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diff --git a/share/examples/sound/kqueue.c b/share/examples/sound/kqueue.c
new file mode 100644
--- /dev/null
+++ b/share/examples/sound/kqueue.c
@@ -0,0 +1,86 @@
+/*
+ * SPDX-License-Identifier: BSD-2-Clause
+ *
+ * Copyright (c) 2025 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/event.h>
+
+#include "oss.h"
+
+int
+main(int argc, char *argv[])
+{
+ int rc, bytes, kq;
+ struct kevent event = {};
+ struct config config = {
+ .device = "/dev/dsp",
+ .mode = O_RDWR,
+ .format = AFMT_S32_NE,
+ .sample_rate = 48000,
+ .mmap = 0,
+ };
+
+ /* Initialize device */
+ oss_init(&config);
+ bytes = config.buffer_info.bytes;
+
+ kq = kqueue();
+ if (kq == -1)
+ err(1, "Failed to allocate kqueue");
+ EV_SET(&event, config.fd, EVFILT_WRITE, EV_ADD | EV_CLEAR, 0, 0, 0);
+ rc = kevent(kq, &event, 1, NULL, 0, NULL);
+ if (rc == -1)
+ err(1, "Failed to register kevent");
+ for (;;) {
+ rc = kevent(kq, NULL, 0, &event, 1, NULL);
+ if (rc == -1 || event.data == 0) {
+ warn("Event error");
+ break;
+ }
+ if (event.flags & EV_ERROR) {
+ warn("Event error: %s", strerror(event.data));
+ break;
+ }
+ if ((rc = read(config.fd, config.buf, bytes)) < bytes) {
+ warn("Requested %d bytes, but read %d!\n", bytes, rc);
+ break;
+ }
+ if ((rc = write(config.fd, config.buf, bytes)) < bytes) {
+ warn("Requested %d bytes, but wrote %d!\n", bytes, rc);
+ break;
+ }
+ }
+ EV_SET(&event, config.fd, EVFILT_WRITE, EV_DELETE, 0, 0, 0);
+ rc = kevent(kq, &event, 1, NULL, 0, NULL);
+ if (rc == -1)
+ err(1, "Failed to unregister kevent");
+ close(kq);
+
+ /* Cleanup */
+ free(config.buf);
+ close(config.fd);
+
+ return (0);
+}
diff --git a/share/examples/sound/oss.h b/share/examples/sound/oss.h
new file mode 100644
--- /dev/null
+++ b/share/examples/sound/oss.h
@@ -0,0 +1,226 @@
+/*
+ * SPDX-License-Identifier: BSD-2-Clause
+ *
+ * Copyright (c) 2025 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 <err.h>
+#include <fcntl.h>
+#include <stdio.h>
+#include <stdlib.h>
+#include <string.h>
+#include <unistd.h>
+
+/*
+ * Minimal configuration for OSS
+ * For real world applications, this structure will probably contain many
+ * more fields
+ */
+struct config {
+ char *device;
+ int mode;
+ int fd;
+ int format;
+ int sample_count;
+ int sample_rate;
+ int sample_size;
+ int chsamples;
+ int mmap;
+ void *buf;
+ oss_audioinfo audio_info;
+ audio_buf_info buffer_info;
+};
+
+/* Calculate frag by giving it the buffer size. The buffer used by OSS is
+ * 2^fragment size, so this function looks for smalles fragment where buffer
+ * of size "x" will fit. For more advanced OSS and real-time applications,
+ * developers need to handle buffers more carefully. The size of the buffer
+ * in OSS is selected using fragment size size_selector and the buffer size
+ * is 2^size_selector for values between 4 and 16. The fragments determines
+ * in how many fragments the buffer will be, hence if the size_selector is 4,
+ * the requested size is 2^4 = 16 and for the fragments of 2, the total buffer
+ * size will be (2 ^ size_selector) * max_fragments
+ * or in this case 32 bytes. Please note that size of buffer is in bytes not
+ * samples. For example, 24bit sample will be represented with 3 bytes. If
+ * you're porting audio app from Linux, you should be aware that 24 bit samples
+ * are represented with 4 bytes (usually int).
