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authorSven Gothel <[email protected]>2023-05-03 16:17:49 +0200
committerSven Gothel <[email protected]>2023-05-03 16:17:49 +0200
commitec167fd05661a5b02dd406c87081f84a0f8dd77d (patch)
tree9c4669e471c9969bda59265381b18d2d416db060 /examples/alstreamcb.cpp
parent0d14d30808cfe7b9e3413353e3eef8a0f201399a (diff)
parentd3875f333fb6abe2f39d82caca329414871ae53b (diff)
Merge branch 'v1.23.1'
Resolved Conflicts: CMakeLists.txt
Diffstat (limited to 'examples/alstreamcb.cpp')
-rw-r--r--examples/alstreamcb.cpp551
1 files changed, 551 insertions, 0 deletions
diff --git a/examples/alstreamcb.cpp b/examples/alstreamcb.cpp
new file mode 100644
index 00000000..a2e7b659
--- /dev/null
+++ b/examples/alstreamcb.cpp
@@ -0,0 +1,551 @@
+/*
+ * 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 "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;
+}