diff options
Diffstat (limited to 'src/output/oss_output_plugin.c')
| -rw-r--r-- | src/output/oss_output_plugin.c | 701 |
1 files changed, 701 insertions, 0 deletions
diff --git a/src/output/oss_output_plugin.c b/src/output/oss_output_plugin.c new file mode 100644 index 000000000..46505873b --- /dev/null +++ b/src/output/oss_output_plugin.c @@ -0,0 +1,701 @@ +/* + * Copyright (C) 2003-2011 The Music Player Daemon Project + * http://www.musicpd.org + * + * This program is free software; you can redistribute it and/or modify + * it under the terms of the GNU General Public License as published by + * the Free Software Foundation; either version 2 of the License, or + * (at your option) any later version. + * + * This program is distributed in the hope that it will be useful, + * but WITHOUT ANY WARRANTY; without even the implied warranty of + * MERCHANTABILITY or FITNESS FOR A PARTICULAR PURPOSE. See the + * GNU General Public License for more details. + * + * You should have received a copy of the GNU General Public License along + * with this program; if not, write to the Free Software Foundation, Inc., + * 51 Franklin Street, Fifth Floor, Boston, MA 02110-1301 USA. + */ + +#include "config.h" +#include "oss_output_plugin.h" +#include "output_api.h" +#include "mixer_list.h" +#include "fd_util.h" + +#include <glib.h> + +#include <sys/stat.h> +#include <sys/ioctl.h> +#include <fcntl.h> +#include <errno.h> +#include <stdlib.h> +#include <unistd.h> +#include <assert.h> + +#undef G_LOG_DOMAIN +#define G_LOG_DOMAIN "oss" + +#if defined(__OpenBSD__) || defined(__NetBSD__) +# include <soundcard.h> +#else /* !(defined(__OpenBSD__) || defined(__NetBSD__) */ +# include <sys/soundcard.h> +#endif /* !(defined(__OpenBSD__) || defined(__NetBSD__) */ + +/* We got bug reports from FreeBSD users who said that the two 24 bit + formats generate white noise on FreeBSD, but 32 bit works. This is + a workaround until we know what exactly is expected by the kernel + audio drivers. */ +#ifndef __linux__ +#undef AFMT_S24_PACKED +#undef AFMT_S24_NE +#endif + +struct oss_data { + struct audio_output base; + + int fd; + const char *device; + + /** + * The current input audio format. This is needed to reopen + * the device after cancel(). + */ + struct audio_format audio_format; +}; + +/** + * The quark used for GError.domain. + */ +static inline GQuark +oss_output_quark(void) +{ + return g_quark_from_static_string("oss_output"); +} + +static struct oss_data * +oss_data_new(void) +{ + struct oss_data *ret = g_new(struct oss_data, 1); + + ret->device = NULL; + ret->fd = -1; + + return ret; +} + +static void +oss_data_free(struct oss_data *od) +{ + g_free(od); +} + +enum oss_stat { + OSS_STAT_NO_ERROR = 0, + OSS_STAT_NOT_CHAR_DEV = -1, + OSS_STAT_NO_PERMS = -2, + OSS_STAT_DOESN_T_EXIST = -3, + OSS_STAT_OTHER = -4, +}; + +static enum oss_stat +oss_stat_device(const char *device, int *errno_r) +{ + struct stat st; + + if (0 == stat(device, &st)) { + if (!S_ISCHR(st.st_mode)) { + return OSS_STAT_NOT_CHAR_DEV; + } + } else { + *errno_r = errno; + + switch (errno) { + case ENOENT: + case ENOTDIR: + return OSS_STAT_DOESN_T_EXIST; + case EACCES: + return OSS_STAT_NO_PERMS; + default: + return OSS_STAT_OTHER; + } + } + + return OSS_STAT_NO_ERROR; +} + +static const char *default_devices[] = { "/dev/sound/dsp", "/dev/dsp" }; + +static bool +oss_output_test_default_device(void) +{ + int fd, i; + + for (i = G_N_ELEMENTS(default_devices); --i >= 0; ) { + fd = open_cloexec(default_devices[i], O_WRONLY, 0); + + if (fd >= 0) { + close(fd); + return true; + } + g_warning("Error opening OSS device \"%s\": %s\n", + default_devices[i], strerror(errno)); + } + + return false; +} + +static struct audio_output * +oss_open_default(GError **error) +{ + int i; + int err[G_N_ELEMENTS(default_devices)]; + enum oss_stat ret[G_N_ELEMENTS(default_devices)]; + + for (i = G_N_ELEMENTS(default_devices); --i >= 0; ) { + ret[i] = oss_stat_device(default_devices[i], &err[i]); + if (ret[i] == OSS_STAT_NO_ERROR) { + struct oss_data *od = oss_data_new(); + if (!ao_base_init(&od->base, &oss_output_plugin, NULL, + error)) { + g_free(od); + return NULL; + } + + od->device = default_devices[i]; + return &od->base; + } + } + + for (i = G_N_ELEMENTS(default_devices); --i >= 0; ) { + const char *dev = default_devices[i]; + switch(ret[i]) { + case OSS_STAT_NO_ERROR: + /* never reached */ + break; + case OSS_STAT_DOESN_T_EXIST: + g_warning("%s not found\n", dev); + break; + case OSS_STAT_NOT_CHAR_DEV: + g_warning("%s is not a character device\n", dev); + break; + case OSS_STAT_NO_PERMS: + g_warning("%s: permission denied\n", dev); + break; + case OSS_STAT_OTHER: + g_warning("Error accessing %s: %s\n", + dev, strerror(err[i])); + } + } + + g_set_error(error, oss_output_quark(), 0, + "error trying to open default OSS device"); + return NULL; +} + +static struct audio_output * +oss_output_init(const struct config_param *param, GError **error) +{ + const char *device = config_get_block_string(param, "device", NULL); + if (device != NULL) { + struct oss_data *od = oss_data_new(); + if (!ao_base_init(&od->base, &oss_output_plugin, param, + error)) { + g_free(od); + return NULL; + } + + od->device = device; + return &od->base; + } + + return oss_open_default(error); +} + +static void +oss_output_finish(struct audio_output *ao) +{ + struct oss_data *od = (struct oss_data *)ao; + + ao_base_finish(&od->base); + oss_data_free(od); +} + +static void +oss_close(struct oss_data *od) +{ + if (od->fd >= 0) + close(od->fd); + od->fd = -1; +} + +/** + * A tri-state type for oss_try_ioctl(). + */ +enum oss_setup_result { + SUCCESS, + ERROR, + UNSUPPORTED, +}; + +/** + * Invoke an ioctl on the OSS file descriptor. On success, SUCCESS is + * returned. If the parameter is not supported, UNSUPPORTED is + * returned. Any other failure returns ERROR and allocates a GError. + */ +static enum oss_setup_result +oss_try_ioctl_r(int fd, unsigned long request, int *value_r, + const char *msg, GError **error_r) +{ + assert(fd >= 0); + assert(value_r != NULL); + assert(msg != NULL); + assert(error_r == NULL || *error_r == NULL); + + int ret = ioctl(fd, request, value_r); + if (ret >= 0) + return SUCCESS; + + if (errno == EINVAL) + return UNSUPPORTED; + + g_set_error(error_r, oss_output_quark(), errno, + "%s: %s", msg, g_strerror(errno)); + return ERROR; +} + +/** + * Invoke an ioctl on the OSS file descriptor. On success, SUCCESS is + * returned. If the parameter is not supported, UNSUPPORTED is + * returned. Any other failure returns ERROR and allocates a GError. + */ +static enum oss_setup_result +oss_try_ioctl(int fd, unsigned long request, int value, + const char *msg, GError **error_r) +{ + return oss_try_ioctl_r(fd, request, &value, msg, error_r); +} + +/** + * Set up the channel number, and attempts to find alternatives if the + * specified number is not supported. + */ +static bool +oss_setup_channels(int fd, struct audio_format *audio_format, GError **error_r) +{ + const char *const msg = "Failed to set channel count"; + int channels = audio_format->channels; + enum oss_setup_result result = + oss_try_ioctl_r(fd, SNDCTL_DSP_CHANNELS, &channels, msg, error_r); + switch (result) { + case SUCCESS: + if (!audio_valid_channel_count(channels)) + break; + + audio_format->channels = channels; + return true; + + case ERROR: + return false; + + case UNSUPPORTED: + break; + } + + for (unsigned i = 1; i < 2; ++i) { + if (i == audio_format->channels) + /* don't try that again */ + continue; + + channels = i; + result = oss_try_ioctl_r(fd, SNDCTL_DSP_CHANNELS, &channels, + msg, error_r); + switch (result) { + case SUCCESS: + if (!audio_valid_channel_count(channels)) + break; + + audio_format->channels = channels; + return true; + + case ERROR: + return false; + + case UNSUPPORTED: + break; + } + } + + g_set_error(error_r, oss_output_quark(), EINVAL, "%s", msg); + return false; +} + +/** + * Set up the sample rate, and attempts to find alternatives if the + * specified sample rate is not supported. + */ +static bool +oss_setup_sample_rate(int fd, struct audio_format *audio_format, + GError **error_r) +{ + const char *const msg = "Failed to set sample rate"; + int sample_rate = audio_format->sample_rate; + enum oss_setup_result result = + oss_try_ioctl_r(fd, SNDCTL_DSP_SPEED, &sample_rate, + msg, error_r); + switch (result) { + case SUCCESS: + if (!audio_valid_sample_rate(sample_rate)) + break; + + audio_format->sample_rate = sample_rate; + return true; + + case ERROR: + return false; + + case UNSUPPORTED: + break; + } + + static const int sample_rates[] = { 48000, 44100, 0 }; + for (unsigned i = 0; sample_rates[i] != 0; ++i) { + sample_rate = sample_rates[i]; + if (sample_rate == (int)audio_format->sample_rate) + continue; + + result = oss_try_ioctl_r(fd, SNDCTL_DSP_SPEED, &sample_rate, + msg, error_r); + switch (result) { + case SUCCESS: + if (!audio_valid_sample_rate(sample_rate)) + break; + + audio_format->sample_rate = sample_rate; + return true; + + case ERROR: + return false; + + case UNSUPPORTED: + break; + } + } + + g_set_error(error_r, oss_output_quark(), EINVAL, "%s", msg); + return false; +} + +/** + * Convert a MPD sample format to its OSS counterpart. Returns + * AFMT_QUERY if there is no direct counterpart. + */ +static int +sample_format_to_oss(enum sample_format format) +{ + switch (format) { + case SAMPLE_FORMAT_UNDEFINED: + case SAMPLE_FORMAT_FLOAT: + return AFMT_QUERY; + + case SAMPLE_FORMAT_S8: + return AFMT_S8; + + case SAMPLE_FORMAT_S16: + return AFMT_S16_NE; + + case SAMPLE_FORMAT_S24: +#ifdef AFMT_S24_PACKED + return AFMT_S24_PACKED; +#else + return AFMT_QUERY; +#endif + + case SAMPLE_FORMAT_S24_P32: +#ifdef AFMT_S24_NE + return AFMT_S24_NE; +#else + return AFMT_QUERY; +#endif + + case SAMPLE_FORMAT_S32: +#ifdef AFMT_S32_NE + return AFMT_S32_NE; +#else + return AFMT_QUERY; +#endif + } + + return AFMT_QUERY; +} + +/** + * Convert an OSS sample format to its MPD counterpart. Returns + * SAMPLE_FORMAT_UNDEFINED if there is no direct counterpart. + */ +static enum sample_format +sample_format_from_oss(int format) +{ + switch (format) { + case AFMT_S8: + return SAMPLE_FORMAT_S8; + + case AFMT_S16_NE: + return SAMPLE_FORMAT_S16; + +#ifdef AFMT_S24_PACKED + case AFMT_S24_PACKED: + return SAMPLE_FORMAT_S24; +#endif + +#ifdef AFMT_S24_NE + case AFMT_S24_NE: + return SAMPLE_FORMAT_S24_P32; +#endif + +#ifdef AFMT_S32_NE + case AFMT_S32_NE: + return SAMPLE_FORMAT_S32; +#endif + + default: + return SAMPLE_FORMAT_UNDEFINED; + } +} + +/** + * Set up the sample format, and attempts to find alternatives if the + * specified format is not supported. + */ +static bool +oss_setup_sample_format(int fd, struct audio_format *audio_format, + GError **error_r) +{ + const char *const msg = "Failed to set sample format"; + int oss_format = sample_format_to_oss(audio_format->format); + enum oss_setup_result result = oss_format != AFMT_QUERY + ? oss_try_ioctl_r(fd, SNDCTL_DSP_SAMPLESIZE, + &oss_format, msg, error_r) + : UNSUPPORTED; + enum sample_format mpd_format; + switch (result) { + case SUCCESS: + mpd_format = sample_format_from_oss(oss_format); + if (mpd_format == SAMPLE_FORMAT_UNDEFINED) + break; + + audio_format->format = mpd_format; + +#ifdef AFMT_S24_PACKED + if (oss_format == AFMT_S24_PACKED) + audio_format->reverse_endian = + G_BYTE_ORDER != G_LITTLE_ENDIAN; +#endif + return true; + + case ERROR: + return false; + + case UNSUPPORTED: + break; + } + + /* the requested sample format is not available - probe for + other formats supported by MPD */ + + static const enum sample_format sample_formats[] = { + SAMPLE_FORMAT_S24_P32, + SAMPLE_FORMAT_S32, + SAMPLE_FORMAT_S24, + SAMPLE_FORMAT_S16, + SAMPLE_FORMAT_S8, + SAMPLE_FORMAT_UNDEFINED /* sentinel */ + }; + + for (unsigned i = 0; sample_formats[i] != SAMPLE_FORMAT_UNDEFINED; ++i) { + mpd_format = sample_formats[i]; + if (mpd_format == audio_format->format) + /* don't try that again */ + continue; + + oss_format = sample_format_to_oss(mpd_format); + if (oss_format == AFMT_QUERY) + /* not supported by this OSS version */ + continue; + + result = oss_try_ioctl_r(fd, SNDCTL_DSP_SAMPLESIZE, + &oss_format, msg, error_r); + switch (result) { + case SUCCESS: + mpd_format = sample_format_from_oss(oss_format); + if (mpd_format == SAMPLE_FORMAT_UNDEFINED) + break; + + audio_format->format = mpd_format; + +#ifdef AFMT_S24_PACKED + if (oss_format == AFMT_S24_PACKED) + audio_format->reverse_endian = + G_BYTE_ORDER != G_LITTLE_ENDIAN; +#endif + return true; + + case ERROR: + return false; + + case UNSUPPORTED: + break; + } + } + + g_set_error(error_r, oss_output_quark(), EINVAL, "%s", msg); + return false; +} + +/** + * Sets up the OSS device which was opened before. + */ +static bool +oss_setup(struct oss_data *od, struct audio_format *audio_format, + GError **error_r) +{ + return oss_setup_channels(od->fd, audio_format, error_r) && + oss_setup_sample_rate(od->fd, audio_format, error_r) && + oss_setup_sample_format(od->fd, audio_format, error_r); +} + +/** + * Reopen the device with the saved audio_format, without any probing. + */ +static bool +oss_reopen(struct oss_data *od, GError **error_r) +{ + assert(od->fd < 0); + + od->fd = open_cloexec(od->device, O_WRONLY, 0); + if (od->fd < 0) { + g_set_error(error_r, oss_output_quark(), errno, + "Error opening OSS device \"%s\": %s", + od->device, strerror(errno)); + return false; + } + + enum oss_setup_result result; + + const char *const msg1 = "Failed to set channel count"; + result = oss_try_ioctl(od->fd, SNDCTL_DSP_CHANNELS, + od->audio_format.channels, msg1, error_r); + if (result != SUCCESS) { + oss_close(od); + if (result == UNSUPPORTED) + g_set_error(error_r, oss_output_quark(), EINVAL, + "%s", msg1); + return false; + } + + const char *const msg2 = "Failed to set sample rate"; + result = oss_try_ioctl(od->fd, SNDCTL_DSP_SPEED, + od->audio_format.sample_rate, msg2, error_r); + if (result != SUCCESS) { + oss_close(od); + if (result == UNSUPPORTED) + g_set_error(error_r, oss_output_quark(), EINVAL, + "%s", msg2); + return false; + } + + const char *const msg3 = "Failed to set sample format"; + assert(sample_format_to_oss(od->audio_format.format) != AFMT_QUERY); + result = oss_try_ioctl(od->fd, SNDCTL_DSP_SAMPLESIZE, + sample_format_to_oss(od->audio_format.format), + msg3, error_r); + if (result != SUCCESS) { + oss_close(od); + if (result == UNSUPPORTED) + g_set_error(error_r, oss_output_quark(), EINVAL, + "%s", msg3); + return false; + } + + return true; +} + +static bool +oss_output_open(struct audio_output *ao, struct audio_format *audio_format, + GError **error) +{ + struct oss_data *od = (struct oss_data *)ao; + + od->fd = open_cloexec(od->device, O_WRONLY, 0); + if (od->fd < 0) { + g_set_error(error, oss_output_quark(), errno, + "Error opening OSS device \"%s\": %s", + od->device, strerror(errno)); + return false; + } + + if (!oss_setup(od, audio_format, error)) { + oss_close(od); + return false; + } + + od->audio_format = *audio_format; + return true; +} + +static void +oss_output_close(struct audio_output *ao) +{ + struct oss_data *od = (struct oss_data *)ao; + + oss_close(od); +} + +static void +oss_output_cancel(struct audio_output *ao) +{ + struct oss_data *od = (struct oss_data *)ao; + + if (od->fd >= 0) { + ioctl(od->fd, SNDCTL_DSP_RESET, 0); + oss_close(od); + } +} + +static size_t +oss_output_play(struct audio_output *ao, const void *chunk, size_t size, + GError **error) +{ + struct oss_data *od = (struct oss_data *)ao; + ssize_t ret; + + /* reopen the device since it was closed by dropBufferedAudio */ + if (od->fd < 0 && !oss_reopen(od, error)) + return 0; + + while (true) { + ret = write(od->fd, chunk, size); + if (ret > 0) + return (size_t)ret; + + if (ret < 0 && errno != EINTR) { + g_set_error(error, oss_output_quark(), errno, + "Write error on %s: %s", + od->device, strerror(errno)); + return 0; + } + } +} + +const struct audio_output_plugin oss_output_plugin = { + .name = "oss", + .test_default_device = oss_output_test_default_device, + .init = oss_output_init, + .finish = oss_output_finish, + .open = oss_output_open, + .close = oss_output_close, + .play = oss_output_play, + .cancel = oss_output_cancel, + + .mixer_plugin = &oss_mixer_plugin, +}; |