Add an example using convolution reverb
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@ -1508,6 +1508,10 @@ if(ALSOFT_EXAMPLES)
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target_link_libraries(alstreamcb PRIVATE ${LINKER_FLAGS} SndFile::SndFile ex-common
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${UNICODE_FLAG})
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add_executable(alconvolve examples/alconvolve.cpp)
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target_link_libraries(alconvolve PRIVATE ${LINKER_FLAGS} common SndFile::SndFile ex-common
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${UNICODE_FLAG})
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if(ALSOFT_INSTALL_EXAMPLES)
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set(EXTRA_INSTALLS ${EXTRA_INSTALLS} alplay alstream alreverb almultireverb allatency
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alhrtf)
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examples/alconvolve.cpp
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536
examples/alconvolve.cpp
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@ -0,0 +1,536 @@
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/*
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* OpenAL Convolution Reverb Example
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*
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* Copyright (c) 2020 by Chris Robinson <chris.kcat@gmail.com>
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*
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* Permission is hereby granted, free of charge, to any person obtaining a copy
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* of this software and associated documentation files (the "Software"), to deal
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* in the Software without restriction, including without limitation the rights
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* to use, copy, modify, merge, publish, distribute, sublicense, and/or sell
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* copies of the Software, and to permit persons to whom the Software is
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* furnished to do so, subject to the following conditions:
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*
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* The above copyright notice and this permission notice shall be included in
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* all copies or substantial portions of the Software.
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*
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* THE SOFTWARE IS PROVIDED "AS IS", WITHOUT WARRANTY OF ANY KIND, EXPRESS OR
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* IMPLIED, INCLUDING BUT NOT LIMITED TO THE WARRANTIES OF MERCHANTABILITY,
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* FITNESS FOR A PARTICULAR PURPOSE AND NONINFRINGEMENT. IN NO EVENT SHALL THE
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* AUTHORS OR COPYRIGHT HOLDERS BE LIABLE FOR ANY CLAIM, DAMAGES OR OTHER
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* LIABILITY, WHETHER IN AN ACTION OF CONTRACT, TORT OR OTHERWISE, ARISING FROM,
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* OUT OF OR IN CONNECTION WITH THE SOFTWARE OR THE USE OR OTHER DEALINGS IN
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* THE SOFTWARE.
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*/
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/* This file contains a streaming audio player, using the convolution reverb
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* effect.
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*/
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#include <string.h>
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#include <stdlib.h>
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#include <stdio.h>
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#include <atomic>
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#include <cassert>
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#include <chrono>
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#include <limits>
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#include <memory>
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#include <stdexcept>
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#include <string>
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#include <thread>
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#include <vector>
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#include "sndfile.h"
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#include "AL/al.h"
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#include "AL/alc.h"
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#include "AL/alext.h"
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#include "common/alhelpers.h"
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#ifndef AL_SOFT_callback_buffer
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#define AL_SOFT_callback_buffer
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typedef unsigned int ALbitfieldSOFT;
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#define AL_BUFFER_CALLBACK_FUNCTION_SOFT 0x19A0
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#define AL_BUFFER_CALLBACK_USER_PARAM_SOFT 0x19A1
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typedef ALsizei (AL_APIENTRY*LPALBUFFERCALLBACKTYPESOFT)(ALvoid *userptr, ALvoid *sampledata, ALsizei numsamples);
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typedef void (AL_APIENTRY*LPALBUFFERCALLBACKSOFT)(ALuint buffer, ALenum format, ALsizei freq, LPALBUFFERCALLBACKTYPESOFT callback, ALvoid *userptr, ALbitfieldSOFT flags);
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typedef void (AL_APIENTRY*LPALGETBUFFERPTRSOFT)(ALuint buffer, ALenum param, ALvoid **value);
