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/*
* 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>
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 AudioResampler* create(int bitDepth, 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(int16_t left, int16_t 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 bitDepth, 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 mBitDepth;
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;
}
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
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