520 lines
16 KiB
C
520 lines
16 KiB
C
/*
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* OpenAL Audio Stream Example
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*
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* Copyright (c) 2011 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 relatively simple streaming audio player. */
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#include <assert.h>
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#include <inttypes.h>
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#include <stdio.h>
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#include <stdlib.h>
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#include <string.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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/* Define the number of buffers and buffer size (in milliseconds) to use. 4
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* buffers at 200ms each gives a nice per-chunk size, and lets the queue last
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* for almost one second.
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*/
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#define NUM_BUFFERS 4
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#define BUFFER_MILLISEC 200
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typedef enum SampleType {
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Int16, Float, IMA4, MSADPCM
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} SampleType;
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typedef struct StreamPlayer {
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/* These are the buffers and source to play out through OpenAL with. */
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ALuint buffers[NUM_BUFFERS];
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ALuint source;
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/* Handle for the audio file */
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SNDFILE *sndfile;
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SF_INFO sfinfo;
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void *membuf;
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/* The sample type and block/frame size being read for the buffer. */
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SampleType sample_type;
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int byteblockalign;
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int sampleblockalign;
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int block_count;
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/* The format of the output stream (sample rate is in sfinfo) */
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ALenum format;
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} StreamPlayer;
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static StreamPlayer *NewPlayer(void);
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static void DeletePlayer(StreamPlayer *player);
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static int OpenPlayerFile(StreamPlayer *player, const char *filename);
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static void ClosePlayerFile(StreamPlayer *player);
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static int StartPlayer(StreamPlayer *player);
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static int UpdatePlayer(StreamPlayer *player);
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/* Creates a new player object, and allocates the needed OpenAL source and
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* buffer objects. Error checking is simplified for the purposes of this
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* example, and will cause an abort if needed.
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*/
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static StreamPlayer *NewPlayer(void)
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{
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StreamPlayer *player;
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player = calloc(1, sizeof(*player));
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assert(player != NULL);
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/* Generate the buffers and source */
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alGenBuffers(NUM_BUFFERS, player->buffers);
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assert(alGetError() == AL_NO_ERROR && "Could not create buffers");
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alGenSources(1, &player->source);
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assert(alGetError() == AL_NO_ERROR && "Could not create source");
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/* Set parameters so mono sources play out the front-center speaker and
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* won't distance attenuate. */
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alSource3i(player->source, AL_POSITION, 0, 0, -1);
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alSourcei(player->source, AL_SOURCE_RELATIVE, AL_TRUE);
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alSourcei(player->source, AL_ROLLOFF_FACTOR, 0);
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assert(alGetError() == AL_NO_ERROR && "Could not set source parameters");
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return player;
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}
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/* Destroys a player object, deleting the source and buffers. No error handling
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* since these calls shouldn't fail with a properly-made player object. */
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static void DeletePlayer(StreamPlayer *player)
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{
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ClosePlayerFile(player);
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alDeleteSources(1, &player->source);
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alDeleteBuffers(NUM_BUFFERS, player->buffers);
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if(alGetError() != AL_NO_ERROR)
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fprintf(stderr, "Failed to delete object IDs\n");
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memset(player, 0, sizeof(*player));
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free(player);
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}
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/* Opens the first audio stream of the named file. If a file is already open,
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* it will be closed first. */
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static int OpenPlayerFile(StreamPlayer *player, const char *filename)
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{
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int byteblockalign=0, splblockalign=0;
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ClosePlayerFile(player);
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/* Open the audio file and check that it's usable. */
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player->sndfile = sf_open(filename, SFM_READ, &player->sfinfo);
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if(!player->sndfile)
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{
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fprintf(stderr, "Could not open audio in %s: %s\n", filename, sf_strerror(NULL));
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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((player->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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player->sample_type = 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(player->sfinfo.channels <= 2
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&& (player->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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player->sample_type = IMA4;
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break;
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case SF_FORMAT_MS_ADPCM:
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if(player->sfinfo.channels <= 2
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&& (player->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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player->sample_type = MSADPCM;
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break;
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}
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if(player->sample_type == IMA4 || player->sample_type == 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(player->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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player->sample_type = 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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player->sample_type = 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(player->sample_type == IMA4)
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{
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splblockalign = (byteblockalign/player->sfinfo.channels - 4)/4*8 + 1;
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if(splblockalign < 1
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|| ((splblockalign-1)/2 + 4)*player->sfinfo.channels != byteblockalign)
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player->sample_type = Int16;
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}
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else
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{
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splblockalign = (byteblockalign/player->sfinfo.channels - 7)*2 + 2;
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if(splblockalign < 2