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typedef void (AL_APIENTRY*LPALGETBUFFER3PTRSOFT)(ALuint buffer, ALenum param, ALvoid **value1, ALvoid **value2, ALvoid **value3);
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typedef void (AL_APIENTRY*LPALGETBUFFERPTRVSOFT)(ALuint buffer, ALenum param, ALvoid **values);
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#endif
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#ifndef AL_SOFT_convolution_reverb
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#define AL_SOFT_convolution_reverb
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#define AL_EFFECT_CONVOLUTION_REVERB_SOFT 0xA000
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#endif
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namespace {
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/* Effect object functions */
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LPALGENEFFECTS alGenEffects;
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LPALDELETEEFFECTS alDeleteEffects;
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LPALISEFFECT alIsEffect;
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LPALEFFECTI alEffecti;
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LPALEFFECTIV alEffectiv;
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LPALEFFECTF alEffectf;
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LPALEFFECTFV alEffectfv;
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LPALGETEFFECTI alGetEffecti;
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LPALGETEFFECTIV alGetEffectiv;
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LPALGETEFFECTF alGetEffectf;
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LPALGETEFFECTFV alGetEffectfv;
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/* Auxiliary Effect Slot object functions */
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LPALGENAUXILIARYEFFECTSLOTS alGenAuxiliaryEffectSlots;
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LPALDELETEAUXILIARYEFFECTSLOTS alDeleteAuxiliaryEffectSlots;
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LPALISAUXILIARYEFFECTSLOT alIsAuxiliaryEffectSlot;
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LPALAUXILIARYEFFECTSLOTI alAuxiliaryEffectSloti;
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LPALAUXILIARYEFFECTSLOTIV alAuxiliaryEffectSlotiv;
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LPALAUXILIARYEFFECTSLOTF alAuxiliaryEffectSlotf;
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LPALAUXILIARYEFFECTSLOTFV alAuxiliaryEffectSlotfv;
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LPALGETAUXILIARYEFFECTSLOTI alGetAuxiliaryEffectSloti;
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LPALGETAUXILIARYEFFECTSLOTIV alGetAuxiliaryEffectSlotiv;
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LPALGETAUXILIARYEFFECTSLOTF alGetAuxiliaryEffectSlotf;
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LPALGETAUXILIARYEFFECTSLOTFV alGetAuxiliaryEffectSlotfv;
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ALuint CreateEffect()
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{
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/* Create the effect object and try to set convolution reverb. */
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ALuint effect{0};
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alGenEffects(1, &effect);
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printf("Using Convolution Reverb\n");
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alEffecti(effect, AL_EFFECT_TYPE, AL_EFFECT_CONVOLUTION_REVERB_SOFT);
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/* Check if an error occured, and clean up if so. */
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if(ALenum err{alGetError()})
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{
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fprintf(stderr, "OpenAL error: %s\n", alGetString(err));
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if(alIsEffect(effect))
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alDeleteEffects(1, &effect);
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return 0;
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}
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return effect;
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}
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ALuint LoadSound(const char *filename)
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{
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/* Open the audio file and check that it's usable. */
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SF_INFO sfinfo{};
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SNDFILE *sndfile{sf_open(filename, SFM_READ, &sfinfo)};
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if(!sndfile)
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{
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fprintf(stderr, "Could not open audio in %s: %s\n", filename, sf_strerror(sndfile));
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return 0;
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}
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constexpr sf_count_t max_samples{std::numeric_limits<int>::max() / sizeof(float)};
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if(sfinfo.frames < 1 || sfinfo.frames > max_samples/sfinfo.channels)
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{
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fprintf(stderr, "Bad sample count in %s (%" PRId64 ")\n", filename, sfinfo.frames);
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sf_close(sndfile);
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return 0;
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}
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/* Get the sound format, and figure out the OpenAL format. Use a float
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* format since impulse responses are keen on having a low noise floor.
