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318 lines
10 KiB
C++
318 lines
10 KiB
C++
/* ScummVM - Graphic Adventure Engine
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*
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* ScummVM is the legal property of its developers, whose names
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* are too numerous to list here. Please refer to the COPYRIGHT
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* file distributed with this source distribution.
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*
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* This program is free software: you can redistribute it and/or modify
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* it under the terms of the GNU General Public License as published by
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* the Free Software Foundation, either version 3 of the License, or
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* (at your option) any later version.
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*
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* This program is distributed in the hope that it will be useful,
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* but WITHOUT ANY WARRANTY; without even the implied warranty of
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* MERCHANTABILITY or FITNESS FOR A PARTICULAR PURPOSE. See the
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* GNU General Public License for more details.
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*
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* You should have received a copy of the GNU General Public License
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* along with this program. If not, see <http://www.gnu.org/licenses/>.
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*
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*/
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/*
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* The code in this file is based on code with Copyright 1998 Fabrice Bellard
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* Fabrice original code is part of SoX (http://sox.sourceforge.net).
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* Max Horn adapted that code to the needs of ScummVM and rewrote it partial,
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* in the process removing any use of floating point arithmetic. Various other
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* improvements over the original code were made.
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*/
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#include "audio/audiostream.h"
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#include "audio/rate.h"
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#include "audio/mixer.h"
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#include "common/util.h"
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namespace Audio {
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/**
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* The default fractional type in frac.h (with 16 fractional bits) limits
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* the rate conversion code to 65536Hz audio: we need to able to handle
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* 96kHz audio, so we use fewer fractional bits in this code.
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*/
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enum {
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FRAC_BITS_LOW = 15,
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FRAC_ONE_LOW = (1L << FRAC_BITS_LOW),
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FRAC_HALF_LOW = (1L << (FRAC_BITS_LOW-1))
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};
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template<bool inStereo, bool outStereo, bool reverseStereo>
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class RateConverter_Impl : public RateConverter {
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private:
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/** Input and output rates */
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st_rate_t _inRate, _outRate;
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/**
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* The intermediate input cache. Bigger values may increase performance,
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* but only until some point (depends largely on cache size, target
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* processor and various other factors), at which it will decrease again.
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*/
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st_sample_t _buffer[512];
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/** Current position inside the buffer */
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const st_sample_t *_bufferPos;
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/** Size of data currently loaded into the buffer */
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int _bufferSize;
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/** How far output is ahead of input when doing simple conversion */
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frac_t _outPos;
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/** Fractional position of the output stream in input stream unit */
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frac_t _outPosFrac;
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/** Last sample(s) in the input stream (left/right channel) */
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st_sample_t _inLastL, _inLastR;
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/** Current sample(s) in the input stream (left/right channel) */
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st_sample_t _inCurL, _inCurR;
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int copyConvert(AudioStream &input, st_sample_t *outBuffer, st_size_t numSamples, st_volume_t vol_l, st_volume_t vol_r);
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int simpleConvert(AudioStream &input, st_sample_t *outBuffer, st_size_t numSamples, st_volume_t vol_l, st_volume_t vol_r);
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int interpolateConvert(AudioStream &input, st_sample_t *outBuffer, st_size_t numSamples, st_volume_t vol_l, st_volume_t vol_r);
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public:
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RateConverter_Impl(st_rate_t inputRate, st_rate_t outputRate);
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virtual ~RateConverter_Impl() {}
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int convert(AudioStream &input, st_sample_t *outBuffer, st_size_t numSamples, st_volume_t vol_l, st_volume_t vol_r) override;
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void setInputRate(st_rate_t inputRate) override { _inRate = inputRate; }
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void setOutputRate(st_rate_t outputRate) override { _outRate = outputRate; }
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st_rate_t getInputRate() const override { return _inRate; }
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st_rate_t getOutputRate() const override { return _outRate; }
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bool needsDraining() const override { return _bufferSize != 0; }
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};
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template<bool inStereo, bool outStereo, bool reverseStereo>
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int RateConverter_Impl<inStereo, outStereo, reverseStereo>::copyConvert(AudioStream &input, st_sample_t *outBuffer, st_size_t numSamples, st_volume_t volL, st_volume_t volR) {
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st_sample_t *outStart, *outEnd;
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outStart = outBuffer;
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outEnd = outBuffer + numSamples * (outStereo ? 2 : 1);
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while (outBuffer < outEnd) {
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// Check if we have to refill the buffer
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if (_bufferSize == 0) {
