18db49a462
Fix indentation to be multiple of 4. Make it easier to search: sp< not sp < to "switch (...)" instead of "switch(...)" (also "if" and "while") Remove redundant blank line at start or EOF. Remove whitespace at end of line. Remove extra blank lines where they don't add value. Use git diff -b or -w to verify. Change-Id: I966b7ba852faa5474be6907fb212f5e267c2874e
103 lines
3.1 KiB
C++
103 lines
3.1 KiB
C++
/*
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* Copyright (C) 2007 The Android Open Source Project
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*
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* Licensed under the Apache License, Version 2.0 (the "License");
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* you may not use this file except in compliance with the License.
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* You may obtain a copy of the License at
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*
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* http://www.apache.org/licenses/LICENSE-2.0
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*
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* Unless required by applicable law or agreed to in writing, software
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* distributed under the License is distributed on an "AS IS" BASIS,
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* WITHOUT WARRANTIES OR CONDITIONS OF ANY KIND, either express or implied.
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* See the License for the specific language governing permissions and
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* limitations under the License.
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*/
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#ifndef ANDROID_AUDIO_RESAMPLER_H
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#define ANDROID_AUDIO_RESAMPLER_H
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#include <stdint.h>
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#include <sys/types.h>
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#include "AudioBufferProvider.h"
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namespace android {
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// ----------------------------------------------------------------------------
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class AudioResampler {
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public:
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// Determines quality of SRC.
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// LOW_QUALITY: linear interpolator (1st order)
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// MED_QUALITY: cubic interpolator (3rd order)
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// HIGH_QUALITY: fixed multi-tap FIR (e.g. 48KHz->44.1KHz)
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// NOTE: high quality SRC will only be supported for
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// certain fixed rate conversions. Sample rate cannot be
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// changed dynamically.
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enum src_quality {
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DEFAULT=0,
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LOW_QUALITY=1,
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MED_QUALITY=2,
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HIGH_QUALITY=3
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};
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static AudioResampler* create(int bitDepth, int inChannelCount,
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int32_t sampleRate, int quality=DEFAULT);
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virtual ~AudioResampler();
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virtual void init() = 0;
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virtual void setSampleRate(int32_t inSampleRate);
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virtual void setVolume(int16_t left, int16_t right);
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virtual void setLocalTimeFreq(uint64_t freq);
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// set the PTS of the next buffer output by the resampler
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virtual void setPTS(int64_t pts);
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virtual void resample(int32_t* out, size_t outFrameCount,
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AudioBufferProvider* provider) = 0;
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virtual void reset();
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virtual size_t getUnreleasedFrames() const { return mInputIndex; }
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protected:
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// number of bits for phase fraction - 30 bits allows nearly 2x downsampling
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static const int kNumPhaseBits = 30;
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// phase mask for fraction
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static const uint32_t kPhaseMask = (1LU<<kNumPhaseBits)-1;
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// multiplier to calculate fixed point phase increment
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static const double kPhaseMultiplier = 1L << kNumPhaseBits;
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AudioResampler(int bitDepth, int inChannelCount, int32_t sampleRate);
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// prevent copying
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AudioResampler(const AudioResampler&);
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AudioResampler& operator=(const AudioResampler&);
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int64_t calculateOutputPTS(int outputFrameIndex);
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const int32_t mBitDepth;
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const int32_t mChannelCount;
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const int32_t mSampleRate;
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int32_t mInSampleRate;
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AudioBufferProvider::Buffer mBuffer;
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union {
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int16_t mVolume[2];
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uint32_t mVolumeRL;
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};
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int16_t mTargetVolume[2];
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size_t mInputIndex;
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int32_t mPhaseIncrement;
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uint32_t mPhaseFraction;
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uint64_t mLocalTimeFreq;
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int64_t mPTS;
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};
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// ----------------------------------------------------------------------------
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}
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; // namespace android
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#endif // ANDROID_AUDIO_RESAMPLER_H
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