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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,
};
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 signed Q19.12.
// 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 = 1L << kNumPhaseBits;
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;
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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