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|| ((splblockalign-2)/2 + 7)*player->sfinfo.channels != byteblockalign)
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player->sample_type = 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(player->sample_type == Int16)
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{
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player->sampleblockalign = 1;
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player->byteblockalign = player->sfinfo.channels * 2;
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}
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else if(player->sample_type == Float)
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{
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player->sampleblockalign = 1;
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player->byteblockalign = player->sfinfo.channels * 4;
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}
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else
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{
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player->sampleblockalign = splblockalign;
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player->byteblockalign = byteblockalign;
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}
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/* Figure out the OpenAL format from the file and desired sample type. */
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player->format = AL_NONE;
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if(player->sfinfo.channels == 1)
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{
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if(player->sample_type == Int16)
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player->format = AL_FORMAT_MONO16;
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else if(player->sample_type == Float)
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player->format = AL_FORMAT_MONO_FLOAT32;
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else if(player->sample_type == IMA4)
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player->format = AL_FORMAT_MONO_IMA4;
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else if(player->sample_type == MSADPCM)
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player->format = AL_FORMAT_MONO_MSADPCM_SOFT;
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}
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else if(player->sfinfo.channels == 2)
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{
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if(player->sample_type == Int16)
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player->format = AL_FORMAT_STEREO16;
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else if(player->sample_type == Float)
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player->format = AL_FORMAT_STEREO_FLOAT32;
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else if(player->sample_type == IMA4)
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player->format = AL_FORMAT_STEREO_IMA4;
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else if(player->sample_type == MSADPCM)
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player->format = AL_FORMAT_STEREO_MSADPCM_SOFT;
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}
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else if(player->sfinfo.channels == 3)
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{
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if(sf_command(player->sndfile, SFC_WAVEX_GET_AMBISONIC, NULL, 0) == SF_AMBISONIC_B_FORMAT)
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{
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if(player->sample_type == Int16)
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player->format = AL_FORMAT_BFORMAT2D_16;
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else if(player->sample_type == Float)
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player->format = AL_FORMAT_BFORMAT2D_FLOAT32;
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}
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}
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else if(player->sfinfo.channels == 4)
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{
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if(sf_command(player->sndfile, SFC_WAVEX_GET_AMBISONIC, NULL, 0) == SF_AMBISONIC_B_FORMAT)
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{
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if(player->sample_type == Int16)
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player->format = AL_FORMAT_BFORMAT3D_16;
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else if(player->sample_type == Float)
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player->format = AL_FORMAT_BFORMAT3D_FLOAT32;
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}
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}
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if(!player->format)
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{
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fprintf(stderr, "Unsupported channel count: %d\n", player->sfinfo.channels);
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sf_close(player->sndfile);
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player->sndfile = NULL;
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return 0;
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}
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player->block_count = player->sfinfo.samplerate / player->sampleblockalign;
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player->block_count = player->block_count * BUFFER_MILLISEC / 1000;
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player->membuf = malloc((size_t)(player->block_count * player->byteblockalign));
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return 1;
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}
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/* Closes the audio file stream */
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static void ClosePlayerFile(StreamPlayer *player)
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{
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if(player->sndfile)
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sf_close(player->sndfile);
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player->sndfile = NULL;
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free(player->membuf);
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player->membuf = NULL;
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if(player->sampleblockalign > 1)
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{
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ALsizei i;
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for(i = 0;i < NUM_BUFFERS;i++)
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alBufferi(player->buffers[i], AL_UNPACK_BLOCK_ALIGNMENT_SOFT, 0);
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player->sampleblockalign = 0;
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player->byteblockalign = 0;
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}
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}
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/* Prebuffers some audio from the file, and starts playing the source */
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static int StartPlayer(StreamPlayer *player)
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{
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ALsizei i;
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/* Rewind the source position and clear the buffer queue */
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alSourceRewind(player->source);
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alSourcei(player->source, AL_BUFFER, 0);
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/* Fill the buffer queue */
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for(i = 0;i < NUM_BUFFERS;i++)
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{
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sf_count_t slen;
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/* Get some data to give it to the buffer */
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if(player->sample_type == Int16)
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{
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slen = sf_readf_short(player->sndfile, player->membuf,
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player->block_count * player->sampleblockalign);
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if(slen < 1) break;
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slen *= player->byteblockalign;
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}
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else if(player->sample_type == Float)
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{
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slen = sf_readf_float(player->sndfile, player->membuf,
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player->block_count * player->sampleblockalign);
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if(slen < 1) break;
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slen *= player->byteblockalign;
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}
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else
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{
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slen = sf_read_raw(player->sndfile, player->membuf,
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player->block_count * player->byteblockalign);
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if(slen > 0) slen -= slen%player->byteblockalign;
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if(slen < 1) break;
