text: /*
   * 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 <cutils/compiler.h>
  
  #include <media/AudioBufferProvider.h>
  #include <system/audio.h>
  
  namespace android {
  // ----------------------------------------------------------------------------
  
  class ANDROID_API 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_QUALITY=0,
          LOW_QUALITY=1,
          MED_QUALITY=2,
          HIGH_QUALITY=3,
          VERY_HIGH_QUALITY=4,
          DYN_LOW_QUALITY=5,
          DYN_MED_QUALITY=6,
          DYN_HIGH_QUALITY=7,
      };
  
      static const float UNITY_GAIN_FLOAT = 1.0f;
  
      static AudioResampler* create(audio_format_t format, int inChannelCount,
              int32_t sampleRate, src_quality quality=DEFAULT_QUALITY);
  
      virtual ~AudioResampler();
  
      virtual void init() = 0;
      virtual void setSampleRate(int32_t inSampleRate);
      virtual void setVolume(float left, float right);
      virtual void setLocalTimeFreq(uint64_t freq);
  
      // set the PTS of the next buffer output by the resampler
      virtual void setPTS(int64_t pts);
  
      // Resample int16_t samples from provider and accumulate into 'out'.
      // A mono provider delivers a sequence of samples.
      // A stereo provider delivers a sequence of interleaved pairs of samples.
      // Multi-channel providers are not supported.
      // In either case, 'out' holds interleaved pairs of fixed-point Q4.27.
      // That is, for a mono provider, there is an implicit up-channeling.
      // Since this method accumulates, the caller is responsible for clearing 'out' initially.
      // FIXME assumes provider is always successful; it should return the actual frame count.
      virtual void resample(int32_t* out, size_t outFrameCount,
              AudioBufferProvider* provider) = 0;
  
      virtual void reset();
      virtual size_t getUnreleasedFrames() const { return mInputIndex; }
  
      // called from destructor, so must not be virtual
      src_quality getQuality() const { return mQuality; }
  
  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;
  
      AudioResampler(int inChannelCount, int32_t sampleRate, src_quality quality);
  
      // prevent copying
      AudioResampler(const AudioResampler&);
      AudioResampler& operator=(const AudioResampler&);
  
      int64_t calculateOutputPTS(int outputFrameIndex);
  
      const int32_t mChannelCount;
      const 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;
      uint64_t mLocalTimeFreq;
      int64_t mPTS;
  
      // returns the inFrameCount required to generate outFrameCount frames.
      //
      // Placed here to be a consistent for all resamplers.
      //
      // Right now, we use the upper bound without regards to the current state of the
      // input buffer using integer arithmetic, as follows:
      //
      // (static_cast<uint64_t>(outFrameCount)*mInSampleRate + (mSampleRate - 1))/mSampleRate;
      //
      // The double precision equivalent (float may not be precise enough):
      // ceil(static_cast<double>(outFrameCount) * mInSampleRate / mSampleRate);
      //
      // this relies on the fact that the mPhaseIncrement is rounded down from
      // #phases * mInSampleRate/mSampleRate and the fact that Sum(Floor(x)) <= Floor(Sum(x)).
      // http://www.proofwiki.org/wiki/Sum_of_Floors_Not_Greater_Than_Floor_of_Sums
      //
      // (so long as double precision is computed accurately enough to be considered
      // greater than or equal to the Floor(x) value in int32_t arithmetic; thus this
      // will not necessarily hold for floats).
      //
      // TODO:
      // Greater accuracy and a tight bound is obtained by:
      // 1) subtract and adjust for the current state of the AudioBufferProvider buffer.
      // 2) using the exact integer formula where (ignoring 64b casting)
      //  inFrameCount = (mPhaseIncrement * (outFrameCount - 1) + mPhaseFraction) / phaseWrapLimit;
      //  phaseWrapLimit is the wraparound (1 << kNumPhaseBits), if not specified explicitly.
      //
      inline size_t getInFrameCountRequired(size_t outFrameCount) {
          return (static_cast<uint64_t>(outFrameCount)*mInSampleRate
                  + (mSampleRate - 1))/mSampleRate;
      }
  
      inline float clampFloatVol(float volume) {
          if (volume > UNITY_GAIN_FLOAT) {
              return UNITY_GAIN_FLOAT;
          } else if (volume >= 0.) {
              return volume;
          }
          return 0.;  // NaN or negative volume maps to 0.
      }
  
  private:
      const src_quality mQuality;
  
      // Return 'true' if the quality level is supported without explicit request
      static bool qualityIsSupported(src_quality quality);
  
      // For pthread_once()
      static void init_routine();
  
      // Return the estimated CPU load for specific resampler in MHz.
      // The absolute number is irrelevant, it's the relative values that matter.
      static uint32_t qualityMHz(src_quality quality);
  };
  
  // ----------------------------------------------------------------------------
  }
  ; // namespace android
  
  #endif // ANDROID_AUDIO_RESAMPLER_H
  