Add an example using convolution reverb

This commit is contained in:
Chris Robinson 2020-08-25 04:59:04 -07:00
parent 801c7a9226
commit 309be1c6f6
2 changed files with 540 additions and 0 deletions

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@ -1508,6 +1508,10 @@ if(ALSOFT_EXAMPLES)
target_link_libraries(alstreamcb PRIVATE ${LINKER_FLAGS} SndFile::SndFile ex-common
${UNICODE_FLAG})
add_executable(alconvolve examples/alconvolve.cpp)
target_link_libraries(alconvolve PRIVATE ${LINKER_FLAGS} common SndFile::SndFile ex-common
${UNICODE_FLAG})
if(ALSOFT_INSTALL_EXAMPLES)
set(EXTRA_INSTALLS ${EXTRA_INSTALLS} alplay alstream alreverb almultireverb allatency
alhrtf)

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