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 #include <cutils/compiler.h> 23 24 #include <media/AudioBufferProvider.h> 25 #include <system/audio.h> 26 27 namespace android { 28 // ---------------------------------------------------------------------------- 29 30 class ANDROID_API AudioResampler { 31 public: 32 // Determines quality of SRC. 33 // LOW_QUALITY: linear interpolator (1st order) 34 // MED_QUALITY: cubic interpolator (3rd order) 35 // HIGH_QUALITY: fixed multi-tap FIR (e.g. 48KHz->44.1KHz) 36 // NOTE: high quality SRC will only be supported for 37 // certain fixed rate conversions. Sample rate cannot be 38 // changed dynamically. 39 enum src_quality { 40 DEFAULT_QUALITY=0, 41 LOW_QUALITY=1, 42 MED_QUALITY=2, 43 HIGH_QUALITY=3, 44 VERY_HIGH_QUALITY=4, 45 DYN_LOW_QUALITY=5, 46 DYN_MED_QUALITY=6, 47 DYN_HIGH_QUALITY=7, 48 }; 49 50 static const float UNITY_GAIN_FLOAT = 1.0f; 51 52 static AudioResampler* create(audio_format_t format, int inChannelCount, 53 int32_t sampleRate, src_quality quality=DEFAULT_QUALITY); 54 55 virtual ~AudioResampler(); 56 57 virtual void init() = 0; 58 virtual void setSampleRate(int32_t inSampleRate); 59 virtual void setVolume(float left, float right); 60 virtual void setLocalTimeFreq(uint64_t freq); 61 62 // set the PTS of the next buffer output by the resampler 63 virtual void setPTS(int64_t pts); 64 65 // Resample int16_t samples from provider and accumulate into 'out'. 66 // A mono provider delivers a sequence of samples. 67 // A stereo provider delivers a sequence of interleaved pairs of samples. 68 // Multi-channel providers are not supported. 69 // In either case, 'out' holds interleaved pairs of fixed-point Q4.27. 70 // That is, for a mono provider, there is an implicit up-channeling. 71 // Since this method accumulates, the caller is responsible for clearing 'out' initially. 72 // FIXME assumes provider is always successful; it should return the actual frame count. 73 virtual void resample(int32_t* out, size_t outFrameCount, 74 AudioBufferProvider* provider) = 0; 75 76 virtual void reset(); getUnreleasedFrames()77 virtual size_t getUnreleasedFrames() const { return mInputIndex; } 78 79 // called from destructor, so must not be virtual getQuality()80 src_quality getQuality() const { return mQuality; } 81 82 protected: 83 // number of bits for phase fraction - 30 bits allows nearly 2x downsampling 84 static const int kNumPhaseBits = 30; 85 86 // phase mask for fraction 87 static const uint32_t kPhaseMask = (1LU<<kNumPhaseBits)-1; 88 89 // multiplier to calculate fixed point phase increment 90 static const double kPhaseMultiplier; 91 92 AudioResampler(int inChannelCount, int32_t sampleRate, src_quality quality); 93 94 // prevent copying 95 AudioResampler(const AudioResampler&); 96 AudioResampler& operator=(const AudioResampler&); 97 98 int64_t calculateOutputPTS(int outputFrameIndex); 99 100 const int32_t mChannelCount; 101 const int32_t mSampleRate; 102 int32_t mInSampleRate; 103 AudioBufferProvider::Buffer mBuffer; 104 union { 105 int16_t mVolume[2]; 106 uint32_t mVolumeRL; 107 }; 108 int16_t mTargetVolume[2]; 109 size_t mInputIndex; 110 int32_t mPhaseIncrement; 111 uint32_t mPhaseFraction; 112 uint64_t mLocalTimeFreq; 113 int64_t mPTS; 114 115 // returns the inFrameCount required to generate outFrameCount frames. 116 // 117 // Placed here to be a consistent for all resamplers. 118 // 119 // Right now, we use the upper bound without regards to the current state of the 120 // input buffer using integer arithmetic, as follows: 121 // 122 // (static_cast<uint64_t>(outFrameCount)*mInSampleRate + (mSampleRate - 1))/mSampleRate; 123 // 124 // The double precision equivalent (float may not be precise enough): 125 // ceil(static_cast<double>(outFrameCount) * mInSampleRate / mSampleRate); 126 // 127 // this relies on the fact that the mPhaseIncrement is rounded down from 128 // #phases * mInSampleRate/mSampleRate and the fact that Sum(Floor(x)) <= Floor(Sum(x)). 129 // http://www.proofwiki.org/wiki/Sum_of_Floors_Not_Greater_Than_Floor_of_Sums 130 // 131 // (so long as double precision is computed accurately enough to be considered 132 // greater than or equal to the Floor(x) value in int32_t arithmetic; thus this 133 // will not necessarily hold for floats). 134 // 135 // TODO: 136 // Greater accuracy and a tight bound is obtained by: 137 // 1) subtract and adjust for the current state of the AudioBufferProvider buffer. 138 // 2) using the exact integer formula where (ignoring 64b casting) 139 // inFrameCount = (mPhaseIncrement * (outFrameCount - 1) + mPhaseFraction) / phaseWrapLimit; 140 // phaseWrapLimit is the wraparound (1 << kNumPhaseBits), if not specified explicitly. 141 // getInFrameCountRequired(size_t outFrameCount)142 inline size_t getInFrameCountRequired(size_t outFrameCount) { 143 return (static_cast<uint64_t>(outFrameCount)*mInSampleRate 144 + (mSampleRate - 1))/mSampleRate; 145 } 146 clampFloatVol(float volume)147 inline float clampFloatVol(float volume) { 148 if (volume > UNITY_GAIN_FLOAT) { 149 return UNITY_GAIN_FLOAT; 150 } else if (volume >= 0.) { 151 return volume; 152 } 153 return 0.; // NaN or negative volume maps to 0. 154 } 155 156 private: 157 const src_quality mQuality; 158 159 // Return 'true' if the quality level is supported without explicit request 160 static bool qualityIsSupported(src_quality quality); 161 162 // For pthread_once() 163 static void init_routine(); 164 165 // Return the estimated CPU load for specific resampler in MHz. 166 // The absolute number is irrelevant, it's the relative values that matter. 167 static uint32_t qualityMHz(src_quality quality); 168 }; 169 170 // ---------------------------------------------------------------------------- 171 } 172 ; // namespace android 173 174 #endif // ANDROID_AUDIO_RESAMPLER_H 175