336 lines
11 KiB
C
336 lines
11 KiB
C
/*
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* OpenAL Source Play Example
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*
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* Copyright (c) 2017 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 an example for playing a sound buffer. */
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#include <assert.h>
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#include <inttypes.h>
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#include <limits.h>
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#include <stdio.h>
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#include <stdlib.h>
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#include "sndfile.h"
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#include "AL/al.h"
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#include "AL/alext.h"
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#include "common/alhelpers.h"
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enum FormatType {
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Int16,
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Float,
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IMA4,
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MSADPCM
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};
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/* LoadBuffer loads the named audio file into an OpenAL buffer object, and
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* returns the new buffer ID.
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*/
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static ALuint LoadSound(const char *filename)
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{
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enum FormatType sample_format = Int16;
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ALint byteblockalign = 0;
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ALint splblockalign = 0;
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sf_count_t num_frames;
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ALenum err, format;
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ALsizei num_bytes;
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SNDFILE *sndfile;
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SF_INFO sfinfo;
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ALuint buffer;
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void *membuf;
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/* Open the audio file and check that it's usable. */
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sndfile = sf_open(filename, SFM_READ, &sfinfo);
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if(!sndfile)
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{
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fprintf(stderr, "Could not open audio in %s: %s\n", filename, sf_strerror(sndfile));
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return 0;
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}
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if(sfinfo.frames < 1)
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{
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fprintf(stderr, "Bad sample count in %s (%" PRId64 ")\n", filename, sfinfo.frames);
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sf_close(sndfile);
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return 0;
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}
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/* Detect a suitable format to load. Formats like Vorbis and Opus use float
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* natively, so load as float to avoid clipping when possible. Formats
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* larger than 16-bit can also use float to preserve a bit more precision.
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*/
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switch((sfinfo.format&SF_FORMAT_SUBMASK))
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{
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case SF_FORMAT_PCM_24:
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case SF_FORMAT_PCM_32:
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case SF_FORMAT_FLOAT:
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case SF_FORMAT_DOUBLE:
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case SF_FORMAT_VORBIS:
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case SF_FORMAT_OPUS:
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case SF_FORMAT_ALAC_20:
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case SF_FORMAT_ALAC_24:
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case SF_FORMAT_ALAC_32:
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case 0x0080/*SF_FORMAT_MPEG_LAYER_I*/:
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case 0x0081/*SF_FORMAT_MPEG_LAYER_II*/:
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case 0x0082/*SF_FORMAT_MPEG_LAYER_III*/:
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if(alIsExtensionPresent("AL_EXT_FLOAT32"))
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sample_format = Float;
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break;
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case SF_FORMAT_IMA_ADPCM:
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/* ADPCM formats require setting a block alignment as specified in the
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* file, which needs to be read from the wave 'fmt ' chunk manually
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* since libsndfile doesn't provide it in a format-agnostic way.
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*/
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if(sfinfo.channels <= 2 && (sfinfo.format&SF_FORMAT_TYPEMASK) == SF_FORMAT_WAV
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&& alIsExtensionPresent("AL_EXT_IMA4")
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&& alIsExtensionPresent("AL_SOFT_block_alignment"))
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sample_format = IMA4;
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break;
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case SF_FORMAT_MS_ADPCM:
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if(sfinfo.channels <= 2 && (sfinfo.format&SF_FORMAT_TYPEMASK) == SF_FORMAT_WAV
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&& alIsExtensionPresent("AL_SOFT_MSADPCM")
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&& alIsExtensionPresent("AL_SOFT_block_alignment"))
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sample_format = MSADPCM;
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break;
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}
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if(sample_format == IMA4 || sample_format == MSADPCM)
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{
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/* For ADPCM, lookup the wave file's "fmt " chunk, which is a
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* WAVEFORMATEX-based structure for the audio format.
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*/
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SF_CHUNK_INFO inf = { "fmt ", 4, 0, NULL };
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SF_CHUNK_ITERATOR *iter = sf_get_chunk_iterator(sndfile, &inf);
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/* If there's an issue getting the chunk or block alignment, load as
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* 16-bit and have libsndfile do the conversion.
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*/
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if(!iter || sf_get_chunk_size(iter, &inf) != SF_ERR_NO_ERROR || inf.datalen < 14)
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sample_format = Int16;
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else
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{
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ALubyte *fmtbuf = calloc(inf.datalen, 1);
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inf.data = fmtbuf;
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if(sf_get_chunk_data(iter, &inf) != SF_ERR_NO_ERROR)
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sample_format = Int16;
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else
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{
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/* Read the nBlockAlign field, and convert from bytes- to
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* samples-per-block (verifying it's valid by converting back
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* and comparing to the original value).
