Add a streaming example using a callback buffer

1 files changed, 436 insertions, 0 deletions

diff --git a/examples/alstreamcb.cpp b/examples/alstreamcb.cpp
new file mode 100644
index 00000000..fbc3d02d
--- /dev/null
+++ b/examples/alstreamcb.cpp
@@ -0,0 +1,436 @@
+/*
+ * OpenAL Callback-based Stream 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 a callback buffer. */
+
+#include <string.h>
+#include <stdlib.h>
+#include <stdio.h>
+
+#include <atomic>
+#include <chrono>
+#include <memory>
+#include <stdexcept>
+#include <string>
+#include <thread>
+#include <vector>
+
+#include "SDL_sound.h"
+#include "SDL_audio.h"
+#include "SDL_stdinc.h"
+
+#include "AL/al.h"
+#include "AL/alc.h"
+
+#include "common/alhelpers.h"
+
+
+#ifndef SDL_AUDIO_MASK_BITSIZE
+#define SDL_AUDIO_MASK_BITSIZE (0xFF)
+#endif
+#ifndef SDL_AUDIO_BITSIZE
+#define SDL_AUDIO_BITSIZE(x) (x & SDL_AUDIO_MASK_BITSIZE)
+#endif
+
+
+#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
+
+namespace {
+
+using std::chrono::seconds;
+using std::chrono::nanoseconds;
+
+LPALBUFFERCALLBACKSOFT alBufferCallbackSOFT;
+
+struct StreamPlayer {
+    /* A lockless ring-buffer (supports single-provider, single-consumer
+     * operation).
+     */
+    std::unique_ptr<ALbyte[]> mBufferData;
+    size_t mBufferDataSize{0};
+    std::atomic<size_t> mReadPos{0};
+    std::atomic<size_t> mWritePos{0};
+
+    /* The buffer to get the callback, and source to play with. */
+    ALuint mBuffer{0}, mSource{0};
+    size_t mStartOffset{0};
+
+    /* Handle for the audio file to decode. */
+    Sound_Sample *mSample{nullptr};
+    Uint32 mAvailableData{0};
+    size_t mDecoderOffset{0};
+
+    /* The format of the callback samples. */
+    ALenum mFormat;
+    ALsizei mSampleRate;
+
+    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(mSample)
+            Sound_FreeSample(mSample);
+    }
+
+    void close()
+    {
+        if(mSample)
+        {
+            alSourceRewind(mSource);
+            alSourcei(mSource, AL_BUFFER, 0);
+            Sound_FreeSample(mSample);
+            mSample = nullptr;
+        }
+    }
+
+    bool open(const char *filename)
+    {
+        close();
+
+        /* Open the file in its normal format. */
+        mSample = Sound_NewSampleFromFile(filename, nullptr, 0);
+        if(!mSample)
+        {
+            fprintf(stderr, "Could not open audio in %s\n", filename);
+            return false;
+        }
+
+        /* Figure out the OpenAL format from the sample's format. */
+        mFormat = AL_NONE;
+        if(mSample->actual.channels == 1)
+        {
+            if(mSample->actual.format == AUDIO_U8)
+                mFormat = AL_FORMAT_MONO8;
+            else if(mSample->actual.format == AUDIO_S16SYS)
+                mFormat = AL_FORMAT_MONO16;
+        }
+        else if(mSample->actual.channels == 2)
+        {
+            if(mSample->actual.format == AUDIO_U8)
+                mFormat = AL_FORMAT_STEREO8;
+            else if(mSample->actual.format == AUDIO_S16SYS)
+                mFormat = AL_FORMAT_STEREO16;
+        }
+        if(!mFormat)
+        {
+            fprintf(stderr, "Unsupported sample format: 0x%04x, %d channels\n",
+                mSample->actual.format, mSample->actual.channels);
+            Sound_FreeSample(mSample);
+            mSample = nullptr;
+
+            return false;
+        }
+        mSampleRate = static_cast<ALsizei>(mSample->actual.rate);
+
+        const auto frame_size = Uint32{mSample->actual.channels} *
+            SDL_AUDIO_BITSIZE(mSample->actual.format) / 8;
+
+        /* Set a 50ms decode buffer size. */
+        Sound_SetBufferSize(mSample, static_cast<Uint32>(mSampleRate)*50/1000 * frame_size);
+        mAvailableData = 0;
+
+        /* Set a 1s ring buffer size. */
+        mBufferDataSize = static_cast<Uint32>(mSampleRate) * size_t{frame_size};
+        mBufferData.reset(new ALbyte[mBufferDataSize]);
+        mReadPos.store(0, std::memory_order_relaxed);
+        mWritePos.store(0, std::memory_order_relaxed);
+        mDecoderOffset = 0;
+
+        return true;
+    }
+
+    /* The actual C-style callback just forwards to the non-static method. Not
+     * strictly needed and the compiler will optimize it to a normal function,
+     * but it allows the callback implementation to have a nice 'this' pointer
+     * with normal member access.
