The branch main has been updated by christos:

URL: 
https://cgit.FreeBSD.org/src/commit/?id=3decd659a7887e83c2c8af35053301dc7d6f7be2

commit 3decd659a7887e83c2c8af35053301dc7d6f7be2
Author:     Christos Margiolis <chris...@freebsd.org>
AuthorDate: 2024-08-24 12:07:45 +0000
Commit:     Christos Margiolis <chris...@freebsd.org>
CommitDate: 2024-08-24 12:07:45 +0000

    sound examples: Simplify audio example
    
    - Merge ossinit.h and basic.c.
    - Rename basic.c to audio.c.
    - Use err(3) instead of fprintf(3) + exit(3).
    - Some style(9) improvements.
    
    Sponsored by:   The FreeBSD Foundation
    MFC after:      2 days
    Reviewed by:    dev_submerge.ch
    Differential Revision:  https://reviews.freebsd.org/D46307
---
 share/examples/Makefile          |   5 +-
 share/examples/sound/oss/audio.c | 310 +++++++++++++++++++++++++++++++++++++++
 share/examples/sound/oss/basic.c |  99 -------------
 3 files changed, 312 insertions(+), 102 deletions(-)

diff --git a/share/examples/Makefile b/share/examples/Makefile
index d57416112226..65b261157b7a 100644
--- a/share/examples/Makefile
+++ b/share/examples/Makefile
@@ -320,9 +320,8 @@ SE_SOUND= \
 SE_DIRS+=      sound/oss
 SE_SOUND_OSS= \
        README \
-       basic.c \
-       midi.c \
-       ossinit.h
+       audio.c \
+       midi.c
 
 SE_DIRS+=      sunrpc
 SE_SUNRPC=     Makefile
diff --git a/share/examples/sound/oss/audio.c b/share/examples/sound/oss/audio.c
new file mode 100644
index 000000000000..4dd3c8b82575
--- /dev/null
+++ b/share/examples/sound/oss/audio.c
@@ -0,0 +1,310 @@
+/*
+ * 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
+ * <chris...@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/oss/basic.c b/share/examples/sound/oss/basic.c
deleted file mode 100644
index 83ecbe6ea9a7..000000000000
--- a/share/examples/sound/oss/basic.c
+++ /dev/null
@@ -1,99 +0,0 @@
-/*
- * SPDX-License-Identifier: BSD-2-Clause
- *
- * 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 "ossinit.h"
-
-int
-main()
-{
-       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
-        */
-       int ret;
-       int bytes = config.buffer_info.bytes;
-       int8_t *ibuf = malloc(bytes);
-       int8_t *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);
-}

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