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freebsd
GitHub Repository: freebsd/freebsd-src
Path: blob/main/share/examples/sound/simple.c
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/*
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* SPDX-License-Identifier: BSD-2-Clause
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*
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* Copyright (c) 2024 The FreeBSD Foundation
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* Copyright (c) 2025 Goran Mekić
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*
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* Portions of this software were developed by Christos Margiolis
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* <[email protected]> under sponsorship from the FreeBSD Foundation.
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*
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* Redistribution and use in source and binary forms, with or without
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* modification, are permitted provided that the following conditions
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* are met:
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* 1. Redistributions of source code must retain the above copyright
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* notice, this list of conditions and the following disclaimer.
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* 2. Redistributions in binary form must reproduce the above copyright
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* notice, this list of conditions and the following disclaimer in the
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* documentation and/or other materials provided with the distribution.
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*
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* THIS SOFTWARE IS PROVIDED BY THE AUTHOR AND CONTRIBUTORS ``AS IS'' AND
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* ANY EXPRESS OR IMPLIED WARRANTIES, INCLUDING, BUT NOT LIMITED TO, THE
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* IMPLIED WARRANTIES OF MERCHANTABILITY AND FITNESS FOR A PARTICULAR PURPOSE
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* ARE DISCLAIMED. IN NO EVENT SHALL THE AUTHOR OR CONTRIBUTORS BE LIABLE
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* FOR ANY DIRECT, INDIRECT, INCIDENTAL, SPECIAL, EXEMPLARY, OR CONSEQUENTIAL
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* DAMAGES (INCLUDING, BUT NOT LIMITED TO, PROCUREMENT OF SUBSTITUTE GOODS
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* OR SERVICES; LOSS OF USE, DATA, OR PROFITS; OR BUSINESS INTERRUPTION)
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* HOWEVER CAUSED AND ON ANY THEORY OF LIABILITY, WHETHER IN CONTRACT, STRICT
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* LIABILITY, OR TORT (INCLUDING NEGLIGENCE OR OTHERWISE) ARISING IN ANY WAY
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* OUT OF THE USE OF THIS SOFTWARE, EVEN IF ADVISED OF THE POSSIBILITY OF
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* SUCH DAMAGE.
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*/
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#include <sys/soundcard.h>
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#include "oss.h"
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/*
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* Split input buffer into channels. The input buffer is in interleaved format,
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* which means if we have 2 channels (L and R), this is what the buffer of 8
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* samples would contain: L,R,L,R,L,R,L,R. The result of this function is a
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* buffer containing: L,L,L,L,R,R,R,R.
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*/
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static void
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to_channels(struct config *config, void *output)
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{
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uint8_t *in = config->buf;
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uint8_t *out = output;
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int i, channel, index, offset, byte;
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/* Iterate over bytes in the input buffer */
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for (byte = 0; byte < config->buffer_info.bytes;
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byte += config->sample_size) {
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/*
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* Get index of a sample in the input buffer measured in
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* samples
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*/
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i = byte / config->sample_size;
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/* Get which channel is being processed */
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channel = i % config->audio_info.max_channels;
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/* Get offset of the sample inside a single channel */
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offset = i / config->audio_info.max_channels;
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/* Get index of a sample in the output buffer */
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index = (channel * config->chsamples + offset) *
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config->sample_size;
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/* Copy singe sample from input to output */
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memcpy(out+index, in+byte, config->sample_size);
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}
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}
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/*
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* Convert channels into interleaved format and put into output buffer
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*/
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static void
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to_interleaved(struct config *config, void *input)
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{
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uint8_t *out = config->buf;
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uint8_t *in = input;
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int i, index, offset, channel, byte;
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/* Iterate over bytes in the input buffer */
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for (byte = 0; byte < config->buffer_info.bytes;
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byte += config->sample_size) {
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/*
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* Get index of a sample in the input buffer measured in
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* samples
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*/
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index = byte / config->sample_size;
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/* Get which channel is being processed */
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channel = index / config->chsamples;
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/* Get offset of the sample inside a single channel */
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offset = index % config->chsamples;
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/* Get index of a sample in the output buffer */
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i = (config->audio_info.max_channels * offset + channel) *
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config->sample_size;
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/* Copy singe sample from input to output */
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memcpy(out+i, in+byte, config->sample_size);
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}
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}
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int
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main(int argc, char *argv[])
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{
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struct config config = {
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.device = "/dev/dsp",
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.mode = O_RDWR,
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.format = AFMT_S32_NE,
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.sample_rate = 48000,
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};
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int32_t *channels;
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int rc, bytes;
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oss_init(&config);
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if (config.format != AFMT_S32_NE)
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errx(1, "Device doesn't support signed 32bit samples. "
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"Check with 'sndctl' if it can be configured for 's32le' format.");
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bytes = config.buffer_info.bytes;
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channels = malloc(bytes);
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for (;;) {
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if ((rc = read(config.fd, config.buf, bytes)) < bytes) {
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warn("Requested %d bytes, but read %d!\n", bytes, rc);
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break;
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}
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/*
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* Strictly speaking, we could omit "channels" and operate only
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* using config->buf, but this example tries to show the real
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* world application usage. The problem is that the buffer is
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* in interleaved format, and if you'd like to do any
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* processing and/or mixing, it is easier to do that if samples
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* are grouped per channel.
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*/
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to_channels(&config, channels);
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to_interleaved(&config, channels);
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if ((rc = write(config.fd, config.buf, bytes)) < bytes) {
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warn("Requested %d bytes, but wrote %d!\n", bytes, rc);
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break;
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}
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}
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free(channels);
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free(config.buf);
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close(config.fd);
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return (0);
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}
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