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CGH0S7
2025-02-06 22:24:29 +08:00
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externals/openal-soft/examples/alstreamcb.cpp vendored Normal file
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/*
* OpenAL Callback-based Stream 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 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 "sndfile.h"
#include "AL/al.h"
#include "AL/alc.h"
#include "AL/alext.h"
#include "common/alhelpers.h"
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};
size_t mSamplesPerBlock{1};
size_t mBytesPerBlock{1};
enum class SampleType {
Int16, Float, IMA4, MSADPCM
};
SampleType mSampleFormat{SampleType::Int16};
/* 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. */
SNDFILE *mSndfile{nullptr};
SF_INFO mSfInfo{};
size_t mDecoderOffset{0};
/* The format of the callback samples. */
ALenum mFormat;
StreamPlayer()
{
alGenBuffers(1, &mBuffer);
if(alGetError() != AL_NO_ERROR)
throw std::runtime_error{"alGenBuffers failed"};
alGenSources(1, &mSource);
if(alGetError() != AL_NO_ERROR)
{
alDeleteBuffers(1, &mBuffer);
throw std::runtime_error{"alGenSources failed"};
}
}
~StreamPlayer()
{
alDeleteSources(1, &mSource);
alDeleteBuffers(1, &mBuffer);
if(mSndfile)
sf_close(mSndfile);
}
void close()
{
if(mSamplesPerBlock > 1)
alBufferi(mBuffer, AL_UNPACK_BLOCK_ALIGNMENT_SOFT, 0);
if(mSndfile)
{
alSourceRewind(mSource);
alSourcei(mSource, AL_BUFFER, 0);
sf_close(mSndfile);
mSndfile = nullptr;
}
}
bool open(const char *filename)
{
close();
/* Open the file and figure out the OpenAL format. */
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;
}
switch((mSfInfo.format&SF_FORMAT_SUBMASK))
{
case SF_FORMAT_PCM_24:
case SF_FORMAT_PCM_32:
case SF_FORMAT_FLOAT:
case SF_FORMAT_DOUBLE:
case SF_FORMAT_VORBIS:
case SF_FORMAT_OPUS:
case SF_FORMAT_ALAC_20:
case SF_FORMAT_ALAC_24:
case SF_FORMAT_ALAC_32:
case 0x0080/*SF_FORMAT_MPEG_LAYER_I*/:
case 0x0081/*SF_FORMAT_MPEG_LAYER_II*/:
case 0x0082/*SF_FORMAT_MPEG_LAYER_III*/:
if(alIsExtensionPresent("AL_EXT_FLOAT32"))
mSampleFormat = SampleType::Float;
break;
case SF_FORMAT_IMA_ADPCM:
if(mSfInfo.channels <= 2 && (mSfInfo.format&SF_FORMAT_TYPEMASK) == SF_FORMAT_WAV
&& alIsExtensionPresent("AL_EXT_IMA4")
&& alIsExtensionPresent("AL_SOFT_block_alignment"))
mSampleFormat = SampleType::IMA4;
break;
case SF_FORMAT_MS_ADPCM:
if(mSfInfo.channels <= 2 && (mSfInfo.format&SF_FORMAT_TYPEMASK) == SF_FORMAT_WAV
&& alIsExtensionPresent("AL_SOFT_MSADPCM")
&& alIsExtensionPresent("AL_SOFT_block_alignment"))
mSampleFormat = SampleType::MSADPCM;
break;
}
int splblocksize{}, byteblocksize{};
if(mSampleFormat == SampleType::IMA4 || mSampleFormat == SampleType::MSADPCM)
{
SF_CHUNK_INFO inf{ "fmt ", 4, 0, nullptr };
SF_CHUNK_ITERATOR *iter = sf_get_chunk_iterator(mSndfile, &inf);
if(!iter || sf_get_chunk_size(iter, &inf) != SF_ERR_NO_ERROR || inf.datalen < 14)
mSampleFormat = SampleType::Int16;
else
{
auto fmtbuf = std::make_unique<ALubyte[]>(inf.datalen);
inf.data = fmtbuf.get();
if(sf_get_chunk_data(iter, &inf) != SF_ERR_NO_ERROR)
mSampleFormat = SampleType::Int16;
else
{
byteblocksize = fmtbuf[12] | (fmtbuf[13]<<8u);
if(mSampleFormat == SampleType::IMA4)
{
splblocksize = (byteblocksize/mSfInfo.channels - 4)/4*8 + 1;
if(splblocksize < 1
|| ((splblocksize-1)/2 + 4)*mSfInfo.channels != byteblocksize)
mSampleFormat = SampleType::Int16;
}
else
{
splblocksize = (byteblocksize/mSfInfo.channels - 7)*2 + 2;
if(splblocksize < 2
|| ((splblocksize-2)/2 + 7)*mSfInfo.channels != byteblocksize)
mSampleFormat = SampleType::Int16;
}
}
}
}
if(mSampleFormat == SampleType::Int16)
{
mSamplesPerBlock = 1;
mBytesPerBlock = static_cast<size_t>(mSfInfo.channels * 2);
}
else if(mSampleFormat == SampleType::Float)
{
mSamplesPerBlock = 1;
mBytesPerBlock = static_cast<size_t>(mSfInfo.channels * 4);
}
else
{
mSamplesPerBlock = static_cast<size_t>(splblocksize);
mBytesPerBlock = static_cast<size_t>(byteblocksize);
}
mFormat = AL_NONE;
if(mSfInfo.channels == 1)
{
if(mSampleFormat == SampleType::Int16)
mFormat = AL_FORMAT_MONO16;
else if(mSampleFormat == SampleType::Float)
mFormat = AL_FORMAT_MONO_FLOAT32;
else if(mSampleFormat == SampleType::IMA4)
mFormat = AL_FORMAT_MONO_IMA4;
else if(mSampleFormat == SampleType::MSADPCM)
