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-rw-r--r--src/output/oss_output_plugin.c701
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,
+};