+ * The idea of a total buffer size that holds max_fragments fragments is to give
+ * some slack and allow application to be about max_fragments - 1 fragments late.
+ * Let's call this the jitter tolerance. The jitter tolerance may be much less if
+ * there is a slight mismatch between the period and the samples per fragment.
+ * Jitter tolerance gets better if we can make either the period or the samples
+ * per fragment considerably smaller than the other. In our case that means we
+ * divide the total buffer size into smaller fragments, keeping overall latency at
+ * the same level.
+ * Official OSS development howto: http://manuals.opensound.com/developer/DSP.html
+ */
+static inline int
+size2frag(int x)
+{
+ int frag = 0;
+
+ while ((1 << frag) < x)
+ frag++;
+
+ return (frag);
+}
+
+static void
+oss_init(struct config *config)
+{
+ int tmp = 0;
+ unsigned long request = SNDCTL_DSP_GETOSPACE;
+
+ if ((config->fd = open(config->device, config->mode)) < 0)
+ err(1, "Error opening the device %s", config->device);
+
+ /* Get device information */
+ if (ioctl(config->fd, SNDCTL_ENGINEINFO, &config->audio_info) != 0)
+ err(1, "Unable to get device info");
+
+ /* Get device capabilities */
+ if (ioctl(config->fd, SNDCTL_DSP_GETCAPS, &config->audio_info.caps) != 0)
+ err(1, "Unable to get capabilities");
+
+ /* Check if device supports triggering */
+ if (!(config->audio_info.caps & PCM_CAP_TRIGGER))
+ errx(1, "Device doesn't support triggering!\n");
+
+ /* Handle memory mapped mode */
+ if (config->mmap) {
+ if (!(config->audio_info.caps & PCM_CAP_MMAP))
+ errx(1, "Device doesn't support mmap mode!\n");
+ tmp = 0;
+ if (ioctl(config->fd, SNDCTL_DSP_COOKEDMODE, &tmp) != 0)
+ err(1, "Unable to set cooked mode");
+ }
+
+ /* Set sample format */
+ if (ioctl(config->fd, SNDCTL_DSP_SETFMT, &config->format) != 0)
+ err(1, "Unable to set sample format");
+
+ /* Set channels */
+ if (ioctl(config->fd, SNDCTL_DSP_CHANNELS, &config->audio_info.max_channels) != 0)
+ err(1, "Unable to set channels");
+
+ /* Set sample rate */
+ if (ioctl(config->fd, SNDCTL_DSP_SPEED, &config->sample_rate) != 0)
+ err(1, "Unable to set sample rate");
+
+ /* Calculate sample size */
+ switch (config->format) {
+ case AFMT_S8:
+ case AFMT_U8:
+ config->sample_size = 1;
+ break;
+ case AFMT_S16_BE:
+ case AFMT_S16_LE:
+ case AFMT_U16_BE:
+ case AFMT_U16_LE:
+ config->sample_size = 2;
+ break;
+ case AFMT_S24_BE:
+ case AFMT_S24_LE:
+ case AFMT_U24_BE:
+ case AFMT_U24_LE:
+ config->sample_size = 3;
+ break;
+ case AFMT_S32_BE:
+ case AFMT_S32_LE:
+ case AFMT_U32_BE:
+ case AFMT_U32_LE:
+ case AFMT_F32_BE:
+ case AFMT_F32_LE:
+ config->sample_size = 4;
+ break;
+ default:
+ errx(1, "Invalid audio format %d", config->format);
+ break;
+ }
+
+ /* Set fragment and sample size.
+ * This part is optional as OSS has default value for fragment and
+ * sample size. From kernel perspective, there are few points OSS
+ * developers should be aware of:
+ * - There is a software facing buffer (bs) and a hardware driver
+ * buffer (b)
+ * - The sizes can be seen with cat /dev/sndstat as
+ * [b:_/_/_] [bs:_/_/_] (needed: sysctl hw.snd.verbose=2)
+ * - OSS ioctl only concern software buffer fragments, not hardware
+ * For USB the block size is according to hw.usb.uaudio.buffer_ms
+ * sysctl, meaning 2ms at 48kHz gives 0.002 * 48000 = 96 samples per
+ * block, all multiples of this work well. Block size should be set
+ * accordingly. OSS driver insists on reading / writing a certain
+ * number of samples at a time, one fragment full of samples. It is
+ * bound to do so in a fixed time frame, to avoid under- and overruns
+ * in communication with the hardware.
