Date: Fri, 27 Aug 2021 12:46:35 GMT From: Ka Ho Ng <khng@FreeBSD.org> To: src-committers@FreeBSD.org, dev-commits-src-all@FreeBSD.org, dev-commits-src-branches@FreeBSD.org Subject: git: 45d1b7b34a5a - stable/12 - sound: Add an example of basic sound application Message-ID: <202108271246.17RCkZIR059996@gitrepo.freebsd.org>
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The branch stable/12 has been updated by khng: URL: https://cgit.FreeBSD.org/src/commit/?id=45d1b7b34a5a431562fa793d2361a51497b19ff1 commit 45d1b7b34a5a431562fa793d2361a51497b19ff1 Author: Goran Mekić <meka@tilda.center> AuthorDate: 2021-08-04 10:04:54 +0000 Commit: Ka Ho Ng <khng@FreeBSD.org> CommitDate: 2021-08-27 12:05:27 +0000 sound: Add an example of basic sound application This is an example demonstrating the usage of the OSS-compatible APIs provided by the sound(4) subsystem. It reads frames from a dsp node and writes them to the same dsp node. Reviewed by: hselasky, bcr Differential revision: https://reviews.freebsd.org/D30149 (cherry picked from commit 21d854658801f6ddb91de3a3c3384e90f5d920f2) --- share/examples/Makefile | 7 ++ share/examples/sound/README | 66 +++++++++++ share/examples/sound/basic.c | 99 ++++++++++++++++ share/examples/sound/ossinit.h | 262 +++++++++++++++++++++++++++++++++++++++++ 4 files changed, 434 insertions(+) diff --git a/share/examples/Makefile b/share/examples/Makefile index 6b58daf5310e..5eab9f5210d7 100644 --- a/share/examples/Makefile +++ b/share/examples/Makefile @@ -30,6 +30,7 @@ LDIRS= BSD_daemon \ printing \ ses \ scsi_target \ + sound \ sunrpc \ ypldap @@ -317,6 +318,12 @@ SE_SCSI_TARGET= \ scsi_target.8 \ scsi_cmds.c +SE_DIRS+= sound +SE_SOUND= \ + basic.c \ + ossinit.h \ + README + SE_DIRS+= sunrpc SE_SUNRPC= Makefile diff --git a/share/examples/sound/README b/share/examples/sound/README new file mode 100644 index 000000000000..0188a26348c8 --- /dev/null +++ b/share/examples/sound/README @@ -0,0 +1,66 @@ +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/basic.c b/share/examples/sound/basic.c new file mode 100644 index 000000000000..c903cbd8bf11 --- /dev/null +++ b/share/examples/sound/basic.c @@ -0,0 +1,99 @@ +/* + * 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 "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); +} 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; +}
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