+     */
+    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)
+    {
+        /* NOTE: The callback *MUST* be real-time safe! That means no blocking,
+         * no allocations or deallocations, no I/O, no page faults, or calls to
+         * functions that could do these things (this includes calling to
+         * libraries like SDL_sound, libsndfile, ffmpeg, etc). Nothing should
+         * unexpectedly stall this call since the audio has to get to the
+         * device on time.
+         */
+        ALsizei got{0};
+
+        size_t roffset{mReadPos.load(std::memory_order_acquire)};
+        while(got < size)
+        {
+            /* If the write offset == read offset, there's nothing left in the
+             * ring-buffer. Break from the loop and give what has been written.
+             */
+            const size_t woffset{mWritePos.load(std::memory_order_relaxed)};
+            if(woffset == roffset) break;
+
+            /* If the write offset is behind the read offset, the readable
+             * portion wrapped around. Just read up to the end of the buffer in
+             * that case, otherwise read up to the write offset. Also limit the
+             * amount to copy given how much is remaining to write.
+             */
+            size_t todo{((woffset < roffset) ? mBufferDataSize : woffset) - roffset};
+            todo = std::min<size_t>(todo, static_cast<ALuint>(size-got));
+
+            /* Copy from the ring buffer to the provided output buffer. Wrap
+             * the resulting read offset if it reached the end of the ring-
+             * buffer.
+             */
+            memcpy(data, &mBufferData[roffset], todo);
+            data = static_cast<ALbyte*>(data) + todo;
+            got += static_cast<ALsizei>(todo);
+
+            roffset += todo;
+            if(roffset == mBufferDataSize)
+                roffset = 0;
+        }
+        /* Finally, store the updated read offset, and return how many bytes
+         * have been written.
+         */
+        mReadPos.store(roffset, std::memory_order_release);
+
+        return got;
+    }
+
+    bool prepare()
+    {
+        alBufferCallbackSOFT(mBuffer, mFormat, mSampleRate, 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;
+        }
+
+        mAvailableData = Sound_Decode(mSample);
+        if(!mAvailableData)
+            fprintf(stderr, "Failed to decode any samples: %s\n", Sound_GetError());
+        return mAvailableData != 0;
+    }
+
+    bool update()
+    {
+        constexpr int BadFlags{SOUND_SAMPLEFLAG_EOF | SOUND_SAMPLEFLAG_ERROR};
+
+        ALenum state;
+        ALint pos;
+        alGetSourcei(mSource, AL_SAMPLE_OFFSET, &pos);
+        alGetSourcei(mSource, AL_SOURCE_STATE, &state);
+
+        size_t woffset{mWritePos.load(std::memory_order_acquire)};
+        if(state != AL_INITIAL)
+        {
+            const auto frame_size = Uint32{mSample->actual.channels} *
+                SDL_AUDIO_BITSIZE(mSample->actual.format) / 8;
+            const size_t roffset{mReadPos.load(std::memory_order_relaxed)};
+            const size_t readable{((woffset >= roffset) ? woffset : (mBufferDataSize+woffset)) -
+                roffset};
+            /* For a stopped (underrun) source, the current playback offset is
+             * the current decoder offset excluding the readable buffered data.
+             * For a playing/paused source, it's the source's offset including
+             * the playback offset the source was started with.
