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Diffstat (limited to 'share/examples/sound/ossinit.h')
| -rw-r--r-- | share/examples/sound/ossinit.h | 262 |
1 files changed, 262 insertions, 0 deletions
diff --git a/share/examples/sound/ossinit.h b/share/examples/sound/ossinit.h new file mode 100644 index 000000000000..b5fbd3e99244 --- /dev/null +++ b/share/examples/sound/ossinit.h @@ -0,0 +1,262 @@ +/* + * SPDX-License-Identifier: BSD-2-Clause-FreeBSD + * + * Copyright (c) 2021 Goran Mekić + * + * Redistribution and use in source and binary forms, with or without + * modification, are permitted provided that the following conditions + * are met: + * 1. Redistributions of source code must retain the above copyright + * notice, this list of conditions and the following disclaimer. + * 2. Redistributions in binary form must reproduce the above copyright + * notice, this list of conditions and the following disclaimer in the + * documentation and/or other materials provided with the distribution. + * + * THIS SOFTWARE IS PROVIDED BY THE AUTHOR AND CONTRIBUTORS ``AS IS'' AND + * ANY EXPRESS OR IMPLIED WARRANTIES, INCLUDING, BUT NOT LIMITED TO, THE + * IMPLIED WARRANTIES OF MERCHANTABILITY AND FITNESS FOR A PARTICULAR PURPOSE + * ARE DISCLAIMED. IN NO EVENT SHALL THE AUTHOR OR CONTRIBUTORS BE LIABLE + * FOR ANY DIRECT, INDIRECT, INCIDENTAL, SPECIAL, EXEMPLARY, OR CONSEQUENTIAL + * DAMAGES (INCLUDING, BUT NOT LIMITED TO, PROCUREMENT OF SUBSTITUTE GOODS + * OR SERVICES; LOSS OF USE, DATA, OR PROFITS; OR BUSINESS INTERRUPTION) + * HOWEVER CAUSED AND ON ANY THEORY OF LIABILITY, WHETHER IN CONTRACT, STRICT + * LIABILITY, OR TORT (INCLUDING NEGLIGENCE OR OTHERWISE) ARISING IN ANY WAY + * OUT OF THE USE OF THIS SOFTWARE, EVEN IF ADVISED OF THE POSSIBILITY OF + * SUCH DAMAGE. + */ + +#include <sys/soundcard.h> +#include <errno.h> +#include <fcntl.h> +#include <stdio.h> +#include <stdlib.h> +#include <string.h> +#include <unistd.h> + + +#ifndef SAMPLE_SIZE +#define SAMPLE_SIZE 16 +#endif + +/* Format can be unsigned, in which case replace S with U */ +#if SAMPLE_SIZE == 32 +typedef int32_t sample_t; +int format = AFMT_S32_NE; /* Signed 32bit native endian format */ +#elif SAMPLE_SIZE == 16 +typedef int16_t sample_t; +int format = AFMT_S16_NE; /* Signed 16bit native endian format */ +#elif SAMPLE_SIZE == 8 +typedef int8_t sample_t; +int format = AFMT_S8_NE; /* Signed 8bit native endian format */ +#else +#error Unsupported sample format! +typedef int32_t sample_t; +int format = AFMT_S32_NE; /* Not a real value, just silencing + * compiler errors */ +#endif + + + +/* + * Minimal configuration for OSS + * For real world applications, this structure will probably contain many + * more fields + */ +typedef struct config { + char *device; + int channels; + int fd; + int format; + int frag; + int sample_count; + int sample_rate; + int sample_size; + int chsamples; + int mmap; + oss_audioinfo audio_info; + audio_buf_info buffer_info; +} config_t; + + +/* + * Error state is indicated by value=-1 in which case application exits + * with error + */ +static inline void +check_error(const int value, const char *message) +{ + if (value == -1) { + fprintf(stderr, "OSS error: %s %s\n", message, strerror(errno)); + exit(1); + } +} + + + +/* Calculate frag by giving it minimal size of buffer */ +static inline int +size2frag(int x) +{ + int frag = 0; + + while ((1 << frag) < x) { + ++frag; + } + return frag; +} + + +/* + * Split input buffer into channels. Input buffer is in interleaved format + * which means if we have 2 channels (L and R), this is what the buffer of + * 8 samples would contain: L,R,L,R,L,R,L,R. The result are two channels + * containing: L,L,L,L and R,R,R,R. + */ +void +oss_split(config_t *config, sample_t *input, sample_t *output) +{ + int channel; + int index; + + for (int i = 0; i < config->sample_count; ++i) { + channel = i % config->channels; + index = i / config->channels; + output[channel * index] = input[i]; + } +} + + +/* + * Convert channels into interleaved format and place it in output + * buffer + */ +void +oss_merge(config_t *config, sample_t *input, sample_t *output) +{ + for (int channel = 0; channel < config->channels; ++channel) { + for (int index = 0; index < config->chsamples; ++index) { + output[index * config->channels + channel] = input[channel * index]; + } + } +} + +void +oss_init(config_t *config) +{ + int error; + int tmp; + + /* Open the device for read and write */ + config->fd = open(config->device, O_RDWR); + check_error(config->fd, "open"); + + /* Get device information */ + config->audio_info.dev = -1; + error = ioctl(config->fd, SNDCTL_ENGINEINFO, &(config->audio_info)); + check_error(error, "SNDCTL_ENGINEINFO"); + printf("min_channels: %d\n", config->audio_info.min_channels); + printf("max_channels: %d\n", config->audio_info.max_channels); + printf("latency: %d\n", config->audio_info.latency); + printf("handle: %s\n", config->audio_info.handle); + if (config->audio_info.min_rate > config->sample_rate || config->sample_rate > config->audio_info.max_rate) { + fprintf(stderr, "%s doesn't support chosen ", config->device); + fprintf(stderr, "samplerate of %dHz!\n", config->sample_rate); + exit(1); + } + if (config->channels < 1) { + config->channels = config->audio_info.max_channels; + } + + /* + * If device is going to be used in mmap mode, disable all format + * conversions. Official OSS documentation states error code should not be + * checked. http://manuals.opensound.com/developer/mmap_test.c.html#LOC10 + */ + if (config->mmap) { + tmp = 0; + ioctl(config->fd, SNDCTL_DSP_COOKEDMODE, &tmp); + } + + /* + * Set number of channels. If number of channels is chosen to the value + * near the one wanted, save it in config + */ + tmp = config->channels; + error = ioctl(config->fd, SNDCTL_DSP_CHANNELS, &tmp); + check_error(error, "SNDCTL_DSP_CHANNELS"); + if (tmp != config->channels) { /* or check if tmp is close enough? */ + fprintf(stderr, "%s doesn't support chosen ", config->device); + fprintf(stderr, "channel count of %d", config->channels); + fprintf(stderr, ", set to %d!\n", tmp); + } + config->channels = tmp; + + /* Set format, or bit size: 8, 16, 24 or 32 bit sample */ + tmp = config->format; + error = ioctl(config->fd, SNDCTL_DSP_SETFMT, &tmp); + check_error(error, "SNDCTL_DSP_SETFMT"); + if (tmp != config->format) { + fprintf(stderr, "%s doesn't support chosen sample format!\n", config->device); + exit(1); + } + + /* Most common values for samplerate (in kHz): 44.1, 48, 88.2, 96 */ + tmp = config->sample_rate; + error = ioctl(config->fd, SNDCTL_DSP_SPEED, &tmp); + check_error(error, "SNDCTL_DSP_SPEED"); + + /* Get and check device capabilities */ + error = ioctl(config->fd, SNDCTL_DSP_GETCAPS, &(config->audio_info.caps)); + check_error(error, "SNDCTL_DSP_GETCAPS"); + if (!(config->audio_info.caps & PCM_CAP_DUPLEX)) { + fprintf(stderr, "Device doesn't support full duplex!\n"); + exit(1); + } + if (config->mmap) { + if (!(config->audio_info.caps & PCM_CAP_TRIGGER)) { + fprintf(stderr, "Device doesn't support triggering!\n"); + exit(1); + } + if (!(config->audio_info.caps & PCM_CAP_MMAP)) { + fprintf(stderr, "Device doesn't support mmap mode!\n"); + exit(1); + } + } + + /* + * If desired frag is smaller than minimum, based on number of channels + * and format (size in bits: 8, 16, 24, 32), set that as frag. Buffer size + * is 2^frag, but the real size of the buffer will be read when the + * configuration of the device is successfull + */ + int min_frag = size2frag(config->sample_size * config->channels); + + if (config->frag < min_frag) { + config->frag = min_frag; + } + + /* + * Allocate buffer in fragments. Total buffer will be split in number + * of fragments (2 by default) + */ + if (config->buffer_info.fragments < 0) { + config->buffer_info.fragments = 2; + } + tmp = ((config->buffer_info.fragments) << 16) | config->frag; + error = ioctl(config->fd, SNDCTL_DSP_SETFRAGMENT, &tmp); + check_error(error, "SNDCTL_DSP_SETFRAGMENT"); + + /* When all is set and ready to go, get the size of buffer */ + error = ioctl(config->fd, SNDCTL_DSP_GETOSPACE, &(config->buffer_info)); + check_error(error, "SNDCTL_DSP_GETOSPACE"); + if (config->buffer_info.bytes < 1) { + fprintf( + stderr, + "OSS buffer error: buffer size can not be %d\n", + config->buffer_info.bytes + ); + exit(1); + } + config->sample_count = config->buffer_info.bytes / config->sample_size; + config->chsamples = config->sample_count / config->channels; +} |