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https://github.com/OpenRCT2/OpenRCT2
synced 2026-01-18 04:23:20 +01:00
A new custom software mixer, supports playing streams at any rate, which is needed to emulate the directsound SetFrequency effects (when a ride malfunctions and the music slows down and stops). The speex resampler is added for this. Currently samples can be played and their volumes and looping set on an arbitrary number of channels. Support for FLAC and other streaming formats can be added.
This commit is contained in:
zsilencer
401
src/mixer.cpp
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401
src/mixer.cpp
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/*****************************************************************************
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* Copyright (c) 2014 Ted John
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* OpenRCT2, an open source clone of Roller Coaster Tycoon 2.
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*
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* This file is part of OpenRCT2.
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*
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* OpenRCT2 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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* 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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* 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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#include <math.h>
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#include <SDL.h>
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#include <string.h>
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extern "C" {
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#include "audio.h"
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#include "config.h"
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}
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#include "mixer.h"
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Sample::Sample()
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{
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data = 0;
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length = 0;
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issdlwav = false;
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}
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Sample::~Sample()
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{
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Unload();
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}
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bool Sample::Load(char* filename)
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{
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Unload();
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SDL_RWops* rw = SDL_RWFromFile(filename, "rb");
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if (!rw) {
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SDL_RWclose(rw);
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return false;
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}
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SDL_AudioSpec audiospec;
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memset(&audiospec, 0, sizeof(audiospec));
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SDL_AudioSpec* spec = SDL_LoadWAV_RW(rw, false, &audiospec, &data, &length);
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if (spec != NULL) {
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format.freq = spec->freq;
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format.format = spec->format;
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format.channels = spec->channels;
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} else {
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issdlwav = true;
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}
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return true;
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}
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bool Sample::LoadCSS1(char* filename, unsigned int offset)
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{
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Unload();
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SDL_RWops* rw = SDL_RWFromFile(filename, "rb");
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if (!rw) {
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return false;
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}
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Uint32 numsounds;
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SDL_RWread(rw, &numsounds, sizeof(numsounds), 1);
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if (offset > numsounds) {
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SDL_RWclose(rw);
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return false;
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}
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SDL_RWseek(rw, offset * 4, RW_SEEK_CUR);
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Uint32 soundoffset;
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SDL_RWread(rw, &soundoffset, sizeof(soundoffset), 1);
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SDL_RWseek(rw, soundoffset, RW_SEEK_SET);
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Uint32 soundsize;
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SDL_RWread(rw, &soundsize, sizeof(soundsize), 1);
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length = soundsize;
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WAVEFORMATEX waveformat;
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SDL_RWread(rw, &waveformat, sizeof(waveformat), 1);
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format.freq = waveformat.nSamplesPerSec;
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format.format = AUDIO_S16;
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format.channels = (Uint8)waveformat.nChannels;
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data = new Uint8[length];
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SDL_RWread(rw, data, length, 1);
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SDL_RWclose(rw);
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return true;
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}
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void Sample::Unload()
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{
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SDL_LockAudio();
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if (data) {
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if (issdlwav) {
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SDL_FreeWAV(data);
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} else {
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delete[] data;
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}
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data = 0;
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}
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issdlwav = false;
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length = 0;
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SDL_UnlockAudio();
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}
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bool Sample::Convert(AudioFormat format)
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{
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if(Sample::format.format != format.format || Sample::format.channels != format.channels || Sample::format.freq != format.freq){
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SDL_AudioCVT cvt;
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if (SDL_BuildAudioCVT(&cvt, Sample::format.format, Sample::format.channels, Sample::format.freq, format.format, format.channels, format.freq) < 0) {
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return false;
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}
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cvt.len = length;
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cvt.buf = new Uint8[cvt.len * cvt.len_mult];
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memcpy(cvt.buf, data, length);
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if (SDL_ConvertAudio(&cvt) < 0) {
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delete[] cvt.buf;
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return false;
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}
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Unload();
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data = cvt.buf;
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length = cvt.len_cvt;
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Sample::format = format;
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}
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return true;
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}
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const Uint8* Sample::Data()
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{
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return data;
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}
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int Sample::Length()
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{
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return length;
