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      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();
     77     virtual size_t getUnreleasedFrames() const { return mInputIndex; }
     78 
     79     // called from destructor, so must not be virtual
     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     //
    142     inline size_t getInFrameCountRequired(size_t outFrameCount) {
    143         return (static_cast<uint64_t>(outFrameCount)*mInSampleRate
    144                 + (mSampleRate - 1))/mSampleRate;
    145     }
    146 
    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