+ */
+ config->buffer_info.fragments = 2;
+ tmp = size2frag(config->sample_size * config->audio_info.max_channels);
+ tmp = ((config->buffer_info.fragments) << 16) | tmp;
+ if (ioctl(config->fd, SNDCTL_DSP_SETFRAGMENT, &tmp) != 0)
+ err(1, "Unable to set fragment size");
+
+ /* Get buffer info */
+ if ((config->mode & O_ACCMODE) == O_RDONLY)
+ request = SNDCTL_DSP_GETISPACE;
+ if (ioctl(config->fd, request, &config->buffer_info) != 0)
+ err(1, "Unable to get buffer info");
+ if (config->buffer_info.fragments < 1)
+ config->buffer_info.fragments = config->buffer_info.fragstotal;
+ if (config->buffer_info.bytes < 1)
+ config->buffer_info.bytes = config->buffer_info.fragstotal * config->buffer_info.fragsize;
+ if (config->buffer_info.bytes < 1) {
+ errx(1, "OSS buffer error: buffer size can not be %d\n",
+ config->buffer_info.bytes);
+ }
+ config->sample_count = config->buffer_info.bytes / config->sample_size;
+ config->chsamples = config->sample_count / config->audio_info.max_channels;
+ config->buf = malloc(config->buffer_info.bytes);
+
+ printf("bytes: %d, fragments: %d, fragsize: %d, fragstotal: %d, "
+ "samples: %d\n",
+ config->buffer_info.bytes, config->buffer_info.fragments,
+ config->buffer_info.fragsize, config->buffer_info.fragstotal,
+ config->sample_count);
+
+ /* Set the trigger */
+ switch (config->mode & O_ACCMODE) {
+ case O_RDONLY:
+ tmp = PCM_ENABLE_INPUT;
+ break;
+ case O_WRONLY:
+ tmp = PCM_ENABLE_OUTPUT;
+ break;
+ case O_RDWR:
+ tmp = PCM_ENABLE_INPUT | PCM_ENABLE_OUTPUT;
+ break;
+ default:
+ errx(1, "Invalid mode %d", config->mode);
+ break;
+ }
+ if (ioctl(config->fd, SNDCTL_DSP_SETTRIGGER, &tmp) != 0)
+ err(1, "Failed to set trigger");
+}
diff --git a/share/examples/sound/oss/README b/share/examples/sound/oss/README
deleted file mode 100644
--- a/share/examples/sound/oss/README
+++ /dev/null
@@ -1,66 +0,0 @@
-Briefly summarised, a general audio application will:
-- open(2)
-- ioctl(2)
-- read(2)
-- write(2)
-- close(2)
-
-In this example, read/write will be called in a loop for a duration of
-record/playback. Usually, /dev/dsp is the device you want to open, but it can
-be any OSS compatible device, even user space one created with virtual_oss. For
-configuring sample rate, bit depth and all other configuring of the device
-ioctl is used. As devices can support multiple sample rates and formats, what
-specific application should do in case there's an error issuing ioctl, as not
-all errors are fatal, is upon the developer to decide. As a general guideline
-Official OSS development howto should be used. FreeBSD OSS and virtual_oss are
-different to a small degree.
-
-For more advanced OSS and real-time applications, developers need to handle
-buffers more carefully. The size of the buffer in OSS is selected using fragment
-size size_selector and the buffer size is 2^size_selector for values between 4
-and 16. The formula on the official site is:
-
-int frag = (max_fragments << 16) | (size_selector);
-ioctl(fd, SNDCTL_DSP_SETFRAGMENT, &frag);
-
-The max_fragments determines in how many fragments the buffer will be, hence if
-the size_selector is 4, the requested size is 2^4 = 16 and for the
-max_fragments of 2, the total buffer size will be
-
-(2 ^ size_selector) * max_fragments
-
-or in this case 32 bytes. Please note that size of buffer is in bytes not
-samples. For example, 24bit sample will be represented with 3 bytes. If you're
-porting audio app from Linux, you should be aware that 24 bit samples are
-represented with 4 bytes (usually int).
