437 lines
15 KiB
C++
437 lines
15 KiB
C++
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/*
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* OpenAL Callback-based Stream 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 a callback buffer. */
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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 <chrono>
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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 "SDL_sound.h"
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#include "SDL_audio.h"
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#include "SDL_stdinc.h"
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#include "AL/al.h"
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#include "AL/alc.h"
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#include "common/alhelpers.h"
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#ifndef SDL_AUDIO_MASK_BITSIZE
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#define SDL_AUDIO_MASK_BITSIZE (0xFF)
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#endif
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#ifndef SDL_AUDIO_BITSIZE
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#define SDL_AUDIO_BITSIZE(x) (x & SDL_AUDIO_MASK_BITSIZE)
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#endif
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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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namespace {
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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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/* A lockless ring-buffer (supports single-provider, single-consumer
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* operation).
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*/
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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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/* The buffer to get the callback, and source to play with. */
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ALuint mBuffer{0}, mSource{0};
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size_t mStartOffset{0};
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/* Handle for the audio file to decode. */
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Sound_Sample *mSample{nullptr};
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Uint32 mAvailableData{0};
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size_t mDecoderOffset{0};
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/* The format of the callback samples. */
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ALenum mFormat;
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ALsizei mSampleRate;
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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(mSample)
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Sound_FreeSample(mSample);
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}
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void close()
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{
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if(mSample)
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{
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alSourceRewind(mSource);
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alSourcei(mSource, AL_BUFFER, 0);
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Sound_FreeSample(mSample);
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mSample = 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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/* Open the file in its normal format. */
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mSample = Sound_NewSampleFromFile(filename, nullptr, 0);
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if(!mSample)
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{
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fprintf(stderr, "Could not open audio in %s\n", filename);
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return false;
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}
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/* Figure out the OpenAL format from the sample's format. */
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mFormat = AL_NONE;
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if(mSample->actual.channels == 1)
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{
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if(mSample->actual.format == AUDIO_U8)
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mFormat = AL_FORMAT_MONO8;
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else if(mSample->actual.format == AUDIO_S16SYS)
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mFormat = AL_FORMAT_MONO16;
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}
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else if(mSample->actual.channels == 2)
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{
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if(mSample->actual.format == AUDIO_U8)
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mFormat = AL_FORMAT_STEREO8;
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else if(mSample->actual.format == AUDIO_S16SYS)
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mFormat = AL_FORMAT_STEREO16;
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}
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if(!mFormat)
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{
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fprintf(stderr, "Unsupported sample format: 0x%04x, %d channels\n",
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mSample->actual.format, mSample->actual.channels);
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Sound_FreeSample(mSample);
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mSample = nullptr;
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return false;
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}
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mSampleRate = static_cast<ALsizei>(mSample->actual.rate);
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const auto frame_size = Uint32{mSample->actual.channels} *
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SDL_AUDIO_BITSIZE(mSample->actual.format) / 8;
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/* Set a 50ms decode buffer size. */
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Sound_SetBufferSize(mSample, static_cast<Uint32>(mSampleRate)*50/1000 * frame_size);
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mAvailableData = 0;
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/* Set a 1s ring buffer size. */
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mBufferDataSize = static_cast<Uint32>(mSampleRate) * size_t{frame_size};
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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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/* The actual C-style callback just forwards to the non-static method. Not
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* strictly needed and the compiler will optimize it to a normal function,
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* but it allows the callback implementation to have a nice 'this' pointer
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* with normal member access.
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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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/* NOTE: The callback *MUST* be real-time safe! That means no blocking,
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* no allocations or deallocations, no I/O, no page faults, or calls to
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* functions that could do these things (this includes calling to
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* libraries like SDL_sound, libsndfile, ffmpeg, etc). Nothing should
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* unexpectedly stall this call since the audio has to get to the
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* device on time.
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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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/* If the write offset == read offset, there's nothing left in the
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* ring-buffer. Break from the loop and give what has been written.
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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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/* If the write offset is behind the read offset, the readable
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* portion wrapped around. Just read up to the end of the buffer in
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* that case, otherwise read up to the write offset. Also limit the
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* amount to copy given how much is remaining to write.
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*/
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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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/* Copy from the ring buffer to the provided output buffer. Wrap
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* the resulting read offset if it reached the end of the ring-
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* buffer.
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*/
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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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/* Finally, store the updated read offset, and return how many bytes
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* have been written.
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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, mSampleRate, 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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mAvailableData = Sound_Decode(mSample);
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if(!mAvailableData)
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fprintf(stderr, "Failed to decode any samples: %s\n", Sound_GetError());
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return mAvailableData != 0;
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}
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bool update()
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{
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constexpr int BadFlags{SOUND_SAMPLEFLAG_EOF | SOUND_SAMPLEFLAG_ERROR};
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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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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 auto frame_size = Uint32{mSample->actual.channels} *
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SDL_AUDIO_BITSIZE(mSample->actual.format) / 8;
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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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/* For a stopped (underrun) source, the current playback offset is
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* the current decoder offset excluding the readable buffered data.
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* For a playing/paused source, it's the source's offset including
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* the playback offset the source was started with.
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*/
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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)) / mSample->actual.rate};
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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(mAvailableData > 0)
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{
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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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/* Note that the ring buffer's writable space is one byte less
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* than the available area because the write offset ending up
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* at the read offset would be interpreted as being empty
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* instead of full.
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*/
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const size_t writable{roffset-woffset-1};
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/* Don't copy the sample data if it can't all fit. */
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if(writable < mAvailableData) break;
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memcpy(&mBufferData[woffset], mSample->buffer, mAvailableData);
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woffset += mAvailableData;
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}
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else
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{
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/* If the read offset is at or behind the write offset, the
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* writeable area (might) wrap around. Make sure the sample
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* data can fit, and calculate how much goes in front and in
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* back.
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*/
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const size_t writable{mBufferDataSize+roffset-woffset-1};
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if(writable < mAvailableData) break;
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const size_t todo1{std::min<size_t>(mAvailableData, mBufferDataSize-woffset)};
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const size_t todo2{mAvailableData - todo1};
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memcpy(&mBufferData[woffset], mSample->buffer, todo1);
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woffset += todo1;
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if(woffset == mBufferDataSize)
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{
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woffset = 0;
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if(todo2 > 0)
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{
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memcpy(&mBufferData[woffset], static_cast<ALbyte*>(mSample->buffer)+todo1,
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todo2);
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woffset += todo2;
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}
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}
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}
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mWritePos.store(woffset, std::memory_order_release);
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mDecoderOffset += mAvailableData;
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if(!(mSample->flags&BadFlags))
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mAvailableData = Sound_Decode(mSample);
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else
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mAvailableData = 0;
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}
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if(state != AL_PLAYING && state != AL_PAUSED)
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{
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/* If the source is not playing or paused, it either underrun
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* (AL_STOPPED) or is just getting started (AL_INITIAL). If the
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* ring buffer is empty, it's done, otherwise play the source with
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* what's available.
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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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/* Store the playback offset that the source will start reading
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* from, so it can be tracked during playback.
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*/
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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;
|
||
|
|
}
|
||
|
|
};
|
||
|
|
|
||
|
|
} // 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;
|
||
|
|
}
|