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}
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Stream::Stream()
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{
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sourcetype = SOURCE_NONE;
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}
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int Stream::GetSome(int offset, const Uint8** data, int length)
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{
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int size = length;
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switch(sourcetype) {
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case SOURCE_SAMPLE:
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if (offset >= sample->Length()) {
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return 0;
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}
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if (offset + length > sample->Length()) {
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size = sample->Length() - offset;
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}
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*data = &sample->Data()[offset];
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return size;
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break;
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}
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return 0;
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}
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int Stream::Length()
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{
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switch(sourcetype) {
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case SOURCE_SAMPLE:
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return sample->Length();
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break;
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}
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return 0;
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}
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void Stream::SetSource_Sample(Sample& sample)
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{
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sourcetype = SOURCE_SAMPLE;
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Stream::sample = &sample;
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}
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const AudioFormat* Stream::Format()
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{
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switch(sourcetype) {
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case SOURCE_SAMPLE:
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return &sample->format;
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break;
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}
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return 0;
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}
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Channel::Channel()
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{
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rate = 1;
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resampler = 0;
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SetVolume(SDL_MIX_MAXVOLUME);
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}
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Channel::~Channel()
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{
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if (resampler) {
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speex_resampler_destroy(resampler);
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resampler = 0;
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}
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}
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void Channel::Play(Stream& stream, int loop = 0)
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{
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Channel::stream = &stream;
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Channel::loop = loop;
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offset = 0;
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}
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void Channel::SetRate(double rate)
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{
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Channel::rate = rate;
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if (Channel::rate < 0.001) {
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Channel::rate = 0.001;
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}
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}
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void Channel::SetVolume(int volume)
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{
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Channel::volume = volume;
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if (Channel::volume > SDL_MIX_MAXVOLUME) {
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Channel::volume = SDL_MIX_MAXVOLUME;
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}
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if (Channel::volume < 0) {
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Channel::volume = 0;
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}
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}
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void Mixer::Init(char* device)
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{
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Close();
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SDL_AudioSpec want, have;
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SDL_zero(want);
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want.freq = 22050;
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want.format = AUDIO_S16;
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want.channels = 2;
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want.samples = 1024;
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want.callback = Callback;
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want.userdata = this;
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deviceid = SDL_OpenAudioDevice(device, 0, &want, &have, 0);
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format.format = have.format;
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format.channels = have.channels;
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format.freq = have.freq;
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char css1filename[260];
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strcpy(css1filename, gGeneral_config.game_path);
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strcat(css1filename, "\\Data\\css1.dat");
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for (int i = 0; i < 63; i++) {
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css1samples[i].LoadCSS1(css1filename, i);
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css1samples[i].Convert(format); // convert to audio output format, saves some cpu usage but requires a bit more memory, optional
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css1streams[i].SetSource_Sample(css1samples[i]);
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}
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SDL_PauseAudioDevice(deviceid, 0);
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}
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void Mixer::Close()
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{
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SDL_CloseAudioDevice(deviceid);
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}
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void SDLCALL Mixer::Callback(void* arg, Uint8* stream, int length)
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{
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Mixer* mixer = (Mixer*)arg;
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memset(stream, 0, length);
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for (int i = 0; i < 10; i++) {
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mixer->MixChannel(mixer->channels[i], stream, length);
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}
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}
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void Mixer::MixChannel(Channel& channel, Uint8* data, int length)
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{
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if (channel.stream) {
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if (!channel.resampler) {
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channel.resampler = speex_resampler_init(format.channels, format.freq, format.freq, 0, 0);
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}
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AudioFormat channelformat = *channel.stream->Format();
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int loaded = 0;
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SDL_AudioCVT cvt;
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cvt.len_ratio = 1;
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do {
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int samplesize = format.channels * format.BytesPerSample();
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int samples = length / samplesize;