-
-FreeBSD kernel will round up max_fragments and size of fragment/buffer, so the
-last thing any OSS code should do is get info about buffer with audio_buf_info
-and SNDCTL_DSP_GETOSPACE. That also means that not all values of max_fragments
-are permitted.
-
-From kernel perspective, there are few points OSS developers should be aware of:
-- There is a software facing buffer (bs) and a hardware driver buffer (b)
-- The sizes can be seen with cat /dev/sndstat as [b:_/_/_] [bs:_/_/_] (needed:
- sysctl hw.snd.verbose=2)
-- OSS ioctl only concern software buffer fragments, not hardware
-
-For USB the block size is according to hw.usb.uaudio.buffer_ms sysctl, meaning
-2ms at 48kHz gives 0.002 * 48000 = 96 samples per block, all multiples of this
-work well. Block size for virtual_oss, if used, should be set accordingly.
-
-OSS driver insists on reading / writing a certain number of samples at a time,
-one fragment full of samples. It is bound to do so in a fixed time frame, to
-avoid under- and overruns in communication with the hardware.
-
-The idea of a total buffer size that holds max_fragments fragments is to give
-some slack and allow application to be about max_fragments - 1 fragments late.
-Let's call this the jitter tolerance. The jitter tolerance may be much less if
-there is a slight mismatch between the period and the samples per fragment.
-
-Jitter tolerance gets better if we can make either the period or the samples
-per fragment considerably smaller than the other. In our case that means we
-divide the total buffer size into smaller fragments, keeping overall latency at
-the same level.
-
-Official OSS development howto: http://manuals.opensound.com/developer/DSP.html
diff --git a/share/examples/sound/oss/audio.c b/share/examples/sound/oss/audio.c
deleted file mode 100644
--- a/share/examples/sound/oss/audio.c
+++ /dev/null
@@ -1,310 +0,0 @@
-/*
- * SPDX-License-Identifier: BSD-2-Clause
- *
- * Copyright (c) 2021 Goran Mekić
- * Copyright (c) 2024 The FreeBSD Foundation
- *
- * Portions of this software were developed by Christos Margiolis
- * <christos@FreeBSD.org> under sponsorship from the FreeBSD Foundation.
- *
- * 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 <err.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)
- err(1, "OSS error: %s\n", message);
-}
-
-
-/* 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.
- */
-static void
-oss_split(config_t *config, sample_t *input, sample_t *output)
-{
- int channel, index, i;
-
- for (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
- */
-static void
-oss_merge(config_t *config, sample_t *input, sample_t *output)
-{
- int channel, index;
-
- for (channel = 0; channel < config->channels; ++channel) {
- for (index = 0; index < config->chsamples; ++index) {
- output[index * config->channels + channel] =
- input[channel * index];
- }
- }
-}
-
-static void
-oss_init(config_t *config)
-{
- int error, tmp, min_frag;
-
- /* 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) {
- errx(1, "%s doesn't support chosen samplerate of %dHz!\n",
- config->device, config->sample_rate);
- }
- 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");
- /* Or check if tmp is close enough? */
- if (tmp != config->channels) {
- errx(1, "%s doesn't support chosen channel count of %d set "
- "to %d!\n", config->device, config->channels, 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) {
- errx(1, "%s doesn't support chosen sample format!\n",