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*/
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ALenum format{};
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if(sfinfo.channels == 1)
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format = AL_FORMAT_MONO_FLOAT32;
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else if(sfinfo.channels == 2)
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format = AL_FORMAT_STEREO_FLOAT32;
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else
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{
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fprintf(stderr, "Unsupported channel count: %d\n", sfinfo.channels);
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sf_close(sndfile);
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return 0;
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}
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auto membuf = std::make_unique<float[]>(static_cast<size_t>(sfinfo.frames * sfinfo.channels));
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sf_count_t num_frames{sf_readf_float(sndfile, membuf.get(), sfinfo.frames)};
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if(num_frames < 1)
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{
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membuf = nullptr;
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sf_close(sndfile);
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fprintf(stderr, "Failed to read samples in %s (%" PRId64 ")\n", filename, num_frames);
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return 0;
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}
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const auto num_bytes = static_cast<ALsizei>(num_frames * sfinfo.channels) *
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ALsizei{sizeof(float)};
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ALuint buffer{0};
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alGenBuffers(1, &buffer);
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alBufferData(buffer, format, membuf.get(), num_bytes, sfinfo.samplerate);
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membuf = nullptr;
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sf_close(sndfile);
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if(ALenum err{alGetError()})
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{
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fprintf(stderr, "OpenAL Error: %s\n", alGetString(err));
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if(buffer && alIsBuffer(buffer))
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alDeleteBuffers(1, &buffer);
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return 0;
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}
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return buffer;
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}
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/* This is largely the same as in alstreamcb.cpp. Comments removed for brevity,
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* see the aforementioned source for more details.
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*/
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using std::chrono::seconds;
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using std::chrono::nanoseconds;
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LPALBUFFERCALLBACKSOFT alBufferCallbackSOFT;
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struct StreamPlayer {
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std::unique_ptr<ALbyte[]> mBufferData;
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size_t mBufferDataSize{0};
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std::atomic<size_t> mReadPos{0};
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std::atomic<size_t> mWritePos{0};
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ALuint mBuffer{0}, mSource{0};
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size_t mStartOffset{0};
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SNDFILE *mSndfile{nullptr};
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SF_INFO mSfInfo{};
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size_t mDecoderOffset{0};
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ALenum mFormat;
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StreamPlayer()
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{
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alGenBuffers(1, &mBuffer);
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if(ALenum err{alGetError()})
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throw std::runtime_error{"alGenBuffers failed"};
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alGenSources(1, &mSource);
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if(ALenum err{alGetError()})
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{
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alDeleteBuffers(1, &mBuffer);
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throw std::runtime_error{"alGenSources failed"};
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}
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}
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~StreamPlayer()
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{
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alDeleteSources(1, &mSource);
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alDeleteBuffers(1, &mBuffer);
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if(mSndfile)
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sf_close(mSndfile);
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}
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void close()
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{
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if(mSndfile)
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{
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alSourceRewind(mSource);
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alSourcei(mSource, AL_BUFFER, 0);
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sf_close(mSndfile);
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mSndfile = nullptr;
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}
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}
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bool open(const char *filename)
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{
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close();
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mSndfile = sf_open(filename, SFM_READ, &mSfInfo);
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if(!mSndfile)
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{
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fprintf(stderr, "Could not open audio in %s: %s\n", filename, sf_strerror(mSndfile));
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return false;
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}
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mFormat = AL_NONE;
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if(mSfInfo.channels == 1)
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mFormat = AL_FORMAT_MONO_FLOAT32;
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else if(mSfInfo.channels == 2)
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mFormat = AL_FORMAT_STEREO_FLOAT32;
