the License. 15 */ 16 17#ifndef ANDROID_AUDIO_RESAMPLER_H 18#define ANDROID_AUDIO_RESAMPLER_H 19 20#include <stdint.h> 21#include <sys/types.h> 22 23#include "AudioBufferProvider.h" 24 25namespace android { 26// ---------------------------------------------------------------------------- 27 28class AudioResampler { 29public: 30 // Determines quality of SRC. 31 // LOW_QUALITY: linear interpolator (1st order) 32 // MED_QUALITY: cubic interpolator (3rd order) 33 // HIGH_QUALITY: fixed multi-tap FIR (e.g. 48KHz->44.1KHz) 34 // NOTE: high quality SRC will only be supported for 35 // certain fixed rate conversions. Sample rate cannot be 36 // changed dynamically. 37 enum src_quality { 38 DEFAULT=0, 39 LOW_QUALITY=1, 40 MED_QUALITY=2, 41 HIGH_QUALITY=3 42 }; 43 44 static AudioResampler* create(int bitDepth, int inChannelCount, 45 int32_t sampleRate, int quality=DEFAULT); 46 47 virtual ~AudioResampler(); 48 49 virtual void init() = 0; 50 virtual void setSampleRate(int32_t inSampleRate); 51 virtual void setVolume(int16_t left, int16_t right); 52 virtual void setLocalTimeFreq(uint64_t freq); 53 54 // set the PTS of the next buffer output by the resampler 55 virtual void setPTS(int64_t pts); 56 57 virtual void resample(int32_t* out, size_t outFrameCount, 58 AudioBufferProvider* provider) = 0; 59 60 virtual void reset(); 61 virtual size_t getUnreleasedFrames() const { return mInputIndex; } 62 63protected: 64 // number of bits for phase fraction - 30 bits allows nearly 2x downsampling 65 static const int kNumPhaseBits = 30; 66 67 // phase mask for fraction 68 static const uint32_t kPhaseMask = (1LU<<kNumPhaseBits)-1; 69 70 // multiplier to calculate fixed point phase increment 71 static const double kPhaseMultiplier = 1L << kNumPhaseBits; 72 73 AudioResampler(int bitDepth, int inChannelCount, int32_t sampleRate); 74 75 // prevent copying 76 AudioResampler(const AudioResampler&); 77 AudioResampler& operator=(const AudioResampler&); 78 79 int64_t calculateOutputPTS(int outputFrameIndex); 80 81 const int32_t mBitDepth; 82 const int32_t mChannelCount; 83 const int32_t mSampleRate; 84 int32_t mInSampleRate; 85 AudioBufferProvider::Buffer mBuffer; 86 union { 87 int16_t mVolume[2]; 88 uint32_t mVolumeRL; 89 }; 90 int16_t mTargetVolume[2]; 91 size_t mInputIndex; 92 int32_t mPhaseIncrement; 93 uint32_t mPhaseFraction; 94 uint64_t mLocalTimeFreq; 95 int64_t mPTS; 96}; 97 98// ---------------------------------------------------------------------------- 99} 100; // namespace android 101 102#endif // ANDROID_AUDIO_RESAMPLER_H 103