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}
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if(player->sampleblockalign > 1)
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alBufferi(player->buffers[i], AL_UNPACK_BLOCK_ALIGNMENT_SOFT,
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player->sampleblockalign);
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alBufferData(player->buffers[i], player->format, player->membuf, (ALsizei)slen,
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player->sfinfo.samplerate);
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}
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if(alGetError() != AL_NO_ERROR)
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{
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fprintf(stderr, "Error buffering for playback\n");
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return 0;
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}
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/* Now queue and start playback! */
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alSourceQueueBuffers(player->source, i, player->buffers);
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alSourcePlay(player->source);
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if(alGetError() != AL_NO_ERROR)
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{
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fprintf(stderr, "Error starting playback\n");
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return 0;
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}
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return 1;
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}
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static int UpdatePlayer(StreamPlayer *player)
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{
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ALint processed, state;
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/* Get relevant source info */
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alGetSourcei(player->source, AL_SOURCE_STATE, &state);
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alGetSourcei(player->source, AL_BUFFERS_PROCESSED, &processed);
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if(alGetError() != AL_NO_ERROR)
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{
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fprintf(stderr, "Error checking source state\n");
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return 0;
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}
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/* Unqueue and handle each processed buffer */
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while(processed > 0)
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{
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ALuint bufid;
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sf_count_t slen;
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alSourceUnqueueBuffers(player->source, 1, &bufid);
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processed--;
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/* Read the next chunk of data, refill the buffer, and queue it
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* back on the source */
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if(player->sample_type == Int16)
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{
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slen = sf_readf_short(player->sndfile, player->membuf,
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player->block_count * player->sampleblockalign);
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if(slen > 0) slen *= player->byteblockalign;
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}
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else if(player->sample_type == Float)
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{
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slen = sf_readf_float(player->sndfile, player->membuf,
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player->block_count * player->sampleblockalign);
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if(slen > 0) slen *= player->byteblockalign;
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}
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else
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{
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slen = sf_read_raw(player->sndfile, player->membuf,
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player->block_count * player->byteblockalign);
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if(slen > 0) slen -= slen%player->byteblockalign;
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}
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if(slen > 0)
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{
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alBufferData(bufid, player->format, player->membuf, (ALsizei)slen,
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player->sfinfo.samplerate);
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alSourceQueueBuffers(player->source, 1, &bufid);
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}
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if(alGetError() != AL_NO_ERROR)
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{
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fprintf(stderr, "Error buffering data\n");
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return 0;
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}
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}
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/* Make sure the source hasn't underrun */
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if(state != AL_PLAYING && state != AL_PAUSED)
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{
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ALint queued;
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/* If no buffers are queued, playback is finished */
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alGetSourcei(player->source, AL_BUFFERS_QUEUED, &queued);
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if(queued == 0)
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return 0;
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alSourcePlay(player->source);
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if(alGetError() != AL_NO_ERROR)
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{
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fprintf(stderr, "Error restarting playback\n");
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return 0;
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}
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}
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return 1;
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}
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int main(int argc, char **argv)
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{
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StreamPlayer *player;
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int i;
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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>] <filenames...>\n", argv[0]);
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return 1;
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}
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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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player = NewPlayer();
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/* Play each file listed on the command line */
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for(i = 0;i < argc;i++)
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{
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const char *namepart;
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if(!OpenPlayerFile(player, argv[i]))
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continue;
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/* Get the name portion, without the path, for display. */
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namepart = strrchr(argv[i], '/');
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if(namepart || (namepart=strrchr(argv[i], '\\')))
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namepart++;
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else
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namepart = argv[i];
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printf("Playing: %s (%s, %dhz)\n", namepart, FormatName(player->format),
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player->sfinfo.samplerate);
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fflush(stdout);
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if(!StartPlayer(player))
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{
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ClosePlayerFile(player);
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continue;
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}
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while(UpdatePlayer(player))
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al_nssleep(10000000);
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/* All done with this file. Close it and go to the next */
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ClosePlayerFile(player);
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}
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printf("Done.\n");
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/* All files done. Delete the player, and close down OpenAL */
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DeletePlayer(player);
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player = NULL;
|
|
|
|
CloseAL();
|
|
|
|
return 0;
|
|
}
|