- config->device);
- }
-
- /* 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))
- errx(1, "Device doesn't support full duplex!\n");
-
- if (config->mmap) {
- if (!(config->audio_info.caps & PCM_CAP_TRIGGER))
- errx(1, "Device doesn't support triggering!\n");
- if (!(config->audio_info.caps & PCM_CAP_MMAP))
- errx(1, "Device doesn't support mmap mode!\n");
- }
-
- /*
- * 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 successful
- */
- 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) {
- errx(1, "OSS buffer error: buffer size can not be %d\n",
- config->buffer_info.bytes);
- }
- config->sample_count = config->buffer_info.bytes / config->sample_size;
- config->chsamples = config->sample_count / config->channels;
-}
-
-int
-main(int argc, char *argv[])
-{
- int ret, bytes;
- int8_t *ibuf, *obuf;
- config_t config = {
- .device = "/dev/dsp",
- .channels = -1,
- .format = format,
- .frag = -1,
- .sample_rate = 48000,
- .sample_size = sizeof(sample_t),
- .buffer_info.fragments = -1,
- .mmap = 0,
- };
-
- /* Initialize device */
- oss_init(&config);
-
- /*
- * Allocate input and output buffers so that their size match frag_size
- */
- bytes = config.buffer_info.bytes;
- ibuf = malloc(bytes);
- obuf = malloc(bytes);
- sample_t *channels = malloc(bytes);
-
- printf("bytes: %d, fragments: %d, fragsize: %d, fragstotal: %d, "
- "samples: %d\n",
- bytes, config.buffer_info.fragments,
- config.buffer_info.fragsize, config.buffer_info.fragstotal,
- config.sample_count);
-
- /* Minimal engine: read input and copy it to the output */
- for (;;) {
- ret = read(config.fd, ibuf, bytes);
- if (ret < bytes) {
- fprintf(stderr, "Requested %d bytes, but read %d!\n",
- bytes, ret);
- break;
- }
- oss_split(&config, (sample_t *)ibuf, channels);
- /* All processing will happen here */
- oss_merge(&config, channels, (sample_t *)obuf);
- ret = write(config.fd, obuf, bytes);
- if (ret < bytes) {
- fprintf(stderr, "Requested %d bytes, but wrote %d!\n",
- bytes, ret);
- break;
- }
- }
-
- /* Cleanup */
- free(channels);
- free(obuf);
- free(ibuf);
- close(config.fd);
-
- return (0);
-}
diff --git a/share/examples/sound/poll.c b/share/examples/sound/poll.c
new file mode 100644
--- /dev/null
+++ b/share/examples/sound/poll.c
@@ -0,0 +1,70 @@
+/*
+ * SPDX-License-Identifier: BSD-2-Clause
+ *
+ * Copyright (c) 2025 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/poll.h>
+
+#include "oss.h"
+
+int
+main(int argc, char *argv[])
+{
+ int rc, bytes;
+ struct pollfd pfds[1];
+ struct config config = {
+ .device = "/dev/dsp",
+ .mode = O_RDWR,
+ .format = AFMT_S32_NE,
+ .sample_rate = 48000,
+ .mmap = 0,
+ };
+
+ /* Initialize device */
+ oss_init(&config);
+ bytes = config.buffer_info.bytes;
+
+ for (;;) {
+ pfds[0].fd = config.fd;
+ pfds[0].events = POLLOUT;
+ rc = poll(pfds, sizeof(pfds) / sizeof(struct pollfd), -1);
+ if (pfds[0].revents != 0) {
+ if ((rc = read(config.fd, config.buf, bytes)) < bytes) {
+ warn("Requested %d bytes, but read %d!\n", bytes, rc);
+ break;
+ }
+ if ((rc = write(config.fd, config.buf, bytes)) < bytes) {
+ err(1, "Requested %d bytes, but wrote %d!\n", bytes, rc);
+ break;
+ }
+ }
+ }
+
+ /* Cleanup */
+ free(config.buf);
+ close(config.fd);
+
+ return (0);
+}
diff --git a/share/examples/sound/select.c b/share/examples/sound/select.c
new file mode 100644
--- /dev/null
+++ b/share/examples/sound/select.c
@@ -0,0 +1,70 @@
+/*
+ * SPDX-License-Identifier: BSD-2-Clause
+ *