+             */
+            const size_t curtime{((state==AL_STOPPED) ? (mDecoderOffset-readable) / frame_size
+                : (static_cast<ALuint>(pos) + mStartOffset/frame_size)) / mSample->actual.rate};
+            printf("\r%3zus (%3zu%% full)", curtime, readable * 100 / mBufferDataSize);
+        }
+        else
+            fputs("Starting...", stdout);
+        fflush(stdout);
+
+        while(mAvailableData > 0)
+        {
+            const size_t roffset{mReadPos.load(std::memory_order_relaxed)};
+            if(roffset > woffset)
+            {
+                /* Note that the ring buffer's writable space is one byte less
+                 * than the available area because the write offset ending up
+                 * at the read offset would be interpreted as being empty
+                 * instead of full.
+                 */
+                const size_t writable{roffset-woffset-1};
+                /* Don't copy the sample data if it can't all fit. */
+                if(writable < mAvailableData) break;
+
+                memcpy(&mBufferData[woffset], mSample->buffer, mAvailableData);
+                woffset += mAvailableData;
+            }
+            else
+            {
+                /* If the read offset is at or behind the write offset, the
+                 * writeable area (might) wrap around. Make sure the sample
+                 * data can fit, and calculate how much goes in front and in
+                 * back.
+                 */
+                const size_t writable{mBufferDataSize+roffset-woffset-1};
+                if(writable < mAvailableData) break;
+
+                const size_t todo1{std::min<size_t>(mAvailableData, mBufferDataSize-woffset)};
+                const size_t todo2{mAvailableData - todo1};
+
+                memcpy(&mBufferData[woffset], mSample->buffer, todo1);
+                woffset += todo1;
+                if(woffset == mBufferDataSize)
+                {
+                    woffset = 0;
+                    if(todo2 > 0)
+                    {
+                        memcpy(&mBufferData[woffset], static_cast<ALbyte*>(mSample->buffer)+todo1,
+                            todo2);
+                        woffset += todo2;
+                    }
+                }
+            }
+            mWritePos.store(woffset, std::memory_order_release);
+            mDecoderOffset += mAvailableData;
+
+            if(!(mSample->flags&BadFlags))
+                mAvailableData = Sound_Decode(mSample);
+            else
+                mAvailableData = 0;
+        }
+
+        if(state != AL_PLAYING && state != AL_PAUSED)
+        {
+            /* If the source is not playing or paused, it either underrun
+             * (AL_STOPPED) or is just getting started (AL_INITIAL). If the
+             * ring buffer is empty, it's done, otherwise play the source with
+             * what's available.
+             */
+            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;
+
+            /* Store the playback offset that the source will start reading
+             * from, so it can be tracked during playback.
+             */
+            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 and SDL_sound startup and shutdown. */
+    struct AudioManager {
+        AudioManager(char ***argv_, int *argc_)
+        {
+            if(InitAL(argv_, argc_) != 0)
+                throw std::runtime_error{"Failed to initialize OpenAL"};
+            Sound_Init();
+        }
+        ~AudioManager() { Sound_Quit(); CloseAL(); }
+    };
+
+    /* Print out usage if no arguments were specified */
+    if(argc < 2)
+    {
+        fprintf(stderr, "Usage: %s [-device <name>] <filenames...>\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;
+    }
+
+    alBufferCallbackSOFT = reinterpret_cast<LPALBUFFERCALLBACKSOFT>(
+        alGetProcAddress("alBufferCallbackSOFT"));
+
+    ALCint refresh{25};
+    alcGetIntegerv(alcGetContextsDevice(alcGetCurrentContext()), ALC_REFRESH, 1, &refresh);
+
+    std::unique_ptr<StreamPlayer> player{new StreamPlayer{}};
+
+    /* Play each file listed on the command line */
+    for(int i{0};i < argc;++i)
+    {
+        if(!player->open(argv[i]))
+            continue;
+
+        /* Get the name portion, without the path, for display. */
+        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->mSampleRate);
+        fflush(stdout);
+
+        if(!player->prepare())
+        {
+            player->close();
+            continue;
+        }
+
+        while(player->update())
+            std::this_thread::sleep_for(nanoseconds{seconds{1}} / refresh);
+        putc('\n', stdout);
+
+        /* All done with this file. Close it and go to the next */
+        player->close();
+    }
+    /* All done. */
+    printf("Done.\n");
+
+    return 0;
+}