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_bufferPos = _buffer;
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_bufferSize = input.readBuffer(_buffer, ARRAYSIZE(_buffer));
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if (_bufferSize <= 0)
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return (outBuffer - outStart) / (outStereo ? 2 : 1);
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}
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// Mix the data into the output buffer
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st_sample_t inL, inR;
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inL = *_bufferPos++;
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inR = (inStereo ? *_bufferPos++ : inL);
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_bufferSize -= (inStereo ? 2 : 1);
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st_sample_t outL, outR;
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outL = (inL * (int)volL) / Audio::Mixer::kMaxMixerVolume;
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outR = (inR * (int)volR) / Audio::Mixer::kMaxMixerVolume;
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if (outStereo) {
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// Output left channel
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clampedAdd(outBuffer[reverseStereo ], outL);
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// Output right channel
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clampedAdd(outBuffer[reverseStereo ^ 1], outR);
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outBuffer += 2;
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} else {
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// Output mono channel
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clampedAdd(outBuffer[0], (outL + outR) / 2);
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outBuffer += 1;
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}
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}
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return (outBuffer - outStart) / (outStereo ? 2 : 1);
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}
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template<bool inStereo, bool outStereo, bool reverseStereo>
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int RateConverter_Impl<inStereo, outStereo, reverseStereo>::simpleConvert(AudioStream &input, st_sample_t *outBuffer, st_size_t numSamples, st_volume_t volL, st_volume_t volR) {
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// How much to increment _outPos by
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frac_t outPos_inc = _inRate / _outRate;
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st_sample_t *outStart, *outEnd;
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outStart = outBuffer;
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outEnd = outBuffer + numSamples * (outStereo ? 2 : 1);
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while (outBuffer < outEnd) {
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// Read enough input samples so that _outPos >= 0
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do {
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// Check if we have to refill the buffer
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if (_bufferSize == 0) {
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_bufferPos = _buffer;
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_bufferSize = input.readBuffer(_buffer, ARRAYSIZE(_buffer));
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if (_bufferSize <= 0)
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return (outBuffer - outStart) / (outStereo ? 2 : 1);
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}
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_bufferSize -= (inStereo ? 2 : 1);
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_outPos--;
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if (_outPos >= 0) {
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_bufferPos += (inStereo ? 2 : 1);
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}
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} while (_outPos >= 0);
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st_sample_t inL, inR;
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inL = *_bufferPos++;
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inR = (inStereo ? *_bufferPos++ : inL);
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// Increment output position
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_outPos += outPos_inc;
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st_sample_t outL, outR;
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outL = (inL * (int)volL) / Audio::Mixer::kMaxMixerVolume;
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outR = (inR * (int)volR) / Audio::Mixer::kMaxMixerVolume;
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if (outStereo) {
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// output left channel
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clampedAdd(outBuffer[reverseStereo ], outL);
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// output right channel
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clampedAdd(outBuffer[reverseStereo ^ 1], outR);
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outBuffer += 2;
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} else {
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// output mono channel
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clampedAdd(outBuffer[0], (outL + outR) / 2);
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outBuffer += 1;
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}
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}
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return (outBuffer - outStart) / (outStereo ? 2 : 1);
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}
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template<bool inStereo, bool outStereo, bool reverseStereo>
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int RateConverter_Impl<inStereo, outStereo, reverseStereo>::interpolateConvert(AudioStream &input, st_sample_t *outBuffer, st_size_t numSamples, st_volume_t volL, st_volume_t volR) {
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// How much to increment _outPosFrac by
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frac_t outPos_inc = (_inRate << FRAC_BITS_LOW) / _outRate;
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st_sample_t *outStart, *outEnd;
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outStart = outBuffer;
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outEnd = outBuffer + numSamples * (outStereo ? 2 : 1);
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while (outBuffer < outEnd) {
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// Read enough input samples so that _outPosFrac < 0
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while ((frac_t)FRAC_ONE_LOW <= _outPosFrac) {
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// Check if we have to refill the buffer
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if (_bufferSize == 0) {
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_bufferPos = _buffer;
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_bufferSize = input.readBuffer(_buffer, ARRAYSIZE(_buffer));
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if (_bufferSize <= 0)
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return (outBuffer - outStart) / (outStereo ? 2 : 1);
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}
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_bufferSize -= (inStereo ? 2 : 1);
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_inLastL = _inCurL;
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_inCurL = *_bufferPos++;
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if (inStereo) {
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_inLastR = _inCurR;
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_inCurR = *_bufferPos++;
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}
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_outPosFrac -= FRAC_ONE_LOW;
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}
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// Loop as long as the _outPos trails behind, and as long as there is
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// still space in the output buffer.
