| /* |
| * 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; |
| |
| virtual void reset(); |
| virtual size_t getUnreleasedFrames() const { return mInputIndex; } |
| |
| 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; |
| |
| 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]; |
| size_t mInputIndex; |
| int32_t mPhaseIncrement; |
| uint32_t mPhaseFraction; |
| }; |
| |
| // ---------------------------------------------------------------------------- |
| } |
| ; // namespace android |
| |
| #endif // ANDROID_AUDIO_RESAMPLER_H |