From 309be1c6f6bc6364d758712c29bb2ccbb1cc3511 Mon Sep 17 00:00:00 2001
From: Chris Robinson <chris.kcat@gmail.com>
Date: Tue, 25 Aug 2020 04:59:04 -0700
Subject: Add an example using convolution reverb
---
examples/alconvolve.cpp | 536 ++++++++++++++++++++++++++++++++++++++++++++++++
1 file changed, 536 insertions(+)
create mode 100644 examples/alconvolve.cpp
(limited to 'examples/alconvolve.cpp')
diff --git a/examples/alconvolve.cpp b/examples/alconvolve.cpp
new file mode 100644
index 00000000..68ab5615
--- /dev/null
+++ b/examples/alconvolve.cpp
@@ -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;
+}
--
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