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 <cutils/compiler.h> 24 #include <utils/Compat.h> 25 26 #include <media/AudioBufferProvider.h> 27 #include <system/audio.h> 28 29 namespace android { 30 // ---------------------------------------------------------------------------- 31 32 class ANDROID_API AudioResampler { 33 public: 34 // Determines quality of SRC. 35 // LOW_QUALITY: linear interpolator (1st order) 36 // MED_QUALITY: cubic interpolator (3rd order) 37 // HIGH_QUALITY: fixed multi-tap FIR (e.g. 48KHz->44.1KHz) 38 // NOTE: high quality SRC will only be supported for 39 // certain fixed rate conversions. Sample rate cannot be 40 // changed dynamically. 41 enum src_quality { 42 DEFAULT_QUALITY=0, 43 LOW_QUALITY=1, 44 MED_QUALITY=2, 45 HIGH_QUALITY=3, 46 VERY_HIGH_QUALITY=4, 47 DYN_LOW_QUALITY=5, 48 DYN_MED_QUALITY=6, 49 DYN_HIGH_QUALITY=7, 50 }; 51 52 static const CONSTEXPR float UNITY_GAIN_FLOAT = 1.0f; 53 54 static AudioResampler* create(audio_format_t format, int inChannelCount, 55 int32_t sampleRate, src_quality quality=DEFAULT_QUALITY); 56 57 virtual ~AudioResampler(); 58 59 virtual void init() = 0; 60 virtual void setSampleRate(int32_t inSampleRate); 61 virtual void setVolume(float left, float right); 62 virtual void setLocalTimeFreq(uint64_t freq); 63 64 // set the PTS of the next buffer output by the resampler 65 virtual void setPTS(int64_t pts); 66 67 // Resample int16_t samples from provider and accumulate into 'out'. 68 // A mono provider delivers a sequence of samples. 69 // A stereo provider delivers a sequence of interleaved pairs of samples. 70 // 71 // In either case, 'out' holds interleaved pairs of fixed-point Q4.27. 72 // That is, for a mono provider, there is an implicit up-channeling. 73 // Since this method accumulates, the caller is responsible for clearing 'out' initially. 74 // 75 // For a float resampler, 'out' holds interleaved pairs of float samples. 76 // 77 // Multichannel interleaved frames for n > 2 is supported for quality DYN_LOW_QUALITY, 78 // DYN_MED_QUALITY, and DYN_HIGH_QUALITY. 79 // 80 // Returns the number of frames resampled into the out buffer. 81 virtual size_t resample(int32_t* out, size_t outFrameCount, 82 AudioBufferProvider* provider) = 0; 83 84 virtual void reset(); 85 virtual size_t getUnreleasedFrames() const { return mInputIndex; } 86 87 // called from destructor, so must not be virtual 88 src_quality getQuality() const { return mQuality; } 89 90 protected: 91 // number of bits for phase fraction - 30 bits allows nearly 2x downsampling 92 static const int kNumPhaseBits = 30; 93 94 // phase mask for fraction 95 static const uint32_t kPhaseMask = (1LU<<kNumPhaseBits)-1; 96 97 // multiplier to calculate fixed point phase increment 98 static const double kPhaseMultiplier; 99 100 AudioResampler(int inChannelCount, int32_t sampleRate, src_quality quality); 101 102 // prevent copying 103 AudioResampler(const AudioResampler&); 104 AudioResampler& operator=(const AudioResampler&); 105 106 int64_t calculateOutputPTS(int outputFrameIndex); 107 108 const int32_t mChannelCount; 109 const int32_t mSampleRate; 110 int32_t mInSampleRate; 111 AudioBufferProvider::Buffer mBuffer; 112 union { 113 int16_t mVolume[2]; 114 uint32_t mVolumeRL; 115 }; 116 int16_t mTargetVolume[2]; 117 size_t mInputIndex; 118 int32_t mPhaseIncrement; 119 uint32_t mPhaseFraction; 120 uint64_t mLocalTimeFreq; 121 int64_t mPTS; 122 123 // returns the inFrameCount required to generate outFrameCount frames. 124 // 125 // Placed here to be a consistent for all resamplers. 126 // 127 // Right now, we use the upper bound without regards to the current state of the 128 // input buffer using integer arithmetic, as follows: 129 // 130 // (static_cast<uint64_t>(outFrameCount)*mInSampleRate + (mSampleRate - 1))/mSampleRate; 131 // 132 // The double precision equivalent (float may not be precise enough): 133 // ceil(static_cast<double>(outFrameCount) * mInSampleRate / mSampleRate); 134 // 135 // this relies on the fact that the mPhaseIncrement is rounded down from 136 // #phases * mInSampleRate/mSampleRate and the fact that Sum(Floor(x)) <= Floor(Sum(x)). 137 // http://www.proofwiki.org/wiki/Sum_of_Floors_Not_Greater_Than_Floor_of_Sums 138 // 139 // (so long as double precision is computed accurately enough to be considered 140 // greater than or equal to the Floor(x) value in int32_t arithmetic; thus this 141 // will not necessarily hold for floats). 142 // 143 // TODO: 144 // Greater accuracy and a tight bound is obtained by: 145 // 1) subtract and adjust for the current state of the AudioBufferProvider buffer. 146 // 2) using the exact integer formula where (ignoring 64b casting) 147 // inFrameCount = (mPhaseIncrement * (outFrameCount - 1) + mPhaseFraction) / phaseWrapLimit; 148 // phaseWrapLimit is the wraparound (1 << kNumPhaseBits), if not specified explicitly. 149 // 150 inline size_t getInFrameCountRequired(size_t outFrameCount) { 151 return (static_cast<uint64_t>(outFrameCount)*mInSampleRate 152 + (mSampleRate - 1))/mSampleRate; 153 } 154 155 inline float clampFloatVol(float volume) { 156 if (volume > UNITY_GAIN_FLOAT) { 157 return UNITY_GAIN_FLOAT; 158 } else if (volume >= 0.) { 159 return volume; 160 } 161 return 0.; // NaN or negative volume maps to 0. 162 } 163 164 private: 165 const src_quality mQuality; 166 167 // Return 'true' if the quality level is supported without explicit request 168 static bool qualityIsSupported(src_quality quality); 169 170 // For pthread_once() 171 static void init_routine(); 172 173 // Return the estimated CPU load for specific resampler in MHz. 174 // The absolute number is irrelevant, it's the relative values that matter. 175 static uint32_t qualityMHz(src_quality quality); 176 }; 177 178 // ---------------------------------------------------------------------------- 179 } // namespace android 180 181 #endif // ANDROID_AUDIO_RESAMPLER_H 182
