Add an example using convolution reverb

2 files changed, 540 insertions, 0 deletions

diff --git a/CMakeLists.txt b/CMakeLists.txt
index f77f5a24..092c041d 100644
--- a/CMakeLists.txt
+++ b/CMakeLists.txt
@@ -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)
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 <[email protected]>
+ *
+ * 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;
+}