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else if(mSfInfo.channels == 6)
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mFormat = AL_FORMAT_51CHN32;
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else
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{
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fprintf(stderr, "Unsupported channel count: %d\n", mSfInfo.channels);
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sf_close(mSndfile);
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mSndfile = nullptr;
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return false;
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}
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mBufferDataSize = static_cast<ALuint>(mSfInfo.samplerate*mSfInfo.channels) * sizeof(float);
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mBufferData.reset(new ALbyte[mBufferDataSize]);
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mReadPos.store(0, std::memory_order_relaxed);
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mWritePos.store(0, std::memory_order_relaxed);
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mDecoderOffset = 0;
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return true;
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}
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static ALsizei AL_APIENTRY bufferCallbackC(void *userptr, void *data, ALsizei size)
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{ return static_cast<StreamPlayer*>(userptr)->bufferCallback(data, size); }
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ALsizei bufferCallback(void *data, ALsizei size)
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{
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ALsizei got{0};
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size_t roffset{mReadPos.load(std::memory_order_acquire)};
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while(got < size)
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{
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const size_t woffset{mWritePos.load(std::memory_order_relaxed)};
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if(woffset == roffset) break;
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size_t todo{((woffset < roffset) ? mBufferDataSize : woffset) - roffset};
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todo = std::min<size_t>(todo, static_cast<ALuint>(size-got));
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memcpy(data, &mBufferData[roffset], todo);
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data = static_cast<ALbyte*>(data) + todo;
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got += static_cast<ALsizei>(todo);
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roffset += todo;
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if(roffset == mBufferDataSize)
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roffset = 0;
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}
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mReadPos.store(roffset, std::memory_order_release);
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return got;
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}
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bool prepare()
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{
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alBufferCallbackSOFT(mBuffer, mFormat, mSfInfo.samplerate, bufferCallbackC, this, 0);
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alSourcei(mSource, AL_BUFFER, static_cast<ALint>(mBuffer));
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if(ALenum err{alGetError()})
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{
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fprintf(stderr, "Failed to set callback: %s (0x%04x)\n", alGetString(err), err);
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return false;
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}
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return true;
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}
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bool update()
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{
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ALenum state;
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ALint pos;
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alGetSourcei(mSource, AL_SAMPLE_OFFSET, &pos);
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alGetSourcei(mSource, AL_SOURCE_STATE, &state);
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const size_t frame_size{static_cast<ALuint>(mSfInfo.channels) * sizeof(float)};
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size_t woffset{mWritePos.load(std::memory_order_acquire)};
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if(state != AL_INITIAL)
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{
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const size_t roffset{mReadPos.load(std::memory_order_relaxed)};
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const size_t readable{((woffset >= roffset) ? woffset : (mBufferDataSize+woffset)) -
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roffset};
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const size_t curtime{((state==AL_STOPPED) ? (mDecoderOffset-readable) / frame_size
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: (static_cast<ALuint>(pos) + mStartOffset/frame_size))
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/ static_cast<ALuint>(mSfInfo.samplerate)};
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printf("\r%3zus (%3zu%% full)", curtime, readable * 100 / mBufferDataSize);
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}
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else
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fputs("Starting...", stdout);
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fflush(stdout);
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while(!sf_error(mSndfile))
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{
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size_t read_bytes;
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const size_t roffset{mReadPos.load(std::memory_order_relaxed)};
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if(roffset > woffset)
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{
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const size_t writable{roffset-woffset-1};
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if(writable < frame_size) break;
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sf_count_t num_frames{sf_readf_float(mSndfile,
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reinterpret_cast<float*>(&mBufferData[woffset]),
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static_cast<sf_count_t>(writable/frame_size))};
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if(num_frames < 1) break;
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read_bytes = static_cast<size_t>(num_frames) * frame_size;
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woffset += read_bytes;
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}
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else
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{
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const size_t writable{!roffset ? mBufferDataSize-woffset-1 :
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(mBufferDataSize-woffset)};