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while (_outPosFrac < (frac_t)FRAC_ONE_LOW && outBuffer < outEnd) {
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// Interpolate
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st_sample_t inL, inR;
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inL = (st_sample_t)(_inLastL + (((_inCurL - _inLastL) * _outPosFrac + FRAC_HALF_LOW) >> FRAC_BITS_LOW));
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inR = (inStereo ?
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(st_sample_t)(_inLastR + (((_inCurR - _inLastR) * _outPosFrac + FRAC_HALF_LOW) >> FRAC_BITS_LOW)) :
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inL);
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st_sample_t outL, outR;
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outL = (inL * (int)volL) / Audio::Mixer::kMaxMixerVolume;
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outR = (inR * (int)volR) / Audio::Mixer::kMaxMixerVolume;
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if (outStereo) {
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// Output left channel
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clampedAdd(outBuffer[reverseStereo ], outL);
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// Output right channel
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clampedAdd(outBuffer[reverseStereo ^ 1], outR);
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outBuffer += 2;
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} else {
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// Output mono channel
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clampedAdd(outBuffer[0], (outL + outR) / 2);
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outBuffer += 1;
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}
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// Increment output position
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_outPosFrac += outPos_inc;
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}
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}
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return (outBuffer - outStart) / (outStereo ? 2 : 1);
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}
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template<bool inStereo, bool outStereo, bool reverseStereo>
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RateConverter_Impl<inStereo, outStereo, reverseStereo>::RateConverter_Impl(st_rate_t inputRate, st_rate_t outputRate) :
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_inRate(inputRate),
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_outRate(outputRate),
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_outPos(1),
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_outPosFrac(FRAC_ONE_LOW),
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_inLastL(0),
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_inLastR(0),
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_inCurL(0),
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_inCurR(0),
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_bufferSize(0),
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_bufferPos(nullptr) {}
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template<bool inStereo, bool outStereo, bool reverseStereo>
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int RateConverter_Impl<inStereo, outStereo, reverseStereo>::convert(AudioStream &input, st_sample_t *outBuffer, st_size_t numSamples, st_volume_t volL, st_volume_t volR) {
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assert(input.isStereo() == inStereo);
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if (_inRate == _outRate) {
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return copyConvert(input, outBuffer, numSamples, volL, volR);
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} else {
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if ((_inRate % _outRate) == 0 && (_inRate < 65536)) {
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return simpleConvert(input, outBuffer, numSamples, volL, volR);
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} else {
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return interpolateConvert(input, outBuffer, numSamples, volL, volR);
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}
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}
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}
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RateConverter *makeRateConverter(st_rate_t inRate, st_rate_t outRate, bool inStereo, bool outStereo, bool reverseStereo) {
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if (inStereo) {
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if (outStereo) {
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if (reverseStereo)
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return new RateConverter_Impl<true, true, true>(inRate, outRate);
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else
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return new RateConverter_Impl<true, true, false>(inRate, outRate);
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} else
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return new RateConverter_Impl<true, false, false>(inRate, outRate);
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} else {
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if (outStereo) {
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return new RateConverter_Impl<false, true, false>(inRate, outRate);
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} else
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return new RateConverter_Impl<false, false, false>(inRate, outRate);
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}
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}
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} // End of namespace Audio
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