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int samplesloaded = loaded / samplesize;
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int samplestoread = (int)ceil((samples - samplesloaded) * channel.rate);
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int lengthloaded = 0;
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if (channel.offset < channel.stream->Length()) {
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bool mustconvert = false;
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if (MustConvert(*channel.stream)) {
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if (SDL_BuildAudioCVT(&cvt, channelformat.format, channelformat.channels, channelformat.freq, Mixer::format.format, Mixer::format.channels, Mixer::format.freq) == -1) {
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break;
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}
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mustconvert = true;
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}
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const Uint8* datastream;
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int readfromstream = (channel.stream->GetSome(channel.offset, &datastream, (int)(((samplestoread) * samplesize) / cvt.len_ratio)) / channelformat.BytesPerSample()) * channelformat.BytesPerSample();
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if (readfromstream == 0) {
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break;
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}
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int volume = channel.volume;
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if (mustconvert) {
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Uint8* dataconverted;
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if (Convert(cvt, datastream, readfromstream, &dataconverted)) {
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if (channel.rate != 1 && format.format == AUDIO_S16) {
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spx_uint32_t in_len = (spx_uint32_t)(ceil((double)cvt.len_cvt / samplesize));
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Uint8* out = new Uint8[length + 200]; // needs some extra, otherwise resampler sometimes doesn't process all the input samples
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spx_uint32_t out_len = samples + 20;
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speex_resampler_set_rate(channel.resampler, format.freq, (int)(format.freq * (1 / channel.rate)));
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speex_resampler_process_interleaved_int(channel.resampler, (const spx_int16_t*)dataconverted, &in_len, (spx_int16_t*)out, &out_len);
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int mixlength = (out_len * samplesize);
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if (loaded + mixlength > length) { // check for overflow
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mixlength = length - loaded;
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}
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lengthloaded = (out_len * samplesize);
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SDL_MixAudioFormat(&data[loaded], out, format.format, mixlength, volume);
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delete[] out;
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} else {
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lengthloaded = (cvt.len_cvt / samplesize) * samplesize;
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int mixlength = lengthloaded;
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if (loaded + cvt.len_cvt > length) {
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mixlength = length - loaded;
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}
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SDL_MixAudioFormat(&data[loaded], dataconverted, format.format, mixlength, volume);
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}
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delete[] dataconverted;
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}
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} else {
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if (channel.rate != 1 && format.format == AUDIO_S16) {
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spx_uint32_t in_len = (spx_uint32_t)(ceil((double)readfromstream / samplesize));
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Uint8* out = new Uint8[length + 200];
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spx_uint32_t out_len = samples + 20;
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speex_resampler_set_rate(channel.resampler, format.freq, (int)(format.freq * (1 / channel.rate)));
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speex_resampler_process_interleaved_int(channel.resampler, (const spx_int16_t*)datastream, &in_len, (spx_int16_t*)out, &out_len);
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int mixlength = (out_len * samplesize);
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if (loaded + mixlength > length) {
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mixlength = length - loaded;
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}
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lengthloaded = (out_len * samplesize);
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SDL_MixAudioFormat(&data[loaded], out, format.format, mixlength, volume);
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delete[] out;
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} else {
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lengthloaded = readfromstream;
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int mixlength = lengthloaded;
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if (loaded + readfromstream > length) {
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mixlength = length - loaded;
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}
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SDL_MixAudioFormat(&data[loaded], datastream, format.format, mixlength, volume);
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}
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}
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channel.offset += readfromstream;
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}
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loaded += lengthloaded;
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if (channel.loop != 0 && channel.offset >= channel.stream->Length()) {
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if (channel.loop != -1) {
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channel.loop--;
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}
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channel.offset = 0;
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}
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} while(loaded < length || (loaded < length && channel.loop != 0 && channel.offset == 0));
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}
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}
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bool Mixer::MustConvert(Stream& stream)
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{
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const AudioFormat* streamformat = stream.Format();
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if (!streamformat) {
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return false;
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}
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if (streamformat->format != format.format || streamformat->channels != format.channels || streamformat->freq != format.freq) {
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return true;
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}
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return false;
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}
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bool Mixer::Convert(SDL_AudioCVT& cvt, const Uint8* data, int length, Uint8** dataout)
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{
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if (length == 0 || cvt.len_mult == 0) {
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return false;
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}
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cvt.len = length;
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cvt.buf = (Uint8*)new char[cvt.len * cvt.len_mult];
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memcpy(cvt.buf, data, length);
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if (SDL_ConvertAudio(&cvt) < 0) {
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delete[] cvt.buf;
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return false;
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}
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*dataout = cvt.buf;
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return true;
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}
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void Mixer_Init(char* device)
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{
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static Mixer mixer;
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mixer.Init(device);
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}
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