+ * Copyright (c) 2025 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/poll.h>
+
+#include "oss.h"
+
+int
+main(int argc, char *argv[])
+{
+ int rc, bytes;
+ fd_set fds;
+ struct config config = {
+ .device = "/dev/dsp",
+ .mode = O_RDWR,
+ .format = AFMT_S32_NE,
+ .sample_rate = 48000,
+ .mmap = 0,
+ };
+
+ /* Initialize device */
+ oss_init(&config);
+ bytes = config.buffer_info.bytes;
+
+ for (;;) {
+ FD_ZERO(&fds);
+ FD_SET(config.fd, &fds);
+ rc = select(config.fd + 1, &fds, NULL, NULL, NULL);
+ if (FD_ISSET(config.fd, &fds)) {
+ if ((rc = read(config.fd, config.buf, bytes)) < bytes) {
+ warn("Requested %d bytes, but read %d!\n", bytes, rc);
+ break;
+ }
+ if ((rc = write(config.fd, config.buf, bytes)) < bytes) {
+ warn("Requested %d bytes, but wrote %d!\n", bytes, rc);
+ break;
+ }
+ }
+ }
+
+ /* Cleanup */
+ free(config.buf);
+ close(config.fd);
+
+ return (0);
+}
diff --git a/share/examples/sound/simple.c b/share/examples/sound/simple.c
new file mode 100644
--- /dev/null
+++ b/share/examples/sound/simple.c
@@ -0,0 +1,112 @@
+/*
+ * SPDX-License-Identifier: BSD-2-Clause
+ *
+ * Copyright (c) 2021 Goran Mekić
+ * Copyright (c) 2024 The FreeBSD Foundation
+ *
+ * Portions of this software were developed by Christos Margiolis
+ * <christos@FreeBSD.org> under sponsorship from the FreeBSD Foundation.
+ *
+ * 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 "oss.h"
+
+/*
+ * 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 of this functions is a
+ * buffer containing: L,L,L,L,R,R,R,R.
+ */
+static void
+to_channels(struct config *config, void *output) {
+ int channel, index, sample;
+ int8_t *in;
+ int8_t *out;
+
+ for (int i = 0; i < config->buffer_info.bytes; i += config->sample_size) {
+ sample = i / config->sample_size;
+ channel = sample % config->audio_info.max_channels;
+ index = sample / config->audio_info.max_channels;
+ in = ((int8_t *)config->buf) + i;
+ out = ((int8_t *)output) + channel * index;
+ memcpy(out, in, config->sample_size);
+ }
+}
+
+/* Convert channels into interleaved format and place it in output buffer */
+static void
+to_interleaved(struct config *config, void *input) {
+ int index;
+ int chs = config->audio_info.max_channels;
+ int chbytes = config->chsamples * config->sample_size;
+ int8_t *out;
+ int8_t *in;
+
+ for (int channel = 0; channel < chs; channel++)
+ for (int i = 0; index < chbytes; index += config->sample_size) {
+ index = i * config->sample_size;
+ in = ((int8_t *)input) + channel * index;
+ out = ((int8_t *)config->buf) + channel + index * chs;
+ memcpy(in, out, config->sample_size);
+ }
+}
+
+int
+main(int argc, char *argv[])
+{
+ int rc, bytes;
+ int32_t *channels;
+ struct config config = {
+ .device = "/dev/dsp",
+ .mode = O_RDWR,
+ .format = AFMT_S32_NE,
+ .sample_rate = 48000,
+ .mmap = 0,
+ };
+
+ /* Initialize device */
+ oss_init(&config);
+ bytes = config.buffer_info.bytes;
+ channels = malloc(bytes);
+
+ for (;;) {
+ if ((rc = read(config.fd, config.buf, bytes)) < bytes) {
+ warn("Requested %d bytes, but read %d!\n", bytes, rc);
+ break;
+ }
+ to_channels(&config, channels);
+ /* All processing will happen here */
+ to_interleaved(&config, channels);
+ if ((rc = write(config.fd, config.buf, bytes)) < bytes) {
+ warn("Requested %d bytes, but wrote %d!\n", bytes, rc);
+ break;
+ }
+ }
+
+ /* Cleanup */
+ free(channels);
+ free(config.buf);
+ close(config.fd);
+
+ return (0);
+}

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