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if(writable < frame_size) break;
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sf_count_t num_frames{sf_readf_float(mSndfile,
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reinterpret_cast<float*>(&mBufferData[woffset]),
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static_cast<sf_count_t>(writable/frame_size))};
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if(num_frames < 1) break;
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read_bytes = static_cast<size_t>(num_frames) * frame_size;
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woffset += read_bytes;
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if(woffset == mBufferDataSize)
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woffset = 0;
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}
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mWritePos.store(woffset, std::memory_order_release);
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mDecoderOffset += read_bytes;
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}
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if(state != AL_PLAYING && state != AL_PAUSED)
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{
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const size_t roffset{mReadPos.load(std::memory_order_relaxed)};
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const size_t readable{((woffset >= roffset) ? woffset : (mBufferDataSize+woffset)) -
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roffset};
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if(readable == 0)
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return false;
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mStartOffset = mDecoderOffset - readable;
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alSourcePlay(mSource);
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if(alGetError() != AL_NO_ERROR)
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return false;
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}
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return true;
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}
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};
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} // namespace
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int main(int argc, char **argv)
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{
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/* A simple RAII container for OpenAL startup and shutdown. */
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struct AudioManager {
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AudioManager(char ***argv_, int *argc_)
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{
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if(InitAL(argv_, argc_) != 0)
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throw std::runtime_error{"Failed to initialize OpenAL"};
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}
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~AudioManager() { CloseAL(); }
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};
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/* Print out usage if no arguments were specified */
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if(argc < 2)
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{
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fprintf(stderr, "Usage: %s [-device <name>] <impulse response sound> [sound files...]\n",
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argv[0]);
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return 1;
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}
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argv++; argc--;
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AudioManager almgr{&argv, &argc};
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if(!alIsExtensionPresent("AL_SOFTX_callback_buffer"))
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{
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fprintf(stderr, "AL_SOFT_callback_buffer extension not available\n");
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return 1;
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}
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/* Define a macro to help load the function pointers. */
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#define LOAD_PROC(T, x) ((x) = reinterpret_cast<T>(alGetProcAddress(#x)))
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LOAD_PROC(LPALBUFFERCALLBACKSOFT, alBufferCallbackSOFT);
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LOAD_PROC(LPALGENEFFECTS, alGenEffects);
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LOAD_PROC(LPALDELETEEFFECTS, alDeleteEffects);
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LOAD_PROC(LPALISEFFECT, alIsEffect);
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LOAD_PROC(LPALEFFECTI, alEffecti);
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LOAD_PROC(LPALEFFECTIV, alEffectiv);
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LOAD_PROC(LPALEFFECTF, alEffectf);
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LOAD_PROC(LPALEFFECTFV, alEffectfv);
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LOAD_PROC(LPALGETEFFECTI, alGetEffecti);
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LOAD_PROC(LPALGETEFFECTIV, alGetEffectiv);
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LOAD_PROC(LPALGETEFFECTF, alGetEffectf);
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LOAD_PROC(LPALGETEFFECTFV, alGetEffectfv);
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LOAD_PROC(LPALGENAUXILIARYEFFECTSLOTS, alGenAuxiliaryEffectSlots);
|
||||
LOAD_PROC(LPALDELETEAUXILIARYEFFECTSLOTS, alDeleteAuxiliaryEffectSlots);
|
||||
LOAD_PROC(LPALISAUXILIARYEFFECTSLOT, alIsAuxiliaryEffectSlot);
|
||||
LOAD_PROC(LPALAUXILIARYEFFECTSLOTI, alAuxiliaryEffectSloti);
|
||||
LOAD_PROC(LPALAUXILIARYEFFECTSLOTIV, alAuxiliaryEffectSlotiv);
|
||||
LOAD_PROC(LPALAUXILIARYEFFECTSLOTF, alAuxiliaryEffectSlotf);
|
||||
LOAD_PROC(LPALAUXILIARYEFFECTSLOTFV, alAuxiliaryEffectSlotfv);
|
||||
LOAD_PROC(LPALGETAUXILIARYEFFECTSLOTI, alGetAuxiliaryEffectSloti);
|
||||
LOAD_PROC(LPALGETAUXILIARYEFFECTSLOTIV, alGetAuxiliaryEffectSlotiv);
|
||||
LOAD_PROC(LPALGETAUXILIARYEFFECTSLOTF, alGetAuxiliaryEffectSlotf);
|
||||
LOAD_PROC(LPALGETAUXILIARYEFFECTSLOTFV, alGetAuxiliaryEffectSlotfv);
|
||||
#undef LOAD_PROC
|
||||
|
||||
/* Load the impulse response sound file into a buffer. */
|
||||
ALuint buffer{LoadSound(argv[0])};
|
||||
if(!buffer) return 1;
|
||||
|
||||
/* Create the convolution reverb effect. */
|
||||
ALuint effect{CreateEffect()};
|
||||
if(!effect)
|
||||
{
|
||||
alDeleteBuffers(1, &buffer);
|
||||
return 1;
|
||||
}
|
||||
|
||||
/* Create the effect slot object. This is what "plays" an effect on sources
|
||||
* that connect to it. */
|
||||
ALuint slot{0};
|
||||
alGenAuxiliaryEffectSlots(1, &slot);
|
||||
|
||||
/* Set the impulse response sound buffer on the effect slot. This allows
|
||||
* effects to access it as needed. In this case, convolution reverb uses it
|
||||
* as the filter source. NOTE: Unlike the effect object, the buffer *is*
|
||||
* kept referenced and may not be changed or deleted as long as it's set,
|
||||
* just like with a source. When another buffer is set, or the effect slot
|
||||
* is deleted, the buffer reference is released.