mFormat = AL_FORMAT_MONO_MSADPCM_SOFT;
}
else if(mSfInfo.channels == 2)
{
if(mSampleFormat == SampleType::Int16)
mFormat = AL_FORMAT_STEREO16;
else if(mSampleFormat == SampleType::Float)
mFormat = AL_FORMAT_STEREO_FLOAT32;
else if(mSampleFormat == SampleType::IMA4)
mFormat = AL_FORMAT_STEREO_IMA4;
else if(mSampleFormat == SampleType::MSADPCM)
mFormat = AL_FORMAT_STEREO_MSADPCM_SOFT;
}
else if(mSfInfo.channels == 3)
{
if(sf_command(mSndfile, SFC_WAVEX_GET_AMBISONIC, NULL, 0) == SF_AMBISONIC_B_FORMAT)
{
if(mSampleFormat == SampleType::Int16)
mFormat = AL_FORMAT_BFORMAT2D_16;
else if(mSampleFormat == SampleType::Float)
mFormat = AL_FORMAT_BFORMAT2D_FLOAT32;
}
}
else if(mSfInfo.channels == 4)
{
if(sf_command(mSndfile, SFC_WAVEX_GET_AMBISONIC, NULL, 0) == SF_AMBISONIC_B_FORMAT)
{
if(mSampleFormat == SampleType::Int16)
mFormat = AL_FORMAT_BFORMAT3D_16;
else if(mSampleFormat == SampleType::Float)
mFormat = AL_FORMAT_BFORMAT3D_FLOAT32;
}
}
if(!mFormat)
{
fprintf(stderr, "Unsupported channel count: %d\n", mSfInfo.channels);
sf_close(mSndfile);
mSndfile = nullptr;
return false;
}
/* Set a 1s ring buffer size. */
size_t numblocks{(static_cast<ALuint>(mSfInfo.samplerate) + mSamplesPerBlock-1)
/ mSamplesPerBlock};
mBufferDataSize = static_cast<ALuint>(numblocks * mBytesPerBlock);
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()
{
if(mSamplesPerBlock > 1)
alBufferi(mBuffer, AL_UNPACK_BLOCK_ALIGNMENT_SOFT, static_cast<int>(mSamplesPerBlock));
alBufferCallbackSOFT(mBuffer, mFormat, mSfInfo.samplerate, bufferCallbackC, this);
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);
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};
/* 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) / mBytesPerBlock * mSamplesPerBlock
: (static_cast<ALuint>(pos) + mStartOffset/mBytesPerBlock*mSamplesPerBlock))
/ 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)
{
/* 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) / mBytesPerBlock};
if(!writable) break;
if(mSampleFormat == SampleType::Int16)
{
sf_count_t num_frames{sf_readf_short(mSndfile,
reinterpret_cast<short*>(&mBufferData[woffset]),
static_cast<sf_count_t>(writable*mSamplesPerBlock))};
if(num_frames < 1) break;
read_bytes = static_cast<size_t>(num_frames) * mBytesPerBlock;
}
else if(mSampleFormat == SampleType::Float)
{
sf_count_t num_frames{sf_readf_float(mSndfile,
reinterpret_cast<float*>(&mBufferData[woffset]),
static_cast<sf_count_t>(writable*mSamplesPerBlock))};
if(num_frames < 1) break;
read_bytes = static_cast<size_t>(num_frames) * mBytesPerBlock;
}
else
{
sf_count_t numbytes{sf_read_raw(mSndfile, &mBufferData[woffset],
static_cast<sf_count_t>(writable*mBytesPerBlock))};
if(numbytes < 1) break;
read_bytes = static_cast<size_t>(numbytes);
}
woffset += read_bytes;
}
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 can go in front before
* wrapping.
*/
const size_t writable{(!roffset ? mBufferDataSize-woffset-1 :
(mBufferDataSize-woffset)) / mBytesPerBlock};
if(!writable) break;
if(mSampleFormat == SampleType::Int16)
{
sf_count_t num_frames{sf_readf_short(mSndfile,
reinterpret_cast<short*>(&mBufferData[woffset]),
static_cast<sf_count_t>(writable*mSamplesPerBlock))};
if(num_frames < 1) break;
read_bytes = static_cast<size_t>(num_frames) * mBytesPerBlock;
}
else if(mSampleFormat == SampleType::Float)
{
sf_count_t num_frames{sf_readf_float(mSndfile,
reinterpret_cast<float*>(&mBufferData[woffset]),
static_cast<sf_count_t>(writable*mSamplesPerBlock))};
if(num_frames < 1) break;
read_bytes = static_cast<size_t>(num_frames) * mBytesPerBlock;
}
else
{
sf_count_t numbytes{sf_read_raw(mSndfile, &mBufferData[woffset],
static_cast<sf_count_t>(writable*mBytesPerBlock))};
if(numbytes < 1) break;
read_bytes = static_cast<size_t>(numbytes);
}
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)
{
/* 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 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>] <filenames...>\n", argv[0]);
return 1;
}
argv++; argc--;
AudioManager almgr{&argv, &argc};
if(!alIsExtensionPresent("AL_SOFT_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->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);
/* All done with this file. Close it and go to the next */
player->close();
}
/* All done. */
printf("Done.\n");
return 0;
}