The Android Open Source Project | edbf3b6 | 2009-03-03 19:31:44 -0800 | [diff] [blame] | 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 | |
| 25 | namespace android { |
| 26 | // ---------------------------------------------------------------------------- |
| 27 | |
| 28 | class AudioResampler { |
| 29 | public: |
| 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 | |
| 53 | virtual void resample(int32_t* out, size_t outFrameCount, |
| 54 | AudioBufferProvider* provider) = 0; |
| 55 | |
| 56 | protected: |
| 57 | // number of bits for phase fraction - 30 bits allows nearly 2x downsampling |
| 58 | static const int kNumPhaseBits = 30; |
| 59 | |
| 60 | // phase mask for fraction |
| 61 | static const uint32_t kPhaseMask = (1LU<<kNumPhaseBits)-1; |
| 62 | |
| 63 | // multiplier to calculate fixed point phase increment |
| 64 | static const double kPhaseMultiplier = 1L << kNumPhaseBits; |
| 65 | |
| 66 | enum format {MONO_16_BIT, STEREO_16_BIT}; |
| 67 | AudioResampler(int bitDepth, int inChannelCount, int32_t sampleRate); |
| 68 | |
| 69 | // prevent copying |
| 70 | AudioResampler(const AudioResampler&); |
| 71 | AudioResampler& operator=(const AudioResampler&); |
| 72 | |
| 73 | int32_t mBitDepth; |
| 74 | int32_t mChannelCount; |
| 75 | int32_t mSampleRate; |
| 76 | int32_t mInSampleRate; |
| 77 | AudioBufferProvider::Buffer mBuffer; |
| 78 | union { |
| 79 | int16_t mVolume[2]; |
| 80 | uint32_t mVolumeRL; |
| 81 | }; |
| 82 | int16_t mTargetVolume[2]; |
| 83 | format mFormat; |
| 84 | size_t mInputIndex; |
| 85 | int32_t mPhaseIncrement; |
| 86 | uint32_t mPhaseFraction; |
| 87 | }; |
| 88 | |
| 89 | // ---------------------------------------------------------------------------- |
| 90 | } |
| 91 | ; // namespace android |
| 92 | |
| 93 | #endif // ANDROID_AUDIO_RESAMPLER_H |