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author | Max Kellermann <max@duempel.org> | 2013-01-29 14:32:32 +0100 |
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committer | Max Kellermann <max@duempel.org> | 2013-01-29 14:32:32 +0100 |
commit | 26a9ce7b2927f2fc79af46c3152fbc41ee602197 (patch) | |
tree | 6510001270201b23f8e2f342940c70f5ea287adb /src/output/oss_output_plugin.c | |
parent | 76417d44464248949e7843eee0d5338a8e0a22ac (diff) | |
download | mpd-26a9ce7b2927f2fc79af46c3152fbc41ee602197.tar.gz mpd-26a9ce7b2927f2fc79af46c3152fbc41ee602197.tar.xz mpd-26a9ce7b2927f2fc79af46c3152fbc41ee602197.zip |
output/{alsa,oss}: convert to C++
Diffstat (limited to 'src/output/oss_output_plugin.c')
-rw-r--r-- | src/output/oss_output_plugin.c | 788 |
1 files changed, 0 insertions, 788 deletions
diff --git a/src/output/oss_output_plugin.c b/src/output/oss_output_plugin.c deleted file mode 100644 index e366a4537..000000000 --- a/src/output/oss_output_plugin.c +++ /dev/null @@ -1,788 +0,0 @@ -/* - * 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_compat.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 - -#ifdef AFMT_S24_PACKED -#include "pcm_export.h" -#endif - -struct oss_data { - struct audio_output base; - -#ifdef AFMT_S24_PACKED - struct pcm_export_state export; -#endif - - 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 current OSS audio format. This is needed to reopen the - * device after cancel(). - */ - int oss_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], g_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, g_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); -} - -#ifdef AFMT_S24_PACKED - -static bool -oss_output_enable(struct audio_output *ao, G_GNUC_UNUSED GError **error_r) -{ - struct oss_data *od = (struct oss_data *)ao; - - pcm_export_init(&od->export); - return true; -} - -static void -oss_output_disable(struct audio_output *ao) -{ - struct oss_data *od = (struct oss_data *)ao; - - pcm_export_deinit(&od->export); -} - -#endif - -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: - case SAMPLE_FORMAT_DSD: - return AFMT_QUERY; - - case SAMPLE_FORMAT_S8: - return AFMT_S8; - - case SAMPLE_FORMAT_S16: - return AFMT_S16_NE; - - 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_P32; -#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; - } -} - -/** - * Probe one sample format. - * - * @return the selected sample format or SAMPLE_FORMAT_UNDEFINED on - * error - */ -static enum oss_setup_result -oss_probe_sample_format(int fd, enum sample_format sample_format, - enum sample_format *sample_format_r, - int *oss_format_r, -#ifdef AFMT_S24_PACKED - struct pcm_export_state *export, -#endif - GError **error_r) -{ - int oss_format = sample_format_to_oss(sample_format); - if (oss_format == AFMT_QUERY) - return UNSUPPORTED; - - enum oss_setup_result result = - oss_try_ioctl_r(fd, SNDCTL_DSP_SAMPLESIZE, - &oss_format, - "Failed to set sample format", error_r); - -#ifdef AFMT_S24_PACKED - if (result == UNSUPPORTED && sample_format == SAMPLE_FORMAT_S24_P32) { - /* if the driver doesn't support padded 24 bit, try - packed 24 bit */ - oss_format = AFMT_S24_PACKED; - result = oss_try_ioctl_r(fd, SNDCTL_DSP_SAMPLESIZE, - &oss_format, - "Failed to set sample format", error_r); - } -#endif - - if (result != SUCCESS) - return result; - - sample_format = sample_format_from_oss(oss_format); - if (sample_format == SAMPLE_FORMAT_UNDEFINED) - return UNSUPPORTED; - - *sample_format_r = sample_format; - *oss_format_r = oss_format; - -#ifdef AFMT_S24_PACKED - pcm_export_open(export, sample_format, 0, false, false, - oss_format == AFMT_S24_PACKED, - oss_format == AFMT_S24_PACKED && - G_BYTE_ORDER != G_LITTLE_ENDIAN); -#endif - - return SUCCESS; -} - -/** - * 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, - int *oss_format_r, -#ifdef AFMT_S24_PACKED - struct pcm_export_state *export, -#endif - GError **error_r) -{ - enum sample_format mpd_format; - enum oss_setup_result result = - oss_probe_sample_format(fd, audio_format->format, - &mpd_format, oss_format_r, -#ifdef AFMT_S24_PACKED - export, -#endif - error_r); - switch (result) { - case SUCCESS: - audio_format->format = mpd_format; - return true; - - case ERROR: - return false; - - case UNSUPPORTED: - break; - } - - if (result != UNSUPPORTED) - return result == SUCCESS; - - /* 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_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; - - result = oss_probe_sample_format(fd, mpd_format, - &mpd_format, oss_format_r, -#ifdef AFMT_S24_PACKED - export, -#endif - error_r); - switch (result) { - case SUCCESS: - audio_format->format = mpd_format; - return true; - - case ERROR: - return false; - - case UNSUPPORTED: - break; - } - } - - g_set_error_literal(error_r, oss_output_quark(), EINVAL, - "Failed to set sample format"); - 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, &od->oss_format, -#ifdef AFMT_S24_PACKED - &od->export, -#endif - 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, g_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"; - result = oss_try_ioctl(od->fd, SNDCTL_DSP_SAMPLESIZE, - od->oss_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, g_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; - -#ifdef AFMT_S24_PACKED - chunk = pcm_export(&od->export, chunk, size, &size); -#endif - - while (true) { - ret = write(od->fd, chunk, size); - if (ret > 0) { -#ifdef AFMT_S24_PACKED - ret = pcm_export_source_size(&od->export, ret); -#endif - return ret; - } - - if (ret < 0 && errno != EINTR) { - g_set_error(error, oss_output_quark(), errno, - "Write error on %s: %s", - od->device, g_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, -#ifdef AFMT_S24_PACKED - .enable = oss_output_enable, - .disable = oss_output_disable, -#endif - .open = oss_output_open, - .close = oss_output_close, - .play = oss_output_play, - .cancel = oss_output_cancel, - - .mixer_plugin = &oss_mixer_plugin, -}; 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