|
||||
*
|
||||
* The effect slot's gain is reduced because the impulse responses I've
|
||||
* tested with result in excessively loud reverb. Is that normal? Even with
|
||||
* this, it seems a bit on the loud side.
|
||||
*
|
||||
* Also note: unlike standard or EAX reverb, there is no automatic
|
||||
* attenuation of a source's reverb response with distance, so the reverb
|
||||
* will remain full volume regardless of a given sound's distance from the
|
||||
* listener. You can use a send filter to alter a given source's
|
||||
* contribution to reverb.
|
||||
*/
|
||||
alAuxiliaryEffectSloti(slot, AL_BUFFER, static_cast<ALint>(buffer));
|
||||
alAuxiliaryEffectSlotf(slot, AL_EFFECTSLOT_GAIN, 1.0f / 16.0f);
|
||||
alAuxiliaryEffectSloti(slot, AL_EFFECTSLOT_EFFECT, static_cast<ALint>(effect));
|
||||
assert(alGetError()==AL_NO_ERROR && "Failed to set effect slot");
|
||||
|
||||
ALCint refresh{25};
|
||||
alcGetIntegerv(alcGetContextsDevice(alcGetCurrentContext()), ALC_REFRESH, 1, &refresh);
|
||||
|
||||
std::unique_ptr<StreamPlayer> player{new StreamPlayer{}};
|
||||
alSource3i(player->mSource, AL_AUXILIARY_SEND_FILTER, static_cast<ALint>(slot), 0,
|
||||
AL_FILTER_NULL);
|
||||
|
||||
for(int i{1};i < argc;++i)
|
||||
{
|
||||
if(!player->open(argv[i]))
|
||||
continue;
|
||||
|
||||
const char *namepart{strrchr(argv[i], '/')};
|
||||
if(namepart || (namepart=strrchr(argv[i], '\\')))
|
||||
++namepart;
|
||||
else
|
||||
namepart = argv[i];
|
||||
|
||||
printf("Playing: %s (%s, %dhz)\n", namepart, FormatName(player->mFormat),
|
||||
player->mSfInfo.samplerate);
|
||||
fflush(stdout);
|
||||
|
||||
if(!player->prepare())
|
||||
{
|
||||
player->close();
|
||||
continue;
|
||||
}
|
||||
|
||||
while(player->update())
|
||||
std::this_thread::sleep_for(nanoseconds{seconds{1}} / refresh);
|
||||
putc('\n', stdout);
|
||||
|
||||
player->close();
|
||||
}
|
||||
/* All done. */
|
||||
printf("Done.\n");
|
||||
|
||||
player = nullptr;
|
||||
alDeleteAuxiliaryEffectSlots(1, &slot);
|
||||
alDeleteEffects(1, &effect);
|
||||
alDeleteBuffers(1, &buffer);
|
||||
|
||||
return 0;
|
||||
}
|
Loading…
x
Reference in New Issue
Block a user