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*/
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byteblockalign = fmtbuf[12] | (fmtbuf[13]<<8);
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if(sample_format == IMA4)
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{
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splblockalign = (byteblockalign/sfinfo.channels - 4)/4*8 + 1;
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if(splblockalign < 1
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|| ((splblockalign-1)/2 + 4)*sfinfo.channels != byteblockalign)
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sample_format = Int16;
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}
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else
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{
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splblockalign = (byteblockalign/sfinfo.channels - 7)*2 + 2;
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if(splblockalign < 2
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|| ((splblockalign-2)/2 + 7)*sfinfo.channels != byteblockalign)
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sample_format = Int16;
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}
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}
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free(fmtbuf);
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}
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}
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if(sample_format == Int16)
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{
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splblockalign = 1;
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byteblockalign = sfinfo.channels * 2;
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}
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else if(sample_format == Float)
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{
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splblockalign = 1;
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byteblockalign = sfinfo.channels * 4;
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}
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/* Figure out the OpenAL format from the file and desired sample type. */
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format = AL_NONE;
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if(sfinfo.channels == 1)
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{
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if(sample_format == Int16)
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format = AL_FORMAT_MONO16;
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else if(sample_format == Float)
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format = AL_FORMAT_MONO_FLOAT32;
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else if(sample_format == IMA4)
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format = AL_FORMAT_MONO_IMA4;
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else if(sample_format == MSADPCM)
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format = AL_FORMAT_MONO_MSADPCM_SOFT;
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}
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else if(sfinfo.channels == 2)
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{
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if(sample_format == Int16)
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format = AL_FORMAT_STEREO16;
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else if(sample_format == Float)
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format = AL_FORMAT_STEREO_FLOAT32;
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else if(sample_format == IMA4)
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format = AL_FORMAT_STEREO_IMA4;
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else if(sample_format == MSADPCM)
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format = AL_FORMAT_STEREO_MSADPCM_SOFT;
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}
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else if(sfinfo.channels == 3)
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{
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if(sf_command(sndfile, SFC_WAVEX_GET_AMBISONIC, NULL, 0) == SF_AMBISONIC_B_FORMAT)
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{
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if(sample_format == Int16)
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format = AL_FORMAT_BFORMAT2D_16;
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else if(sample_format == Float)
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format = AL_FORMAT_BFORMAT2D_FLOAT32;
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}
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}
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else if(sfinfo.channels == 4)
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{
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if(sf_command(sndfile, SFC_WAVEX_GET_AMBISONIC, NULL, 0) == SF_AMBISONIC_B_FORMAT)
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{
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if(sample_format == Int16)
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format = AL_FORMAT_BFORMAT3D_16;
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else if(sample_format == Float)
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format = AL_FORMAT_BFORMAT3D_FLOAT32;
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}
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}
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if(!format)
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{
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fprintf(stderr, "Unsupported channel count: %d\n", sfinfo.channels);
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sf_close(sndfile);
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return 0;
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}
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if(sfinfo.frames/splblockalign > (sf_count_t)(INT_MAX/byteblockalign))
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{
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fprintf(stderr, "Too many samples in %s (%" PRId64 ")\n", filename, sfinfo.frames);
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sf_close(sndfile);
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return 0;
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}
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/* Decode the whole audio file to a buffer. */
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membuf = malloc((size_t)(sfinfo.frames / splblockalign * byteblockalign));
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if(sample_format == Int16)
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num_frames = sf_readf_short(sndfile, membuf, sfinfo.frames);
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else if(sample_format == Float)
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num_frames = sf_readf_float(sndfile, membuf, sfinfo.frames);
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else
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{
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sf_count_t count = sfinfo.frames / splblockalign * byteblockalign;
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num_frames = sf_read_raw(sndfile, membuf, count);
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if(num_frames > 0)
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num_frames = num_frames / byteblockalign * splblockalign;
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}
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if(num_frames < 1)
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{
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free(membuf);
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sf_close(sndfile);
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fprintf(stderr, "Failed to read samples in %s (%" PRId64 ")\n", filename, num_frames);
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return 0;
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}
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num_bytes = (ALsizei)(num_frames / splblockalign * byteblockalign);
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printf("Loading: %s (%s, %dhz)\n", filename, FormatName(format), sfinfo.samplerate);
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fflush(stdout);
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/* Buffer the audio data into a new buffer object, then free the data and
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* close the file.
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*/
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buffer = 0;
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alGenBuffers(1, &buffer);
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if(splblockalign > 1)
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alBufferi(buffer, AL_UNPACK_BLOCK_ALIGNMENT_SOFT, splblockalign);
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alBufferData(buffer, format, membuf, num_bytes, sfinfo.samplerate);
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free(membuf);
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sf_close(sndfile);
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/* Check if an error occured, and clean up if so. */
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err = alGetError();
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if(err != AL_NO_ERROR)
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{
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fprintf(stderr, "OpenAL Error: %s\n", alGetString(err));
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if(buffer && alIsBuffer(buffer))
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alDeleteBuffers(1, &buffer);
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return 0;
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}
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return buffer;
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}
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int main(int argc, char **argv)
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{
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ALuint source, buffer;
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ALfloat offset;
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ALenum state;
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/* Print out usage if no arguments were specified */
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if(argc < 2)
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{
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fprintf(stderr, "Usage: %s [-device <name>] <filename>\n", argv[0]);
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return 1;
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}
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/* Initialize OpenAL. */
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argv++; argc--;
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if(InitAL(&argv, &argc) != 0)
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return 1;
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/* Load the sound into a buffer. */
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buffer = LoadSound(argv[0]);
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if(!buffer)
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{
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CloseAL();
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return 1;
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}
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/* Create the source to play the sound with. */
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source = 0;
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alGenSources(1, &source);
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alSourcei(source, AL_BUFFER, (ALint)buffer);
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assert(alGetError()==AL_NO_ERROR && "Failed to setup sound source");
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/* Play the sound until it finishes. */
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alSourcePlay(source);
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do {
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al_nssleep(10000000);
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alGetSourcei(source, AL_SOURCE_STATE, &state);
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/* Get the source offset. */
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alGetSourcef(source, AL_SEC_OFFSET, &offset);
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printf("\rOffset: %f ", offset);
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fflush(stdout);
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} while(alGetError() == AL_NO_ERROR && state == AL_PLAYING);
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printf("\n");
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/* All done. Delete resources, and close down OpenAL. */
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alDeleteSources(1, &source);
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alDeleteBuffers(1, &buffer);
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CloseAL();
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return 0;
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}
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